Cisco Unified Communications Manager

Express System Administrator Guide

June 2007

Americas Headquarters
Cisco Systems, Inc.
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Fax: 408 527-0883

Customer Order Number: OL-10663-02

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Cisco Unified Communications Manager Express System Administrator Guide

© 2007 Cisco Systems, Inc. All rights reserved.
 C O N T E N T S

FEATURE MAP 31

Cisco Unified CME Features Roadmap 37

Cisco Unified CME Overview 47

Contents 47

Information About Cisco Unified CME 47

Cisco Unified CME Overview 48
Licenses 49
PBX or Keyswitch Model 50
Where to Go Next 52

Additional References 52
Related Documents 52
Related Websites 54
MIBs 54
Obtaining Documentation, Obtaining Support, and Security Guidelines 54

Before You Begin 55

Contents 55

Information About Planning Your Configuration 55

System Design 55
Configuration Methods Summary 56
Cisco Unified Communications Express - QCT 58
Cisco Unified CME GUI 59
How to Configure Cisco Unified CME 60
Prerequisites for Configuring Cisco Unified CME 60
Restrictions for Configuring Cisco Unified CME 61
Using Cisco IOS Commands to Create or Modify the Configuration 61
Using Cisco Unified Communications Express - QCT to Generate a Telephony Configuration 62
Using Cisco Unified CME GUI to Modify or Maintain Configuration 63
Feature Summary 64

Planning Worksheets 67

Installing and Upgrading Cisco Unified CME Software 69

Contents 69

Prerequisites for Installing Cisco Unified CME Software 69

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Information About Cisco Unified CME Software 70

Basic Files 70
GUI Files 70
Phone Firmware Files 70
IXML Template 72
Music-on-Hold (MOH) File 72
Script Files 72
Bundled TSP Archive 73
File Naming Conventions 73
Cisco Unified Communications Express - Quick Configuration Tool 73

How to Install and Upgrade Cisco Unified CME Software 74

Installing Cisco Unified CME Software 74
SCCP: Upgrading or Downgrading Phone Firmware Between Versions 76
SIP: Upgrading or Downgrading Phone Firmware Between Versions 77
SCCP: Converting Phone Firmware to SIP 81
SIP: Converting Phone to SCCP 84
SCCP: Verifying the Phone Firmware Version on an IP Phone 88
Troubleshooting Tips 88
Additional References 90
Related Documents 90
Technical Assistance 90

Defining Network Parameters 91

Contents 91

Prerequisites for Defining Network Parameters 91

Restrictions for Defining Network Parameters 92

Information About Defining Network Parameters 92
DHCP Service 92
Network Time Protocol for the Cisco Unified CME Router 93
DTMF Relay 93
SIP Register Support 93
Out-of-Dialog REFER 94
How to Define Network Parameters 95
Enabling Calls in Your VoIP Network 96
Defining DHCP 98
Enabling Network Time Protocol on the Cisco Unified CME Router 104
Configuring DTMF Relay for H.323 Networks in Multisite Installations 105
Configuring SIP Trunk Support 106
Verifying SIP Trunk Support Configuration 108

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Changing the TFTP Address on a DHCP Server 109

Enabling OOD-R 110
Verifying OOD-R Configuration 112
Troubleshooting OOD-R 113
Configuration Examples for Network Parameters 114
NTP Server: Example 114
DTMF Relay for H.323 Networks: Example 114
OOD-R: Example 115
Where to Go Next 115

Additional References 115

Related Documents 115
Technical Assistance 116
Feature Information for Network Parameters 117

Configuring System-Level Parameters 119

Contents 119

Prerequisites for System-Level Parameters 119

Information About Configuring System-Level Parameters 120

Network Time Protocol for SIP Phones 120
Per-Phone Configuration Files 120
Redundant Cisco Unified CME Router 121
Timeouts 122
How to Configure System-Level Parameters 122
Configuring Bulk Registration 123
SCCP: Setting Up Cisco Unified CME 125
SCCP: Setting Date and Time Parameters 127
SCCP: Blocking Automatic Registration 128
SCCP: Defining Per-Phone Configuration Files and Alternate Location 129
SCCP: Changing Defaults for Time Outs 130
SCCP: Configuring a Redundant Router 132
SIP: Setting Up Cisco Unified CME 135
SIP: Setting Date and Time Parameters 137
SIP: Setting Network Time Protocol 139
SIP: Changing Session-Level Application for SIP Phones 140
Configuration Examples for System-Level Parameters 141
System-Level Parameters: Example 141
Blocking Automatic Registration: Example 142
Redundant Router: Example 143
Where to Go Next 143

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Additional References 143

Related Documents 143
Technical Assistance 144
Feature Information for System-Level Parameters 145

Configuring Phones to Make Basic Calls 147

Contents 147

Prerequisites for Configuring Phones to Make Basic Calls 148

Restrictions for Configuring Phones to Make Basic Calls 148

Information About Configuring Phones to Make Basic Calls 148

Phones in Cisco Unified CME 148
Directory Numbers 149
PSTN FXO Trunk Lines 154
Analog Phones 154
Remote Teleworker Phones 155
Digit Collection on SIP Phones 157
Session Transport Protocol for SIP Phones 158
How to Configure Phones for a PBX System 158
SCCP: Creating Directory Numbers 158
SCCP: Assigning Directory Numbers to Phones 160
SIP: Creating Directory Numbers 162
SIP: Assigning Directory Numbers to Phones 163
SIP: Configuring Dial Plans 165
SIP: Verifying Dial Plan Configuration 168
SIP: Enabling KPML 169
SIP: Selecting Session-Transport Protocol for a Phone 171
SIP: Disabling SIP Proxy Registration for a Directory Number 172
Configuring Codec for Local Calling Between SIP and SCCP Phones 174

How to Configure Phones for a Key System 176

SCCP: Creating Directory Numbers for a Simple Key System 176
SCCP: Configuring Trunk Lines for a Key System 179
SCCP: Configuring Individual IP Phones for Key System 187
How to Configure Cisco ATA, Analog Phone Support, Remote Phones, and Cisco IP Communicator 188
Configuring Cisco ATA Support 189
Verifying Cisco ATA Support 190
Troubleshooting Cisco ATA Support 190
Using Call Pickup and Group Call Pickup with Cisco ATA 192
SCCP: Configuring Analog Phone Support 193
SCCP: Verifying Analog Phone Support 196

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SCCP: Enabling a Remote Phone 196

SCCP: Verifying Remote Phones 198
SCCP: Configuring Cisco IP Communicator Support 198
SCCP: Verifying Cisco IP Communicator Support 199
SCCP: Troubleshooting Cisco IP Communicator Support 199
Configuration Examples for Making Basic Calls 199
Configuring SCCP Phones for Making Basic Calls: Example 199
Configuring SIP Phones for Making Basic Calls: Example 204
Disabling a Bulk Registration for a SIP Phone: Example 206
Cisco ATA: Example 207
SCCP Analog Phone: Example 207
Remote Teleworker Phones: Example 208
Where to Go Next 208

Additional References 209

Related Documents 209
Technical Assistance 209
Feature Information for Configuring Phones to Make Basic Calls 210

Creating Phone Configurations Using Extension Assigner 213

Contents 213

Prerequisites for Extension Assigner 213

Restrictions for Extension Assigner 214

Information About Extension Assigner 214

Extension Assigner Overview 214
Files Included in this Release 218
How to Configure Extension Assigner 219
Determining Which Extension Numbers to Assign to the New Phones and Plan Your
Configuration 220
Downloading the Tcl Script 220
Configuring the Tcl Script 221
Specifying the Extension That Installation Technicians Call to Assign Extension Numbers 224
Specifying Whether the Extension Used to Access Extension Assigner Is Dialed Automatically 225
Configuring the Extension Assigner Feature 226
Configuring Temporary Extension Numbers for Phones That Use Extension Assigner 227
Configuring Extension Numbers That Installation Technicians Can Assign to Phones 229
Configuring Ephones with Temporary MAC Addresses 231
Configuring the Router to Automatically Save Your Configuration 234
Provide the Installation Technician with the Needed Information 236
Assigning New Extension Numbers 236

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Unassigning the Current Extension Number 237

Reassigning the Current Extension Number 237
Verifying Extension Assigner 238
Configuration Examples for Extension Assigner 239

Additional References 242

Related Documents 242
Technical Assistance 242
Feature Information for Extension Assigner 243

Generating Configuration Files for Phones 245

Contents 245

Information About Configuration Files 245

Configuration Files for Phones in Cisco Unified CME 245
Per-Phone Configuration Files 246
How to Generate Configuration Files for Phones 247
SCCP: Generating Configuration Files for SCCP Phones 247
SCCP: Verifying Configuration Files for SCCP Phones 248
SIP: Generating Configuration Profiles for SIP Phones 250
SIP: Verifying Configuration Profiles for SIP Phones 251
Where to Go Next 254

Additional References 255

Related Documents 255
Technical Assistance 255

Resetting and Restarting Phones 257

Contents 257

Information About Resetting and Restarting Phones 257

Differences between Resetting and Restarting IP Phones 257

How to Reset and Restart Phones 258

SCCP: Using the reset Command 259
SCCP: Using the restart Command 260
SIP: Using the reset Command 261
SIP: Using the restart Command 263
Verifying Basic Calling 264
Additional References 265
Related Documents 265
Technical Assistance 265

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Configuring Dialing Plans 267

Contents 267

Information About Dialing Plans 267

Phone Number Plan 268
Dial-Plan Patterns 269
Direct Inward Dialing Trunk Lines 270
Voice Translation Rules and Profiles 270
Secondary Dial Tone 270
How to Configure Dialing Plans 271
SCCP: Configuring Dial-Plan Patterns 271
SIP: Configuring Dial-Plan Patterns 272
Verifying Dial-Plan Patterns 274
Defining Voice Translation Rules in Cisco CME 3.2 and Later Versions 275
SCCP: Applying Voice Translation Rules in Cisco CME 3.2 and Later Versions 277
SCCP: Applying Translation Rules Before Cisco CME 3.2 278
SIP: Applying Voice Translation Rules in Cisco Unified CME 4.1 and Later 280
SIP: Applying Voice Translation Rules before Cisco Unified CME 4.1 281
Verifying Voice Translation Rules and Profiles 282
Activating a Secondary Dial Tone 283
Configuration Examples for Dialing Plan Features 284
Secondary Dial Tone: Example 284
Additional References 285
Related Documents 285
Technical Assistance 285
Feature Information for Dialing Plan Features 286

Configuring Localization Support 287

Contents 287

Information About Localization 287

System-Defined Locales 288
User-Defined Locales 288
Multiple Locales 288
How to Configure Localization Support 289
Installing System-Defined Locales for Cisco Unified IP Phone 7911, 7941, 7961, 7970, and 7971 289
Installing User-Defined Locales 292
Verifying User-Defined Locales 294
Configuring Multiple Locales 295
Verifying Multiple Locales 298
Configuration Examples for Localization 299

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Multiple User and Network Locales: Example 299

User-Defined Locales: Example 300
Where to Go Next 301

Additional References 301

Related Documents 301
Technical Assistance 301
Feature Information for Localization Support 302

Configuring Transcoding Resources 303

Contents 303

Prerequisites for Transcoding Resources 303

Restrictions for Transcoding Resources 304

Information About Transcoding Resources 304

Transcoding Support 304
Transcoding When a Remote Phone Uses G.729r8 305
Secure DSP Farm Transcoding 306
How to Configure Transcoding Resources 306
Determining DSP Resources for Transcoding 306
Provisioning NMs or NM Farms for Transcoding 309
Configuring DSP Farms for NM-HDs and NM-HDV2s 310
Configuring DSP Farms for NM-HDVs 314
Modifying the Number of Transcoding Sessions for NM-HDVs 316
Configuring the Cisco Unified CME Router to Act as the DSP Farm Host 317
Registering the DSP Farm with Cisco Unified CME in Secure Mode 320
Verifying DSP Farm Operation 329
Tuning DSP Farm Performance 332
Configuration Examples for Transcoding Resources 333
DSP Farms for NM-HDVs: Example 334
DSP Farms for NM-HDs and NM-HDV2s: Example 334
Cisco Unified CME Router as the DSP Farm Host: Example 335

Where to go Next 335

Additional References 335

Related Documents 335
Technical Assistance 336
Feature Information for Transcoding Resources 337

Enabling the GUI 339

Contents 339

Prerequisites for Enabling the GUI 339

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Restrictions for Enabling the GUI 340

Information About Enabling the GUI 340

Cisco Unified CME GUI Support 340
AAA Authentication 341
How to Enable the GUI 341
Enabling the HTTP Server 341
Enabling GUI Access for the System Administrator 343
Accessing the Cisco Unified CME GUI 344
Creating a Customized XML File for Customer Administrator GUI 345
Enabling GUI Access for Customer Administrators 346
Enabling GUI Access for Phone Users 348
Troubleshooting the Cisco Unified CME GUI 349
Configuration Examples for Enabling the GUI 350
HTTP and Account Configuration: Example 350
XML Configuration File Template: Example 350
XML Configuration File: Example 351
Additional References 352
Related Documents 352
Technical Assistance 353
Feature Information for Enabling the GUI 354

Integrating Voice Mail 355

Contents 355

Prerequisites 355

Information About Voice-Mail Integration 355

Cisco Unity Connection Integration 356
Cisco Unity Express Integration 356
Cisco Unity Integration 356
DTMF Integration for Legacy Voice-Mail Applications 357
Mailbox Selection Policy 357
RFC 2833 DTMF MTP Passthrough 357
MWI Line Selection 358
AMWI (Audible Message Waiting Indicator) 358
SIP MWI Prefix Specification 359
SIP MWI - QSIG Translation 359
How to Configure Voice-Mail Integration 360
SCCP: Configuring a Voice Mailbox Pilot Number 361
SCCP: Configuring a Mailbox Selection Policy 362
SIP: Configuring a Voice Mailbox Pilot Number 365

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Enabling DTMF Integration 367

SCCP: Configuring a Phone Line for MWI Outcall 374
SCCP: Configuring Audible MWI 375
SIP: Enabling MWI at System-Level 376
SIP: Configuring a Directory Number for MWI 377
Enabling SIP MWI Prefix Specification 380
Verifying Voice-Mail Integration 381
Configuration Examples for Voice-Mail Integration 381
Enabling DTMF Integration for Legacy Voice-Mail Applications: Example 382
Enabling Mailbox Selection Policy for SCCP Phones: Example 382
Configuring a Voice Mailbox: Example 382
Enabling DTMF Integration Using RFC 2833: Example 382
Enabling DTMF Integration Using SIP Notify: Example 383
Configuring a SCCP Phone Line for MWI: Example 383
Enabling SIP MWI Prefix Specification: Example 384
Configuring SIP Directory Number for MWI Outcall: Example 384
Configuring a SIP Directory Number for MWI Unsolicited Notify: Example 384
Configuring a SIP Directory Number for MWI Subscribe/NOTIFY: Example 385
Additional References 385
Related Documents 385
Technical Assistance 385
Feature Information for Voice-Mail Integration 386

Configuring Security 387

Contents 387

Prerequisites for Security 388

Restrictions for Security 388

Information About Security 389

Phone Authentication Overview 390
Public Key Infrastructure 391
Phone Authentication Components 391
Phone Authentication Process 394
Startup Messages 395
Configuration File Maintenance 395
CTL File Maintenance 396
CTL Client and Provider 396
Manually Importing MIC Root Certificate 397
Feature Design of Media Encryption 397
Secure Cisco Unified CME 398

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Secure Supplementary Services 399

Secure Transcoding for Remote Phones with DSP Farm Transcoding Configured 400
Secure Cisco Unified CME with Cisco Unity Express 401
Secure Cisco Unified CME with Cisco Unity 401
How to Configure Security 402
Configuring the Cisco IOS Certification Authority 402
Verifying the Cisco IOS Certification Authority 406
Configuring the Registration Authority 406
Verifying the Registration Authority 409
Authenticating Certificates for Server Functions 409
Verifying Certificates for Server Functions 412
Manually Importing MIC Root Certificate 412
Configuring Telephony-Service Security Parameters 414
Verifying Telephony-Service Security Parameters 419
Configuring the CTL Client 419
Verifying the CTL Client 424
Configuring the CTL Provider 424
Verifying the CTL Provider 426
Configuring the CAPF Server 426
Verifying the CAPF Server 429
Entering the Authentication String on the Phone 430
Verifying the Authentication String on the Phone 431
Configuring Secure Calls Between Cisco Unified CMEs Across an H.323 Trunk 431
Configuring Cisco Unified CME SRTP Fallback for H.323 Dial Peers 433
Configuring Cisco Unity for Secure Cisco Unified CME Operation 435
Configuration Examples for Security 438
Cisco IOS CA Server: Example 438
Enabling a Registration Authority: Example 438
Manually Importing MIC Root Certificate on the Cisco Unified CME Router: Example 439
Obtaining a Certificate for Cisco Unified CME Server Functions: Example 442
Telephony-Service Security Parameters: Example 442
CTL Client Running on Cisco Unified CME Router: Example 442
CTL Client Running on Another Router: Example 442
CAPF Server: Example 443
Secure Cisco Voice Gateway with Media Encryption (SRTP) on Cisco Unified CME: Example 445
Secure Cisco Unified CME: Example 451
Where to Go Next 458

Additional References 459

Related Documents 459

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Technical Assistance 459

Feature Information for Security 460

Configuring Automatic Line Selection 461

Contents 461

Information About Automatic Line Selection 461

Automatic Line Selection for Incoming and Outgoing Calls 461

How to Configure Automatic Line Selection 462

Enabling Automatic Line Selection 462
Verifying Automatic Line Selection 464
Configuration Examples for Automatic Line Selection 464
Automatic Line Selection: Example 464
Additional References 465
Related Documents 465
Technical Assistance 465
Feature Information for Automatic Line Selection 466

Configuring Call Blocking 467

Contents 467

Information About Call Blocking 467

Call Blocking Based on Date and Time (After-Hours Toll Bar) 468
Call Blocking Override 468
Class of Restriction 469
How to Configure Call Blocking 469
SCCP: Applying Class of Restriction to a Directory Number 470
SIP: Applying Class of Restriction to Directory Number 471
Verifying Class of Restriction 472
Configuring Call Blocking Based on Date and Time 474
Configuring Call Blocking Exemption for a Dial Peer 475
SCCP: Configuring Call Blocking Exemption for an Individual Phone 476
SIP: Configuring Call Blocking Exemption for an Individual Phone or Directory Number 477
Verifying Call Blocking Based on Date and Time 478
Configuration Examples for Call Blocking 479
Call Blocking: Example 479
Class of Restriction: Example 480
Where to Go Next 481

Additional References 481

Related Documents 481
Technical Assistance 482

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Feature Information for Call Blocking 483

Configuring Call Park 485

Contents 485

Information About Call Park 485

Basic Call Park 486
Dedicated Call-Park Slots 488
Call-Park Blocking 489
Call-Park Redirect 489
How to Configure Call Park 490
Enabling Call Park 490
Verifying Call Park 494
Troubleshooting Call Park 495
Configuration Examples for Call Park 495
Basic Call Park: Example 495
Phone Blocked From Using Call Park: Example 495
Call-Park Redirect: Example 496
Where to Go Next 496

Additional References 497

Related Documents 497
Technical Assistance 497
Feature Information for Call Park 498

Configuring Call Transfer and Forwarding 499

Contents 499

Information About Call Transfer and Forwarding 499

Call Forwarding 500
B2BUA Call Forwarding for SIP Devices 501
Call Forward All Synchronization for SIP Phones 501
Call Transfer 502
H.450.2 and H.450.3 Support 503
Transfer Method Recommendations by Cisco Unified CME Version 506
H.450.12 Support 507
Hairpin Call Routing 507
H.450 Tandem Gateways 510
Dial Peers 512
QSIG Supplementary Services 512
Disabling SIP Supplementary Services for Call Forward and Call Transfer 513
Typical Network Scenarios for Call Transfer and Call Forwarding 514

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How to Configure Call Transfer and Forwarding 518

Enabling Call Transfer and Forwarding at System-Level 519
SCCP: Enabling Call Forwarding for a Directory Number 523
SCCP: Enabling Call Transfer for a Directory Number 526
SCCP: Configuring Call Transfer Options for Phones 527
SCCP: Verifying Call Transfer 528
Enabling H.450.12 Capabilities 529
Enabling H.323-to-H.323 Connection Capabilities 531
Forwarding Calls Using Local Hairpin Routing 533
Enabling H.450.7 and QSIG Supplementary Services at a System-Level 535
Enabling H.450.7 and QSIG Supplementary Services on a Dial Peer 536
Disabling SIP Supplementary Services for Call Forward and Call Transfer 538
Enabling Interworking with Cisco Unified Communications Manager 540
SIP: Configuring SIP-to-SIP Phone Call Forwarding 546
SIP: Configuring Call-Forwarding-All Soft Key URI 548
SIP: Specifying Number of 3XX Responses To be Handled 549
SIP: Configuring Call Transfer 550
Configuration Examples for Call Transfer and Forwarding 552
H.450.2 and H.450.3: Example 553
Basic Call Forwarding: Example 553
Call Forwarding Blocked for Local Calls: Example 553
Selective Call Forwarding: Example 553
Call Transfer: Example 554
H.450.12: Example 554
H.450.7 and QSIG Supplementary Services: Example 555
Cisco Unified CME and Cisco Unified Communications Manager in Same Network: Example 555
H.450 Tandem Gateway Working with Cisco Unified CME and
Cisco Unified Communications Manager: Example 558
Forwarding Calls to Cisco Unity Express: Example 559
Where to Go Next 559

Additional References 560

Related Documents 560
Technical Assistance 560
Feature Information for Call Transfer and Forwarding 561

Configuring Call-Coverage Features 563

Contents 563

Information About Call Coverage Features 563

Call-Coverage Summary 564
Call Hunt 565

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Call Pickup 566

Call Waiting 568
Callback Busy Subscriber 569
Hunt Groups 569
Night Service 579
Overlaid Ephone-dns 581
How to Configure Call Coverage Features 585
SCCP: Configuring Call Hunt 586
SCCP: Verifying Call Hunt 587
SIP: Configuring Call Hunt 589
SCCP: Enabling Local-Group Call Pickup at a System-Level 590
SCCP: Creating Pickup Groups 591
SCCP: Verifying Call Pickup 592
SCCP: Configuring Call-Waiting Indicator Tone 593
SCCP: Verifying Call-Waiting Indicator Tone 594
SIP: Enabling Call Waiting 595
SCCP: Configuring Hunt Groups 596
SCCP: Verifying Hunt Groups 603
SIP: Configuring Hunt Groups 605
SCCP: Configuring Night Service 609
SCCP: Verifying Night Service 613
SCCP: Configuring Overlaid Ephone-dns 615
SCCP: Verifying Overlaid Ephone-dns 618
Configuration Examples for Call Coverage Features 619
Call Hunt: Examples 619
Call Pickup: Examples 621
Call-Waiting Beep: Example 622
Call-Waiting Ring: Example 622
Hunt Group: Examples 622
Night Service: Examples 625
Overlaid Ephone-dns Examples 626
Where to Go Next 633

Additional References 634

Related Documents 634
Technical Assistance 635
Feature Information for Call Coverage Features 636

Configuring Caller ID Blocking 639

Contents 639

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Restrictions for Caller ID Blocking 639

Information about Caller ID Blocking 639

Caller ID Blocking on Outbound Calls 640
How to Configure Caller ID Blocking 640
Blocking Caller ID For Local Calls From a Directory Number 640
Blocking Caller ID For All Outbound Calls 642
Verifying Caller ID Blocking 643
Configuration Examples for Caller ID Blocking 644
Caller ID Blocking Code: Example 644
Caller ID Blocking for Outbound Calls: Example 644
Additional References 644
Related Documents 644
Technical Assistance 645
Feature Information for Caller ID Blocking 646

Configuring Conferencing 647

Contents 647

Restrictions for Conferencing 647

Information About Conferencing 648

Conferencing Overview 648
Secure Conferencing Limitation 648
Ad Hoc Conferencing 648
Meet-Me Conferencing 650
How to Configure Conferencing 651
Modifying the Default Configuration for Three-Party Ad Hoc Conferencing 651
SCCP: Configuring Conferencing Options on a Phone 653
SIP: Configuring Conferencing Options on a Phone 655
Verifying Three-Party Ad Hoc Conferencing 656
SCCP: Configuring Multi-Party Ad Hoc and Meet-Me Conferencing 657
SCCP: Verifying Multi-Party Ad Hoc and Meet-Me Conferencing 670
Configuration Examples for Conferencing 671
Basic Conferencing: Example 671
End of Conference Options: Example 671
DSP Farm and Cisco Unified CME on the Same Router: Example 672
DSP Farm and Cisco Unified CME on Different Routers: Example 677
Where to Go Next 686

Additional References 686

Related Documents 686
Technical Assistance 686

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Feature Information for Conferencing 687

Configuring Directory Services 689

Contents 689

Information About Directory Services 689

Local Directory 690
External Directory 690
Called-Name Display 690
How to Configure Directory Services 691
Configuring Local Directory Service 691
SCCP: Defining a Name for a Directory Number 692
SCCP: Adding an Entry to a Local Directory 693
SCCP: Configuring External Directory Service 694
SCCP: Enabling Called-Name Display 696
Verifying Called-Name Display 697
SIP: Defining a Name for a Directory Number 698
SIP: Configuring External Directory Service 699
Verifying Directory Services 700
Configuration Examples for Directory Services 701
Local Directory 701
Called-Name Display 702
Additional References 706
Related Documents 706
Technical Assistance 706
Feature Information for Directory Services 707

Configuring Do Not Disturb 709

Contents 709

Information About Do Not Disturb 709

Do Not Disturb 709
How to Configure Do Not Disturb 710
SCCP: Blocking Do Not Disturb 710
SCCP: Verifying Do Not Disturb 711
SIP: Configuring Do Not Disturb 712
Configuration Examples for Do Not Disturb 714
Where to Go Next 714

Additional References 715

Related Documents 715
Technical Assistance 715

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Feature Information for Do Not Disturb 716

Configuring Extension Mobility 717

Contents 717

Information About Extension Mobility 717

Extension Mobility 717
How to Enable Extension Mobility 718
Prerequisites 718
Restrictions 718
Configuring a Logout Profile for an IP Phone 719
Enabling an IP Phone for Extension Mobility 721
Configuring a User Profile 723
Configuration Examples for Extension Mobility 725
Logout Profile: Example 725
Enabling an IP Phone for Extension Mobility: Example 725
Voice-User Profile: Example 726
Where to Go Next 726

Additional References 727

Related Documents 727
Technical Assistance 727
Feature Information for Extension Mobility 728

Configuring Feature Access Codes 729

Contents 729

Information About Feature Access Codes 729

Feature Access Codes 730
How to Configure Feature Access Codes 731
Enabling Feature Access Codes 731
Verifying Feature Access Codes 732
Configuration Examples for Feature Access Codes 733
FAC: Example 733
Additional References 734
Related Documents 734
Technical Assistance 734
Feature Information for Feature Access Codes 735

Configuring Fax Relay 737

Contents 737

Prerequisites for Fax Relay 737

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Restrictions for Fax Relay 738

Information About Fax Relay 738

Fax Relay and Equipment 738
Feature Design of Cisco Fax Relay 739

How to Configure Fax Relay 740

Configuring Fax Relay 740
Verifying and Troubleshooting Fax Relay Configuration 742

Configuration Examples for Fax Relay 742

Fax Relay: Example 742
Additional References 743
Related Documents 743
Technical Assistance 743
Feature Information for Fax Relay 744

Configuring Headset Auto-Answer 745

Contents 745

Information About Headset Auto-Answer 745

Auto-Answering Calls Using a Headset 746
Difference Between a Line and a Button 746
How to Configure Headset Auto-Answer 748
Enabling Headset Auto-Answer 748
Verifying Headset Auto-Answer 749
Configuration Examples for Headset Auto-answer 749

Additional References 750

Related Documents 750
Technical Assistance 750
Feature Information for Headset Auto-Answer 751

Configuring Intercom Lines 753

Contents 753

Information About Intercom Lines 753

Intercom Auto-Answer Lines 753
How to Configure Intercom Lines 755
SCCP: Configuring an Intercom Auto-Answer Line 755
SIP: Configuring an Intercom Auto-Answer Line 757
Configuration Examples for Intercom Lines 759
Intercom Lines: Example 759
Where to Go Next 759

Additional References 760

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Related Documents 760

Technical Assistance 760
Feature Information for Intercom Lines 761

Configuring Loopback Call Routing 763

Contents 763

Information About Loopback Call Routing 763

Loopback Call Routing 763
How to Configure Loopback Call Routing 764
Enabling Loopback Call Routing 764
Verifying Loopback Call Routing 768
Configuration Examples for Loopback Call Routing 768
Enabling Loopback Call Routing: Example 768
Additional References 769
Related Documents 769
Technical Assistance 769
Feature Information for Loopback Call Routing 770

Configuring Music on Hold 771

Contents 771

Prerequisites for Music on Hold 771

Restrictions for Music on Hold 771

Information About Music on Hold 772

Music on Hold 772
How to Configure Music on Hold 774
Configuring Music on Hold from an Audio File 774
Configuring Music on Hold from a Live Feed 776
Verifying Music on Hold 780
Configuration Examples for Music on Hold 781
MOH from an Audio File: Example 781
MOH from a Live Feed: Example 781
Additional References 782
Related Documents 782
Technical Assistance 782
Feature Information for Music on Hold 783

Configuring Paging 785

Contents 785

Information About Paging 785

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Audio Paging 785

How to Configure Paging 787

Configuring a Simple Paging Group 787
Configuring a Combined Paging Group 788
Verifying Paging 791
Configuration Examples for Paging 791

Where to Go Next 794

Additional References 794

Related Documents 794
Technical Assistance 794
Feature Information for Paging 795

Configuring Presence Service 797

Contents 797

Prerequisites for Presence Service 797

Restrictions for Presence Service 798

Information About Presence Service 798

Presence Service 798
How to Configure Presence Service 799
Enabling Presence for Internal Lines 800
Enabling a Directory Number to be Watched 801
Enabling a SCCP Phone to Monitor BLF Status for Speed-Dials and Call Lists 803
Enabling a SIP Phone to Monitor BLF Status for Speed-Dials and Call Lists 806
Configuring Presence to Watch External Lines 808
Verifying Presence Configuration 810
Troubleshooting Presence 811
Configuration Examples for Presence 813
Presence in Cisco Unified CME: Example 813

Additional References 817

Related Documents 817
Technical Assistance 817
Feature Information for Presence Service 818

Configuring Ring Tones 819

Contents 819

Information About Ring Tones 819

Distinctive Ringing 820
Customized Ring Tones 820
On-Hold Indicator 820

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How to Configure Ring Tones 821

SCCP: Enabling Distinctive Ringing 821
SCCP: Enabling Customized Ring Tones 822
SCCP: Enabling On-Hold Indicator 824
SIP: Enabling Distinctive Ringing 825
Configuration Examples for Ring Tones 826
Distinctive Ringing for Internal Calls: Example 826
On-Hold Indicator: Example 826
Additional References 827
Related Documents 827
Technical Assistance 827
Feature Information for Ring Tones 828

Customizing Soft Keys 829

Contents 829

Information About Soft Keys 829

Soft Keys on IP Phones 830
Account Code Entry 831
Hookflash Soft Key 831
Feature Blocking 832
How to Customize Soft Keys 832
SCCP: Modifying Soft-Key Display 832
SIP: Modifying Soft-Key Display 835
Verifying Soft-Key Configuration 837
Enabling Flash Soft Key 838
Verifying Flash Soft-Key Configuration 839
Configuring Feature Blocking 840
Verifying Feature Blocking 842
Configuration Examples for Soft-Keys 842
Modifying Soft-Key Display: Example 842
Modifying the HLog Soft Key for Ephone Hunt Groups: Example 843
Enabling Flash Soft Key for PSTN Calls: Example 843
Park and Transfer Blocking: Example 843
Conference Blocking: Example 843
Where to Go Next 844
Additional References 844
Related Documents 844
Technical Assistance 844
Feature Information for Soft Keys 845

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Configuring Speed Dial 847

Contents 847

Information About Speed Dial 847

Speed Dial Summary 848
Speed Dial Buttons and Abbreviated Dialing 849
Bulk-Loading Speed Dial Numbers 849
Monitor-Line Button for Speed Dial 850
DSS (Direct Station Select) Service 851
How to Configure Speed Dial 851
Enabling a Local Speed Dial Menu 852
SCCP: Enabling DSS Service 854
SCCP: Enabling a Personal Speed Dial Menu 855
SCCP: Defining Speed-Dial Buttons and Abbreviated Dialing 856
SCCP: Enabling Bulk-Loading Speed-Dial 858
SCCP: Verifying Bulk Speed-Dial Parameters 860
SIP: Defining Speed-Dial Buttons 860
SIP: Configuring a Personal Speed-Dial Menu 861
Configuration Examples for Speed Dial 863
Enabling a Local Speed Dial Menu: Example 863
Personal Speed Dial Menu: Example 863
Speed-Dial Buttons and Abbreviated Dialing: Example 864
Bulk-Loading Speed Dial: Example 864
Where to Go Next 864

Additional References 865

Related Documents 865
Technical Assistance 865
Feature Information for Speed Dial 866

Configuring Video Support for SCCP-Based Endpoints 867

Contents 867

Prerequisites for Video Support for SCCP-Based Endpoints 867

Restrictions for Video Support for SCCP-Based Endpoints 868

Information About Video Support for SCCP-Based Endpoints 869

Video Support Overview 870
Matching Endpoint Capabilities 870
Retrieving Video Codec Information 870
Call Fallback to Audio-Only 870
Call Setup for Video Endpoints 871
Flow of the RTP Video Stream 871

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How to Configure Video for SCCP-Based Endpoints 872

Enabling Slow Connect Procedures 872
Enabling Video Capabilities Globally 873
Enabling Video Capabilities on a Specific Phone 874
Setting Video Parameters 876
Verifying Video Support for SCCP-Based Endpoints 877
Troubleshooting Video Support for SCCP-Based Endpoints 878

Additional References 878

Related Documents 878
Technical Assistance 879
Feature Information for Video Support for SCCP-Based Endpoints 880

Creating Templates 881

Contents 881

Information About Templates 881

Phone Templates 882
Ephone-dn Templates 882
How to Configure Templates 882
SCCP: Enabling Ephone Templates 883
SCCP: Enabling Ephone-dn Templates 884
SCCP: Verifying Templates 885
SIP: Creating and Applying Templates to SIP Phones 886

Configuration Examples for Creating Templates 888

Using Ephone Template to Block The Use of Park and Transfer Soft Keys 888
Using Ephone-dn Template to Set Call Forwarding 889
Where to Go Next 889

Additional References 889

Related Documents 889
Technical Assistance 890
Feature Information for Creating Templates 891

Modifying Cisco Unified IP Phone Options 893

Contents 893

Information About Cisco Unified IP Phone Options 893

Customized Background Images for Cisco Unified IP Phone 7970 894
Fixed Line/Feature Buttons for Cisco Unified IP Phone 7931G 894
Header Bar Display 894
Phone Labels 895
Programmable Vendor Parameters for Phones 895

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System Message Display 895

URL Provisioning for Feature Buttons 896

How to Configure Cisco Unified IP Phone Options 896

SCCP: Selecting Button Layout for a Cisco Unified IP Phone 7931G 897
SCCP: Modifying Header Bar Display 898
SIP: Modifying Header Bar Display 899
Verifying Header Bar Display 901
Troubleshooting Header Bar Display 901
SCCP: Creating Labels for Directory Numbers 901
SIP: Creating Labels for Directory Numbers 903
Verifying Labels 904
SCCP: Modifying System Message Display 904
Verifying System Message Display 906
Troubleshooting System Message Display 906
SCCP: Provisioning URLs for Feature Buttons 907
SIP: Provisioning URLs for Feature Buttons 908
Troubleshooting URL Provisioning for Feature Buttons 909
SCCP: Modifying Vendor Parameters for All Phones 910
SCCP: Modifying Vendor Parameters For a Specific Phone 911
Troubleshooting Vendor Parameter Configuration 913
Configuration Examples for Cisco Unified IP Phone Options 913
Text Labels for Ephone-dns: Example 914
Phone Header Bar Display: Example 914
System Text Message Display: Example 914
System File Display: Example 914
URL Provisioning for Directories, Services, and Messages Buttons: Example 914
Programmable VendorConfig Parameters: Example 915
Additional References 915
Related Documents 915
Technical Assistance 916
Feature Information for Cisco Unified IP Phone Options 917

Configuring Interoperability with External Services 919

Contents 919

Information About Interoperability with External Services 919

Interoperability with Unified CCX 920
How to Configure Interoperability with External Services 921
Configuring Cisco Unified CME to Enable Interoperability with Unified CCX 922
Identifying Agent Directory Numbers in Cisco Unified CME for Session Manager 924

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Verifying Registrations and Subscriptions in Cisco Unified CME 926

Re-creating a Session Manager in Cisco Unified CME 926
Reconfiguring a Cisco CRS Route Point as a SIP Endpoint in Cisco Unified CME 928

Configuration Examples for Interoperability with Unified CCX 930

Where to Go Next 939

Additional References 940

Related Documents 940
Technical Assistance 940
Feature Information for Interoperability with External Services 941

Configuring SRST Fallback Support 943

Contents 943

Prerequisites for SRST Fallback Support 943

Restrictions for SRST Fallback Support 944

Information About SRST Fallback Support 944

SRST Fallback Support Using Cisco Unified CME 944
Prebuilding Cisco Unified CME Phone Configurations 948
How to Configure SRST Fallback Support 948
Enabling SRST Mode 948
Verifying SRST Mode 950
Prebuilding Cisco Unified CME Phone Configurations 951
Modifying Call Pickup for Fallback Support 951
Configuration Examples for SRST Fallback Support 953
Enabling SRST Mode: Example 953
Provisioning Directory Numbers for Fallback Support: Example 954
Configuring Templates for Fallback Support: Example 954
Enabling Hunt Groups for Fallback Support: Example 955
Modifying Call Pickup for Fallback Support: Example 955
Additional References 955
Related Documents 955
Technical Assistance 956
Feature Information for SRST Fallback Support 957

Configuring the XML API 959

Contents 959
Information About XML API 959
XML API Definition 959
XML API Provision Using IXI 960
How to Configure XML API 960

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Defining XML Transport Parameters 960

Defining XML Application Parameters 962
Defining Authentication for XML Access 963
Defining XML Event Table Parameters 964
Troubleshooting the XML Interface 965
Configuration Examples for XML API 965
XML Transport Parameters: Example 965
XML Application Parameters: Example 965
XML Authentication: Example 966
XML Event Table: Example 966
Where to Go Next 966

Additional References 966

Related Documents 966
Technical Assistance 967
Feature Information for XML API 968

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 F E AT U R E M A P

A C

Abbreviated Dialing Speed Dial 856 Call Blocking Based on Date and Time 467
Account Code Entry 831 Call Blocking Override 467
Ad Hoc Conferencing 647 Call Forwarding 500
Adding Directory Entries 689 Call Forwarding Support 499
After-Hours Call Blocking 467 Call Hold 819
After-Hours Toll Bar 467 Call Hunt 565
Agent Availability, Hunt Groups 575 Call Park 485
Analog Phones 147 Call Pickup 566
API, XML 959 Call Transfer 499
ATA(Cisco Analog Telephone Adapters) 147 Call Transfer Blocking 503
Audio Paging 785 Call Transfer Support 499
Authentication, Phone 387 Call Waiting 568
Auto-Answer, Headset 745 Call Waiting for Overlaid Ephone-dns 581
Automatic Agent Status Not-Ready, Ephone Hunt Callback Busy Subscriber 569
Groups 575 Called-Name Display 689
Automatic Line Selection 461 Caller ID Blocking 639
Auto-Registration Blocking 129 Call-Park Blocking 489
Call-Park Redirect 489
Call-Waiting Beep 568
B Call-Waiting Ring 569
Channel Huntstop 565
Backup Router 119 Cisco IP Communicator 147
BLF notification 797 Cisco VG 224 147
Blocking Call Transfer 503 Conference Gain Control 649
Blocking Caller ID 639 Conference Initiator Drop-Off Control 647
Blocking Call-Park 489 Conferencing 647
Blocking Calls Based on Date and Time 467 Configuration Files 120, 245
Blocking Features 467, 829 Customized Background Images 893
Blocking Local Directory 689 Customizing Feature Buttons 893
Blocking, Automatic Registration 129
Bulk-Loading Speed-Dial Numbers 858
Busy Timeout 119 D

Dedicated Call-Park Slots 488

Dedicated FXO Trunk Lines 147
DHCP Setup 69
Dial Plans, SIP 147

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 Index

Dial Tone, Secondary 267 Forwarding Support 499

Dial-Peer Preference 565 FXO Hookflash 829
Directed Call Park 486 FXO Lines, Dedicated 147
Directories 689 FXS Ports 147
Directory Disable 689
Display
Called-Name 689 G
Phone Header Bar 893
Phone System Message 893 Group Call Pickup 566
Distinctive Ringing 819
Do Not Disturb (DND) 709
DSP Farms 306 H
DSS (Direct Station Select) 502, 851
DTMF Integration Patterns for Voice Mail 367 Hairpin Call Routing 507
DTMF Relay for H.323 Networks 105 Hardware-Based Conferencing 647
DTMF Relay for SIP Trunks 106 Header Bar Display 893
Dynamic Membership, Ephone Hunt Groups 575 Headset Auto-Answer 745
Hold 819
Hold Notification 820
E Hookflash, FXO 829
Hunt Groups 569
End-of-Conference Options 649 Huntstop 565
Ephone Hunt Group Agent Availability Options Huntstop, Channel 565
575
Ephone Hunt Groups 569
Ephone-dn Overview 147 I
Ephone-dn, Overlaid 581
Extension Assigner 213 Intercom Lines 753
Extension Mobility 717 Interdigit Timeout 119
Externally Stored Configuration Files 120, 245 International Languages and Tones 287
Interoperability with Unified CCX 919

F
J
FAC (feature access code) 729
fallback support for Cisco Unified Communica- Join Ephone Hunt Groups 575
tions Manager 943
Fax Relay 737
Feature Blocking 829 K
Feature Buttons,URL Provisioning 893
Keep-Conference Options 649
Feature Control 829
Key System 50, 176
Feature Ring 709
Keyswitch 50, 176
Files, Configuration 245
KPML 147
Fixed Line/Feature Button Set 893
Flash Soft Key 829
Forwarding 499

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 Index

L P

Labels, Phone 893 Paging 785

Languages and Tones 287 Parallel Voice Hunt Groups 574
Leave Ephone Hunt Groups 575 Park 485
Line Selection for MWI 355 PBX System 50, 158
Line Selection, Automatic 461 Peer Ephone Hunt Groups 572
Local Directory 689 Per-Phone Configuration Files 120, 246
Local Speed Dial 852 Personal Speed Dial 855
Locales, Multiple 288 Phone Authentication 387
Locales, Network and User 287 Phone Display
Longest-Idle Ephone Hunt Groups 573 Called-Name 689
Loopback Call Routing 763 Header Bar 893
System Message 893
Phone Feature Buttons, Customizing 893
M Phone Header Bar Display 893
Phone Labels 893
Mailbox Selection Policy 362 Phone Screen, Customized Background Images
Media Encryption 387 893
Meet-Me Conferencing 647 Phone Soft Keys 829
Monitor Mode 147 Phones
Monitor-Line Speed Dial 850 Analog 147
MTP (Media Termination Point) 156 Remote Teleworker 155
Multi-party Conferencing Enhancements 647 Pickup Groups 566
Music on Hold (MOH) 771 Preference, Ephone-dn Dial Peer 565
MWI Line Selection 355 Presence 797
MWI Prefix Specification for SIP Voice Mail 380 Programmable Vendor Parameters 893

N Q
Network Locales 287 QSIG Supplementary Services 499
Network Time Protocol 91, 119, 139
Night Service 579
R

O Redundant Router 119

Remote Teleworker Phones 155
On-hold Notification 820 Resetting Phones 257
Out-of-Dialog REFER 91 Restarting Phones 257
Overlaid Ephone-dns 581 Ringing Timeout 119
Overlaid Ephone-dns Rollover Buttons 585 Ringing, Distinctive 819
Rollover Buttons for Overlaid Ephone-dns 585

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 Index

S U

Secondary Dial Tone 267 URL Provisioning for Feature Buttons 893
Secondary Router 119 User-Defined Locales 287
Security 387
Selective Call Forwarding 500
Sequential Ephone Hunt Groups 571 V
Session Transport Protocol 147
Shared Lines 152 Vendor Configuration Parameters 895
Shared-line Overlay Ephone-dns 581 Video Support 867
Silent Ring 563 Voice Hunt Groups 569
SIP Dial Plans 147 Voice Mail Integration 355
SIP Trunks 91 Voice Translation Rules and Profiles 267
Soft Keys 829
Software-based Conferencing 647
Speed Dial 847 X
Abbreviated Dialing 856
Bulk Loading 858 XML Application Programming Interface 959
Local Speed Dial 852 XML Configuration Files 69
Monitor-Line Button 850
Personal Speed Dial 855
Speed-Dial Buttons 856
SRST Fallback Support Using Cisco Unified CME
943
SRTP 387
System Message Display 893
System-Defined Locales 287

Tandem Gateway 510

Teleworker Remote Phones 155
Timeouts 119
Toll Bar and Toll Bar Override 467
Tone, Secondary Dial 119
Transcoding Support 303
Transfer 499
Translation Rules and Profiles 267
Trunks, SIP 91

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 Cisco Unified CME Features Roadmap

Last Updated: June 18, 2007

This roadmap lists the features documented in the Cisco Unified Communications Manager Express
System Administrator Guide and maps them to the modules in which they appear.

Feature and Release Support

Table 1 lists feature support for Cisco Unified CME versions. Only features that were introduced or
modified in Cisco Unified CME .4.0 or a later version appear in the table. Not all features may be
supported in your Cisco Unified CME software version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 1 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 1 Supported Cisco Unified CME Features

Release Feature Name Feature Description Where Documented

Cisco Unified CME 4.2
4.2 Extension Mobility Provides the benefit of phone mobility for end users by Configuring Extension
enabling the user to log into any local Cisco Unified IP phone Mobility
that is enabled for extension mobility.
Interoperability with Enables interoperability between Cisco Unified CME and Configuring
Cisco Unified Contact Cisco Customer Response Solutions (CRS) 5.0 and later Interoperability with
Center Express (Cisco versions with Cisco Unified Contact Center Express External Services
UCCX) (Unified CCX), including Cisco Unified IP IVR, enhanced
call processing, device and call monitoring, and unattended
call transfers to multiple call center agents and basic
extension mobility.

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 Cisco Unified CME Features Roadmap

Table 1 Supported Cisco Unified CME Features (continued)

Release Feature Name Feature Description Where Documented

Media Encryption Provides the following secure voice call capabilities: • Configuring
(SRTP) on Cisco Unified • Secure call control signaling and media streams in Security
Communications Cisco Unified CME networks using Secure Real-Time • SCCP Controlled
Manager Express Transport Protocol (SRTP) and H.323 protocols. Analog (FXS) Ports
with Supplementary
• Secure supplementary services for Cisco Unified CME
Features in Cisco
networks using H.323 trunks.
IOS Gateways
• Secure Cisco VG224 Analog Phone Gateway endpoints.
Cisco Unified CME 4.1
4.1 Call Forward All When a user enables Call Forward All on a SIP phone using Configuring Call
Synchronization the CfwdAll soft key, the uniform resource identifier (URI) Transfer and Forwarding
for the service is sent to Cisco Unified CME. When Call
Forward All is configured in Cisco Unified CME, the
configuration is sent to the SIP phone which updates the
CfwdAll soft key to indicate that Call forward All is enabled.
Cisco Unified IP Phones SCCP support was added for the following phone: Cisco Unified
Communications
• Cisco Unified IP Phone 7921G
Manager Express 4.1
SIP support was added for the following phones: Supported Firmware,
• Cisco Unified IP Phone 3951 Platforms, Memory, and
Voice Products
• Cisco Unified IP Phone 7911G
• Cisco Unified IP Phone 7941G and 7941G-GE
• Cisco Unified IP Phone 7961G and 7961G-GE
• Cisco Unified IP Phone 7970G and 7971G-GE
No additional configuration is required for these phones.
They are supported in the appropriate Cisco IOS commands.
Directory Services Local directory and local speed dial features are supported for Configuring Directory
SIP phones. Services
Disabling SIP You can disable REFER messages for call transfers and Configuring Call
Supplementary Services redirect responses for call forwarding from being sent by Transfer and Forwarding
for Call Forward and Cisco Unified CME if a destination gateway does not support
Call Transfer supplementary services.
Disabling supplementary services is supported if all
endpoints use SCCP or all endpoints use SIP.
KPML Key Press Markup Language (KPML) reports SIP phone Configuring Phones to
users input digit by digit to Cisco Unified CME, which Make Basic Calls
performs pattern recognition by matching a destination
pattern to a dial peer as it collects the dialed digits.
Multi-Party • Enhanced ad-hoc conferences are hardware-based and Configuring
Conferencing allow more than three parties. Conferencing
Enhancements
• Meet-me conferences consist of at least three parties
dialing a meet-me conference number.

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 Cisco Unified CME Features Roadmap

Table 1 Supported Cisco Unified CME Features (continued)

Release Feature Name Feature Description Where Documented

Network Time Protocol SIP phones registered to a Cisco Unified CME router can Defining Network
synchronize to a Network Time Protocol (NTP) server, Parameters
known as the clock master.
Out-of-Dialog REFER Out-of-dialog REFER (OOD-R) allows remote applications Defining Network
to establish calls by sending a REFER message to Parameters
Cisco Unified CME without an initial INVITE. After the
REFER is sent, the remainder of the call setup is independent
of the application and the media stream does not flow through
the application.
Presence with BLF Presence supports BLF notification features for speed-dial Configuring Presence
Status buttons and directory call lists for missed calls, placed calls, Service
and received calls. SIP and SCCP phones that support the
BLF speed-dial and BLF call-list features can subscribe to
status change notification for internal and external directory
numbers.
Restarting Phones SIP phones can be quickly reset by using the restart Resetting and Restarting
command. Phones contact the TFTP server for updated Phones
configuration information and reregister without contacting
the DHCP server.
Session Transport TCP can be used as the transport protocol for supported SIP Configuring Phones to
phones connected to Cisco Unified CME. Previously only Make Basic Calls
UDP was supported.
SIP Dial Plans Dial plans enable SIP phones to perform local digit collection Configuring Phones to
and recognize dial patterns as user input is collected. After a Make Basic Calls
pattern is recognized, the SIP phone sends an INVITE
message to Cisco Unified CME to initiate the call.
Soft Keys You can customize the display and order of soft keys that Customizing Soft Keys
appear on individual SIP phones during the connected, hold,
idle, and seized call states.
Translation Rules SIP phones in a Cisco Unified CME system support Configuring Dialing
translation rules with functionality similar to phones running Plans
SCCP. Translation rules can be applied to incoming calls for
directory numbers on a SIP phone.
Cisco Unified CME 4.0(3)
4.0(3) AMWI Cisco Unified IP Phone 7911 and Cisco Unified IP Phone Integrating Voice Mail
7931G can be configured to receive AMWI (Audible Message
Line Indicator) and visual MWI notification from an external
voice-messaging system.
Cisco Unified IP Phones Support was added for the following phones: Cisco Unified
Communications
• Cisco Unified IP Phone 7906G
Manager Express 4.0(3)
• Cisco IP Unified IP Phone 7931G Supported Firmware,
Platforms, Memory, and
Voice Products

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 Cisco Unified CME Features Roadmap

Table 1 Supported Cisco Unified CME Features (continued)

Release Feature Name Feature Description Where Documented

DSS DSS (Direct Station Select) feature allows the phone user to Configuring Speed Dial
press a single speed-dial line button to transfer an incoming
call when the call is in the connected state. This feature is
supported on all phones on which monitor line buttons for
speed dial or speed-dial line buttons are configured.
Extension Assigner Allows installation technicians to assign extension numbers Creating Phone
to phones without administrative access to Configurations Using
Cisco Unified CME, typically during the installation of new Extension Assigner
phones or the replacement of broken phones.
Fax Relay SCCP-enhanced features add support for Cisco Fax Relay and Configuring Fax Relay
Super Group 3 (SG3) to G3 fax relay. This feature allows the
fax stream between two SG3 fax machines to negotiate down
to G3 speeds (less than 14.4 kbps) allowing SG3 fax machines
to interoperate over fax relay with G3 fax machines.
Cisco Unified CME 4.0(1)
4.0(1) Call Forwarding Automatic call forwarding during night Configuring Call
service—Ephone-dns (extensions) can be designated to Transfer and Forwarding
automatically forward their calls to a specified number during
the time that night service is in effect.
Blocking call forwarding of local calls—Forwarding of
local (internal) calls from other Cisco Unified CME ephones
can be blocked. External calls will continue to be forwarded
as specified by the configuration for the ephone-dns.
Selective call forwarding—Call forwarding for busy and
no-answer ephone-dns can be applied selectively based on the
number that a caller dials for a particular ephone-dn: the
primary number, the secondary number, or either of those
numbers expanded through the use of a dial-plan pattern.
Call Park Call park blocked per ephone—Individual ephones can be Configuring Call Park
blocked from parking calls at call-park slots. If a blocked
ephone has a dedicated park slot, it can park calls at the
dedicated park slot, but not at any other park slot.
Call park redirect—You can specify that calls use the H.450
or SIP Refer method of call forwarding or transfer to park
calls and to pick up calls from park. The default is that hairpin
call forwarding or transfer is used to park calls and to pick up
calls from park.
Dedicated call-park slots—A private call-park slot can be
configured for each ephone. Optional parameters include
timeout intervals, after which the parked call can be
automatically recalled to the parking phone or transferred to
another number.
Direct pickup of parked call on monitored park slot —A
call that is parked on a monitored call-park slot can be picked
up by pressing the assigned monitor button.

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Table 1 Supported Cisco Unified CME Features (continued)

Release Feature Name Feature Description Where Documented

Call Pickup Directed call pickup disable—The no service Configuring
directed-pickup command globally disables directed call Call-Coverage Features
pickup and changes the action of the PickUp soft key to
invoke local group pickup rather than directed call pickup.
Call Transfer Call transfer blocking—When call transfers to phones Configuring Call
outside the Cisco Unified CME system have been globally Transfer and Forwarding
enabled, you can block them for individual ephones.
Call transfer destination digits limited—When call
transfers to phones outside the Cisco Unified CME system
have been globally enabled, you can limit the number of digits
that can be dialed when transferring a call.
transfer-system command—The command default has been
changed from the blind keyword to the full-consult keyword,
making H.450.2 consultative transfer the default method.
QSIG supplementary services support—H.450
supplementary services features allow Cisco Unified CME
phones to use QSIG to interwork with PBX phones. IP phones
can use a PBX message center with proper MWI notifications.
Cisco Unified IP Phones Support was added for the following phones: Cisco Unified
• Cisco Unified IP Phone 7911G Communications
Manager Express 4.0
• Cisco Unified IP Phone 7941G and 7941G-GE Supported Firmware,
• Cisco Unified IP Phone 7961G and 7961G-GE Platforms, Memory, and
Voice Products
No additional configuration is required for these phones.
They are supported in the appropriate Cisco IOS commands.
Conferencing Drop last party or keep parties connected—New options Configuring
specify whether the last party that joined a conference can be Conferencing
dropped from the conference and whether the remaining two
parties should be allowed to continue their connection after
the conference initiator has left the conference.
Improved conference display—A Cisco Unified IP phone
that is connected to a three-way conference displays
“Conference.” No special configuration is required.
Feature Access Codes Feature Access Code (FAC) support—The same FACs that Configuring Feature
are used by analog phones can be enabled for IP phones. In Access Codes
addition, standard FACs can be customized and aliases can be
created to simplify the dialing of a FAC and any additional
digits that are required to activate the feature.
Headset Auto-Answer Headset auto-answer—When the headset key on a phone is Configuring Headset
activated, lines on the phone that are specified for headset Auto-Answer
auto-answer will automatically connect to incoming calls
after playing an alerting tone to notify the phone user of the
incoming call. This feature is available on Cisco Unified IP
Phones 7940G, 7960G, 7970G, and 7971G-GE.

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Table 1 Supported Cisco Unified CME Features (continued)

Release Feature Name Feature Description Where Documented

Hunt Groups Agent status control—Hunt group agents can put their Configuring
phones in a not-ready state to temporarily suspend the Call-Coverage Features
receiving of hunt group calls by using the HLog soft key. A
new FAC can toggle ready and not-ready state.
Automatic agent not-ready status—The criterion for
placing a hunt group agent into not-ready status (previously
called automatic logout) was changed. If an agent does not
answer the number of consecutive hunt-group calls that you
specify in the auto logout command, the agent’s ephone-dn
is put into not-ready status (logged out) and will not receive
further hunt group calls.
Call hold statistics—New fields describing the length of time
that calls spend in the hold state are in the statistical reports
for Cisco Unified CME B-ACD applications. See the
show ephone-hunt statistics command and the hunt-group
report url command in Cisco Unified CME B-ACD and Tcl
Call-Handling Applications.
Dynamic hunt group membership—Agents can join or
leave a hunt group using standard or custom FACs when
wildcard slots are configured for hunt groups and the agents’
ephone-dns are authorized to join hunt groups. An agent
joining a hunt group uses a wildcard slot, and an agent leaving
a group relinquishes the slot so that another agent can use it.
Change in hops command default—The maximum number of
hops allowed by a hunt group is automatically adjusted to
reflect the dynamically changing number of members. No
special configuration is required.
Enhanced display of ephone hunt-group information—A
text string can be added to provide information in
configuration output and to display on IP phones when a
hunt-group call is ringing or answered. This text string can be
used to indicate the name or purpose of the hunt group.
A text string can be displayed on IP phones when all
hunt-group members are logged out. This text string can be
used to indicate where calls are being sent at that time; for
example, to night service or voice mail.
Local call forwarding restriction in sequential ephone
hunt groups—In sequential ephone-hunt groups, local
(internal) calls to the hunt group can be prevented from being
forwarded beyond the first ephone-dn in the hunt group.
Longest-idle hunt group improvement—A new command,
the from-ring command, specifies that on-hook time stamps
should be updated when a call rings an agent as well as when
a call is answered by an agent.

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Table 1 Supported Cisco Unified CME Features (continued)

Release Feature Name Feature Description Where Documented

Hunt Groups Maximum number of agents per hunt group has increased Configuring
from 10 to 20. No special configuration is required. Call-Coverage Features
Maximum number of hunt groups per Cisco Unified CME
system has increased from 10 to 100. No special configuration
is required.
No-answer timeout enhancements—No-answer timeouts in
ephone hunt groups can be set individually for each
ephone-dn in the list. A maximum cumulative no-answer
timeout can be also be set.
Restricting presentation of calls to idle or on-hook
phones—The presentation of hunt group calls can be
restricted to hunt-group members on phones that are idle or
on-hook. This enhancement considers all lines on the phone,
both members of the hunt group and nonmembers, when
restricting presentation of hunt group calls.
Return to a secondary destination in an ephone hunt
group after call park—Calls parked by hunt group agents
can be returned to a different entry point in the hunt group.
Return to transferring party on no answer in an ephone
hunt group—A call that was transferred into a hunt group
and was not answered can be returned to the party that
transferred it to the hunt group instead of being sent to voice
mail or another final destination.
Localization Multiple user locales and network locales—Up to five user Configuring
and network locales are supported. Localization Support
User-defined user locales and network locales—
User-defined locales can be added for supported phones.
Music on Hold Music on hold (MOH) for internal calls—Internal callers Configuring Music on
(those making calls between extensions in the same Hold
Cisco Unified CME system) hear music when they are on
hold or are being transferred. The mulitcast moh command
must be used to enable the flow of packets to the subnet on
which the phones are located.
Internal extensions that are connected through an analog
voice gateway or through a WAN (remote extensions) do not
hear MOH on internal calls.
The ability to disable multicast MOH per phone was
introduced, using the no multicast-moh command in ephone
or ephone-template configuration mode.

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Table 1 Supported Cisco Unified CME Features (continued)

Release Feature Name Feature Description Where Documented

Overlaid Ephone-dns Overlaid ephone-dns—The maximum number of overlaid Configuring
ephone-dns per ephone button has increased from 10 to 25. Call-Coverage Features
No special configuration is required.
Overlaid ephone-dn call-waiting display—The number of
waiting calls that can be displayed for overlaid ephone-dns
that have call waiting configured has been increased to six for
the Cisco IP Phone 7940G, 7941G, 7941G-GE, 7960G,
7961G, 7961G-GE, 7970G, and 7971G-GE.
The overlaid ephone-dns must be configured on the phone
using the button command and the c keyword.
Overlaid ephone-dn call overflow to other buttons—One
or more buttons can be dedicated to serve as expansion, or
overflow, buttons for another button on the same
Cisco Unified IP phone that has overlaid ephone-dns. A call
to an overlay button that is busy with an active call will roll
over to the next available expansion button.
Phone Support Cisco IP Communicator is a software-based application that Configuring Phones to
appears on a user’s computer monitor as a graphical, Make Basic Calls
display-based IP phone with a color screen, a key pad, feature
buttons, and soft keys. Cisco Unified CME supports Cisco IP
Communicator 2.0 and later versions.
Remote teleworker phone—Teleworkers can connect
remote phones over a WAN and be directly supported by
Cisco Unified CME.
Ring Tones Distinctive ringing—An extension’s ring patterns can be set Configuring Ring Tones
to distinguish among internal, external, and feature calls.
Security Cisco Unified CME phone authentication is a security Configuring Security
infrastructure for providing secure Skinny Client Control
Protocol (SCCP) signaling between Cisco Unified CME and
IP phones.
Soft keys Feature blocking—The features associated with the Customizing Soft Keys
following soft keys can be individually blocked per ephone:
CFwdAll, Confrn, GpickUp, Park, PickUp, and Trnsfer. The
soft key is not removed, but it does not function.
Soft-key control for hold state—The soft keys that are
available while a call is on hold can be modified. The
NewCall and Resume soft keys are normally available when a
phone has a call on hold, but a template can be applied to the
phone to remove these soft keys.
Speed Dial Bulk-loading of speed-dial numbers—Text files with lists Configuring Speed Dial
of speed-dial numbers can be loaded into system flash or a
URL. The files can hold up to 10,000 numbers and can be
applied to all ephones or to specific ephones.

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Table 1 Supported Cisco Unified CME Features (continued)

Release Feature Name Feature Description Where Documented

System-Level Disabling automatic phone registration—Normally, Configuring
Parameters Cisco Unified CME allocates an ephone slot to any ephone System-Level
that connects to the system. To prevent unauthorized Parameters
registrations, the no auto-reg-ephone command prevents any
ephone from registering with Cisco Unified CME if its MAC
address is not explicitly listed in the configuration.
External storage of configuration files and per-phone
configuration files—Phone configuration files can be stored
on an external TFTP server to offload the TFTP server
function of the Cisco Unified CME router. This additional
storage space permits the use of per-phone configuration
files, which can be used to specify different user locales and
network locales for phones.
Failover to Redundant Router—Sites can be set up with a
primary and secondary Cisco Unified CME router to provide
redundant Cisco Unified CME capability. Phones
automatically register at the secondary router if the primary
router fails and later rehome to the primary router when it is
operational again.
Templates Maximum number of ephone templates that can be defined Creating Templates
has increased from 5 to 20. No special configuration is
required.
New commands available for ephone templates—Ephone
templates were previously introduced to allow system
administrators to control the display of soft keys in various
call states on individual ephones. Their role has been
expanded to allow you to define a set of ephone parameter
values that can be assigned to one or more phones in a single
step.
Ephone-dn templates are introduced to allow administrators
to easily apply sets of configured parameters to individual
ephone-dns. Up to 15 ephone-dn templates can be defined.
Video Support Video support for SCCP-based endpoints—This feature Configuring Video
adds video support to allow you to pass a video stream with a Support for SCCP-Based
voice call, between video-capable SCCP endpoints and Endpoints
between SCCP and H.323 endpoints. Through the
Cisco Unified CME router, the video-capable endpoints can
communicate with each other locally, to a remote H.323
endpoint through a gateway, or through an H.323 network.

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Table 1 Supported Cisco Unified CME Features (continued)

Release Feature Name Feature Description Where Documented

Voice Mail Line-selectable MWI—Previously, the message-waiting Integrating Voice Mail
indication (MWI) lamp on a phone could only indicate when
messages were waiting for the primary number on a phone.
Now any phone line can be designated during configuration.
Mailbox selection policy for voice-mail servers—A policy
can be set for selecting the mailbox to use for calls that are
diverted one or more times within a Cisco Unified CME
system before being sent to a Cisco Unity Express,
Cisco Unity, or PBX voice-mail pilot number.
Prefix option for SIP unsolicited MWI Notify
messages—Central voice-message servers that provide
mailboxes for multiple Cisco Unified CME sites may use site
codes or prefixes to distinguish among similarly numbered
ranges of extensions at different sites.
You can specify the prefix for your site so that central
mailbox numbers are correctly converted to your extension
numbers.
XML Interface XML interface enhancements—An eXtensible Markup Configuring the XML
Language (XML) application program interface (API) is API
provided to supply data from Cisco Unified CME to
management software. In Cisco Unified CME 4.0 and later
versions, all Cisco Unified CME features have XML support.

CCVP, the Cisco Logo, and the Cisco Square Bridge logo are trademarks of Cisco Systems, Inc.; Changing the Way We Work, Live, Play, and Learn is a
service mark of Cisco Systems, Inc.; and Access Registrar, Aironet, BPX, Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, CCSP, Cisco, the Cisco
Certified Internetwork Expert logo, Cisco IOS, Cisco Press, Cisco Systems, Cisco Systems Capital, the Cisco Systems logo, Cisco Unity,
Enterprise/Solver, EtherChannel, EtherFast, EtherSwitch, Fast Step, Follow Me Browsing, FormShare, GigaDrive, GigaStack, HomeLink, Internet
Quotient, IOS, iPhone, IP/TV, iQ Expertise, the iQ logo, iQ Net Readiness Scorecard, iQuick Study, LightStream, Linksys, MeetingPlace, MGX,
Networking Academy, Network Registrar, Packet, PIX, ProConnect, RateMUX, ScriptShare, SlideCast, SMARTnet, StackWise, The Fastest Way to
Increase Your Internet Quotient, and TransPath are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the United States and certain other
countries.

All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply a
partnership relationship between Cisco and any other company. (0612R)

Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and
figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and
coincidental.

© 2007 Cisco Systems, Inc. All rights reserved.

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Last Updated: June 18, 2007

Cisco Unified Communications Manager Express (formerly known as Cisco Unified CallManager
Express) 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.

Contents
• Information About Cisco Unified CME, page 47
• Where to Go Next, page 52
• Additional References, page 52
• Obtaining Documentation, Obtaining Support, and Security Guidelines, page 54

Information About Cisco Unified CME

To design and configure a Cisco Unified Communications Manager Express (Cisco Unified CME)
system, you should understand the following concepts:
• Cisco Unified CME Overview, page 48
• Licenses, page 49
• PBX or Keyswitch Model, page 50

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Information About Cisco Unified CME

Cisco Unified CME Overview

Cisco Unified CME is a feature-rich entry-level IP telephony solution that is integrated directly into
Cisco IOS software. Cisco Unified CME 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 Unified 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 Unified CME offers most of the core
telephony features required in the small office, and also 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.
A Cisco Unified CME system is extremely flexible because it is modular. A Cisco Unified 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.
Figure 1 shows a typical deployment of a Cisco Unified CME router with several phones and devices
connected to it. The Cisco Unified 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 Unified CME for the Small- and Medium-Size Office

Telephone Telephone

Fax

Cisco Unified CME router

PSTN

Cisco Unified IP phones

RADIUS
billing
IP IP IP server
146626

PCs
Gatekeeper

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Information About Cisco Unified CME

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

Figure 2 Cisco Unified CME for Service Providers

Telephone Telephone
IP
PSTN network
Fax

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

Cisco Unified IP phones

IP IP IP
Voice-mail
Gatekeeper server

146627
PCs

A Cisco Unified CME system uses the following basic building blocks:
• Ephone or voice register pool—A software concept that usually represents a physical telephone,
although it is also used to represent a port that connects to a voice-mail system, and provides the
ability to configure a physical phone using Cisco IOS software. Each phone can have multiple
extensions associated with it and a single extension can be assigned to multiple phones. Maximum
number of ephones and voice register pools supported in a Cisco Unified CME system is equal to
the maximum number of physical phones that can be connected to the system.
• Directory number—A software concept that represents the line that connects a voice channel to a
phone. A directory number represents a virtual voice port in the Cisco Unified CME system, so the
maximum number of directory numbers supported in Cisco Unified CME is the maximum number
of simultaneous call connections that can occur. This concept is different from the maximum number
of physical lines in a traditional telephony system.

Licenses
You must purchase a base Cisco Unified CME feature license and phone user licenses that entitle you to
use Cisco Unified CME.

Note To support H.323 call transfers and forwards to network devices that do not support the H.450 standard,
such as Cisco Unified Communications Manager, a tandem gateway is required in the network. The
tandem gateway must be running Cisco IOS release 12.3(7)T or a later release 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.

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Information About Cisco Unified CME

PBX or Keyswitch Model

When setting up a Cisco Unified 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 Unified CME provides a
significant amount of flexibility in this area, but requires that you have a clear understanding of the
model that you choose.

PBX Model
The simplest case is the PBX model, in which most of the IP phones in your system have a single unique
extension number. Incoming PSTN calls are routed to a receptionist at an attendant console or to an
automated attendant. Phone users may be in separate offices or be geographically separated and
therefore often use the telephone to contact each other.
For this model, we recommend that you configure directory numbers as dual-lines so that 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 directory numbers enable your configuration to
support call waiting, call transfer with consultation, and three-party conferencing (G.711 only).
Figure 3 shows a PSTN call that is received at the Cisco Unified CME router, which sends it to the
designated receptionist or automated attendant (1), which then routes it to the requested extension (2).

Figure 3 Incoming Call Using PBX Model

FXO ports
1

2
Cisco Unified CME Receptionist or
automated attendant

IP IP IP

146456
Extension Extension Extension
1001 1002 1003

For configuration information, see the “How to Configure Phones for a PBX System” section on
page 158.

Keyswitch Model
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. Phone users
are generally in close proximity and have little need to use the telephone to contact each other.
For example, a 3x3 keyswitch system has three PSTN lines shared across three telephones, such that all
three PSTN lines appear on each of the three telephones. This permits an incoming call on any PSTN
line to be directly answered by any telephone—without the aid of a receptionist, auto-attendant or the
use of (expensive) DID lines. Also, the lines act as shared lines—a call can be put on hold on one phone
and resumed on another phone without invoking call transfer.

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In the keyswitch model, the same directory numbers are assigned to all IP phones. 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 dual-line option because the
PSTN lines to which the directory numbers 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 configure the keyswitch model by creating a set of directory numbers that correspond one-to-one
with your PSTN lines. Then you configure your PSTN ports to route incoming calls to those ephone-dns.
The maximum number of PSTN lines that you can assign in this model can be limited by the number of
available buttons on your IP phones. If so, the overlay option may be useful for extending the number of
lines that can be accessed by a phone.
Figure 4 shows an incoming call from the PSTN (1), which is routed to extension 1001 on all three
phones (2).

Figure 4 Incoming PSTN Call Using Keyswitch Model

FXO ports
1

Cisco Unified CME

2

IP IP IP
Extension Extension Extension
1001 1001 1001
146457

1002 1002 1002

1003 1003 1003

For configuration information, see the “How to Configure Phones for a Key System” section on
page 176.

Hybrid Model
PBX and keyswitch configurations can be mixed on the same IP phone and can include both unique
per-phone extensions for PBX-style calling and shared lines for keyswitch-style call operations.
Single-line and dual-line directory numbers can be combined on the same phone.
In the simplest keyswitch deployments, individual telephones do not have private extension numbers.
Where key system telephones do have individual lines, the lines are sometimes referred to as intercoms
rather than as extensions. The term “Intercom” is derived from Internal Communication; there is no
assumption of the common ''intercom press-to-talk'' behavior of auto dial or auto answer in this context,
although those options may exist.
For key systems that have individual intercom (extension) lines, PSTN calls can usually be transferred
from one key system phone to another using the intercom (extension) line. When Call Transfer is
invoked in the context of a connected PSTN line, the outbound consultation call is usually placed from
the transferrer phone to the transfer-to phone using one of the phone's intercom (extension) line buttons.
When the transferred call is connected to the transfer-to phone and the transfer is committed (the

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Where to Go Next

transferrer hangs up), the intercom lines on both phones are normally released and the transfer-to call
continues in the context of the original PSTN line button (all PSTN lines are directly available on all
phones). This behavior allows the transferred call to be put on hold (on the PSTN line button) and then
subsequently resumed from another phone that shares that PSTN line.
For example, you can design a 3x3 keyswitch system as shown in Figure 4 and then add another, unique
extension on each phone (Figure 5). This setup will allow each phone to have a “private” line to use to
call the other phones or to make outgoing calls.

Figure 5 Incoming PSTN Call Using Hybrid PBX-Keyswitch Model

FXO ports
1

Cisco Unified CME

2

IP IP IP
Extension Extension Extension
1001 1001 1001
1002 1002 1002

146458
1003 1003 1003
1004 1005 1006

Where to Go Next
Before configuring Cisco Unified CME, see “Before You Begin” on page 55.

Additional References
The following sections provide references related to Cisco Unified CME.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Cisco IOS voice troubleshooting • Cisco IOS Voice Troubleshooting and Monitoring Guide

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Additional References

Related Topic Document Title

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)
Dynamic Host Configuration Protocol (DHCP) • “DHCP” part of the Cisco IOS IP Addressing Services
Configuration Guide
Fax and modem configurations • Cisco IOS Fax and Modem Services over IP Application Guide
FXS ports FXS Ports in H.323 Mode
• “Configuring Analog Voice Ports” section of the Cisco IOS
Voice Port Configuration Guide
• Caller ID
• Cisco IOS Fax and Modem Services over IP Application Guide

FXS Ports in SCCP Mode on Cisco VG 224 Analog Phone Gateway

• SCCP Controlled Analog (FXS) Ports with Supplementary
Features in Cisco IOS Gateways
• Cisco VG 224 Analog Phone Gateway data sheet
H.323 • Cisco IOS H.323 Configuration Guide
Network management software using • Provisioning Manager - Managed Cisco CallManager Express
Cisco Packet Telephony Center - Virtual Switch Routers
(Cisco PTC - VS)
Network Time Protocol (NTP) • “Performing Basic System Management” chapter of Cisco IOS
Network Management Configuration Guide
Phone documentation for Cisco phones • Cisco 7900 Series IP Phones
• Cisco ATA 180 Series Analog Telephone Adaptors
• Cisco IP Communicator
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides
Public key infrastructure (PKI) • “Part 5: Implementing and Managing a PKI” in the Cisco IOS
Security Configuration Guide
SIP • Cisco IOS SIP Configuration Guide
TAPI and TSP documentation • See links at Cisco Unified CME Documentation Roadmap
Tcl IVR and VoiceXML • Cisco IOS Tcl IVR and VoiceXML Application Guide -
12.3(14)T and later
• Default Session Application Enhancements
• Tcl IVR API Version 2.0 Programmer’s Guide
• Cisco VoiceXML Programmer’s Guide
VLAN class-of-service (COS) marking • Enterprise QoS Solution Reference Network Design Guide

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Obtaining Documentation, Obtaining Support, and Security Guidelines

Related Topic Document Title

Voice-mail integration • Cisco Unified CallManager Express 3.0 Integration Guide for
Cisco Unity 4.0
• Integrating Cisco CallManager Express with
Cisco Unity Express
XML • XML Provisioning Guide for Cisco CME/SRST
• Cisco IP Phone Services Application Development Notes

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.

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

Obtaining Documentation, Obtaining Support, and Security

Guidelines
For information on obtaining documentation, obtaining support, providing documentation feedback,
security guidelines, and also recommended aliases and general Cisco documents, see the monthly
What’s New in Cisco Product Documentation, which also lists all new and revised Cisco technical
documentation, at:
http://www.cisco.com/en/US/docs/general/whatsnew/whatsnew.html

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Last Updated: June 19, 2007

This module describes general decisions that you should make before you configure Cisco Unified
Communications Manager Express (Cisco Unified CME).

Contents
• Information About Planning Your Configuration, page 55
• How to Configure Cisco Unified CME, page 60
• Feature Summary, page 64
• Planning Worksheets, page 67

Information About Planning Your Configuration

Before configuring Cisco Unified CME, you should understand the following concepts:
• System Design, page 55
• Configuration Methods Summary, page 56
• Cisco Unified Communications Express - QCT, page 58
• Cisco Unified CME GUI, page 59

System Design
Traditional telephony systems are based on physical connections and are therefore limited in the types
of phone services that they can offer. Because phone configurations and directory numbers in a
Cisco Unified CME system are software entities 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 Unified CME systems can be designed in many ways. The key is to determine the total number of
simultaneous calls you want to handle at your site and at each phone at your site, and how many different
directory numbers and phones you want to have. Even a Cisco Unified CME system has its limits,
however. Consider the following factors in your system design:

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• Maximum number of phones—This number corresponds to the maximum number of devices that
can be attached. The maximum is platform- and version-dependent. To find the maximum for your
platform and version, see the appropriate Cisco CME Supported Firmware, Platforms, Memory, and
Voice Products document at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap0918
6a0080189132.html.
• Maximum number of directory numbers—This number corresponds to the maximum number of
simultaneous call connections that can occur. The maximum is platform- and version-dependent. To
find the maximum for your platform and version, see the appropriate Cisco CME Supported
Firmware, Platforms, Memory, and Voice Products document at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap0918
6a0080189132.html.
• Telephone number scheme—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 20
different telephone numbers.
The flexibility of a Cisco Unified CME system is due largely to the different types of directory numbers
(DNs) that you can assign to phones in your system. By understanding types of DNs and considering
how they can be combined, you can create the complete call coverage that your business requires. For
more information about DNs, see“Configuring Phones to Make Basic Calls” on page 147.
After setting up the DNs and phones that you need, you add optional Cisco Unified CME features to
create a telephony environment that enhances your business objectives. Cisco Unified 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.
When creating number plans, dialing schemes, and call coverage patterns in Cisco Unified CME, there
are several factors that you must consider:
• Is there an existing PBX or Key System that you are replacing and want to emulate?
• Number of phones and phone users to be supported?
• Do you want to use single-line or dual-line DNs?
• What protocols does your voice network support?
• Which call transfer and forwarding methods must be supported?
• What existing or preferred billing method do you want to use for transferred and forwarded calls?
• Do you need to optimize network bandwidth or minimize voice delay?
Because these factors can limit your choices for some of the configuration decisions that you will make
when you create of a dialing plan, see the Cisco Unified CallManager Express Solution Reference
Network Design Guide to help you understand the effect these factors have on your Cisco Unified CME
implementation.

Configuration Methods Summary

Your choice of configuration method depends on whether you want to create an initial configuration for
your IP telephony system or you want to perform ongoing maintenance, such as routinely making
additions and changes associated with employee turnover. Table 2 compares the different methods for
configuring Cisco Unified CME:

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.
Table 2 Comparison of Configuration Methods for Cisco Unified CME

Configuration Method Benefits Restrictions

Cisco IOS command line • Generates commands for running Requires knowledge of Cisco IOS commands
interface configuration which can be saved on Cisco and Cisco Unified CME.
router to be configured.
For information about
supported features, see • Use for setting up or modifying all
Table 4. parameters and features during initial
For information about using configuration and ongoing maintenance.
Cisco IOS commands, see
the “Using Cisco IOS
Commands to Create or
Modify the Configuration”
section on page 61.
Cisco Unified Communicati • Configuration wizard-like GUI. • Factory-default configuration must be
ons Express - QCT, page 58. loaded in nonvolatile memory and in the
• Auto-discovers the hardware setup of the
running configuration of the Cisco router
For information about using Cisco router.
to be configured.
Cisco Unified Communicati • Accept default values to quickly configure
ons Express - QCT, see the a typical basic IP telephony system or Note Cisco Unified Communications
“Using modify preconfigured values, enter data, Express - QCT can reset the running
Cisco Unified Communicati and choose options to customize the configuration on a router to
ons Express - QCT to configuration. factory-default.
Generate a Telephony
Configuration” section on • Generates commands for configuring an IP • Cannot be used to modify or maintain
page 62. telephony system which can be uploaded to system configuration.
the Cisco router, along with the firmware
• Configures limited subset of advanced
files for all Cisco Unified IP phones to be
features.
connected to the Cisco router.
• Configuration can be saved as template to
be reused for additional systems.
• Use for initial configuration including:
basic information, such as VLANs,
common DHCP pool, NTF servers,
inbound and outbound destination
numbers, and translation rules; and a
subset of advanced features including
paging, intercom, call park, hunt group,
caller ID blocking, Class of Restriction,
enabling video, and SIP truncking.
• Can import user configuration data (names
and extension numbers) from an external
file.
• Supports barcode-scanned input of MAC
addresses and phone types.

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Table 2 Comparison of Configuration Methods for Cisco Unified CME

Configuration Method Benefits Restrictions

Cisco Unified CME GUI, • Graphical user interface • Cannot provision voice features such as
page 59. • Use for ongoing system maintenance digit translation, call routing, and class of
For information about using restriction.
• Modifies, adds, and deletes phones and
the Cisco Unified CME • Cannot provision data features such as
extensions; configures voice-mail; IP
GUI, see the “Using DHCP, IP addressing, and VLANs.
phone URLs; secondary dial tone pattern;
Cisco Unified CME GUI to
timeouts; transfer patterns; and the • Can only provision IP phones that are
Modify or Maintain
music-on-hold file. registered to Cisco Unified CME. Cannot
Configuration” section on
use bulk administration to import
page 63. • Three configurable levels of access.
multiple phones at the same time. Cannot
manage IP phone firmware.
• Requires manual upgrade of files in flash
if Cisco Unified CME version is
upgraded.

Cisco Unified Communications Express - QCT

Cisco Unified Communications Express - QCT 3.0 and later versions is an optional web-based
application provided by Cisco Systems to simplify installation of a brand new Cisco Unified CME. After
you fill in the required fields, Cisco Unified Communications Express - QCT generates the
corresponding commands and uploads the basic telephony configuration to your Cisco router.
Cisco Unified Communications Express - QCT supports auto-discovery of installed hardware, so you
can deploy an IP telephony configuration without having detailed knowledge of the modules installed in
the router. Cisco Unified Communications Express - QCT prompts for common parameters needed to
configure a Cisco Integrated Services router to support a complete IP telephony solution as a
private-branch-exchange (PBX) with direct-inward-dial extensions or as a square mode key system.
Before using Cisco Unified Communications Express - QCT, the factory-default configuration must be
loaded in nonvolatile memory and in the running configuration of the Cisco router to be configured. You
can use Cisco Unified Communications Express - QCT to restore the factory default configuration on
router to be configured.
After using Cisco Unified Communications Express - QCT, you can add any feature not supported by
Cisco Unified Communications Express - QCT to the configuration by using Cisco IOS commands.
Cisco Unified Communications Express - QCT cannot be used to perform routine additions and changes
associated with employee turnover. You can use the Cisco Unified CME GUI or Cisco IOS commands
to modify the basic telephony configuration generated by Cisco Unified Communications Express -
QCT.
Table 3 contains a list of parameters and features that are configured by using
Cisco Unified Communications Express - QCT. For configuration information, see the “Using
Cisco Unified Communications Express - QCT to Generate a Telephony Configuration” section on
page 62.

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Table 3 Parameters and Features Supported by Cisco Unified Communications Express - QCT

Basic Configuration Features

• Installing firmware for • Configuring system-level • Setting up features:
Cisco Unified IP phones parameters
– Call park
• Setting up router for – Basic firewall
– Caller ID blocking
Cisco Unified CME – COR
– Hunt groups
– Automatic detection of
– Date and time
hardware configuration – Intercom
• Configuring phones for basic – Paging
– Connection port and type
calling
– General router information – Pickup groups
– General phone information
– Voice and data VLAN for PBX or square mode key • Configuring support for
system Cisco Unity Express
• Defining network parameters
– Directory numbers and phone • Enabling VTA integration (video)
– Enabling calls on VoIP
user names
– DHCP pool
– DID numbers
– DNS servers
• Creating international and
– NTP domestic dialing plans
– SIP trunking • Configuring transcoding
(Cisco Unified CME 4.1 and
later versions only) • Generating configuration files

– TFTP address

Cisco Unified CME GUI

The Cisco Unified CME GUI provides a web-based interface to manage most system-level and
phone-level features. In particular, the GUI facilitates the routine additions and changes associated with
employee turnover, allowing these changes to be performed by nontechnical staff.
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 additions and changes without having
access to systemwide features. This person does not have to be familiar with Cisco IOS software.
• Phone user—Able to program a small set of features on his or her own phone and search the
Cisco Unified CME directory.
The Cisco Unified CME GUI uses HTTP to transfer information between the Cisco Unified CME router
and 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. Additional customer
administrators and phone users can be added by using Cisco IOS command line interface or by using
GUI screens.
Cisco Unified 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.

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How to Configure Cisco Unified CME

The GUI supports authentication, authorization, and accounting (AAA) authentication for system
administrators through a remote server capability. If authentication through the server fails, the local
router is searched.
Cisco Unified CME GUI must be installed and set up before it can be used. Instructions for using the
Cisco Unified GUI are in online help for the GUI.
For information about using the Cisco Unified CME GUI, see the “Using Cisco Unified CME GUI to
Modify or Maintain Configuration” section on page 63.

How to Configure Cisco Unified CME

This section contains the following tasks:
• Using Cisco IOS Commands to Create or Modify the Configuration, page 61
• Using Cisco Unified Communications Express - QCT to Generate a Telephony Configuration,
page 62
• Using Cisco Unified CME GUI to Modify or Maintain Configuration, page 63

Prerequisites for Configuring Cisco Unified CME

• Base Cisco Unified CME feature license and phone user licenses that entitle you to use
Cisco Unified CME are purchased.

Note To support H.323 call transfers and forwards to network devices that do not support the H.450 standard,
such as Cisco Unified Communications Manager, a tandem gateway is required in the network. The
tandem gateway must be running Cisco IOS release 12.3(7)T or a later release 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.

• Your IP network is operational and you can access Cisco web.

• VoIP networking must be operational. For quality and security purposes, we recommend you 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 Unified 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.
• You have a valid Cisco.com account.
• Cisco router and all recommended services hardware for Cisco Unified CME is installed.
• Recommended Cisco IOS IP Voice or higher image is downloaded to flash memory in the router.
To determine which Cisco IOS software release supports the recommended Cisco Unified CME
version, see the Cisco Unified CME and Cisco IOS Software Compatibility Matrix.
• You have access to a TFTP server for downloading files.

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Restrictions for Configuring Cisco Unified CME

• Cisco Unified CME cannot register as a member of a Cisco Unified Communications Manager
cluster.
• 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.
• After a three-way conference is established, a participant cannot use call transfer to join the
remaining conference participants to a different number.
• Cisco Unified CME does not support the following:
– 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, Cisco Unified Communications Manager Auto Attendant, or Cisco Personal
Assistant
– Telephony Application Programming Interface (TAPI)
Cisco Unified CME implements only a small subset of TAPI functionality. It supports 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 Unified Contact Center (formerly Cisco IPCC).
Also, this TAPI version does not have direct media- and voice-handling capabilities.

Using Cisco IOS Commands to Create or Modify the Configuration

To use Cisco IOS commands to configure Cisco Unified CME, see the Cisco Unified CME System
Administrator Guide.

Prerequisites
• Hardware and software to establish a physical or virtual console connection to the Cisco router using
a terminal or PC running terminal emulation is available and operational.
• To establish a physical console connection, attach a terminal or PC running terminal emulation to
the console port of the router. For more information on cabling, and details about how to connect a
terminal to the console port or the AUX port, see Cabling Guide for Console and Aux Ports on Cisco
Routers.
For connecting to the router to be configured, use the following terminal settings:
– 9600 baud rate
– No parity
– 8 data bits
– 1 stop bit
– No flow control

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• We recommend that you complete the worksheets to gather required site-specific information for
the Cisco router to be configured. See the worksheet set at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/prod_configuration_guide09186a00807
59201.html#wp1007671.

Note Even though the worksheet set is for configuring a Typical (recommended) installation of an IP
telephony system using Cisco Unified Communications Express - QCT, the information is required to
create an initial configuration using any of the configuration methods.

Using Cisco Unified Communications Express - QCT to Generate a Telephony

Configuration
To u se Cisco Unified Communications Express - QCT to configure a PBX or Square Mode (key system)
IP telephony system, with or without Cisco Unity Express voice mail, or to reset the configuration of a
Cisco router to factory default, see the Cisco Unified Communications Express - QCT User Guide.

Prerequisites
• Your IP network is operational and you can access Cisco web.
• A PC with Microsoft® Internet Explorer 5.5 or later is connected, using a serial cable, to the console
port of Cisco router to be configured. If you need assistance in connecting your PC to your router's
console port, refer to the Install and Upgrade Guide for the Cisco router.
• The Block Pop-up Windows feature for Microsoft® Internet Explorer must be disabled.
• You must be a member of Administrators group under User Account settings for your PC.
• You must have a valid Cisco CCO account.
• The factory-default configuration is loaded in nonvolatile memory and in the running configuration
of the Cisco router to be configured.

Note Cisco Unified Communications Express - QCT can be used to restore the factory default configuration
on router to be configured.

• If you are using Cisco Unified Communications Express - QCT to upload firmware files for
Cisco Unified IP phones after uploading the generated configuration, all Cisco firmware files to be
uploaded must be installed in the folder named Phoneloads, within the local folder in which
Cisco Unified Communications Express - QCT is installed.
• Worksheets for Cisco Unified Communications Express - QCT are complete with required
site-specific information for the Cisco router to be configured.

Restrictions
• Cisco Unified Communications Express - QCT cannot be used to perform routine additions and
changes associated with employee turnover.
• Cisco Unified Communications Express - QCT can configure only a subset of features of
Cisco Unified CME. You must use Cisco IOS commands to add features or to modify the
configuration.

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What to Do Next
After using Cisco Unified Communications Express - QCT to generate a basic telephony configuration,
you can skip the following modules in the Cisco Unified CME System Administrator Guide when you
use Cisco IOS commands to modify the configuration:
• Defining Network Parameters
• Configuring System-Level Parameters
• Configuring Phones to Make Basic Calls
• Creating Phone Configurations Using Extension Assigner
• Configuring Dialing Plans
• Adding Features, but only for the following:
– Configuring Call Blocking
– Configuring Call Park
– Configuring Call Transfer and Forwarding
– Configuring Call-Coverage Features
– Configuring Caller ID Blocking
– Configuring Conferencing
– Configuring Intercom Lines
– Configuring Music on Hold
– Configuring Paging
• Enabling the GUI
• Configuring Voice-Mail Support

Using Cisco Unified CME GUI to Modify or Maintain Configuration

To use the Cisco Unified CME GUI to modify the configuration, see “Enabling the GUI” on page 339.

Prerequisites
• Cisco CME 3.2 or a later version.
• Files required for the operation of the GUI must be copied into flash memory on the router. For
information about files, see “Installing and Upgrading Cisco Unified CME Software” on page 69.

Restrictions
• The web browser that you use to access the GUI must be Microsoft Internet Explorer 5.5 or a later
version. No other type of browser can be used to access the GUI.
• Cannot provision voice features such as digit translation, call routing, and class of restriction.
• Cannot provision data features such as DHCP, IP addressing, and VLANs.
• Can only provision IP phones that are registered to Cisco Unified CME. Cannot use bulk
administration to import multiple phones at the same time. Cannot manage IP phone firmware.

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Feature Summary

• Requires manual upgrade of files in flash memory of router if Cisco Unified CME is upgraded to
later version.
• Other minor limitations, such as:
– 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.

Feature Summary
Table 4 contains a list of commonly configured features in Cisco Unified CME and the module in which
they appear in this guide. For a detailed list of features, with links to corresponding information in this
guide, see “Cisco Unified CME Features Roadmap” on page 37.

Table 4 Parameters and Features Supported by Cisco IOS Commands

Parameters and Features Where to Find Configuration Information

Cisco Unified CME Software
Installing and upgrading software, including: Installing and Upgrading Cisco Unified CME
• Cisco Unified CME Software

• Cisco Unified CME GUI

• Firmware files for Cisco Unified IP phones
Basic Configuration
• Enabling Calls in Your VoIP Network Defining Network Parameters
• Defining DHCP
• Setting Network Time Protocol
• Configuring DTMF Relay for H.323 Networks in Multisite
Installations
• Configuring SIP Trunk Support
• Changing the TFTP Address on a DHCP Server
• Enabling OOD-R
• Configuring Bulk Registration Configuring System-Level Parameters
• Setting Up Cisco Unified CME
• Setting Date and Time Parameters
• Blocking Automatic Registration
• Defining Alternate Location and Type of Configuration Files
• Changing Defaults for Time Outs
• Configuring a Redundant Router

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Feature Summary

Table 4 Parameters and Features Supported by Cisco IOS Commands

Parameters and Features Where to Find Configuration Information

• Creating Directory Numbers, Assigning Directory Numbers to Phones Configuring Phones to Make Basic Calls
• Creating Phone Configurations Using Extension Assigner
• Generating Configuration Files for Phones
• Resetting and Restarting Phones
Dialing Plans
• Dial-plan patterns Configuring Dialing Plans
• Translation rules and profiles
• Secondary dial tones
Transcoding Support
• DSP farms Configuring Transcoding Resources.
• NMs or NM farms
• Transcoding sessions
Localization Support
• Use locale Configuring Localization Support.
• Network locale
Cisco Unified CME GUI Enabling the GUI
Features
• Automatic line selection Adding Features
• Call blocking
• Call park
• Call transfer and forwarding
• Caller ID blocking
• Conferencing
• Directory services
• Do Not Disturb (DND)
• Feature Access Codes (FAC)
• Headset auto-answer
• Intercom lines
• Loopback call routing
• Music on Hold (MOH)
• Paging
• Presence service
• Ring tones
• Soft keys
• Speed dial

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Feature Summary

Table 4 Parameters and Features Supported by Cisco IOS Commands

Parameters and Features Where to Find Configuration Information

• Call hunt Configuring Call-Coverage Features.
• Call pickup
• Call waiting
• Callback busy subscriber
• Hunt groups
• Night service
• Overlaid Ephone-dns
Authentication Support
• Phone authentication startup messages Configuring Security.
• CTL file
• CTL client and provider
• MIC root certificate
Phone Options
• Customized Background Images for Cisco Unified IP Phone 7970 Modifying Cisco Unified IP Phone Options.
• Fixed Line/Feature Buttons for Cisco Unified IP Phone 7931G
• Header Bar Display
• PC Port Disable
• Phone Labels
• Programmable vendorConfig Parameters
• System Message Display
• URL Provisioning for Feature Buttons
Video Support Configuring Video Support for SCCP-Based
Endpoints
Voice-Mail Support
• Cisco Unity Connection Integrating Voice Mail.
• Cisco Unity Express
• Cisco Unity
• DTMF integration for legacy voice-mail applications
• Mailbox selection policy
• RFC 2833 Dual Tone Multifrequency (DTMF) MTP Passthrough
• MWI
Cisco Unified CME as SRST fallback Configuring SRST Fallback Support.

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Planning Worksheets

Planning Worksheets
Before configuring Cisco Unified CME, we recommend that you complete the worksheets to gather
required site-specific information for the Cisco router to be configured. See the worksheet set at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/prod_configuration_guide09186a00807592
01.html#wp1007671.

Note Even though the worksheet set is for configuring a Typical (recommended) installation of an IP
telephony system using Cisco Unified Communications Express - QCT, the information is required to
create an initial configuration using any of the configuration methods.

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 Installing and Upgrading Cisco Unified CME
Software

Last Updated: March 26, 2007

This chapter explains how to install Cisco Unified Communications Manager Express
(Cisco Unified CME) software and how to upgrade phone firmware for Cisco Unified IP phones.

Contents
• Prerequisites for Installing Cisco Unified CME Software, page 69
• Information About Cisco Unified CME Software, page 70
• How to Install and Upgrade Cisco Unified CME Software, page 74
• Additional References, page 90

Prerequisites for Installing Cisco Unified CME Software

Hardware
Ensure that you have the correct hardware and that your IP network is functioning correctly. See the
documentation for your particular hardware at
http://www.cisco.com/en/US/products/hw/routers/tsd_products_support_category_home.html and the
“Installing Hardware and Software” chapter of Installing Cisco IPC Express:
Cisco CallManager Express and Cisco Unity Express.

Cisco IOS Software

Download and install the appropriate IP VOICE image from the Cisco Software Center. Consult the
following sources to determine the correct Cisco IOS software version:
• Feature Navigator at http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp
• Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap0918
6a0080189132.html

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Information About Cisco Unified CME Software

Information About Cisco Unified CME Software

This section contains a list of the types of files that must be downloaded and installed in the router flash
memory to use with Cisco Unified CME. The files listed in this section are included in zipped or tar
archives that are downloaded from the Cisco Unified CME software download website at
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp.
• Basic Files, page 70
• GUI Files, page 70
• Phone Firmware Files, page 70
• IXML Template, page 72
• Music-on-Hold (MOH) File, page 72
• Script Files, page 72
• Bundled TSP Archive, page 73
• File Naming Conventions, page 73
• Cisco Unified Communications Express - Quick Configuration Tool, page 73

Basic Files
A tar archive contains the basic files you need for Cisco Unified CME. Be sure to download the correct
version for the Cisco IOS software release that is running on your router. The basic tar archive generally
also contains the phone firmware files that you require, although you may occasionally need to download
individual phone firmware files. For information about installing Cisco Unified CME, see the
“Installing Cisco Unified CME Software” section on page 74.

GUI Files
A tar archive contains the files that you need to use the Cisco Unified CME graphical user interface
(GUI), which provides a mouse-driven interface for provisioning phones after basic installation is
complete. For installation information, see the “Installing Cisco Unified CME Software” section on
page 74.

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

Phone Firmware Files

Phone firmware files provide code to enable phone displays and operations. These files are specialized
for each phone type and protocol, SIP or SCCP, and are periodically revised. You must be sure to have
the appropriate phone firmware files for the types of phones, protocol being used, and
Cisco Unified CME version at your site.

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New IP phones are shipped from Cisco with a default manufacturing SCCP image. When a IP phone
downloads its configuration profile, the phone compares the phone firmware mentioned in the
configuration profile with the firmware already installed on the phone. If the firmware version differs
from the one that is currently loaded on the phone, the phone contacts the TFTP server to upgrade to the
new phone firmware and downloads the new firmware before registering with Cisco Unified CME.
Generally, phone firmware files are included in the Cisco Unified CME software archive that you
download. They can also be posted on the software download website as individual files or archives.
Early versions of Cisco phone firmware for SCCP and SIP IP phones had filenames as follows:
• SCCP firmware—P003xxyy.bin
• SIP firmware—P0S3xxyy.bin
In both bases, x represents the major version, and y represented the minor version. The third character
represents the protocol, “0” for SCCP or “S” for SIP.
In later versions, the following conventions are used:
• SCCP firmware—P003xxyyzzww, where x represents the major version, y represents the major
subversion, z represents the maintenance version, and w represents the maintenance subversion.
• SIP firmware—P0S3-xx-y-zz, where x represents the major version, y represents the minor version,
and z represents the subversions.
• The third character in a filename—Represents the protocol, “0” for SCCP or “S” for SIP.
There are exceptions to the general guidelines. For Cisco ATA, the filename begins with AT. For
Cisco Unified IP Phone 7002, 7905, and 7912, the filename can begin with CP.
Signed and unsigned versions of phone firmware are available for certain phone types. Signed binary
files support image authentication, which increases system security. We recommend signed versions if
your version of Cisco Unified CME supports them. Signed binary files have .sbn file extensions, and
unsigned files have .bin file extensions.
For Java-based IP phones, such as the Cisco Unified IP Phone 7911, 7941, 7941GE, 7961, 796GE, 7970,
and 7971, the firmware consists of multiple files including JAR and tone files. All of the firmware files
for each phone type must be downloaded the TFTP server before they can be downloaded to the phone.
The following example shows a list of phone firmware files that are installed in flash memory for the
Cisco Unified IP Phone 7911:
tftp server-flash:SCCP11.7-2-1-0S.loads
tftp server-flash:term06.default.loads
tftp server-flash:term11.default.loads
tftp server-flash:cvm11.7-2-0-66.sbn
tftp server-flash:jar11.7-2-0-66.sbn
tftp server-flash:dsp11.1-0-0-73.sbn
tftp server-flash:apps11.1-0-0-72.sbn
tftp server-flash:cnu11.3-0-0-81.sbn

However, you only specify the filename for the image file when configuring Cisco Unified CME. For
Java-based IP phones, the following naming conventions are used for image files:
• SCCP firmware—TERMnn.xx-y-z-ww or SCCPnn.xx-y-zz-ww, where n represents the phone type,
x represents the major version, y represents the major subversion, z represents the maintenance
version, and w represents the maintenance subversion.
The following example shows how to configure Cisco Unified CME so that the Cisco Unified IP Phone
7911 can download the appropriate SCCP firmware from flash memory:
Router(config)# telephony-service
Router(config-telephony)#load 7911 SCCP11.7-2-1-0S

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Table 5 contains firmware-naming convention examples, in alphabetical order:

.
Table 5 Firmware-Naming Conventions

SCCP Phones SIP Phones

Image Version Image Version
P00303030300 3.3(3) P0S3-04-4-00 4.4
P00305000200 5.0(2) P0S3-05-2-00 5.2
P00306000100 6.0(1) P0S3-06-0-00 6.0
SCCP41.8-0-4ES4-0-1S 8.0(4) SIP70.8-0-3S 8.0(3)
TERM41.7-0-3-0S 7.0(3) — —

The phone firmware filenames for each phone type and Cisco Unified CME version are listed in the
appropriate Cisco CME Supported Firmware, Platforms, Memory, and Voice Products document at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
For information about installing firmware files, see the “Installing Cisco Unified CME Software”
section on page 74.
For information about configuring Cisco Unified CME for upgrading between versions or converting
between SCCP and SIP, see the “How to Install and Upgrade Cisco Unified CME Software” section on
page 74.

IXML Template
The file called xml.template can be copied and modified to allow or restrict specific GUI functions to
customer administrators, a class of administrative users with limited capabilities in a
Cisco Unified CME system. This file is included in both tar archives (cme-basic-... and cme-gui-...). To
install the file, see the “Installing Cisco Unified CME Software” section on page 74.

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 tar archive with basic files (cme-basic-...). To install the file, see the
“Installing Cisco Unified CME Software” section on page 74.

Script Files
Archives containing Tcl script files are listed individually on the Cisco Unified CME software download
website. For example, the file named app-h450-transfer.2.0.0.9.zip.tar contains a script that adds H.450
transfer and forwarding support for analog FXS ports.
The Cisco Unified CME Basic Automatic Call Distribution and Auto Attendant Service (B-ACD)
requires a number of script files and audio files, which are contained in a tar archive with the name
cme-b-acd-.... For a list of files in the archive and for more information about the files, see the

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appropriate Cisco CME B-ACD and TCL Call-Handling Applications document at

http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
For information about installing TcL script file or an archive, see “Installing Cisco Unified CME
Software” on page 74.

Bundled TSP Archive

An archive is available at the Cisco Unified CME software download website that contains several
Telephony Application Programming Interface (TAPI) Telephony Service Provider (TSP) files. These
files are needed to set up individual PCs for Cisco Unified IP phone users who wish to make use of
Cisco Unified CME-TAPI integration with TAPI-capable PC software. To install the files from the
archive, see the installation instructions in the TSP documentation at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_programming_reference_guide09
186a00801c5f9c.html.

File Naming Conventions

Most of the files available at the Cisco Unified CME software download website are archives that must
be uncompressed before individual files can be copied to the router. In general, the following naming
conventions apply to files on the Cisco Unified CME software download website:
cme-basic-... Basic Cisco Unified CME files, including phone firmware files for a
particular Cisco Unified CME version or versions.
cme-gui-... Files required for the Cisco Unified CME GUI.
cmterm..., P00..., 7970.. Phone firmware files.
Note Not all firmware files to be downloaded to a phone are specified in
the load command. For a list of file names to be installed in flash
memory, and which file names are to be specified by using the load
command, see Cisco Unified CME Supported Firmware,
Platforms, Memory, and Voice Products at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/produc
ts_documentation_roadmap09186a0080189132.html.
cme-b-acd... Files required for Cisco Unified CME B-ACD service.

Cisco Unified Communications Express - Quick Configuration Tool

Cisco Unified Communications Express - Quick Configuration Tool (Cisco Unified
Communications Express - QCT) 3.0 or a later version is a GUI application provided for Cisco Partners
and Resellers designed to simplify and expedite the configuration of Cisco Unified Communications
Manager Express (Cisco Unified CME) and Cisco Unity Express by reducing the number of Cisco IOS
commands with which the user must be familiar in order to deploy a Cisco Unified Communications
Express telephony system.
Use Cisco Unified Communications Express - QCT to set up a simple, typical basic PBX or key system
telephony configuration of 50 or fewer IP phone users on any Cisco Unified CME supported Cisco
Integrated Services Router platform, including the Cisco 2800 and the Cisco 3800 product families. In

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addition, Cisco Unified Communications Express - QCT recognizes any Advanced Integrated Module
(AIM) or NM-CUE module installed in the router, thus providing voice-mail and Auto Attendant (AA)
capability to the Cisco Unified CME system.
After all the necessary information is entered, Cisco Unified Communications Express - QCT generates
all of the required configuration commands which you can upload to the Cisco router to be configured
or save as a template file to use to configure additional systems with similar system parameters.
For information about installing and using Cisco Unified Communications Express - QCT, see the
Cisco Unified Communications Express - QCT User Guide.

How to Install and Upgrade Cisco Unified CME Software

This section contains the following procedures:
• Installing Cisco Unified CME Software, page 74 (required)
• SCCP: Upgrading or Downgrading Phone Firmware Between Versions, page 76 (required)
• SIP: Upgrading or Downgrading Phone Firmware Between Versions, page 77 (required)
• SCCP: Converting Phone Firmware to SIP, page 81 (required)
• SIP: Converting Phone to SCCP, page 84 (required)
• SCCP: Verifying the Phone Firmware Version on an IP Phone, page 88 (optional)
• Troubleshooting Tips, page 88 (optional)

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

Installing Cisco Unified CME Software

To install Cisco Unified CME in flash memory, perform the following steps.

SUMMARY STEPS

1. Go to Software Download site.

2. Download archive.
3. Extract files to be downloaded.
4. Use the copy or archive tar command to copy file to flash memory.
5. Use the show flash: command to list files in flash memory.

DETAILED STEPS

Step 1 Go to http://www.cisco.com/cgi-bin/tablebuild.pl/ip-key.
Step 2 Select the file to download.
Step 3 Download zip file to tftp server.

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Step 4 Use the zip program to extract the file to be installed, then:
a. If the file is an individual file, use the copy command to copy the files to router flash:
Router# copy tftp://x.x.x.x/P00307020300.sbn flash:

b. If the file is a tar file, use the archive tar command to extract the files to flash memory.
Router# archive tar /xtract source-url flash:/file-url
Step 5 Verify the installation. Use the show flash: command to list the files installed in in flash memory.
Router# show flash:

31 128996 Sep 19 2005 12:19:02 -07:00 P00307020300.bin

32 461 Sep 19 2005 12:19:02 -07:00 P00307020300.loads
33 681290 Sep 19 2005 12:19:04 -07:00 P00307020300.sb2
34 129400 Sep 19 2005 12:19:04 -07:00 P00307020300.sbn

What to Do Next
• If you installed Cisco Unified CME software and Cisco Unified CME is not configured on your
router, see “Defining Network Parameters” on page 91.
• If Cisco Unified IP phones presently connected to Cisco Unified CME are using the SCCP protocol
to receive and place calls and the firmware version must be upgraded to a recommended version, or
if the phones to be connected to Cisco Unified CME are brand new, out-of-the-box, the phone
firmware preloaded at the factory must be upgraded to the recommended version before your phones
can complete registration, see the “SCCP: Upgrading or Downgrading Phone Firmware Between
Versions” section on page 76.
• If Cisco Unified IP phones presently connected to Cisco Unified CME are using the SIP protocol to
receive and place calls and the firmware version must be upgraded to a recommended version, see
the “SIP: Upgrading or Downgrading Phone Firmware Between Versions” section on page 77.
• If Cisco Unified IP phones presently connected to Cisco Unified CME are using the SCCP protocol
to receive and place calls and you now want some or all of these phones to use the SIP protocol, the
phone firmware for each phone type must be upgraded from SCCP to the recommended SIP version
before the phones can register. See the “SCCP: Converting Phone Firmware to SIP” section on
page 81.
• If Cisco Unified IP phones to be connected to Cisco Unified CME are using the SIP protocol and
are brand new, out-of-the-box, the phone firmware preloaded at the factory must be upgraded to the
recommended SIP version before your SIP phones can complete registration. See the “SCCP:
Converting Phone Firmware to SIP” section on page 81.
• If Cisco Unified IP phones presently connected to Cisco Unified CME are using the SIP protocol to
receive and place calls and you now want some or all of these phones to use the SCCP protocol, the
phone firmware for each phone type must be upgraded from SIP to the recommended SCCP version
before the phones can register. See the “SIP: Converting Phone to SCCP” section on page 84.

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SCCP: Upgrading or Downgrading Phone Firmware Between Versions

To downgrade or upgrade firmware versions on a Cisco Unified IP phone running SCCP, perform the
following steps.

Prerequisites
Phone firmware for Cisco Unified IP phones to be connected to Cisco Unified CME, including all
versions required during an upgrade or downgrade sequence, must be loaded in the flash memory of the
TFTP server from which the phones download their configuration profiles. For information about
installing firmware files in flash memory, see the “Installing Cisco Unified CME Software” section on
page 74.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. load phone-type firmware-file
5. create cnf
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 tftp-server flash: file-name Enables TFTP file sharing for new phone firmware files.
• A separate tftp-server flash command is required for
Example: each firmware file to be downloaded to this phone.
Router(config)# tftp-server
flash:P00307020300.loads Router(config)#
tftp-server flash:P00307020300.sb2
Router(config)# tftp-server
flash:P00307020300.sbn
Router(config)# tftp-server
flash:P00307020300.bin
Step 4 telephony service Enters telephone-service configuration mode.

Example:
Router(config)# telephony service

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Command or Action Purpose

Step 5 load phone-type firmware-file Associates a phone type with a phone firmware file.
• A separate load command is required for each IP phone
Example: type.
Router(config-telephony)# load 7960-7940
P00307020300

Step 6 create cnf-files Builds XML configuration files required for SCCP phones.

Example:
Router(config-telephony)# create cnf-files
Step 7 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-telephony)# end

What to Do Next
• If the Cisco Unified IP phone to be upgraded is not configured in Cisco Unified CME, see “How to
Configure Phones for a PBX System” on page 158.
• If the Cisco Unified IP phone is already configured in Cisco Unified CME and can make and receive
calls, you are ready to reboot the Cisco Unified IP phones to download the phone firmware to the
phone. See “Resetting and Restarting Phones” on page 257.

SIP: Upgrading or Downgrading Phone Firmware Between Versions

To upgrade or downgrade phone firmware for Cisco Unified IP phones running SIP between versions,
perform the steps in this section.
The upgrade and downgrade sequences for SIP phones differ per phone type as follows:
• Upgrading/downgrading the phone firmware for Cisco Unified IP Phone 7905G, Cisco Unified IP
Phone 7912G, and Cisco ATA Analog Telephone Adapter is straightforward; modify the load
command to upgrade directly to the target load.
• The phone firmware version upgrade sequence for Cisco Unified IP Phone 7940Gs and 7960Gs is
from version [234].x to 4.4, to 5.3, to 6.x, to 7.x. You cannot go directly from version [234].x to
version 7.x.
• To downgrade phone firmware for Cisco Unified IP Phone 7940Gs and 7960Gs, first upgrade to
version 7.x, then modify the load command to downgrade directly to the target phone firmware.

Prerequisites
Phone firmware for Cisco Unified IP phones to be connected to Cisco Unified CME, including all
versions required during an upgrade or downgrade sequence, must be loaded in the flash memory of the
TFTP server from which the phones will download their configuration profiles. For information about
installing firmware files in flash memory, see the “Installing Cisco Unified CME Software” section on
page 74.

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Restrictions
• Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7912G, and Cisco ATA—Signed load starts
from SIP v1.1. After you upgrade the firmware to a signed load, you cannot downgrade the firmware
to an unsigned load.
• Cisco Unified IP Phone 7940G and Cisco Unified IP Phone 7960G—Signed load starts from SIP
v5.x. Once you upgrade the firmware to a signed load, you cannot downgrade the firmware to an
unsigned load.
• The procedures for upgrading phone firmware files for SIP phones is the same for all
Cisco Unified IP phones. For other limits on firmware upgrade between versions, see the phone
firmware upgrade matrix at:
http://www.cisco.com/en/US/products/sw/voicesw/ps4967/prod_installation_guides_list.html.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. mode cme
5. load phone-type firmware-file
6. upgrade
7. Repeat Steps 5 and 6.
8. file text
9. create profile
10. exit
11. voice register pool tag
12. reset
13. exit
14. voice register global
15. no upgrade
16. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 mode cme Enables mode for provisioning SIP phones in
Cisco Unified CME.
Example:
Router(config-register-global)# mode cme
Step 5 load phone-type firmware-file Associates a phone type with a phone firmware file.
• A separate load command is required for each IP phone
Example: type.
Router(config-register-global)# load 7960-7940
P0S3-06-0-00
• firmware-file—Filename to be associated with the
specified Cisco Unified IP phone type.
• Do not use the .sbin or .loads file extension except for
Cisco ATA and Cisco Unified IP Phone 7905 and 7912
Step 6 upgrade Generates a file with the universal application loader image
for upgrading phone firmware and performs the TFTP
server alias binding.
Example:
Router(config-register-global)# upgrade
Step 7 Repeat previous two steps. (Optional) Repeat for each version required in multistep
upgrade sequences only.
Example:
Router(config-register-global)# load 7960-7940
P0S3-07-4-00
Router(config-register-global)# upgrade
Step 8 file text (Optional) Generates ASCII text files for Cisco Unified IP
Phone 7905s and 7905Gs, Cisco Unified IP Phone 7912s
and 7912Gs, Cisco ATA-186, or Cisco ATA-188.
Example:
Router(config-register-global)# file text • Default—System generates binary files to save disk
space.
Step 9 create profile Generates provisioning files required for SIP phones and
writes the file to the location specified with the tftp-path
command.
Example:
Router(config-register-global;)# create profile
Step 10 exit Exits from the current command mode to the next highest
mode in the configuration mode hierarchy.
Example:
Router(config-register-global)# exit
Step 11 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for SIP phones.
Example: • pool-tag—Unique sequence number of the SIP phone
Router(config)# voice register pool 1 to be configured. Range is 1 to 100 or the upper limit as
defined by max-pool command.

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Command or Action Purpose

Step 12 reset Performs a complete reboot of the single SIP phone
specified with the voice register pool command and
contacts the DHCP server and the TFTP server for updated
Example:
Router(config-register-pool)# reset
information.
Step 13 exit Exits from the current command mode to the next highest
mode in the configuration mode hierarchy.
Example:
Router(config-register-pool)# exit
Step 14 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 15 no upgrade Return to the default for the upgrade command.

Example:
Router(config-register-global)# no upgrade
Step 16 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end

Examples
The following example shows the configuration steps for upgrading firmware for a Cisco Unified IP
Phone 7960G or Cisco Unified IP Phone 7940G from SIP 5.3 to SIP 6.0, then from SIP 6.0 to SIP 7.4:
Router(config)# voice register global
Router(config-register-global)# mode cme
Router(config-register-global)# load 7960 P0S3-06-0-00
Router(config-register-global)# upgrade
Router(config-register-global)# load 7960 P0S3-07-4-00
Router(config-register-global)# create profile

The following example shows the configuration steps for downgrading firmware for a Cisco Unified IP
Phone 7960/40 from SIP 7.4 to SIP 6.0:
Router(config)# voice register global
Router(config-register-global)# mode cme
Router(config-register-global)# load 7960 P0S3-06-0-00
Router(config-register-global)# upgrade
Router(config-register-global)# create profile

What to Do Next
• If the Cisco Unified IP phone to be upgraded is not configured in Cisco Unified CME, see “How to
Configure Phones for a PBX System” on page 158.
• If the Cisco Unified IP phone is already configured in Cisco Unified CME and can make and receive
calls, you are ready to reboot the Cisco Unified IP phones to download the phone firmware to the
phone. See “Resetting and Restarting Phones” on page 257.

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SCCP: Converting Phone Firmware to SIP

To upgrade the phone firmware for a particular phone from SCCP to SIP, follow the steps in this task.
If Cisco Unified IP phones presently connected to Cisco Unified CME are using the SCCP protocol to
receive and place calls and you now want some or all of these phones to use the SIP protocol, the phone
firmware for each phone type must be upgraded from SCCP to the recommended SIP version before the
phones can register. If Cisco Unified IP phones to be connected to Cisco Unified CME are brand new,
out-of-the-box, the SCCP phone firmware preloaded at the factory must be upgraded to the
recommended SIP version before your SIP phones can complete registration.

Note If codec values for the dial peers of a connection do not match, the call fails. The default codec for the
POTS dial peer for an SCCP phone is G.711 and the default codec for a VoIP dial peer for a SIP phone
is G.729. If neither the SCCP phone nor the SIP phone in Cisco Unified CME has been specifically
configured to change the codec, calls between the two IP phones on the same router will produce a busy
signal caused by the mismatched default codecs. To avoid codec mismatch, specify the codec for IP
phones in Cisco Unified CME. For configuration information, see the “Configuring Codec for Local
Calling Between SIP and SCCP Phones” section on page 174.

Prerequisites
• Phone firmware for Cisco Unified IP phones to be connected to Cisco Unified CME, including all
versions required during an upgrade or downgrade sequence, must be loaded in the flash memory of
the TFTP server from which the phones download their configuration profiles. For information
about installing firmware files in flash memory, see the “Installing Cisco Unified CME Software”
section on page 74.
• Cisco Unified IP Phone 7940Gs and Cisco Unified IP Phone 7960Gs—If these IP phones are
already configured in Cisco Unified CME to use the SCCP protocol, the SCCP phone firmware on
the phone must be version 5.x. If required, upgrade the SCCP phone firmware to 5.x before
upgrading to SIP.

SUMMARY STEPS

1. enable
2. configure terminal
3. no ephone ephone-tag
4. exit
5. no ephone-dn dn-tag
6. exit
7. voice register global
8. mode cme
9. load phone-type firmware-file
10. upgrade
11. Repeat previous two steps.
12. create profile

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13. file text

14. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 no ephone ephone-tag (Optional) Disables the ephone and removes the ephone
configuration.
Example: • Required only if the Cisco Unified IP phone to be
Router (config)# no ephone 23 configured is already connected to Cisco Unified CME
and is using SCCP protocol.
• ephone-tag—Particular IP phone to which this
configuration change will apply.
Step 4 exit (Optional) Exits from the current command mode to the
next highest mode in the configuration mode hierarchy.
Example: • Required only if you performed the previous step.
Router(config-ephone)# exit
Step 5 no ephone-dn dn-tag (Optional) Disables the ephone-dn and removes the
ephone-dn configuration.
• Required only if this directory number is not now nor
will be associated to any SCCP phone line, intercom
line, paging line, voice-mail port, or message-waiting
indicator (MWI) connected to Cisco Unified CME.
• dn-tag—Particular configuration to which this change
will apply.
Step 6 exit (Optional) Exits from the current command mode to the
next highest mode in the configuration mode hierarchy.
Example: • Required only if you performed the previous step.
Router(config-ephone-dn)# exit
Step 7 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 8 mode cme Enables mode for provisioning SIP phones in
Cisco Unified CME.
Example:
Router(config-register-global)# mode cme

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Command or Action Purpose

Step 9 load phone-type firmware-file Associates a phone type with a phone firmware file.
• A separate load command is required for each IP phone
Example: type.
Router(config-register-global)# load 7960-7940
P0S3-06-3-00
Step 10 upgrade Generates a file with the universal application loader image
for upgrading phone firmware and performs the TFTP
server alias binding.
Example:
Router(config-register-global)# upgrade
Step 11 Repeat previous two steps (Optional) Repeat for each version required in multistep
upgrade sequences only.
Example:
Router(config-register-global)# load 7960-7940
P0S3-07-4-00
Router(config-register-global)# upgrade

Step 12 create profile Generates provisioning files required for SIP phones and
writes the file to the location specified with the tftp-path
command.
Example:
Router(config-register-global;)# create profile
Step 13 file text (Optional) Generates ASCII text files for Cisco Unified IP
Phones 7905 and 7905G, Cisco Unified IP Phone 7912 and
Cisco Unified IP Phone 7912G, Cisco ATA-186, or
Example:
Router(config-register-global)# file text
Cisco ATA-188.
• Default—System generates binary files to save disk
space.
Step 14 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end

Examples
The following example shows the configuration steps for converting firmware on an Cisco Unified IP
phone already connected in Cisco Unified CME and using the SCCP protocol, from SCCP 5.x to SIP 7.4:
Router(config)# telephony-service
Router(config-telephony)# no create cnf
CNF files deleted
Router(config-telephony)# voice register global
Router(config-register-global)# mode cme
Router(config-register-global)# load 7960 P0S3-07-4-00
Router(config-register-global)# upgrade
Router(config-register-global)# create profile

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How to Install and Upgrade Cisco Unified CME Software

What to Do Next
After you configure the upgrade command, refer to the following statements to determine which task
to perform next.
• If the Cisco Unified IP phone to be upgraded is already connected in Cisco Unified CME and you
removed the SCCP configuration file for the phone but have not configured this phone for SIP in
Cisco Unified CME, see “How to Configure Phones for a PBX System” on page 158.
• If the Cisco Unified IP phones to be upgraded are already configured in Cisco Unified CME, see
“Resetting and Restarting Phones” on page 257.

SIP: Converting Phone to SCCP

To upgrade the phone firmware for a particular phone from SIP to SCCP, follow the steps in this task.
If Cisco Unified IP phones presently connected to Cisco Unified CME are using the SIP protocol to
receive and place calls and you now want some or all of these phones to use the SCCP protocol, the
phone firmware for each phone type must be upgraded from SIP to SCCP before the phones can register.

Note If codec values for the dial peers of a connection do not match, the call fails. The default codec for the
POTS dial peer for an SCCP phone is G.711 and the default codec for a VoIP dial peer for a SIP phone
is G.729. If neither the SCCP phone nor the SIP phone in Cisco Unified CME has been specifically
configured to change the codec, calls between the two IP phones on the same router will produce a busy
signal caused by the mismatched default codecs. To avoid codec mismatch, specify the codec for SIP and
SCCP phones in Cisco Unified CME. For more information, see “How to Configure Phones for a PBX
System” on page 158.

Prerequisites
• Phone firmware for Cisco Unified IP phones to be connected to Cisco Unified CME, including all
versions required during an upgrade or downgrade sequence, must be loaded in the flash memory of
the TFTP server from which the phones will download their configuration profiles. For information
about installing firmware files in flash memory, see the “Installing Cisco Unified CME Software”
section on page 74.
• Cisco Unified IP Phone 7940Gs and Cisco Unified IP Phone 7960Gs—If these IP phones are
already configured in Cisco Unified CME to use the SIP protocol, the SIP phone firmware must be
version 7.x. See the “SIP: Upgrading or Downgrading Phone Firmware Between Versions” section
on page 77.

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Removing a SIP Configuration Profile

To remove the SIP configuration profile before downloading the SCCP phone firmware to convert a
phone from SIP to SCCP, perform the steps in this task.

SUMMARY STEPS

1. enable
2. configure terminal
3. no voice register pool pool-tag
4. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 no voice register pool pool-tag Disables voice register pool and removes the voice pool
configuration.
Example: • pool-tag—Unique sequence number for a particular
Router(config)# no voice register pool 1 SIP phone to which this configuration change will
apply.
Step 4 end Exits from the current command mode to the next highest
mode in the configuration mode hierarchy.
Example:
Router(config-register-pool)# end

Generating an SCCP XML Configuration File for Upgrading from SIP to SCCP
To create an ephone entry and generate a new SCCP XML configuration file for upgrading a particular
Cisco Unified IP phone in Cisco Unified CME from SIP to SCCP, perform the steps in this task.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag
4. exit
5. tftp-server flash firmware-file
6. telephony service

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7. load phone-type firmware-file

8. create cnf-files
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 1 ephone-dn during configuration tasks. The maximum
number of ephone-dns in Cisco Unified CME is version
and platform specific. Type ? to display range.
Step 4 exit Exits from the current command mode to the next highest
mode in the configuration mode hierarchy.
Example:
Router(config-ephone-dn)# exit
Step 5 tftp-server flash: file-name Enables TFTP file sharing for new phone firmware files.
• A separate tftp-server flash command is required for
Example: each firmware file to be downloaded to this phone.
Router(config)# tftp-server
flash:P00307020300.loads
Router(config)# tftp-server
flash:P00307020300.sb2
Router(config)# tftp-server
flash:P00307020300.sbn
Router(config)# tftp-server
flash:P00307020300.bin
Step 6 telephony service Enters telephone-service configuration mode.

Example:
Router(config)# telephony service
Step 7 load phone-type firmware-file Associates a phone type with a phone firmware file.
• A separate load command is required for each IP phone
Example: type.
Router(config-telephony)# load 7960-7940
P00307020300
• firmware-file—Filename to be associated with the
specified IP phone type.
• Do not use the .sbin or .loads file extension except for
Cisco ATA and Cisco Unified IP Phone 7905 and 7912

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Command or Action Purpose

Step 8 create cnf-files Builds XML configuration files required for SCCP phones.

Example:
Router(config-telephony)# create cnf-files
Step 9 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-telephony)# end

Examples
The following example shows the configuration steps for upgrading firmware for a Cisco Unified IP
Phone 7960G from SIP to SCCP. First the SIP firmware is upgraded to SIP 6.3 and from SIP 6.3 to SIP
7.4; then, the phone firmware is upgraded from SIP 7.4 to SCCP 7.2(3). The SIP configuration profile is
deleted and a new ephone configuration profile is created for the Cisco Unified IP phone.
Router(config)# voice register global
Router(config-register-global)# mode cme
Router(config-register-global)# load 7960 P0S3-06-0-00
Router(config-register-global)# upgrade
Router(config-register-global)# load 7960 P0S3-07-4-00
Router(config-register-global)# exit
Router(config)# no voice register pool 1
Router(config-register-pool)# exit
Router(config)# voice register global
Router(config-register-global)# no upgrade
Router(config-register-global)# exit
Router(config)# ephone-dn 1
Router(config-ephone-dn)# exit
Router(config)# tftp-server flash:P00307020300.loads
Router(config)# tftp-server flash:P00307020300.sb2
Router(config)# tftp-server flash:P00307020300.sbn
Router(config)# tftp-server flash:P00307020300.bin
Router(config)# telephony service
Router(config-telephony)# load 7960-7940 P00307000100
Router(config-telephony)# create cnf-files

What to Do Next
After you configure the upgrade command:
• If the Cisco Unified IP phone to be upgraded is already connected in Cisco Unified CME and you
removed the SIP configuration file for the phone and have not configured the SCCP phone in
Cisco Unified CME, see “How to Configure Phones for a PBX System” on page 158.
• If the Cisco Unified IP phones to be upgraded are already configured in Cisco Unified CME, see
“Resetting and Restarting Phones” on page 257.

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SCCP: Verifying the Phone Firmware Version on an IP Phone

To verify which version firmware is on an IP phone, perform the following steps.

SUMMARY STEPS

1. show flash:
2. show ephone phone-load

DETAILED STEPS

Step 1 show flash:

Use this command to learn the filenames associated with that phone firmware
Router# show flash:

31 128996 Sep 19 2005 12:19:02 -07:00 P00307020300.bin

32 461 Sep 19 2005 12:19:02 -07:00 P00307020300.loads
33 681290 Sep 19 2005 12:19:04 -07:00 P00307020300.sb2
34 129400 Sep 19 2005 12:19:04 -07:00 P00307020300.sbn

Step 2 show ephone phone-load

Use this command to verify which phone firmware is installed on a particular ephone. The DeviceName
includes the MAC address for the IP phone.
Router# show ephone phone-load

DeviceName CurrentPhoneload PreviousPhoneload LastReset

=====================================================================
SEP000A8A2C8C6E 7.3(3.02) Initialized

Troubleshooting Tips
Use the debug tftp event command to troubleshoot an attempt to upgrade or convert Cisco phone
firmware files for SIP phones. The following sample from the debug tftp event command shows how
the Cisco phone firmware for a Cisco Unified IP Phone 7940G is upgraded from SCCP 5.X to SIP 6.3.
The configuration profiles are downloaded when a phone is rebooted or reset.
Router# debug tftp event
…
Router(config)#telephony-service
Router(config-telephony)#no create cnf
CNF files deleted
Router(config-telephony)#voice register global
Router(config-register-global)#load 7960 P0S3-06-3-00
Router(config-register-global)#upgrade
Router(config-register-global)#create profile
Router(config-register-global)#
*May 6 17:37:03.737: %IPPHONE-6-UNREGISTER_NORMAL: ephone-1:SEP000ED7DF7932 IP:1.5.49.84
Socket:4
DeviceType:Phone has unregistered normally.
*May 6 17:37:35.949: TFTP: Looking for OS79XX.TXT
*May 6 17:37:36.413: TFTP: Opened system:/cme/sipphone/OS79XX.TXT, fd 4, size 13 for
process 81

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*May 6 17:37:36.413: TFTP: Finished system:/cme/sipphone/OS79XX.TXT, time 00:00:00 for

process 81
*May 6 17:37:40.533: TFTP: Looking for P0S3-06-3-00.sbn
*May 6 17:37:40.541: TFTP: Opened flash:P0S3-06-3-00.sbn, fd 4, size 487198 for process 81
*May 6 17:37:48.225: TFTP: Finished flash:P0S3-06-3-00.sbn, time 00:00:07 for process 81
*May 6 17:40:26.925: TFTP: Looking for OS79XX.TXT
*May 6 17:40:26.925: TFTP: Opened system:/cme/sipphone/OS79XX.TXT, fd 4, size 13 for
process 81
*May 6 17:40:26.925: TFTP: Finished system:/cme/sipphone/OS79XX.TXT, time 00:00:00 for
process 81
*May 6 17:40:26.929: TFTP: Looking for SIPDefault.cnf
*May 6 17:40:26.929: TFTP: Opened system:/cme/sipphone/SIPDefault.cnf, fd 4, size 1558 for
process 81
*May 6 17:40:26.937: TFTP: Finished system:/cme/sipphone/SIPDefault.cnf, time 00:00:00 for
process 81
*May 6 17:40:27.053: TFTP: Looking for SIP000ED7DF7932.cnf
*May 6 17:40:27.053: TFTP: Opened system:/cme/sipphone/SIP000ED7DF7932.cnf, fd 4, size 789
for process 81
*May 6 17:40:27.057: TFTP: Finished system:/cme/sipphone/SIP000ED7DF7932.cnf, time
00:00:00 for process 81

The following sample from the debug tftp event command shows how the Cisco phone firmware for a
Cisco Unified IP Phone 7940G is upgraded from SIP 6.3 to SIP 7.0 after the phone is rebooted or reset:
Router# debug tftp event
…
Router(config-register-global)#load 7960 P003-07-4-00
Router(config-register-global)#upgrade
Router(config-register-global)#load 7960 P0S3-07-4-00
Router(config-register-global)#create profile
Router(config-register-global)#end
Router-2012#
*May 6 17:42:35.581: TFTP: Looking for OS79XX.TXT
*May 6 17:42:35.585: TFTP: Opened system:/cme/sipphone/OS79XX.TXT, fd 5, size 13 for
process 81
*May 6 17:42:35.585: TFTP: Finished system:/cme/sipphone/OS79XX.TXT, time 00:00:00 for
process 81
*May 6 17:42:35.969: TFTP: Looking for P003-07-4-00.sbn
*May 6 17:42:35.977: TFTP: Opened slot0:P003-07-4-00.sbn, fd 5, size 129876 for process 81
*May 6 17:42:37.937: TFTP: Finished slot0:P003-07-4-00.sbn, time 00:00:01 for process 81
*May 6 17:44:31.037: TFTP: Looking for CTLSEP000ED7DF7932.tlv
*May 6 17:44:31.057: TFTP: Looking for SEP000ED7DF7932.cnf.xml
*May 6 17:44:31.089: TFTP: Looking for SIP000ED7DF7932.cnf
*May 6 17:44:31.089: TFTP: Opened system:/cme/sipphone/SIP000ED7DF7932.cnf, fd 5, size 789
for process 81
*May 6 17:44:31.089: TFTP: Finished system:/cme/sipphone/SIP000ED7DF7932.cnf, time
00:00:00 for process 81
*May 6 17:44:31.125: TFTP: Looking for P0S3-07-4-00.loads
*May 6 17:44:31.133: TFTP: Opened slot0:P0S3-07-4-00.loads, fd 5, size 461 for process 81
*May 6 17:44:31.141: TFTP: Finished slot0:P0S3-07-4-00.loads, time 00:00:00 for process 81
*May 6 17:44:31.673: TFTP: Looking for P0S3-07-4-00.sb2
*May 6 17:44:31.681: TFTP: Opened slot0:P0S3-07-4-00.sb2, fd 5, size 592626 for process 81
*May 6 17:44:33.989: TFTP: Finished slot0:P0S3-07-4-00.sb2, time 00:00:02 for process 81

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
• Cisco Unified Communications Express - QCT User Guide
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Last Updated: March 26, 2007

This chapter describes how to define parameters that enable Cisco Unified Communications Manager
Express (Cisco Unified CME) to work with your network.

Note If you used Cisco Unified Communications Express - QCT to generate a basic telephony configuration,
you can skip this module unless you want to modify the configuration to relay DHCP requests from IP
phones to a DHCP server on a different router.

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Network Parameters”
section on page 117.

Contents
• Prerequisites for Defining Network Parameters, page 91
• Information About Defining Network Parameters, page 92
• How to Define Network Parameters, page 95
• Configuration Examples for Network Parameters, page 114
• Where to Go Next, page 115
• Additional References, page 115
• Feature Information for Network Parameters, page 117

Prerequisites for Defining Network Parameters

• IP routing must be enabled.
• VoIP networking must be operational. For quality and security purposes, we recommend you 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 Unified 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.

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Restrictions for Defining Network Parameters

• If applicable, PSTN lines are configured and operational.

• If applicable, the WAN links are configured and operational.
• Trivial File Transfer Protocol (TFTP) must be enabled on the router to allow IP phones to download
phone firmware files.
• To support IP phones that are running SIP to be directly connected to the Cisco Unified CME router,
Cisco Unified CME 3.4 or later must be installed on the router. For installation information, see
“Installing and Upgrading Cisco Unified CME Software” on page 69.
• To provide voice-mail support for phones connected to the Cisco Unified CME router, install and
configure voice mail on your network.

Restrictions for Defining Network Parameters

In Cisco Unified CME 4.0 and later versions, Layer-3-to-Layer-2 VLAN Class of Service (CoS) priority
marking is not automatically processed. Cisco Unified CME 4.0 and later versions will continue to mark
Layer 3, but Layer 2 marking is now only handled in the Cisco IOS software. Any Quality of Service
(QoS) design that requires Layer 2 marking will have to be explicitly configured, either on a Catalyst
switch that supports this capability or on the Cisco Unified CME router under the Ethernet interface
configuration. For configuration information, see the Enterprise QoS Solution Reference Network
Design Guide.

Information About Defining Network Parameters

To configure network parameters, you should understand the following concepts:
• DHCP Service, page 92
• Network Time Protocol for the Cisco Unified CME Router, page 93
• DTMF Relay, page 93
• SIP Register Support, page 93
• Out-of-Dialog REFER, page 94

DHCP Service
When a Cisco Unified IP phone is connected to the Cisco Unified 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 Unified IP phone and providing the IP address of the TFTP server through DHCP
option 150. Then the phone registers with the Cisco Unified CME server and attempts to get
configuration and phone firmware files from the TFTP server.
For configuration information, perform only one of the following procedures to set up DHCP service for
your IP phones:
• If your Cisco Unified CME router is the DHCP server and you can use a single shared address pool
for all your DHCP clients, see the “Defining a Single DHCP IP Address Pool” section on page 98.
• If your Cisco Unified CME router is the DHCP server and you need separate pools for non-IP-phone
DHCP clients, see the “Defining a Separate DHCP IP Address Pool for Each DHCP Client” section
on page 100.

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• If the Cisco Unified 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, see the “Defining a DHCP Relay” section on
page 102.

Network Time Protocol for the Cisco Unified CME Router

Network Time Protocol (NTP) allows you to synchronize your Cisco Unified CME router to a single
clock on the network, known as the clock master. NTP is disabled on all interfaces by default, but it is
essential for Cisco Unified CME so you must ensure that it is enabled. For information about
configuring NTP for the Cisco Unified CME router, see the “Enabling Network Time Protocol on the
Cisco Unified CME Router” section on page 104.

DTMF Relay
IP phones connected to Cisco Unified 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 DTMF 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 information about configuring a DTMF relay in a multisite installation,
see the “Configuring DTMF Relay for H.323 Networks in Multisite Installations” section on page 105.
To use remote voice-mail or IVR applications on SIP networks from Cisco Unified CME phones, the
DTMF digits used by the Cisco Unified CME phones must be converted to the RFC 2833 in-band DTMF
relay mechanism used by SIP phones. The SIP DTMF relay method is needed in the following situations:
• When SIP is used to connect a Cisco Unified CME system to a remote SIP-based IVR or voice-mail
application.
• When SIP is used to connect a Cisco Unified CME system to a remote SIP-PSTN voice gateway that
goes through the PSTN to a voice-mail or IVR application.
The requirement for 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 use voice mail on a SIP network that connects to a Cisco Unity Express system, which uses a
nonstandard SIP Notify format, the DTMF digits used by the Cisco Unified CME phones must be
converted to the Notify format. Additional configuration may be required for backward compatibility
with Cisco CME 3.0 and 3.1. For configuration information about enabling DTMF relay for SIP
networks, see “Configuring SIP Trunk Support” section on page 106.

SIP Register Support

SIP register support enables 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.

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Information About Defining Network Parameters

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 configuration information, see the “Basic SIP Configuration” chapter in the
Cisco IOS SIP Configuration Guide.

Note No commands allow registration between the H.323 and SIP protocols.

By default, SIP gateways do not generate SIP Register messages, so the gateway must be configured to
register the gateway’s E.164 telephone numbers with an external SIP registrar. For information about
configuring the SIP gateway to register phone numbers with Cisco Unified CME, see the “Configuring
SIP Trunk Support” section on page 106.

Out-of-Dialog REFER
Out-of-dialog REFER (OOD-R) allows remote applications to establish calls by sending a REFER
message to Cisco Unified CME without an initial INVITE. After the REFER is sent, the remainder of
the call setup is independent of the application and the media stream does not flow through the
application. The application using OOD-R triggers a call setup request that specifies the Referee address
in the Request-URI and the Refer-Target in the Refer-To header. The SIP messaging used to
communicate with Cisco Unified CME is independent of the end-user device protocol which can be SIP,
SCCP, H.323, or POTS. Click-to-dial is an example of an application that can be created using OOD-R.
A click-to-dial application allows users to combine multiple steps into one click for a call setup. For
example, a user can click a web-based directory application from their PC to look up a telephone
number, off-hook their desktop phone, and dial the called number. The application initiates the call setup
without the user having to out-dial from their own phone. The directory application sends a REFER
message to Cisco Unified CME which sets up the call between both parties based on this REFER.
Figure 6 shows an example of OOD-R being used by a click-to-dial application. In this scenario, the
following events occur (refer to the event numbers in the illustration):
1. Remote user clicks to dial.
2. Application sends out-of-dialog REFER to Cisco Unified CME 1.
3. Cisco Unified CME 1 connects to SIP phone 1 (Referee).
4. Cisco Unified CME 1 sends INVITE to Cisco Unified CME 2.
5. Cisco Unified CME 2 sends INVITE to SIP phone 2 (Refer-Target) and the call is accepted.
6. Voice path is created between the two SIP phones.

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Figure 6 Click-to-Dial Application using Out-of-Dialog REFER

Cisco Unified CME 2

Directory SIP 4
services
application
6

1 IP
6 5
2

IP 3 IP
IP phone 1 IP phone 2
6

Cisco Unified CME 1

155789
PSTN

The initial OOD-R request can be authenticated and authorized using RFC 2617-based digest
authentication. To support authentication, Cisco Unified CME retrieves the credential information from
a text file stored in flash. This mechanism is used by Cisco Unified CME in addition to phone-based
credentials. The same credential file can be shared by other services that require request-based
authentication and authorization such as presence service. Up to five credential files can be configured
and loaded into the system. The contents of these five files are mutually exclusive, meaning the username
and password pairs must be unique across all the files. The username and password pairs must also be
different than those configured for SCCP or SIP phones in a Cisco Unified CME system.
For configuration information, see the “Enabling OOD-R” section on page 110.

How to Define Network Parameters

This section contains the following tasks. You may not need to perform all of these procedures.
• Enabling Calls in Your VoIP Network, page 96 (required)
• Defining DHCP, page 98 (required)
• Enabling Network Time Protocol on the Cisco Unified CME Router, page 104 (required)
• Configuring DTMF Relay for H.323 Networks in Multisite Installations, page 105 (optional)
• Configuring SIP Trunk Support, page 106 (optional)
• Verifying SIP Trunk Support Configuration, page 108 (optional)
• Changing the TFTP Address on a DHCP Server, page 109 (optional)
• Enabling OOD-R, page 110 (optional)
• Verifying OOD-R Configuration, page 112 (optional)
• Troubleshooting OOD-R, page 113 (optional)

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Enabling Calls in Your VoIP Network

To enable calls between endpoints in Cisco Unified CME, perform the following steps.

Restrictions
Cisco Unified CME 3.4 and later versions support Media Flow-through mode only; enabling SIP-to-SIP
calls is required before you can successfully make SIP-to-SIP calls.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice service voip
4. allow-connections from-type to to-type
5. sip
6. registrar server [expires [max sec] [min sec]
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip Enters voice service configuration mode and specifies Voice
over IP (VoIP) encapsulation.
Example:
Router(config)# voice service voip
Step 4 allow-connections from-type to to-type Enables calls between specific types of endpoints in a VoIP
network.
Example: • A separate allow-connections command is required for
Router(config-voi-srv)# allow-connections h323 each type of endpoint to be supported.
to h323
Router(config-voi-srv)# allow-connections h323
to SIP
Router(config-voi-srv)# allow-connections SIP
to SIP
Step 5 sip (Optional) Enters SIP configuration mode.
• Required if you are connecting IP phones running SIP
Example: directly in Cisco CME 3.4 and later.
Router(config-voi-srv)# sip

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Command or Action Purpose

Step 6 registrar server [expires [max sec][min sec]] (Optional) Enables SIP registrar functionality in
Cisco Unified CME.
Example: • Required if you are connecting IP phones running SIP
Router(config-voi-sip)# registrar server directly in Cisco CME 3.4 and later.
expires max 600 min 60
• Cisco Unified CME does not maintain a persistent
database of registration entries across reloads. If the
WAN is down and you reboot your Cisco Unified CME
router, when the router reloads it will have no database
of SIP phone registrations. The SIP phones will have to
register again, which could take several minutes, as SIP
phones do not use a keepalive functionality. We
recommend that you change the expiry
• max sec—(Optional) Range: 600 to 86400.
Default: 3600. Recommended value: 600.
• min sec—(Optional) Range: 60 to 3600. Default: 60.
Step 7 exit Exits dial-peer configuration mode.

Example:
Router(config-voi-sip)# exit
Step 8 sip-ua Enters SIP user-agent configuration mode.

Example:
Router(config)# sip-ua
Step 9 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—Range: 500 to 3000.
max-duration 2000 Default: 2000.
Step 10 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:
Router(config-sip-ua)# registrar
ipv4:10.8.17.40 expires 3600 secondary
Step 11 retry register number Sets the total number of SIP Register messages that the
gateway should send.
Example: • number—Number of Register message retries.
Router(config-sip-ua)# retry register 10 Range: 1 to 10. Default: 10.

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Command or Action Purpose

Step 12 timers register time Sets how long the SIP user agent (UA) waits before sending
Register requests.
Example: • time—Waiting time, in milliseconds.
Router(config-sip-ua)# timers register 500 Range: 100 to 1000. Default: 500.
Step 13 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-voi-sip)# end

Defining DHCP
To set up DHCP service for your DHCP clients, perform only one of the following procedures:
• If your Cisco Unified CME router is the DHCP server and you can use a single shared address pool
for all your DHCP clients, see Defining a Single DHCP IP Address Pool, page 98.
• If your Cisco Unified CME router is the DHCP server and you need separate pools for each IP phone
and each non-IP-phone DHCP client, see Defining a Separate DHCP IP Address Pool for Each
DHCP Client, page 100.
• If the Cisco Unified 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, see Defining a DHCP Relay, page 102.

Defining a Single DHCP IP Address Pool

To create a shared pool of IP addresses for all DHCP clients, perform the following step.

Note Do not perform this task if you already have a DHCP server on the LAN that can be used to provide
addresses to the Cisco Unified CME phones. See the “Enabling Network Time Protocol on the Cisco
Unified CME Router” section on page 104.

Prerequisites

Your Cisco Unified CME router is a DHCP server.

Restrictions

A single DHCP IP address pool cannot be used if non-IP-phone clients, such as PCs, must use a different
TFTP server address.

SUMMARY STEPS

1. enable
2. configure terminal
3. ip dhcp pool pool-name
4. network ip-address [mask | /prefix-length]
5. option 150 ip ip-address

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6. default-router ip-address
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 4 network ip-address [mask | /prefix-length] Specifies the IP address of the DHCP address pool
to be configured.
Example:
Router(config-dhcp)# network 10.0.0.0 255.255.0.0
Step 5 option 150 ip ip-address Specifies the TFTP server address from which the
Cisco Unified IP phone downloads the image
configuration file.
Example:
Router(config-dhcp)# option 150 ip 10.0.0.1 • This is your Cisco Unified CME router’s
address.
Step 6 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 Unified CME router is the only
router on the network, this address should be the
Cisco Unified 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
purposes. If the Cisco Unified CME IP source
address becomes unreachable, IP phones will
attempt to register to the address specified in
this command.
Step 7 end Returns to privileged EXEC mode.

Example:
Router(config-dhcp)# end

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What to Do Next

• If you are configuring Cisco Unified CME for the first time on this router, you are ready to configure
NTP for the Cisco Unified CME router. See the “Enabling Network Time Protocol on the Cisco
Unified CME Router” section on page 104.
• If you are finished modifying network parameters for an already configured Cisco Unified CME
router, see “Generating Configuration Files for Phones” on page 245.

Defining a Separate DHCP IP Address Pool for Each DHCP Client

To create a DHCP IP address pool for each DHCP client, including non-IP-phone clients such as PCs,
perform the following steps.

Note Do not perform this task if you already have a DHCP server on the LAN that can be used to provide
addresses to the Cisco Unified CME phones. See the “Enabling Network Time Protocol on the Cisco
Unified CME Router” section on page 104.

Prerequisites

Your Cisco Unified CME router is a DHCP server.

Restrictions

To use a separate DHCP IP address pool for each DHCP client, you must make an entry for every IP
phone.

SUMMARY STEPS

1. enable
2. configure terminal
3. ip dhcp pool pool-name
4. host ip-address subnet-mask
5. client-identifier mac-address
6. option 150 ip ip-address
7. default-router ip-address
8. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 4 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
Step 5 client-identifier mac-address Specifies the MAC address of the phone, which is
printed on a label on each Cisco Unified IP phone.
Example: • A separate client-identifier command is
Router(config-dhcp)# client-identifier 01238.380.3056 required for each DHCP client.
• Add “01” prefix number before the MAC
address.
Step 6 option 150 ip ip-address Specifies the TFTP server address from which the
Cisco Unified IP phone downloads the image
configuration file.
Example:
Router(config-dhcp)# option 150 ip 10.0.0.1 • This is your Cisco Unified CME router’s
address.
Step 7 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 Unified CME router is the only
router on the network, this address should be
the Cisco Unified 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 purposes. If the Cisco Unified CME
IP source address becomes unreachable, IP
phones will attempt to register to the address
specified in this command.

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Command or Action Purpose

Step 8 end Returns to privileged EXEC mode.

Example:
Router(config-dhcp)# end

What to Do Next

• If you are configuring Cisco Unified CME for the first time on this router, you are ready to configure
NTP for the Cisco Unified CME router. See the “Enabling Network Time Protocol on the Cisco
Unified CME Router” section on page 104.
• If you are finished modifying network parameters for an already configured Cisco Unified CME
router, see “Generating Configuration Files for Phones” on page 245.

Defining a DHCP Relay

To set up DHCP relay on the LAN interface where the Cisco Unified IP phones are connected and enable
the DHCP relay to relay requests from the phones to the DHCP server, perform the following steps.

Prerequisites

There is a DHCP server that is not on this Cisco Unified CME router on the LAN that can provide
addresses to the Cisco Unified CME phones.

Restrictions

This Cisco Unified CME router cannot be the DHCP server.

SUMMARY STEPS

1. enable
2. configure terminal
3. service dhcp
4. interface type number
5. ip helper-address ip-address
6. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 service dhcp Enables the Cisco IOS DHCP server feature on the
router.
Example:
Router(config)# service dhcp
Step 4 interface type number Enters interface configuration mode for the
specified interface.
Example:
Router(config)# interface vlan 10
Step 5 ip helper-address ip-address Specifies the helper address for any unrecognized
broadcast for TFTP server and DNS server
requests.
Example:
Router(config-if)# ip helper-address 10.0.0.1 • A separate ip helper-address command is
required for each server 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 6 end Returns to privileged EXEC mode.

Example:
Router(config-if)# end

What to Do Next

• If you are configuring Cisco Unified CME for the first time on this router, you are ready to configure
NTP for the Cisco Unified CME router. See the “Enabling Network Time Protocol on the Cisco
Unified CME Router” section on page 104.
• If you are finished modifying network parameters for an already configured Cisco Unified CME
router, see “Generating Configuration Files for Phones” on page 245.

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Enabling Network Time Protocol on the Cisco Unified CME Router

To enable NTP for the Cisco Unified CME router, perform this task.

SUMMARY STEPS

1. enable
2. configure terminal
3. clock timezone zone hours-offset [minutes-offset]
4. clock summer-time zone recurring [week day month hh:mm week day month hh:mm [offset]]
5. ntp server ip-address
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 clock timezone zone hours-offset [minutes-offset] Sets the local time zone.

Example:
Router(config)# clock timezone pst -8
Step 4 clock summer-time zone recurring [week day month hh:mm (Optional) Specifies daylight savings time.
week day month hh:mm [offset]]
• Default: summer time is disabled. If the clock
summer-time zone recurring command is
Example: specified without parameters, the summer
Router(config)# clock summer-time pdt recurring time rules default to United States rules.
Default of the offset argument is 60.
Step 5 ntp server ip-address Synchronize software clock of router with the
specified NTP server.
Example:
Router(config)# ntp server 10.1.2.3
Step 6 exit Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

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What to Do Next

• If you are configuring Cisco Unified CME for the first time on this router and if you have a multisite
installation, you are ready to configure a DTMF relay. See the “Configuring DTMF Relay for H.323
Networks in Multisite Installations” section on page 105.
• If Cisco Unified CME will interact with a SIP Gateway, you must set up support for the gateway.
See the Configuring SIP Trunk Support, page 106.
• If you are configuring Cisco Unified CME for the first time on this router and you are ready to
configure system parameters. See “Configuring System-Level Parameters” on page 119.
• If you are finished modifying network parameters for an already configured Cisco Unified CME
router, see “Generating Configuration Files for Phones” on page 245.

Configuring DTMF Relay for H.323 Networks in Multisite Installations

To configure DTMF relay for H.323 networks in a multisite installation only, perform the following
steps.

Note To configure DTMF relay on SIP networks, see the “Configuring SIP Trunk Support” on page 106.

SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag voip
4. dtmf-relay h245-alphanumeric
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 dial-peer voice tag voip Enters dial-peer configuration mode.

Example:
Router(config)# dial-peer voice 2 voip

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Command or Action Purpose

Step 4 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 5 end Returns to privileged EXEC mode.

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

What to Do Next
• To set up support for a SIP trunk, see the Configuring SIP Trunk Support, page 106.
• If you are configuring Cisco Unified CME for the first time on this router and you are ready to
configure system parameters. See “Configuring System-Level Parameters” on page 119.
• If you are finished modifying network parameters for an already configured Cisco Unified CME
router, see “Generating Configuration Files for Phones” on page 245.

Configuring SIP Trunk Support

To enable DTMF relay on a dial-peer for a SIP gateway and set up the gateway to register phone numbers
with Cisco Unified CME, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag voip
4. dtmf-relay rtp-nte
5. dtmf-relay sip-notify
6. exit
7. sip-ua
8. notify telephone-event max-duration msec
9. registrar {dns:host-name | ipv4:ip-address} expires seconds [tcp] [secondary]
10. retry register number
11. timers register msec
12. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 dial-peer voice tag voip Enters dial-peer configuration mode.

Example:
Router(config)# dial-peer voice 2 voip
Step 4 dtmf-relay rtp-nte Forwards DTMF tones by using Real-Time Transport
Protocol (RTP) with the Named Telephone Event (NTE)
payload type and enables DTMF relay using the RFC 2833
Example:
Router(config-dial-peer)# dtmf-relay rtp-nte
standard method.
Step 5 dtmf-relay sip-notify Forwards DTMF tones using SIP NOTIFY messages.

Example:
Router(config-dial-peer)# dtmf-relay sip-notify
Step 6 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit
Step 7 sip-ua Enters SIP user-agent configuration mode.

Example:
Router(config)# sip-ua
Step 8 notify telephone-event max-duration msec Sets the maximum milliseconds allowed between two
consecutive NOTIFY messages for a single DTMF event.
Example: • max-duration time—Range: 500 to 3000.
Router(config-sip-ua)# notify telephone-event Default: 2000.
max-duration 2000
Step 9 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:
Router(config-sip-ua)# registrar
ipv4:10.8.17.40 expires 3600 secondary
Step 10 retry register number Sets the total number of SIP Register messages that the
gateway should send.
Example: • number—Number of Register message retries.
Router(config-sip-ua)# retry register 10 Range: 1 to 10. Default: 10.

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Command or Action Purpose

Step 11 timers register msec Sets how long the SIP user agent (UA) waits before sending
Register requests.
Example: • time—Waiting time, in milliseconds.
Router(config-sip-ua)# timers register 500 Range: 100 to 1000. Default: 500.
Step 12 end Returns to privileged EXEC mode.

Example:
Router(config-sip-ua)# end

Verifying SIP Trunk Support Configuration

To verify SIP trunk configuration, perform the following steps:

SUMMARY STEPS

1. show sip-ua status

2. show sip-ua timers
3. show sip-ua register status
4. show sip-ua statistics

DETAILED STEPS

Step 1 show sip-ua status

Use this command to display 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
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

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Step 2 show sip-ua timers

This command displays the waiting time before Register requests are sent; that is, the value that has been
set with the timers register command.
Step 3 show sip-ua register status
This command displays the status of local E.164 registrations.
Step 4 show sip-ua statistics
ThIs command displays the Register messages that have been sent.

Changing the TFTP Address on a DHCP Server

To change the TFTP IP address after it has already been configured, perform the following steps.

Prerequisites
Your Cisco Unified CME router is a DHCP server.

Restrictions
If the DHCP server is on a different router than Cisco Unified CME, reconfigure the external DHCP
server with the new IP address of the TFTP server.

SUMMARY STEPS

1. enable
2. configure terminal
3. ip dhcp pool pool-name
4. option 150 ip ip-address
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 ip dhcp pool pool-name Enters DHCP pool configuration mode to create or
modify a DHCP pool.
Example: • pool-name—Previously configured unique
Router(config)# ip dhcp pool pool2 identifier for the pool to be configured.
Step 4 option 150 ip ip-address Specifies the TFTP server IP address from which
the Cisco Unified IP phone downloads the image
configuration file, XmlDefault.cnf.xml.
Example:
Router(config-dhcp)# option 150 ip 10.0.0.1
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-dhcp)# end

Enabling OOD-R
To enable OOD-R support on the Cisco Unified CME router, perform the following steps.

Prerequisites
• Cisco Unified CME 4.1 or a later version.
• Cisco IOS Release 12.4(11)XJ or a later release.
• The application that initiates OOD-R, such as a click-to-dial application, and its directory server
must be installed and configured.
– For information on the SIP REFER and NOTIFY methods used between the directory server
and Cisco Unified CME, see RFC 3515, The Session Initiation Protocol (SIP) Refer Method.
– For information on the message flow Cisco Unified CME uses when initiating a session
between the Referee and Refer-Target, see RFC 3725, Best Current Practices for Third Party
Call Control (3pcc).

Restrictions
• The call waiting, conferencing, hold, and transfer call features are not supported while the
Refer-Target is ringing.
• In a SIP to SIP scenario, no ringback is heard by the Referee when Refer-Target is ringing.

SUMMARY STEPS

1. enable
2. configure terminal
3. sip-ua
4. refer-ood enable [request-limit]
5. exit
6. voice register global

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7. authenticate ood-refer
8. authenticate credential tag location
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 sip-ua Enters SIP user-agent configuration mode to configure the
user agent.
Example:
Router(config)# sip-ua
Step 4 refer-ood enable [request-limit] Enables OOD-R processing.
• request-limit—Maximum number of concurrent
Example: incoming OOD-R requests that the router can process.
Router(config-sip-ua)# refer-ood enable 300 Range: 1 to 500. Default: 500.
Step 5 exit Exits SIP user-agent configuration mode.

Example:
Router(config-sip-ua)# exit
Step 6 voice register global Enters voice register global configuration mode to set
global parameters for all supported SIP phones in a
Cisco Unified CME or Cisco Unified SRST environment.
Example:
Router(config)# voice register global
Step 7 authenticate ood-refer (Optional) Enables authentication of incoming OOD-R
requests using RFC 2617-based digest authentication.
Example:
Router(config-register-global)# authenticate
ood-refer

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Command or Action Purpose

Step 8 authenticate credential tag location (Optional) Specifies the credential file to use for
authenticating incoming OOD-R requests.
Example: • tag—Number that identifies the credential file to use
Router(config-register-global)# authenticate 1 for OOD-R authentication. Range: 1 to 5.
flash:cred1.csv
• location—Name and location of the credential file in
URL format. Valid storage locations are TFTP, HTTP,
and flash memory.
Step 9 end Exits to privileged EXEC mode.

Example:
Router(config-register-global)# end

Verifying OOD-R Configuration

Step 1 show running-config
This command verifies your configuration.
Router# show running-config
!
voice register global
mode cme
source-address 10.1.1.2 port 5060
load 7971 SIP70.8-0-1-11S
load 7970 SIP70.8-0-1-11S
load 7961GE SIP41.8-0-1-0DEV
load 7961 SIP41.8-0-1-0DEV
authenticate ood-refer
authenticate credential 1 tftp://172.18.207.15/labtest/cred1.csv
create profile sync 0004550081249644
.
.
.
sip-ua
refer-ood enable

Step 2 show sip-ua status refer-ood

This command displays OOD-R configuration settings.
Router# show sip-ua status refer-ood

Maximum allow incoming out-of-dialog refer 500

Current existing incoming out-of-dialog refer dialogs: 1
outgoing out-of-dialog refer dialogs: 0

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Troubleshooting OOD-R
Step 1 debug ccsip messages
This command displays the SIP messages exchanged between the SIP UA client and the router.
Router# debug ccsip messages

SIP Call messages tracing is enabled

Aug 22 18:15:35.757: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:
REFER sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.204.144:59607;branch=z9hG4bK1238
From: <sip:[email protected]>;tag=308fa4ba-4509
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 REFER
Max-Forwards: 70
Contact: <sip:[email protected]:59607>
User-Agent: CSCO/7
Timestamp: 814720186
Refer-To: sip:[email protected]
Referred-By: <sip:[email protected]>
Content-Length: 0

Aug 22 18:15:35.773: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 172.18.204.144:59607;branch=z9hG4bK1238
From: <sip:[email protected]>;tag=308fa4ba-4509
To: <sip:[email protected]>;tag=56D02AC-1E8E
Date: Tue, 22 Aug 2006 18:15:35 GMT
Call-ID: [email protected]
Timestamp: 814720186
CSeq: 101 REFER
Content-Length: 0
Contact: <sip:[email protected]:5060>

Step 2 debug voip application oodrefer

This command displays debugging messages for the OOD-R feature.
Router# debug voip application oodrefer

voip application oodrefer debugging is on

Aug 22 18:16:21.625: //-1//AFW_:/C_ServiceThirdParty_Event_Handle:

Aug 22 18:16:21.625: //-1//AFW_:/AFW_ThirdPartyCC_New:
Aug 22 18:16:21.625: //-1//AFW_:EE461DC520000:/C_PackageThirdPartyCC_NewReq: ThirdPartyCC
module listened by TclModule_45F39E28_0_91076048
Aug 22 18:16:21.625: //-1//AFW_:EE461DC520000:/OCOpen_SetupRequest: Refer Dest1: 1011,
Refer Dest2: 1001; ReferBy User: root
Aug 22 18:16:21.693: //-1//AFW_:EE461DC520000:/OCHandle_SignalEvent_1:
Aug 22 18:16:21.693: //-1//AFW_:/Third_Party_CC_Send_Notify: Third_Party_CC_Send_Notify:
sending notify respStatus=2, final=FALSE, failureCause=16
Aug 22 18:16:21.693: //-1//AFW_:/Third_Party_CC_Send_Notify: AppNotify successful!
Aug 22 18:16:26.225: //-1//AFW_:EE461DC520000:/OCHandle_SignalEvent_1:
Aug 22 18:16:26.229: //-1//AFW_:EE461DC520000:/OCHandle_SignalEvent_1:
Aug 22 18:16:26.249: //-1//AFW_:EE461DC520000:/OCHandle_SignalEvent_2:
Aug 22 18:16:29.341: //-1//AFW_:EE461DC520000:/OCHandle_SignalEvent_2:

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Aug 22 18:16:29.341: //-1//AFW_:/Third_Party_CC_Send_Notify: Third_Party_CC_Send_Notify:

sending notify respStatus=4, final=TRUE, failureCause=16
Aug 22 18:16:29.341: //-1//AFW_:/Third_Party_CC_Send_Notify: AppNotify successful!
Aug 22 18:16:29.349: //-1//AFW_:EE461DC520000:/OCHandle_Handoff: BAG contains:
Aug 22 18:16:29.349: LEG[895 ][LEG_INCCONNECTED(5)][Cause(0)]
Aug 22 18:16:29.349: CON[7 ][CONNECTION_CONFED(2)] {LEG[895
][LEG_INCCONNECTED(5)][Cause(0)],LEG[896 ][LEG_OUTCONNECTED(10)][Cause(0)]}
Aug 22 18:16:29.349: LEG[896 ][LEG_OUTCONNECTED(10)][Cause(0)]
Aug 22 18:16:29.365: //-1//AFW_:EE461DC520000:/OCAnyState_IgnoreEvent: Event Ignored
Aug 22 18:16:29.365: //-1//AFW_:/C_ServiceThirdParty_Event_Handle:
Aug 22 18:16:29.365: //-1//AFW_:EE461DC520000:/C_ServiceThirdParty_Event_Handle: Received
event APP_EV_NOTIFY_DONE[174] in Main Loop
Aug 22 18:16:29.365: //-1//AFW_:EE461DC520000:/OCAnyState_IgnoreEvent: Event Ignored
Aug 22 18:16:29.365: //-1//AFW_:/C_ServiceThirdParty_Event_Handle:
Aug 22 18:16:29.365: //-1//AFW_:EE461DC520000:/C_ServiceThirdParty_Event_Handle: Received
event APP_EV_NOTIFY_DONE[174] in Main Loop
Aug 22 18:16:29.369: //-1//AFW_:EE461DC520000:/OCHandle_SubscribeCleanup:
Aug 22 18:16:29.369: //-1//AFW_:EE461DC520000:/Third_Party_CC_Cleaner:
Aug 22 18:16:29.453: //-1//AFW_:EE461DC520000:/OCClosing_AnyEvent:
Aug 22 18:16:29.453: //-1//AFW_:EE461DC520000:/Third_Party_CC_Cleaner:
Aug 22 18:16:29.453: //-1//AFW_:EE461DC520000:/OCClosing_AnyEvent:
Aug 22 18:16:29.453: //-1//AFW_:EE461DC520000:/Third_Party_CC_Cleaner:

Configuration Examples for Network Parameters

• NTP Server: Example, page 114
• DTMF Relay for H.323 Networks: Example, page 114
• OOD-R: Example, page 115

NTP Server: Example

The following example defines the pst timezone as 8 hours offset from UTC, using a recurring daylight
savings time called pdt, and synchronizes the clock with the NTP server at 10.1.2.3.
clock timezone pst -8
clock summer-time pdt recurring
ntp server 10.1.2.3

DTMF Relay for H.323 Networks: 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

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OOD-R: Example
!
voice register global
mode cme
source-address 11.1.1.2 port 5060
load 7971 SIP70.8-0-1-11S
load 7970 SIP70.8-0-1-11S
load 7961GE SIP41.8-0-1-0DEV
load 7961 SIP41.8-0-1-0DEV
authenticate ood-refer
authenticate credential 1 tftp://172.18.207.15/labtest/cred1.csv
create profile sync 0004550081249644
.
.
.
sip-ua
authentication username jack password 021201481F
refer-ood enable
!

Where to Go Next
• If you are configuring Cisco Unified CME for the first time on this router, you are ready to
configure system-level parameters. See “Configuring System-Level Parameters” on page 119.
• If you modified network parameters for an already configured Cisco Unified CME router, you are
ready to generate the configuration file to save the modifications. See “Generating Configuration
Files for Phones” on page 245

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

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Additional References

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Network Parameters

Table 6 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 6 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 6 Feature Information for Network Parameters

Cisco Unified CME

Feature Name Version Modification
Out-of-Dialog Refer 4.1 Out-of Dialog REFER support was added.

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Last Updated: March 26, 2007

This chapter describes the system-level settings to configure before you add devices and configure
Cisco Unified Communications Manager Express (Cisco Unified CME) features.

Note If you used Cisco Unified Communications Express - QCT to generate a basic telephony configuration,
you can skip this module.

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for System-Level
Parameters” section on page 145.

Contents
• Prerequisites for System-Level Parameters, page 119
• Information About Configuring System-Level Parameters, page 120
• How to Configure System-Level Parameters, page 122
• Configuration Examples for System-Level Parameters, page 141
• Where to Go Next, page 143
• Additional References, page 143
• Feature Information for System-Level Parameters, page 145

Prerequisites for System-Level Parameters

• To support Cisco Unified IP phones that are running SIP to be connected directly to the
Cisco Unified CME router, Cisco CME 3.4 or a later version must be installed on the router. For
installation information, see “Installing and Upgrading Cisco Unified CME Software” on page 69.
• Cisco Unified CME must be configured to work with your IP network. For configuration
information, see “Defining Network Parameters” on page 91.

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Information About Configuring System-Level Parameters

To configure system-level parameters, you should understand the following concepts:
• Network Time Protocol for SIP Phones, page 120
• Per-Phone Configuration Files, page 120
• Redundant Cisco Unified CME Router, page 121
• Timeouts, page 122

Network Time Protocol for SIP Phones

SIP phones registered to a Cisco Unified CME router can synchronize to an NTP server. SIP phones can
synchronize to the Cisco Unified CME router, however some routers can lose their clock after a reboot
causing phones to display the wrong time. Synchronizing to an NTP server ensures that SIP phones
maintain the correct time. You enable NTP for all SIP phones by using the ntp-server command in voice
register global configuration mode. For configuration information, see the “SIP: Setting Network Time
Protocol” section on page 139.

Per-Phone Configuration Files

In Cisco Unified CME 4.0 and later versions, you can use an external TFTP server to offload the TFTP
server function on the Cisco Unified CME router. You can also use flash memory or slot 0 on the
Cisco Unified CME router for this purpose. This additional storage capacity allows you to use different
configuration files for each phone type or for each phone, which allows you to specify different user
locales and network locales for different phones. Before this version, you could specify only a single
default user and network locale for a Cisco Unified CME system.
You can specify one of the following four locations to store configuration files:
• System—This is the default. When the system is the storage location, there can be only one default
configuration file and it is used for all phones in the system. All phones, therefore, use the same user
locale and network locale. User-defined locales are not supported. To use the system location, either
do not use the cnf-file location command to specify a location or use the no cnf-file location {flash:
| slot0: | tftp url} command to reset the option from a previous, different location.
• Flash or slot 0—When flash or slot 0 memory on the router is the storage location, you can create
additional configuration files to apply per phone type or per individual phone. Up to five user and
network locales can be used in these configuration files. To store configuration files in flash or slot
0, use the cnf-file location flash: or cnf-file location slot0: command.

Note When the storage location chosen is flash and the file system type on this device is Class B
(LEFS), check free space on the device periodically and use the squeeze command to free the
space used up by deleted files. Unless you use the squeeze command, the space used by the
moved or deleted configuration files cannot be used by other files. Rewriting flash memory
space during the squeeze operation may take several minutes. We recommend using this
command during scheduled maintenance periods or off-peak hours.

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• TFTP—When an external TFTP server is the storage location, you can create additional
configuration files that can be applied per phone type or per individual phone. Up to five user and
network locales can be used in these configuration files. To store configuration files on an external
TFTP server, use the cnf-file location tftp url command.
You can then specify one of the following ways to create configuration files:
• Per system—This is the default. All phones use a single configuration file. The default user and
network locale in a single configuration file are applied to all phones in the Cisco Unified CME
system. Multiple locales and user-defined locales are not supported. To use the per-system option,
either do not use the cnf-file command or use the no cnf-file command to reset the option from a
different configuration.
• Per phone type—This setting creates separate configuration files for each phone type. For example,
all Cisco Unified IP Phone 7960s use XMLDefault7960.cnf.xml, and all Cisco Unified IP
Phone 7905s use XMLDefault7905.cnf.xml. All phones of the same type use the same configuration
file, which is generated using the default user and network locale. To create configuration files per
phone type, use the cnf-file perphonetype command. This option is not supported if you store the
configuration files in the system location.
• Per phone—This setting creates a separate configuration file for each phone, by MAC address. For
example, a Cisco Unified IP Phone 7960 with the MAC address 123.456.789 creates the per-phone
configuration file SEP123456789.cnf.xml. The configuration file for a phone generates with the
default user and network locale unless a different user and network locale is applied to the phone
using an ephone template. To create configuration files per phone type, use the cnf-file perphone
command. This option is not supported if you store the configuration files in the system location.
For configuration information, see the “SCCP: Defining Per-Phone Configuration Files and Alternate
Location” section on page 129.

Redundant Cisco Unified CME Router

A second Cisco Unified CME router can be configured to provide call-control services if the primary
Cisco Unified CME router fails. The secondary Cisco Unified CME router takes over and provides
services seamlessly until the primary router becomes operational again.
When a phone registers to the primary router, it receives a configuration file from the primary router.
Along with other information, the configuration file contains the IP addresses of the primary and
secondary Cisco Unified CME routers. The phone uses these addresses to initiate a keepalive (KA)
message to each router. The phone sends a KA message after every KA interval (30 seconds by default)
to the router with which it is registered and after every two KA intervals (60 seconds by default) to the
other router. The KA interval can be adjusted with the keepalive command.
If the primary router fails, a phone will not receive an acknowledgment (ACK) to its KA message to the
primary router. If the phone does not get an ACK from the primary router for three consecutive KAs, it
registers with the secondary Cisco Unified CME router.
During the time that the phone is registered to the secondary router, it keeps sending a KA probe to the
primary router to see if it has come back up, now every 60 seconds by default or two times the normal
KA interval. After the primary Cisco Unified CME router is operating normally, the phone starts
receiving ACKs for its probes. After the phone receives ACKs from the primary router for three
consecutive probes, it switches back to the primary router and reregisters with it. The reregistration of
phones with the primary router is also called rehoming.
The physical setup for redundant Cisco Unified CME routers is as follows. The FXO line from the PSTN
is split using a splitter. From the splitter, one line goes to the primary Cisco Unified CME router and the
other goes to the secondary Cisco Unified CME router. When a call comes in on the FXO line, it is

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presented to both the primary and secondary Cisco Unified CME routers. The primary router is
configured by default to answer the call immediately. The secondary Cisco Unified CME router is
configured to answer the call after three rings using the voice-port ring number 3 command. If the
primary router is operational, it answers the call immediately and changes the call state so that the
secondary router does not try to answer it. If the primary router is unavailable and does not answer the
call, the secondary router sees the new call coming in and answers after three rings.
The secondary Cisco Unified CME router should be connected in some way on the LAN, either through
the same switch or through another switch that may or may not be connected to the primary
Cisco Unified CME router directly. As long as both routers and the phones are connected on the LAN
with the appropriate configurations in place, the phones can register to whichever router is active.
Configure primary and secondary Cisco Unified CME routers identically, with the exception that the
FXO voice port from the PSTN on the secondary router should be configured to answer after more rings
than the primary router, as previously explained. The ip source-address command is used on both
routers to specify the IP addresses of the primary and secondary routers.

Timeouts
The following system-level timeout parameters have default values that are generally adequate:
• Busy Timeout—Amount of time that can elapse after a transferred call reaches a busy signal before
the call is disconnected.
• Interdigit Timeout—Amount of time that can elapse between the receipt of individual dialed digits
before the dialing process times out and is terminated. 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, see Dial Peer Configuration on
Voice Gateway Routers.
• Ringing Timeout—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.
• Keepalive—Interval determines how often a message is sent between the router and
Cisco Unified IP phones, over the session to ensure that the keepalive timeout is not exceeded. If no
other traffic is sent over the session during the interval, a keepalive message is sent.

How to Configure System-Level Parameters

• Configuring Bulk Registration, page 123 (optional)

SCCP
• SCCP: Setting Up Cisco Unified CME, page 125 (required)
• SCCP: Setting Date and Time Parameters, page 127 (required)
• SCCP: Blocking Automatic Registration, page 128 (optional)
• SCCP: Defining Per-Phone Configuration Files and Alternate Location, page 129 (optional)
• SCCP: Changing Defaults for Time Outs, page 130 (optional)
• SCCP: Configuring a Redundant Router, page 132 (optional)

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SIP
• SIP: Setting Up Cisco Unified CME, page 135 (required)
• SIP: Setting Date and Time Parameters, page 137 (required)
• SIP: Setting Network Time Protocol, page 139 (required)
• SIP: Changing Session-Level Application for SIP Phones, page 140 (optional)

Configuring Bulk Registration

To configure bulk registration for registering a block of phone numbers with an external registrar so that
calls can be routed to Cisco Unified CME from a SIP network, follow the steps in this section.
Numbers that match the defined number pattern can register with the external registrar. The block of
numbers that is registered can include any phone that is attached to Cisco Unified CME using SIP or
SCCP, or any analog phone that is directly attached to an FXS port on a Cisco Unified CME router.

Note To specify that an individual directory number not register with the external registrar by using this
command. For configuration information, see the “SIP: Disabling SIP Proxy Registration for a Directory
Number” section on page 172.

Prerequisites
Cisco Unified CME 3.4 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. mode cme
5. bulk number
6. exit
7. sip-ua
8. registrar {dns:address | ipv4:destination-address} expires seconds [tcp] [secondary ] no
registrar [secondary ]
9. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 mode cme Enables mode for provisioning SIP phones in
Cisco Unified CME.
Example:
Router(config-register-global)# mode cme
Step 5 bulk number Sets bulk registration for E.164 numbers that will register
with SIP proxy server.
Example: • number—Unique sequence of up to 32 characters
Router(config-register-global)# bulk 408526.... including wild cards and patterns that represents E.164
n umbers that will register with Sip proxy server.
Step 6 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-register-pool)# exit
Step 7 sip-ua Enters Session Initiation Protocol (SIP) user agent (ua)
configuration mode for configuring the user agent.
Example:
Router(config)# sip-ua
Step 8 registrar {dns:address | Enables SIP gateways to register E.164 numbers with SIP
ipv4:destination-address} expires seconds [tcp] proxy server.
[secondary] no registrar [secondary]

Example:
Router(config-sip-ua)# registrar server
ipv4:1.5.49.240
Step 9 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-sip-ua)# end

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Examples
The following example shows that all phone numbers that match the pattern “408555..” can register with
a SIP proxy server (IP address 1.5.49.240):
voice register global
mode cme
bulk 408555….
sip-ua
registrar ipv4:1.5.49.240

SCCP: Setting Up Cisco Unified CME

To identify filenames and location of phone firmware for phone types to be connected, specify the port
for phone registration, and specify number of phones and directory numbers to be supported, perform
the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. tftp-server flash:filename
4. telephony-service
5. load phone-type firmware-file
6. max-ephones max-phones
7. max-dn max-directory-numbers [preference preference-order] [no-reg primary | both]
8. ip source-address ip-address port port [any-match | strict-match]
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 tftp-server flash:filename Permits IP phones served by the Cisco Unified CME
router to access the specified file in flash memory.
Example: • A separate tftp-server flash command is required
Router(config)# tftp-server for each phone type.
flash:P00307020300.bin
Step 4 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 5 load phone-type firmware-file Identifies a Cisco Unified IP phone firmware file to be
used by phones of the specified type when they register.
Example: • A separate load command is required for each phone
Router(config-telephony)# load 7960-7940 type.
P00307020300
Note If you are loading a firmware file larger than
384 KB, you must first load a file for that phone
type that is smaller than 384 KB, then load the
larger file.
Step 6 max-ephones max-phones Limits number of phones to be supported by this router.
• Maximum number is platform and version-specific.
Example: Type? for value.
Router(config-telephony)# max-ephones 24
Step 7 max-dn max- directory- numbers [preference Limits number of directory numbers to be supported by
preference-order] [no-reg primary | both] this router.
• Maximum number is platform and version-specific.
Example: Type? for value.
Router(config-telephony)# max-dn 200 no-reg
primary
Step 8 ip source-address ip-address [port port] [any-match Identifies the IP address and port number that the
| strict-match] Cisco Unified CME router uses for IP phone registration.
• port port—(Optional) TCP/IP port number to use
Example: for SCCP. Range is 2000 to 9999. Default is 2000.
Router(config-telephony)# ip source-address
10.16.32.144 • any-match—(Optional) Disables strict IP address
checking for registration. This is the default.
• 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 9 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

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SCCP: Setting Date and Time Parameters

To specify the format of the date and time that appears on all SCCP phones in Cisco Unified CME,
follow the steps in this section.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. date-format {mm-dd-yy | dd-mm-yy | yy-dd-mm | yy-mm-dd}
5. time-format {12 | 24}
6. time-zone number
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 date-format {mm-dd-yy | dd-mm-yy | yy-dd-mm | (Optional) Sets the date format for phone display.
yy-mm-dd}
• Default: mm-dd-yy.

Example:
Router(config-telephony)# date-format yy-dd-mm
Step 5 time-format {12 | 24} (Optional) Selects a 12 or 24-hour clock for the time
display format on phone display.
Example: • Default: 12.
Router(config-telephony)# time-format 24

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Command or Action Purpose

Step 6 time-zone number Sets time zone used for all SCCP phones.
• Required for Cisco Unified IP Phone 7970G and
Example: 7971G-GE.
Router(config-telephony)# time-zone 2
• Default: 5, Pacific Standard/Daylight Time (-480).
Step 7 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

SCCP: Blocking Automatic Registration

To prevent Cisco Unified IP phones that are not explicitly configured in Cisco Unified CME from
registering with the Cisco Unified CME router, perform the following steps.

Prerequisite
Cisco Unified CME 4.0 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. auto-reg-ephone
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 4 auto-reg-ephone Enables all Cisco Unified IP phones that are running
SCCP to register regardless of whether the phone is
explicitly configured in Cisco Unified CME.
Example:
Router(config-telephony)# no auto-reg-ephone • Default: Enabled
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

SCCP: Defining Per-Phone Configuration Files and Alternate Location

To define a location other than system for storing configuration files and to specify what type of
configuration files to generate, perform the following steps. To use multiple network and user locales,
or user-defined locales, you must define per-phone or per-phone type configuration files

Prerequisites
Cisco Unified CME 4.0 or a later version.

Restrictions
• Externally stored and per-phone configuration files are not supported on the Cisco Unified IP Phone
7902G, 7910, 7910G, or 7920, or the Cisco Unified IP Conference Station 7935 and 7936.
• TFTP does not support file deletion. When configuration files are updated, they overwrite any
existing configuration files with the same name. If you change the configuration file location, files
are not deleted from the TFTP server.
• Generating configuration files on flash or slot 0 can take up to a minute, depending on the number
of files being generated.
• For smaller routers such as Cisco 2600 series routers, you must manually enter the squeeze
command to erase files after changing the configuration file location or entering any commands that
trigger the deletion of configuration files. Unless you use the squeeze command, the space used by
the moved or deleted configuration files is not usable by other files.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. cnf-file location {flash: | slot0: | tftp tftp-url}
5. cnf-file {perphonetype | perphone}
6. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 cnf-file location {flash: | slot0: | tftp Specifies a location other than system:/its for storing
tftp-url} phone configuration files.
• Required for per-phone or per-phone type
Example: configuration files.
Router(config-telephony)# cnf-file location
flash:
Step 5 cnf-file {perphonetype | perphone} Specifies whether to use a separate file for each type of
phone or for each individual phone.
Example:
Router(config-telephony)# cnf-file perphone
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

What to Do Next
If you changed the configuration file storage location, use the option 150 ip command to update the
address. See “Changing the TFTP Address on a DHCP Server” on page 109.

SCCP: Changing Defaults for Time Outs

To configure values for system-level intervals for which default values are typically adequate, perform
the following steps:

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. timeouts busy seconds

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5. timeouts interdigit seconds

6. timeouts ringing seconds
7. keepalive seconds
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 timeouts busy seconds (Optional) Sets the amount of time after which calls that are
transferred to busy destinations are disconnected.
Example: • seconds—Number of seconds. Range is 0 to 30. Default
Router(config-telephony)# timeouts busy 20 is 10.
Step 5 timeouts interdigit seconds (Optional) Configures the interdigit timeout value for all
Cisco Unified IP phones attached to the router.
Example: • seconds—Number of seconds before the interdigit
Router(config-telephony)# timeouts interdigit timer expires. Range is 2 to 120. Default is 10.
30
Step 6 timeouts ringing seconds (Optional) Sets the duration, in seconds, for which the
Cisco Unified CME system allows ringing to continue if a
call is not answered. Range is 5 to 60000. Default is 180.
Example:
Router(config-telephony)# timeouts ringing 30
Step 7 keepalive seconds (Optional) Sets the time interval, in seconds, between
keepalive messages that are sent to the router by
Cisco Unified IP phones.
Example:
Router(config-telephony)# keepalive 45 • The default is usually adequate. If the interval is set too
large, it is possible that notification will be delayed
when a system goes down.
• Range: 10 to 65535. Default: 0.
Step 8 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

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SCCP: Configuring a Redundant Router

To configure a secondary Cisco Unified CME router to act as a backup if the primary router fails,
perform the following steps on both the primary and secondary Cisco Unified CME routers.

Prerequisites
• Cisco Unified CME 4.0 or a later version.
• The secondary router must have a running configuration identical to that of the primary router.
• The physical configuration of the secondary router must be as described in the “Redundant Cisco
Unified CME Router” section on page 121.
• Phones that use this feature must be configured with the type command, which guarantees that the
appropriate phone configuration file will be present.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. ip source-address ip-address port port [secondary ip-address [rehome seconds]] [any-match |
strict-match]
5. exit
6. voice-port slot-number/port
7. signal ground-start
8. incoming alerting ring-only
9. ring number number
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 4 ip source-address ip-address [port port] [secondary Identifies the IP address and port number that the
ip-address [rehome seconds]] [any-match | Cisco Unified CME router uses for IP phone registration.
strict-match]
• ip-address—Address of the primary
Cisco Unified CME router.
Example:
Router(config-telephony)# ip source-address • port port—(Optional) TCP/IP port number to use for
10.0.0.1 secondary 10.2.2.25 SCCP. Range is 2000 to 9999. Default is 2000.
• secondary ip-address—Indicates a backup
Cisco Unified CME router.
• rehome seconds—Not used by Cisco Unified CME.
Used only by phones registered to
Cisco Unified SRST.
• any-match—(Optional) Disables strict IP address
checking for registration. This is the default.
• strict-match—(Optional) Router rejects IP phone
registration attempts if the IP server address used by
the phone does not exactly match the source address.
Step 5 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 6 voice-port slot-number/port Enters voice-port configuration mode for the FXO voice
port for DID calls from the PSTN.
Example:
Router(config)# voice-port 2/0
Step 7 signal ground-start Specifies ground-start signaling for a voice port.

Example:
Router(config-voiceport)# signal ground-start
Step 8 incoming alerting ring-only Instructs the FXO ground-start voice port to detect
incoming calls by detecting incoming ring signals.
Example:
Router(config-voiceport)# incoming alerting
ring-only

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Command or Action Purpose

Step 4 ip source-address ip-address [port port] [secondary Identifies the IP address and port number that the
ip-address [rehome seconds]] [any-match | Cisco Unified CME router uses for IP phone registration.
strict-match]
• ip-address—Address of the primary
Cisco Unified CME router.
Example:
Router(config-telephony)# ip source-address • port port—(Optional) TCP/IP port number to use for
10.0.0.1 secondary 10.2.2.25 SCCP. Range is 2000 to 9999. Default is 2000.
• secondary ip-address—Indicates a backup
Cisco Unified CME router.
• rehome seconds—Not used by Cisco Unified CME.
Used only by phones registered to
Cisco Unified SRST.
• any-match—(Optional) Disables strict IP address
checking for registration. This is the default.
• strict-match—(Optional) Router rejects IP phone
registration attempts if the IP server address used by
the phone does not exactly match the source address.
Step 5 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 6 voice-port slot-number/port Enters voice-port configuration mode for the FXO voice
port for DID calls from the PSTN.
Example:
Router(config)# voice-port 2/0
Step 7 signal ground-start Specifies ground-start signaling for a voice port.

Example:
Router(config-voiceport)# signal ground-start
Step 8 incoming alerting ring-only Instructs the FXO ground-start voice port to detect
incoming calls by detecting incoming ring signals.
Example:
Router(config-voiceport)# incoming alerting
ring-only

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Command or Action Purpose

Step 9 ring number number (Required only for the secondary router) Sets the
maximum number of rings to be detected before
answering an incoming call over an FXO voice port.
Example:
Router(config-voiceport)# ring number 3 • number—Number of rings detected before
answering the call. Range is 1 to 10. Default is 1.
Note For an incoming FXO voice port on a secondary
Cisco Unified CME router, set this value higher
than is set on the primary router. We recommend
setting this value to 3 on the secondary router.
Step 10 end Returns to privileged EXEC mode.

Example:
Router(config-voiceport)# end

SIP: Setting Up Cisco Unified CME

To identify filenames and location of phone firmware for phone types to be connected, specify the port
for phone registration, and specify the number of phones and directory numbers to be supported, perform
the following steps.

Note If your Cisco Unified CME system supports SCCP and SIP phones, do not connect your SIP phones to
your network until after you have verified the configuration profile for the SIP phone.

Prerequisites
• Cisco CME 3.4 or a later version.

Restrictions
Java-based phones, such as the Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE,
7970G, and 7971GE, are supported only in Cisco Unified CME 4.1 and later.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. mode cme
5. source-address ip-address
6. load phone-type firmware-file
7. tftp path {system: | flash: | slot0: | tftp tftp-url}
8. max-pool max-phones
9. max-dn max-directory-numbers

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10. authenticate [all] [realm string]

11. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 mode cme Enables mode for provisioning SIP phones in
Cisco Unified CME.
Example:
Router(config-register-global)# mode cme
Step 5 source-address ip-address [port port] Enables the Cisco Unified CME router to receive messages
from SIP phones through the specified IP address and port.
Example: • port—(Optional) TCP/IP port number.
Router(config-register-global)# source-address Range: 2000 to 9999. Default: 2000.
10.6.21.4
Step 6 load phone-type firmware-file Associates a phone type with a phone firmware file.
• A separate load command is required for each phone
Example: type.
Router(config-register-global)# load 7960-7940
P0S3-07-3-00
Step 7 tftp-path {system: | flash: | slot0: | tftp: Defines the location from which the SIP phones will
tftp-url} download configuration profile files.
• Default: system.
Example:
Router(config-register-global)# tftp-path
http://mycompany.com/files
Step 8 max-pool max-phones Limits number of SIP phones to be supported by the
Cisco Unified CME router.
Example: • Default: 0.
Router(config-register-global)# max-pool 10
Step 9 max-dn max-directory-numbers (Optional) Limits number of directory numbers for SIP
phones to be supported by the Cisco Unified CME router.
Example: • Default: 150 or maximum allowed on platform, by
Router(config-register-global)# max-dn 20 version. Type ? for value.

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Command or Action Purpose

Step 10 authenticate [all][realm string] (Optional) Enables authentication for registration requests
in which the MAC address of the SIP phone cannot be
identified by using other methods.
Example:
Router(config-register-global)# authenticate
all realm company.com
Step 11 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end

SIP: Setting Date and Time Parameters

To specify the format of the date and time that appears on all SIP phones in Cisco Unified CME, follow
the steps in this section.

Prerequisites
• Cisco CME 3.4 or a later version.
• The mode cme command is enabled.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. timezone number
5. date-format [d/m/d | m/d/y | y-d-m | y/d/m | y/m/d | yy-m-d]

6. time-format {12 | 24}

7. dst auto-adjust
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 timezone number Selects the time zone used for SIP phones in
Cisco Unified CME.
Example: • Default: 5, Pacific Standard/Daylight Time. Type ? to
Router(config-register-global)# timezone 8 display a list of time zones.
Step 5 date-format [d/m/y | m/d/y | y-m-d | y/d/m | (Optional) Selects the date display format on SIP phones in
y/m/d | yy-m-d] Cisco Unified CME.
• Default: m-d-y.
Example:
Router(config-register-global)# date-format
yy-m-d
Step 6 time-format {12 | 24} (Optional) Selects the time display format on SIP phones in
Cisco Unified CME.
Example: • Default: 12.
Router(config-register-global)# time-format 24
Step 7 dst auto-adjust (Optional) Enables automatic adjustment of daylight
savings time on SIP phones in Cisco Unified CME.
Example: • To modify start and stop times for daylight savings
Router(config-register-global)# dst auto-adjust time, use the dst command.
Step 8 dst {start | stop} month [day day-of-month | (Optional) Sets the time period for daylight savings time on
week week-number | day day-of-week] time SIP phones in Cisco Unified CME.
hour:minutes
• Required if automatic adjustment of daylight savings
time is enabled by using the dst auto-adjust command.
Example:
Router(config-register-global)# dst start jan • Default is Start: First week of April, Sunday, 2:00 a.m
day 1 time 00:00 and Stop: Last week of October, Sunday 2:00 a.m.
Router(config-register-global)# dst stop mar
day 31 time 23:59
Step 9 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end

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SIP: Setting Network Time Protocol

To enable NTP for Java-based phones, such as the Cisco Unified IP Phone 7911G, 7941G, 7941GE,
7961G, 7961GE, 7970G, and 7971GE, connected to Cisco Unified CME running SIP, perform this task.

Prerequisites
• Cisco Unified CME 4.1 or a later version.
• The firmware load 8.2(1) or a later version is installed for SIP phones to download. For upgrade
information, see the “SIP: Upgrading or Downgrading Phone Firmware Between Versions” section
on page 77.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. ntp-server ip-address [mode {anycast | directedbroadcast | multicast | unicast}]
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to
set global parameters for all supported SIP phones
in a Cisco Unified CME environment.
Example:
Router(config)# voice register global
Step 4 ntp-server ip-address [mode {anycast | Clock on this router is synchronized with the
directedbroadcast | multicast | unicast}] specified NTP server.

Example:
Router(config-register-global)# ntp-server 10.1.2.3
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-register-global)# end

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SIP: Changing Session-Level Application for SIP Phones

To change a the default session-level application for all SIP phones, perform the following steps.

Prerequisites
• Cisco CME 3.4 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. application application-name
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 application application-name (Optional) Changes the default application for all dial peers
associated with the SIP phones in Cisco Unified CME to the
specified application.
Example:
Router(config-register-global)# application Note The application command in voice register pool
sipapp2 configuration mode takes precedence over this
command in voice register global configuration
mode.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end

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Configuration Examples for System-Level Parameters

This section contains the following examples:
• System-Level Parameters: Example, page 141
• Blocking Automatic Registration: Example, page 142
• Redundant Router: Example, page 143

System-Level Parameters: Example

The following example shows the system-level configuration for a Cisco Unified CME that can support
up to 500 directory numbers on 100 phones. It sets up TFTP file sharing for phone firmware files for
Cisco Unified IP Phone 7905, 7912, 7914, 7920, 7940, and 7960 phone types and loads those files.
tftp-server flash:ATA030100SCCP040211A.zup
! ATA 186/188 firmware
tftp-server flash:CP7902080001SCCP051117A.sbin
! 7902 firmware
tftp-server flash:CP7905080001SCCP051117A.sbin
! 7905 firmware
tftp-server flash:CP7912080001SCCP051117A.sbin
! 7912 firmware
tftp-server flash:cmterm_7920.4.0-02-00.bin
! 7914 firmware
tftp-server flash:P00503010100.bin
! 7920 firmware
tftp-server flash:S00104000100.sbn
! 7935 firmware
tftp-server flash:cmterm_7936.3-3-5-0.bin
! 7936 firmware
tftp-server flash:P0030702T023.bin
tftp-server flash:P0030702T023.loads
tftp-server flash:P0030702T023.sb2
! 7960/40 firmware
!
telephony-service
max-ephones 100
max-dn 500
load ata ATA030100SCCP040211A
load 7902 CP7902080001SCCP051117A
load 7905 CP7905080001SCCP051117A
load 7912 CP7912080001SCCP051117A
load 7914 S00104000100
load 7920 cmterm_7920.4.0-02-00
load 7935 P00503010100
load 7936 cmterm_7936.3-3-5-0
load 7960-7940 P0030702T023
ip source-address 10.16.32.144 port 2000
create cnf-files version-stamp Jan 01 2002 00:00:00
transfer-system full-consult

Cisco Unified IP Phone 7911, 7941, 7941-GE, 7961, 7961-GE, 7970, and 7971 require multiple files to
be shared using TFTP. The following configuration example adds support for these phones.
tftp-server flash:SCCP11.7-2-1-0S.loads
tftp-server flash:term11.default.loads
tftp-server flash:apps11.1-0-0-72.sbn
tftp-server flash:cnu11.3-0-0-81.sbn
tftp-server flash:cvm11.7-2-0-66.sbn

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tftp-server flash:dsp11.1-0-0-73.sbn
tftp-server flash:jar11.7-2-0-66.sbn
! 7911 firmware
!
tftp-server flash:TERM41.7-0-3-0S.loads
tftp-server flash:TERM41.DEFAULT.loads
tftp-server flash:TERM61.DEFAULT.loads
tftp-server flash:CVM41.2-0-2-26.sbn
tftp-server flash:cnu41.2-7-6-26.sbn
tftp-server flash:Jar41.2-9-2-26.sbn
! 7941/41-GE, 7961/61-GE firmware
!
tftp-server flash:TERM70.7-0-1-0s.LOADS
tftp-server flash:TERM70.DEFAULT.loads
tftp-server flash:TERM71.DEFAULT.loads
tftp-server flash:CVM70.2-0-2-26.sbn
tftp-server flash:cnu70.2-7-6-26.sbn
tftp-server flash:Jar70.2-9-2-26.sbn
! 7970/71 firmware
!
telephony-service
load 7911 SCCP11.7-2-1-0S
load 7941 TERM41.7-0-3-0S
load 7961 TERM41.7-0-3-0S
load 7941GE TERM41.7-0-3-0S
load 7961GE TERM41.7-0-3-0S
load 7970 TERM70.7-0-1-0s
load 7971 TERM70.7-0-1-0s
create cnf-files version-stamp Jan 01 2002 00:00:00
.
.
.

Blocking Automatic Registration: Example

The following example shows how to disable automatic ephone registration, display a log of attempted
registrations, and then clear the log.
Router(config)# telephony-service
Router(config-telephony)# no auto-reg-ephone
Router(config-telephony)# exit
Router(config)# exit
Router# show ephone attempted-registrations

Attempting Mac address:

Num Mac Address DateTime DeviceType

-----------------------------------------------------------------------------

1 C863.8475.5417 22:52:05 UTC Thu Apr 28 2005 SCCP Gateway (AN)

2 C863.8475.5408 22:52:05 UTC Thu Apr 28 2005 SCCP Gateway (AN)
.....
25 000D.28D7.7222 22:26:32 UTC Thu Apr 28 2005 Telecaster 7960
26 000D.BDB7.A9EA 22:25:59 UTC Thu Apr 28 2005 Telecaster 7960
...
47 C863.94A8.D40F 22:52:17 UTC Thu Apr 28 2005 SCCP Gateway (AN)
48 C863.94A8.D411 22:52:18 UTC Thu Apr 28 2005 SCCP Gateway (AN)

49 C863.94A8.D400 22:52:15 UTC Thu Apr 28 2005 SCCP Gateway (AN)

Router# clear telephony-service ephone-attempted-registrations

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Redundant Router: Example

The following example is configured on the primary Cisco Unified CME router. It establishes the router
at 10.5.2.78 as a secondary router. The voice port 3/0/0 is the FXO port for incoming calls from the
PSTN. It is set to use ground-start signaling and detect incoming calls by counting incoming ring signals.
telephony-service
ip source-address 10.0.0.1 port 2000 secondary 10.5.2.78

voice-port 3/0/0
signal ground-start
incoming alerting ring-only

The secondary Cisco Unified CME router is configured with the same commands, except that the ring
number command is set to 3 instead of using the default of 1.
telephony-service
ip source-address 10.0.0.1 port 2000 secondary 10.5.2.78

voice-port 3/0/0
signal ground-start
incoming alerting ring-only
ring number 3

Where to Go Next
After configuring system-level parameters, you are ready to configure phones in Cisco Unified CME for
making basic calls. See “Configuring Phones to Make Basic Calls” on page 147.

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

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Additional References

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for System-Level Parameters

Feature Information for System-Level Parameters

Table 7 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 7 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 7 Feature Information for System-Level Parameters

Cisco Unified CME

Feature Name Versions Feature Information
Blocking Automatic Registration 4.0 IP phones that are not explicitly configured in
Cisco Unified CME are blocked from registering.
Bulk Registration 3.4 Bulk registration for registering a block of phone numbers
with an external registrar was introduced.
Network Time Protocol for SIP Phones 4.1 SIP phones can synchronize to an NTP server.
Per-Phone Configuration Files and 4.0 Defines a location other than system for storing
Alternate Location configuration files and specifies the type of configuration
files to generate.
Redundant Router 4.0 Redundant router capability was introduced.
SIP phones in Cisco Unified CME 3.4 Support for SIP endpoints directly connected to
Cisco Unified CME was introduced.

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Last Updated: July 5, 2007

This module describes how to configure Cisco Unified IP phones in a Cisco Unified Communications
Manager Express (Cisco Unified CME) system so that you can make and receive basic calls.

Note If you used Cisco Unified Communications Express - QCT to generate a basic telephony configuration,
you can skip this module unless you want to modify the configuration to add phones.

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Configuring Phones
to Make Basic Calls” section on page 210.

Contents
• Prerequisites for Configuring Phones to Make Basic Calls, page 148
• Restrictions for Configuring Phones to Make Basic Calls, page 148
• Information About Configuring Phones to Make Basic Calls, page 148
• How to Configure Phones for a PBX System, page 158
• How to Configure Phones for a Key System, page 176
• How to Configure Cisco ATA, Analog Phone Support, Remote Phones, and Cisco IP Communicator,
page 188
• Configuration Examples for Making Basic Calls, page 199
• Where to Go Next, page 208
• Additional References, page 209
• Feature Information for Configuring Phones to Make Basic Calls, page 210

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Prerequisites for Configuring Phones to Make Basic Calls

Prerequisites for Configuring Phones to Make Basic Calls

• Cisco IOS software and Cisco Unified CME software, including phone firmware files for
Cisco Unified IP phones to be connected to Cisco Unified CME, must be installed in router flash
memory. See “Installing and Upgrading Cisco Unified CME Software” on page 69.
• For Cisco Unified IP phones that are running SIP and are connected directly to Cisco Unified CME,
Cisco Unified CME 3.4 or later must be installed on the router. See “Installing and Upgrading
Cisco Unified CME Software” on page 69.
• Procedures in “Defining Network Parameters” on page 91 and “Configuring System-Level
Parameters” on page 119 must be completed before you start the procedures in this section.

Restrictions for Configuring Phones to Make Basic Calls

When you are configuring dial peers or ephone-dns, including park slots and conferencing extensions,
on Cisco Integrated Services Router Voice Bundles, the following message may appear to warn you that
free memory is not available:
%DIALPEER_DB-3-ADDPEER_MEM_THRESHOLD: Addition of dial-peers limited by available
memory
To configure more dial peers or ephone-dns, increase the DRAM in the system. A moderately complex
configuration may exceed the default 256 MB DRAM and require 512 MB DRAM. Note that many
factors contribute to memory usage, in addition to the number of dial peers and ephone-dns configured.

Information About Configuring Phones to Make Basic Calls

To configure phones to make basic calls, you should understand the following concepts:
• Phones in Cisco Unified CME, page 148
• Directory Numbers, page 149
• PSTN FXO Trunk Lines, page 154
• Analog Phones, page 154
• Remote Teleworker Phones, page 155
• Digit Collection on SIP Phones, page 157
• Session Transport Protocol for SIP Phones, page 158

Phones in Cisco Unified CME

An ephone, or “Ethernet phone,” for SCCP or a voice-register pool for SIP is the software configuration
for a phone in Cisco Unified CME. This phone can be either a Cisco Unified IP phone or an analog
phone. Each physical phone in your system must be configured as an ephone or voice-register pool on
the Cisco Unified CME router to receive support in the LAN environment. Each phone has a unique tag,
or sequence number, to identify it during configuration.

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Directory Numbers
A directory number, also known as an ephone-dn for SCCP or a voice-register dn for SIP, is the software
configuration in Cisco Unified CME that represents the line connecting a voice channel to a phone. A
directory number has one or more extension or telephone numbers associated with it to allow call
connections to be made. Generally, a directory number is equivalent to a phone line, but not always.
There are several types of directory numbers, which have different characteristics.
Each directory number has a unique dn-tag, or sequence number, to identify it during configuration.
Directory numbers are assigned to line buttons on phones during configuration.
One virtual voice port and one or more dial peers are automatically created for each directory number,
depending on the configuration for SCCP phones, or for SIP phones, when the phone registers in
Cisco Unified CME.
The number of directory numbers that you create corresponds to the number of simultaneous calls that
you can have, because each directory number 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
directory numbers with the same destination number pattern.
The directory number is the basic building block of a Cisco Unified CME system. Six different types of
directory number 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 directory numbers low and provide service to a large number of people, you might use shared
directory numbers. Or if you have a limited quantity of extension numbers that you can use and you need
to have a large quantity of simultaneous calls, you might create two or more directory numbers with the
same number. The key is knowing how each type of directory number works and its advantages.
Not all types of directory numbers can be configured for all phones or for all protocols. In the remaining
information about directory numbers, we have used SCCP in the examples presented but that does not
imply exclusivity. The following sections describe the types of directory number in a
Cisco Unified CME system:
• Single-Line, page 149
• Dual-Line, page 150
• Two Directory Numbers with One Telephone or Extension Number, page 151
• Dual-Number, page 152
• Shared, page 152
• Monitor Mode for Shared Lines, page 153
• Overlaid, page 153

Single-Line
A single-line directory number has the following characteristics:
• Makes one call connection at a time using one phone line button. A single-line directory number 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 Unified 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.

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• When used with multiple-line features like call waiting, call transfer, and conferencing, there must
be more than one single-line directory number on a phone.
• Can be combined with dual-line directory numbers on the same phone.
Note that you must make the choice to configure each directory number in your system as either dual-line
or single-line when you initially create configuration entries. If you need to change from single-line to
dual-line later, you must delete the configuration for the directory number, then recreate it.
Figure 7 shows a single-line directory number for an SCCP phone in Cisco Unified CME.

Figure 7 Single-Line Directory Number

ephone-dn 11
number 1001
IP V ephone 1

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

Dual-Line
A dual-line directory number has the following characteristics:
• One voice port with two channels.
• Supported on IP phones that are running SCCP; not supported on IP phones that are running SIP.
• Can make two call connections at the same time using one phone line button. A dual-line directory
number has two channels for separate call connections.
• Can have one number or two numbers (primary and secondary) associated with it.
• Should be used for a directory number that needs to use one line 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 directory numbers on the same phone.
Note that you must make the choice to configure each directory number in your system as either dual-line
or single-line when you initially create configuration entries. If you need to change from single-line to
dual-line later, you must delete the configuration for the directory number, then recreate it.
Figure 8 shows a dual-line directory number for an SCCP phone in Cisco Unified CME.

Figure 8 Dual-Line Directory Number

ephone-dn 12 dual-line
number 1002
IP V ephone 2
88889

Phone 2 button 1:12

Button 1 is extension 1002

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Two Directory Numbers with One Telephone or Extension Number

Two directory numbers 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 dual-line (SCCP only) or single-line directory numbers.
• 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 9 on page 151 shows a phone with two buttons that have the same number, extension 1003. Each
button has a different directory number (button 1 is directory number 13 and button 2 is directory number
14), so each button can make one independent call connection if the directory numbers are single-line
and two call connections (for a total of four) if the directory numbers are dual-line.
Figure 10 shows two phones that each have a button with the same number. Because the buttons have
different directory numbers, 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 1003, and the phone user on phone 5 can receive
a different call on extension 1003 at the same time.
The two directory numbers-with-one-number situation is different than a shared line, which also has two
buttons with one number but has only one directory number for both of them. A shared directory number
will have the same call connection at all the buttons on which the shared directory number appears. If a
call on a shared directory number is answered on one phone and then placed on hold, the call can be
retrieved from the second phone on which the shared directory number appears. But when there are two
directory numbers 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 10, 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” section on page 152.
The examples in Figure 9 and Figure 10 show how two directory numbers with one number are used to
provide a small hunt group capability. In Figure 9, if the directory number on button 1 is busy or does
not answer, an incoming call to extension 1003 rolls over to the directory number associated with button
2 because the appropriate related commands are configured. Similarly, if button 1 on phone 4 is busy, an
incoming call to 1003 rolls over to button 1 on phone 5.

Figure 9 Two Directory Numbers 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

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Figure 10 Two Directory Numbers 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
A dual-number directory number 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 directory number.
• Can make two call connections at a time if it is a dual-line directory number (SCCP only).
• Should be used when you want to have two different numbers for the same button without using
more than one directory number.
Figure 11 shows a directory number that has two numbers, extension 1006 and extension 1007.

Figure 11 Dual-Number Directory

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

Shared
A shared directory number has the following characteristics:
• Line appears on two different phones but uses the same directory number, and extension or phone
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 directory number, if the directory number is connected to a call on
one phone, that directory number 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 12 shows a shared directory number on phones that are running SCCP. Extension 1008 appears
on both phone 7 and phone 8.

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Figure 12 Shared Directory Number

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

Monitor Mode for Shared Lines

Monitor mode for shared lines provides a visible line status indicating whether the line is in-use or not.
When the line is in use, it cannot be used for incoming or outgoing calls.
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 is completed. For all other call states, the monitor line lamp is on
or lit.

Overlaid
An overlaid directory number has the following characteristics:
• Is a member of an overlay set, which includes all the directory numbers 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.
Overlaid directory numbers provide call coverage similar to shared directory numbers because the same
number can appear on more than one phone. The advantage of using two directory numbers in an overlay
arrangement rather than as a simple shared line 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 it were a shared directory number.
For information about configuring call coverage using overlaid ephone-dns, see “Configuring
Call-Coverage Features” on page 563.
You can overlay up to 25 lines on a single button. A typical use of overlaid directory numbers would be
to 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. For configuration information, see the “SCCP:
Creating Directory Numbers for a Simple Key System” section on page 176

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PSTN FXO Trunk Lines

In Cisco CME 3.2 and later, IP phones running SCCP can be configured to have buttons for dedicated
PSTN FXO trunk lines, also known as FXO lines. FXO 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. When a 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. FXO lines can use PSTN service provider voice mail: when the line button is pressed, the line is
seized, allowing the user to hear the stutter dial tone provided by the PSTN to indicate that voice
messages are available.
Because 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 nonPSTN destinations, such as local IP phones, a second directory
number must be provisioned.
Calls placed to or received on an FXO line have restricted Cisco Unified CME services and cannot be
transferred by Cisco Unified CME. However, phone users are able to access hookflash-controlled PSTN
services using the Flash soft key.
In Cisco Unified CME 4.0, the following FXO trunk enhancements were introduced to improve the
keyswitch emulation behavior of PSTN lines on phones running SCCP, in a Cisco Unified CME system.
• FXO port monitoring—Allows the line button on IP phones to reliably show the status of an FXO
port when the port is in use. The status indicator, either a lamp or an icon, depending on the phone
model, accurately displays the status of the FXO port during the duration of the call, even after the
call is forwarded or transferred. The same FXO port can be monitored by multiple phones using
multiple trunk ephone-dns.
• Transfer recall—If a transfer-to phone does not answer after a specified timeout, the call is returned
to the phone that initiated the transfer and it resumes ringing on the FXO line button. The directory
number must be dual-lined.
• Transfer-to button optimization—When an FXO call is transferred to a private extension button on
another phone, and that phone has a shared line button for the FXO port, after the transfer is
committed and the call is answered, the connected call displays on the FXO line button of the
transfer-to phone. This frees up the private extension line on the transfer-to phone. The directory
number n must be dual-line.
• Dual-line ephone-dns— Directory numbers for FXO lines can now be configured for dual-line to
support the FXO monitoring, transfer recall, and transfer-to button optimization features.
For configuration information, see the “SCCP: Configuring Trunk Lines for a Key System” section on
page 179.

Analog Phones
Cisco Unified CME supports analog phones using Cisco Analog Telephone Adaptors (ATAs) or FXS
ports in SCCP mode or H.323 mode, and supports fax machines on Cisco ATA or FXS ports in H.323
mode. The FXS ports used for analog phones or fax can be on the Cisco Unified CME router or on a
Cisco VG 224 voice gateway or Integrated Services Router (ISR). This section provides information on
the following topics:
• Cisco ATAs in SCCP Mode, page 155
• FXS Ports in SCCP Mode, page 155
• FXS Ports in H.323 Mode, page 155

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Cisco ATAs in SCCP Mode

You can configure the Cisco ATA 186 or Cisco ATA 188 to cost-effectively support analog phones using
SCCP in Cisco IOS Release 12.2(11)T and later. Each Cisco ATA enables two analog phones to function
as IP phones. For configuration information, see the “Configuring Cisco ATA Support” section on
page 189.

FXS Ports in SCCP Mode

FXS ports on Cisco VG 224 Voice Gateways and Cisco 2800 Series and Cisco 3800 Series ISRs can be
configured for SCCP supplementary features. For information about using SCCP supplementary features
on analog FXS ports on a Cisco IOS gateway under the control of a Cisco Unified CME router, see SCCP
Controlled Analog (FXS) Ports with Supplementary Features in Cisco IOS Gateways.

FXS Ports in H.323 Mode

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 Unified CME features. Note that when using Cisco Unified CME, you can configure FXS ports in
H.323 mode for call waiting or hookflash transfer, but not both at the same time.
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 Guide
• “Caller ID” section of the Cisco IOS Voice Configuration Library for your Cisco IOS release
• “Modem Support for VoIP” section of the Cisco IOS Voice Configuration Library for your
Cisco IOS release
• Cisco IOS Fax and Modem Services over IP Application Guide for your Cisco IOS release

Remote Teleworker Phones

IP phones or instances of Cisco IP Communicator can be connected to a Cisco Unified CME system over
a wide area network (WAN) to support teleworkers who have offices that are remote from the
Cisco Unified CME router. The maximum number of remote phones that can be supported is determined
by the available bandwidth.
IP addressing is a determining factor in the most critical aspect of remote teleworker phone design. The
following two scenarios represent the most common designs, the second one is the most common for
small and medium businesses:
• Remote site IP phones and the hub Cisco Unified CME router use globally routable IP addresses.
• Remote site IP phones use NAT with nonroutable private IP addresses and the hub
Cisco Unified CME router uses a globally routable address (see Figure 13). This scenario results in
one-way audio unless you use one of the following workarounds:
– Configure static NAT mapping on the remote site router (for example, a Cisco 831 Ethernet
Broadband Router) to convert between a private address and a globally routable address. This
solution uses fewer Cisco Unified CME resources, but voice is unencryped across the WAN.

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– Configure an IPsec VPN tunnel between the remote site router (or example, a Cisco 831) and
the Cisco Unified CME router. This solution requires an Advanced IP Services or higher image
on the Cisco Unified CME router if this router is used to terminate the VPN tunnel. Voice will
be encrypted across the WAN. This method will also work with the Cisco VPN client on a PC
to support Cisco IP Communicator.

Figure 13 Remote Site IP Phones Using NAT

IP WAN PSTN

146625
Teleworker Cisco 831
remote phone NAT firewall Cisco Unified CME
router (VPN)
router

Media Termination Point for Remote Phones

Media termination point (MTP) configuration is used to ensure that Real-Time Transport Protocol (RTP)
media packets from remote phones always transit through the Cisco Unified CME router. Without the
MTP feature, a phone that is connected in a call with another phone in the same Cisco Unified CME
system sends its media packets directly to the other phone, without the packets going through the
Cisco Unified CME router. MTP forces the packets to be sourced from the Cisco Unified CME router.
When this configuration is used to instruct a phone to always send its media packets to the
Cisco Unified CME router, the router acts as an MTP or proxy and forwards the packets to the
destination phone. If a firewall is present, it can be configured to pass the RTP packets because the router
uses a specified UDP port for media packets. In this way, RTP packets from remote IP phones can be
delivered to IP phones on the same system though they must pass through a firewall.
You must use the mtp command to explicitly enable MTP for each remote phone that sends media
packets to Cisco Unified CME.
One factor to consider is whether you are using multicast music on hold (MOH) in your system.
Multicast packets generally cannot be forwarded to phones that are reached over a WAN. The multicast
MOH feature checks to see if MTP is enabled for a phone and if it is, MOH is not sent to that phone. If
you have a WAN configuration that can forward multicast packets and you can allow RTP packets
through your firewall, you can decide not to use MTP.
For configuration information, see the “SCCP: Enabling a Remote Phone” section on page 196.

G.729r8 Codec on Remote Phones

You can select the G.729r8 codec on a remote IP phone to help save network bandwidth. The default
codec is G.711 mu-law. If you use the codec g729r8 command without the dspfarm-assist keyword, the
use of the G.729 codec is preserved only for calls between two phones on the Cisco Unified CME router
(such as between an IP phone and another IP phone or between an IP phone and an FXS analog phone).
The codec g729r8 command has no affect on a call directed through a VoIP dial peer unless the
dspfarm-assist keyword is also used.
For configuration information, see the “SCCP: Enabling a Remote Phone” section on page 196.
For information about transcoding behavior when using the G.729r8 codec, see the “Transcoding When
a Remote Phone Uses G.729r8” section on page 305.

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Digit Collection on SIP Phones

Digit strings dialed by phone users must be collected and matched against predefined patterns to place
calls to the destination corresponding to the user's input. Before Cisco Unified CME 4.1, SIP phone
users had to press the DIAL soft key or # key, or wait for the interdigit-timeout to trigger call processing.
In Cisco United CME 4.1 and later, two methods of collecting and matching digits are supported for SIP
phones, depending on the model of phone:
• KPML Digit Collection, page 157
• SIP Dial Plans, page 157

KPML Digit Collection

Key Press Markup Language (KPML) uses SIP SUBSCRIBE and NOTIFY methods to report user input
digit by digit. Each digit dialed by the phone user generates its own signaling message to
Cisco Unified CME, which performs pattern recognition by matching a destination pattern to a dial peer
as it collects the dialed digits. This process of relaying each digit immediately is similar to the process
used by SCCP phones. It eliminates the need for the user to press the Dial soft key or wait for the
interdigit timeout before the digits are sent to Cisco Unified CME for processing.
KPML is supported on Cisco Unified IP Phones 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and
7971GE. For configuration information, see the “SIP: Enabling KPML” section on page 169.

SIP Dial Plans

A dial plan is a set of dial patterns that SIP phones use to determine when digit collection is complete
after a user goes off-hook and dials a destination number. Dial plans allow SIP phones to perform local
digit collection and recognize dial patterns as user input is collected. After a pattern is recognized, the
SIP phone sends an INVITE message to Cisco Unified CME to initiate the call to the number matching
the user's input. All of the digits entered by the user are presented as a block to Cisco Unified CME for
processing. Because digit collection is done by the phone, dial plans reduce signaling messages overhead
compared to KPML digit collection.
SIP dial plans eliminate the need for a user to press the Dial soft key or # key, or to wait for the interdigit
timeout to trigger an outgoing INVITE. You configure a SIP dial plan and associate the dial plan with a
SIP phone. The dial plan is downloaded to the phone in the configuration file.
You can configure SIP dial plans and associate them with the following SIP phones:
• Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE—These
phones use dial plans and support KPML. If both a dial plan and KPML are enabled, the dial plan
has priority.
If a matching dial plan is not found and KPML is disabled, the user must wait for the interdigit
timeout before the SIP NOTIFY message is sent to Cisco Unified CME. Unlike other SIP phones,
these phones do not have a Dial soft key to indicate the end of dialing, except when on-hook dialing
is used. In this case, the user can press the Dial soft key at any time to send all the dialed digits to
Cisco Unified CME.
• Cisco Unified IP Phone 7905, 7912, 7940, and 7960—These phones use dial plans and do not
support KPML. If you do not configure a SIP dial plan for these phones, or if the dialed digits do
not match a dial plan, the user must press the Dial soft key or wait for the interdigit timeout before
digits are sent to Cisco Unified CME.
When you reset a phone, the phone requests its configuration files from the TFTP server, which builds
the appropriate configuration files depending on the type of phone.

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• Cisco Unified IP Phone 7905 and 7912—The dial plan is a field in their configuration files.
• Cisco Unified IP Phone 7911G, 7940, 7941G, 7941GE, 7960, 7961G, 7961GE, 7970G, and
7971GE—The dial plan is a separate XML file that is pointed to from the normal configuration file.
For configuration information for Cisco Unified CME, see the “SIP: Configuring Dial Plans” section on
page 165.

Session Transport Protocol for SIP Phones

In Cisco Unified CME 4.1 and later versions, you can select TCP as the transport protocol for connecting
supported SIP phones to Cisco Unified CME. Previously only UDP was supported. TCP is selected for
individual SIP phones by using the session-transport command in voice register pool or voice register
template configuration mode. For configuration information, see the “SIP: Selecting Session-Transport
Protocol for a Phone” section on page 171.

How to Configure Phones for a PBX System

This section contains the following tasks:
• SCCP: Creating Directory Numbers, page 158 (required)
• SCCP: Assigning Directory Numbers to Phones, page 160 (required)
• SIP: Creating Directory Numbers, page 162 (required)
• SIP: Assigning Directory Numbers to Phones, page 163 (required)
• SIP: Configuring Dial Plans, page 165 (optional)
• SIP: Verifying Dial Plan Configuration, page 168 (optional)
• SIP: Enabling KPML, page 169 (optional)
• SIP: Selecting Session-Transport Protocol for a Phone, page 171 (optional)
• SIP: Disabling SIP Proxy Registration for a Directory Number, page 172 (required)
• Configuring Codec for Local Calling Between SIP and SCCP Phones, page 174 (required)

SCCP: Creating Directory Numbers

To create a directory number in Cisco Unified CME for a SCCP phone, intercom line, voice port, or a
message-waiting indicator (MWI), perform the following steps for each directory number to be created.
Each ephone-dn becomes a virtual line, or extension, on which call connections can be made. Each
ephone-dn configuration automatically creates one or more virtual dial peers and virtual voice ports to
make those call connections.

Note To create and assign directory numbers to be included in an overlay set, see “SCCP: Configuring
Overlaid Ephone-dns” on page 615.

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Prerequisites
• The maximum number of directory numbers must be configured for other than the default, by using
the max-dn command.

Restrictions
• The Cisco Unified IP Phone 7931G is a SCCP keyset phone and when configured for a key system,
does not support the dual-line option for a directory number. To configure a Cisco Unified IP Phone
7931G, see the “How to Configure Phones for a Key System” section on page 176.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. name name
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an
ephone-dn, and optionally assigns it dual-line status.
Example: • Configuring a dual-line supports features such as call
Router(config)# ephone-dn 55 dual-line waiting, call transfer, and conferencing with a single
ephone-dn.
• 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.
Step 4 number number [secondary number] [no-reg [both Configures a valid extension number for this ephone-dn
| primary]] instance.
• Configuring a secondary number supports features
Example: such as call waiting, call transfer, and conferencing
Router(config-ephone-dn)# number 2345 with a single ephone-dn.

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Command or Action Purpose

Step 5 name name (Optional) Associates a name with this directory number.
This name is used for caller-ID displays and in the local
directory listings.
Example:
Router(config-ephone-dn)# name Smith, John • Must follow the name order that is specified with the
directory command.
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

What to Do Next
After creating directory numbers, you can assign one or more directory number to a Cisco Unified IP
phone. See “SCCP: Assigning Directory Numbers to Phones” section on page 160.

SCCP: Assigning Directory Numbers to Phones

This task sets up the initial ephone-dn-to-ephone relationships—that is, how and which extensions
appear on each phone. To create and modify phone-specific parameters for individual SCCP phones,
perform the following steps for each SCCP phone to be connected in Cisco Unified CME.

Note To create and assign directory numbers to be included in an overlay set, see “SCCP: Configuring
Overlaid Ephone-dns” on page 615.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. mac-address [mac-address]
5. type phone-type [addon 1 module-type [2 module-type]]
6. button button-number{separator}dn-tag [,dn-tag...] [button-number{x}overlay-button-number]
[button-number...]
7. keypad-normalize
8. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 is version and platform-specific.
Type ? to display range.
Step 4 mac-address [mac-address] Specifies the MAC address of the IP phone that is being
configured.
Example: • mac-address—(Optional) For Cisco Unified CME 3.0
Router(config-ephone)# mac-address 2946.3f2.311 and later, not required to register phones before
configuring the phone because Cisco Unified CME
can detect MAC addresses and automatically populate
phone configurations with the MAC addresses and
phone types for individual phones. Not supported for
voice-mail ports.
Step 5 type phone-type [addon 1 module-type Specifies the type of phone.
[2 module-type]]
• Cisco Unified CME 4.0 and later versions—The only
types to which you can apply an add-on module are
Example: 7960, 7961, 7961GE, and 7970.
Router(config-ephone)# type 7960 addon 1 7914
• Cisco CME 3.4 and earlier versions—The only type to
which you can apply an add-on module is 7960.
Step 6 button button-number{separator}dn-tag Associates a button number and line characteristics with an
[,dn-tag...] extension (ephone-dn). Maximum number of buttons is
[button-number{x}overlay-button-number]
[button-number...]
determined by phone type.
Note The Cisco Unified IP Phone 7910 has only one line
button, but can be given two ephone-dn tags.
Example:
Router(config-ephone)# button 1:10 2:11 3b12
4o13,14,15
Step 7 keypad-normalize (Optional) Imposes a 200-millisecond delay before each
keypad message from an IP phone.
Example:
Router(config-ephone)# keypad-normalize
Step 8 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

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What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• After configuring phones in Cisco Unified CME to make basic calls, you are ready to generate
configuration files for the phones to be connected. See “SCCP: Generating Configuration Files for
SCCP Phones” on page 247.

Examples
The following example assigns extension 2225 in the Accounting Department to button 1 on ephone 2.
ephone-dn 25
number 2225
name Accounting

ephone 2
mac-address 00E1.CB13.0395
type 7960
button 1:25

SIP: Creating Directory Numbers

To create a directory number in Cisco Unified CME for a SIP phone, intercom line, voice port, or a
message-waiting indicator (MWI), perform the following steps for each directory number to be created.
Each directory number becomes a virtual line, or extension, on which call connections can be made.
Each directory number configuration automatically creates one or more virtual dial peers and virtual
voice ports to make those call connections.

Prerequisites
• Cisco CME 3.4 or a later version.
• The maximum number of directory numbers supported by a router is version and platform
dependent. To configure more directory numbers than the default, use the max-dn (voice register
global) command before performing this procedure. For configuration information, see “SIP:
Setting Up Cisco Unified CME” on page 135.

Restrictions
• Call forward all, presence, and message-waiting indication (MWI) features in Cisco Unified CME
4.1 and later versions require that SIP phones are configured with a directory number (using dn
keyword in number command); direct line numbers are not supported.
• Shared lines are not supported by SIP phones.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register dn dn-tag

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4. number number
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or a message-waiting indicator (MWI).
Example:
Router(config)# voice register dn 17
Step 4 number number Defines a valid number for a directory number.

Example:
Router(config-register-dn)# number 7001
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-dn)# end

SIP: Assigning Directory Numbers to Phones

This task sets up which extensions appear on each phone. To create and modify phone-specific
parameters for individual SIP phones, perform the following steps for each SIP phone to be connected
in Cisco Unified CME.

Note If your Cisco Unified CME system supports SCCP and also SIP phones, do not connect your SIP phones
to your network until after you have verified the configuration profile for the SIP phone.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register pool pool-tag
4. id mac address
5. type phone-type
6. number tag dn dn-tag

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7. username name password string

8. dtmf-relay [cisco-rtp] [rtp-nte] [sip-notify]
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone in
Cisco Unified CME.
Example:
Router(config)# voice register pool 3
Step 4 id {network address mask mask | ip address mask Explicitly identifies a locally available individual SIP phone to
mask | mac address} support a degree of authentication.

Example:
Router(config-register-pool)# id mac
0009.A3D4.1234
Step 5 type phone-type Defines a phone type for the SIP phone being configured.

Example:
Router(config-register-pool)# type 7960-7940
Step 6 number tag dn dn-tag Associates a directory number with the SIP phone being
configured.
Example: • dn dn-tag—Identifies the directory number for this SIP
Router(config-register-pool)# number 1 dn 17 phone as defined by the voice register dn command.
Step 7 username username password string (Optional) Required only if authentication is enabled with
the authenticate command. Creates an authentication
credential.
Example:
Router(config-register-pool)# username smith Note This command is not for SIP proxy registration. The
password 123zyx password will not be encrypted. All lines in a phone
will share the same credential.

• username—Identifies a local Cisco Unified IP phone

user. Default: Admin.

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Command or Action Purpose

Step 8 dtmf-relay {[cisco-rtp] [rtp-nte] [sip-notify]} (Optional) Specifies a list of DTMF relay methods that can
be used by the SIP phone being configured to relay DTMF
tones.
Example:
Router(config-register-pool)# dtmf-relay Note SIP phones natively support in-band DTMF relay as
rtp-nte specified in RFC 2833.
Step 9 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-pool)# end

What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• If you want to select the session-transport protocol for a SIP phone, see the “SIP: Selecting
Session-Transport Protocol for a Phone” section on page 171.
• If you are finished configuring phones to make basic calls, you are ready to generate configuration
files for the phones to be connected. See “SIP: Generating Configuration Profiles for SIP Phones”
on page 250.

SIP: Configuring Dial Plans

Dial plans enable SIP phones to recognize digit strings dialed by users. After the phone recognizes a dial
pattern, it automatically sends a SIP INVITE message to Cisco Unified CME to initiate the call and does
not require the user to press the Dial key or wait for the interdigit timeout. To define a dial plan for a SIP
phone, perform the following steps.

Prerequisites
• Cisco Unified CME 4.1 or a later version.
• mode cme command must be enabled in Cisco Unified CME.

Restrictions
• If you create a dial plan by downloading a custom XML dial pattern file to flash and using the
filename command, and the XML file contains an error, the dial plan might not work properly on a
phone. We recommend creating a dial pattern file using the pattern command.
• To remove a dial plan that was created using a custom XML file with the filename command, you
must remove the dial plan from the phone, create a new configuration profile, and then use the reset
command to reboot the phone. You can use the restart command after removing a dial plan from a
phone only if the dial plan was created using the pattern command.
• To use KPML if a matching dial plan is not found, when both a dial plan and KPML are enabled on
a phone, you must configure a dial pattern with a single wildcard character (.) as the last pattern in
the dial plan. For example:
voice register dialplan 10

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type 7940-7960-others
pattern 1 66...
pattern 2 91.......
pattern 3 .

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register dialplan dialplan-tag
4. type phone-type
5. pattern tag string [button button-number] [timeout seconds] [user {ip | phone}]
or
filename filename
6. exit
7. voice register pool pool-tag
8. dialplan dialplan-tag
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register dialplan dialplan-tag Enters voice register dialplan configuration mode to define
a dial plan for SIP phones.
Example:
Router(config)# voice register dialplan 1

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Command or Action Purpose

Step 4 type phone-type Defines a phone type for the SIP dial plan.
• 7905-7912—Cisco Unified IP Phone 7905, 7905G,
Example: 7912, or 7912G.
Router(config-register-dialplan)# type
7905-7912
• 7940-7960-others—Cisco Unified IP Phone 7911,
7940, 7940G, 7941, 7941GE, 7960, 7960G, 7961,
7961GE, 7970, or 7971.
• The phone type specified with this command must
match the type of phone for which the dial plan is used.
If this phone type does not match the type assigned to
the phone with the type command in voice register pool
mode, the dial-plan configuration file is not generated.
• You must enter this command before using the pattern
or filename command in the next step.
Step 5 pattern tag string [button button-number] Defines a dial pattern for a SIP dial plan.
[timeout seconds] [user {ip | phone}]
or • tag—Number that identifies the dial pattern.
Range: 1 to 24.
filename filename
• string—Dial pattern, such as the area code, prefix, and
first one or two digits of the telephone number, plus
Example: wildcard characters or dots (.) for the remainder of the
Router(config-register-dialplan)# pattern 1
dialed digits.
52...
or • button button-number—(Optional) Button to which
Router(config-register-dialplan)# filename the dial pattern applies.
dialsip • timeout seconds—(Optional) Time, in seconds, that
the system waits before dialing the number entered by
the user. Range: 0 to 30. To have the number dialed
immediately, specify 0. If you do not use this
parameter, the phone's default interdigit timeout value
is used (10 seconds).
• user—(Optional) Tag that automatically gets added to
the dialed number. Do not use this keyword if
Cisco Unified CME is the only SIP call agent.
• ip—Uses the IP address of the user.
• phone—Uses the phone number of the user.
• Repeat this command for each pattern that you want to
include in this dial plan.
or
Specifies a custom XML file that contains the dial patterns
to use for the SIP dial plan.
• You must load the custom XML file must into flash and
the filename cannot include the .xml extension.
• The filename command is not supported for the
Cisco Unified IP Phone 7905 or 7912.

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Command or Action Purpose

Step 6 exit Exits dialplan configuration mode.

Example:
Router(config-register-dialplan)# exit
Step 7 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone.
Example: • pool-tag—Unique sequence number of the SIP phone
Router(config)# voice register pool 4 to be configured. Range is version and
platform-dependent; type ? to display range. You can
modify the upper limit for this argument with the
max-pool command.
Step 8 dialplan dialplan-tag Assigns a dial plan to a SIP phone.
• dialplan-tag—Number that identifies the dial plan to
Example: use for this SIP phone. This is the number that was used
Router(config-register-pool)# dialplan 1 with the voice register dialplan command in Step 3.
Range: 1 to 24.
Step 9 end Exits to privileged EXEC mode.

Example:
Router(config-register-global)# end

Examples
The following example shows the configuration for dial plan 1 which is assigned to SIP phone 1.
voice register dialplan 1
type 7940-7960-others
pattern 1 2... timeout 10 user ip
pattern 2 1234 user ip button 4
pattern 3 65...
pattern 4 1...!
!
voice register pool 1
id mac 0016.9DEF.1A70
type 7961GE
number 1 dn 1
number 2 dn 2
dialplan 1
dtmf-relay rtp-nte
codec g711ulaw

What to Do Next
If you are done modifying parameters for SIP phones, you must generate a new configuration profile and
restart the phones. See “Generating Configuration Files for Phones” on page 245.

SIP: Verifying Dial Plan Configuration

Step 1 show voice register dialplan tag

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This command displays the configuration information for a specific SIP dial plan.
Router# show voice register dialplan 1

Dialplan Tag 1
Config:
Type is 7940-7960-others
Pattern 1 is 2..., timeout is 10, user option is ip, button is default
Pattern 2 is 1234, timeout is 0, user option is ip, button is 4
Pattern 3 is 65..., timeout is 0, user option is phone, button is default
Pattern 4 is 1..., timeout is 0, user option is phone, button is default

Step 2 show voice register pool tag

This command displays the dial plan assigned to a specific SIP phone.
Router# show voice register pool 29

Pool Tag 29
Config:
Mac address is 0012.7F54.EDC6
Number list 1 : DN 29
Proxy Ip address is 0.0.0.0
DTMF Relay is disabled
Call Waiting is enabled
DnD is disabled
keep-conference is enabled
dialplan tag is 1
kpml signal is enabled
service-control mechanism is not supported
.
.
.
Step 3 show voice register template tag
This command displays the dial plan assigned to a specific template.
Router# show voice register template 3

Temp Tag 3
Config:
Attended Transfer is disabled
Blind Transfer is enabled
Semi-attended Transfer is enabled
Conference is enabled
Caller-ID block is disabled
DnD control is enabled
Anonymous call block is disabled
Voicemail is 62000, timeout 15
Dialplan Tag is 1
Transport type is tcp

SIP: Enabling KPML

To enable KPML digit collection on a SIP phone, perform the following steps.

Prerequisites
• Cisco Unified CME 4.1 or a later version.

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Restrictions
• This feature is supported only on Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE,
7970G, and 7971GE.
• A dial plan assigned to a phone has priority over KPML.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register pool pool-tag
4. digit collect kpml
5. end
6. show voice register dial-peer

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone.
Example: • pool-tag—Unique sequence number of the SIP phone
Router(config)# voice register pool 4 to be configured. Range is version and
platform-dependent; type ? to display range. You can
modify the upper limit for this argument with the
max-pool command.
Step 4 digit collect kpml Enables KPML digit collection for the SIP phone.
Note This command is enabled by default for supported
Example: phones in Cisco Unified CME.
Router(config-register-pool)# digit collect
kpml
Step 5 end Exits to privileged EXEC mode.

Example:
Router(config-register-pool)# end
Step 6 show voice register dial-peers Displays details of all dynamically created VoIP dial peers
associated with the Cisco Unified CME SIP register
including the defined digit collection method.
Example:
Router# show voice register dial-peers

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What to Do Next
If you are done modifying parameters for SIP phones, you must generate a new configuration profile and
restart the phones. See “Generating Configuration Files for Phones” on page 245.

SIP: Selecting Session-Transport Protocol for a Phone

To change the session-transport protocol for a SIP phone to TCP, from the default of UDP, perform the
following steps.

Prerequisites
• Cisco Unified CME 4.1 or a later version.
• SIP phone to which configuration is to be applied must be already configured. For configuration
information, see the “SIP: Assigning Directory Numbers to Phones” section on page 163.

Restrictions
• TCP is not supported as a session-transport protocol for the Cisco Unified IP Phone 7905, 7912,
7940, or 7960. If TCP is assigned to an unsupported phone using this command, calls to that phone
will not complete successfully. The phone can originate calls, but it uses UDP, although TCP has
been assigned.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register pool pool-tag
4. session-transport {tcp | udp}
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone in
Cisco Unified CME.
Example:
Router(config)# voice register pool 3

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Command or Action Purpose

Step 4 session-transport {tcp | udp} (Optional) Specifies the transport layer protocol that a SIP
phone uses to connect to Cisco Unified CME.
Example: • This command can also be configured in voice register
Router(config-register-pool)# session-transport template configuration mode and applied to one or
tcp more phones. The voice register pool configuration has
priority over the voice register template configuration.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-pool)# end

What to Do Next
• If you want to disable SIP Proxy registration for an individual directory number, see the “SIP:
Disabling SIP Proxy Registration for a Directory Number” section on page 172.
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• If you are finished configuring phones to make basic calls, you are ready to generate configuration
files for the phones to be connected. See “SIP: Generating Configuration Profiles for SIP Phones”
on page 250

SIP: Disabling SIP Proxy Registration for a Directory Number

To prevent a particular directory number from registering with an external SIP proxy server, perform the
following steps.

Prerequisites
• Cisco Unified CME 3.4 or a later version.
• Bulk registration is configured at system level. For configuration information, see “Configuring
Bulk Registration” on page 123.

Restrictions
• Phone numbers that are registered under voice register dn must belong to a SIP phone that is itself
registered in Cisco Unified CME.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register dn dn-tag
4. number number

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5. no-reg
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or an MWI.
Example:
Router(config-register-global)# voice register
dn 1
Step 4 number number Defines a valid number for a directory number to be
assigned to a SIP phone in Cisco Unified CME.
Example:
Router(config-register-dn)# number 4085550152
Step 5 no-reg Causes directory number being configured to not register
with an external proxy server.
Example:
Router(config-register-dn)# no-reg
Step 6 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-dn)# end

What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• If you are finished configuring phones to make basic calls, you are ready to generate configuration
files for the phones to be connected. See “SIP: Generating Configuration Profiles for SIP Phones”
on page 250

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Configuring Codec for Local Calling Between SIP and SCCP Phones
To designate a codec for individual phones to ensure connectivity between SIP and SCCP phones
connected to the same Cisco Unified CME router, perform the following steps for each SIP or SCCP
phone.

Note If codec values for the dial peers of an internal connection do not match, the call fails.

Prerequisites
• Cisco Unified CME 3.4 or a later version.
• Cisco Unified IP phone to which codec is to be applied must be already configured. For
configuration information for SIP phones, see the “SIP: Assigning Directory Numbers to Phones”
section on page 163. For configuration information for SCCP phones, see the “SCCP: Assigning
Directory Numbers to Phones” section on page 160.

Restrictions
• Required only if you have SIP and SCCP phones connected to the same Cisco Unified CME router.
• Modify the configuration for either SIP or SCCP phones to ensure that the codec for all phones
match. Do not modify the configuration for SIP and SCCP phones.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone ephone-tag
or
voice register pool-tag
4. codec codec-type
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 ephone ephone-tag Enters ephone configuration mode to set phone-specific
or parameters for a SCCP phone in Cisco Unified CME.
voice register pool pool-tag or
Enters voice register pool configuration mode to set
Example: phone-specific parameters for a SIP phone in
Router(config)# ephone 3 Cisco Unified CME.
or
Router(config)# voice register pool 3
Step 4 codec codec-type Specifies the codec for the dial peer dynamically created
when the SIP phone being configured, registers.
Example: • This command overrides any previously configured
Router(config-ephone)# codec g729r8 codec selection set with the voice-class codec
or command.
Router(config-register-pool)# codec g711alaw • If G.729 is the desired codec for Cisco ATA-186 and
Cisco ATA-188, then only one port of the Cisco ATA
device should be configured in Cisco Unified CME. If
a call is placed to the 2nd port of the Cisco ATA device,
it will be disconnected gracefully. If a you want to use
both Cisco ATA ports simultaneously, then configure
G.711 in Cisco Unified CME.
• SCCP only—This command can also be configured in
ephone- template configuration mode and applied to
one or more phones.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone)# end
or
Router(config-register-pool)# end

What to Do Next
• If you want to select the session-transport protocol for a SIP phone, see the “SIP: Selecting
Session-Transport Protocol for a Phone” section on page 171.
• If you are finished configuring SIP phones to make basic calls using, you are ready to generate
configuration files for the phones to be connected. See “SIP: Generating Configuration Profiles for
SIP Phones” on page 250.
• If you are finished configuring SCCP phones to make basic calls, you are ready to generate
configuration files for the phones to be connected. See “SCCP: Generating Configuration Files for
SCCP Phones” on page 247.

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How to Configure Phones for a Key System

This section contains the following tasks:
• SCCP: Creating Directory Numbers for a Simple Key System, page 176 (required)
• SCCP: Configuring Trunk Lines for a Key System, page 179 (required)
• SCCP: Configuring Individual IP Phones for Key System, page 187 (required)

SCCP: Creating Directory Numbers for a Simple Key System

To create a set of directory numbers with the same number to be associated with multiple line buttons
on an IP phone and provide support for call waiting and call transfer on a key system phone, perform the
following steps.

Restrictions
• Do not configure directory numbers for a key system for dual-line mode because this does not
conform to the key system one-call-per-line button usage model for which the phone is designed.
• Provisioning support for the Cisco Unified IP Phone 7931 is available only in Cisco Unified CME
4.0(2) and later versions.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag
4. number number [secondary number] [no-reg [both | primary]]
5. preference preference-order
6. no huntstop
or
huntstop
7. mwi-type {visual | audio | both}
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode to create a
directory number.
Example:
Router(config)# ephone-dn 11
Step 4 number number [secondary number] [no-reg [both | Configures a valid phone or extension number for this
primary]] directory number.

Example:
Router(config-ephone-dn)# number 101
Step 5 preference preference-order Sets dial-peer preference order for a directory number
associated with a Cisco Unified IP phone.
Example:
• Default: 0.
Router(config-ephone-dn)# preference 1
• Increment the preference order for all subsequent
instances within a set of ephone dns with the same
number to be associated with a key system phone.
That is, the first instance of the directory number is
preference 0 by default and you must specify 1 for
the second instance of the same number, 2 for the
next, and so on. This allows you to create multiple
buttons with the same number on an IP phone.
• Required to support call waiting and call transfer on
a key system phone.
Step 6 no huntstop Explicitly enables call hunting behavior for a directory
number.
Example: • Configure no huntstop for all instances, except the
Router(config-ephone-dn)# no huntstop final instance, within a set of ephone dns with the
same number to be associated with a key system
or phone.
huntstop
• Required to allow call hunting across multiple line
buttons with the same number on an IP phone.
Example: or
Router(config-ephone-dn)# huntstop
Disables call hunting behavior for a directory number.
• Configure the huntstop command for the final
instance within a set of ephone dns with the same
number to be associated with a key system phone.
• Required to limit the call hunting to a set of multiple
line buttons with the same number on an IP phone.

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Command or Action Purpose

Step 7 mwi-type {visual | audio | both} Specifies the type of MWI notification to be received.
• This command is supported only by Cisco Unified
Example: IP Phone 7931s and Cisco Unified IP Phone 7911s.
Router(config-ephone-dn)# mwi-type audible
• This command can also be configured in
ephone-dn-template configuration mode. The value
set in ephone-dn configuration mode has priority
over the value set in ephone-dn-template mode.
Step 8 end Exits to privileged EXEC mode.

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

Examples

The following example shows the configuration for six instances of directory number 101, assigned to
the first six buttons of an IP phone.
ephone-dn 10
number 101
no huntstop

ephone-dn 11
number 101
preference 1
no huntstop

ephone-dn 12
number 101
preference 2
no huntstop

ephone-dn 13
number 101
preference 3
no huntstop

ephone-dn 14
number 101
preference 4
no huntstop

ephone-dn 15
number 101
preference 5

ephone 1
mac-address 0001.2345.6789
type 7931
button 1:10 2:11 3:12 4:13 5:14 6:15

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SCCP: Configuring Trunk Lines for a Key System

To set up trunk lines for your key system, perform only one of the following procedures:
• To only enable direct status monitoring of the FXO port on the line button of the IP phone, see the
“SCCP: Configuring a Simple Key System Phone Trunk Line Configuration” section on page 179
• To enable direct status monitoring and allow transferred PSTN FXO line calls to be automatically
recalled if the transfer target does not answer, see the “SCCP: Configuring an Advanced Key System
Phone Trunk Line Configuration” section on page 182.

SCCP: Configuring a Simple Key System Phone Trunk Line Configuration

Perform the steps in this section to:
• Create directory numbers corresponding to each FXO line that allows phones to have shared or
private lines connected directly to the PSTN.
• Enable direct status monitoring of the FXO port on the line button of the IP phone. The line button
indicator, either a lamp or an icon depending on the phone, shows the in-use status of the FXO port
during the duration of the call.

Prerequisites

• FXO port for a private line automatic ringdown (PLAR) off-premises extension (OPX) connection
must be configured; for example:
voice-port 1/0/0
connection plar-opx 801 <<----Private number

• Dial peers for FXO port must be configured; for example:

dial-peer voice 111 pots
destination-pattern 811 <<----Trunk-tag
port 1/0/0

Restrictions

• A directory number with a trunk line cannot be configured for call forward, busy, or no answer.
• 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 Unified CME router. Only the trunk tag is 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.
• FXO trunk lines do not support bulk speed dial.

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• FXO port monitoring has the following restrictions:

– Not supported before Cisco Unified CME 4.0.
– Supported only for analog FXO loop-start and ground-start ports and T1/E1 FXO CAS ports.
FXS loop-start and ground-start ports and PRI/BRI PSTN trunks are not supported.
– Not supported for analog ports on the Cisco VG 224 or Cisco ATA 180 Series.
– T1 CAS DS0 group must be configured per time slot (cannot bundle more than one time slot
into a ds0-group).
• Transfer recall and transfer-to button optimization are supported on dual-line directory numbers
only in Cisco Unified CME 4.0 and later.
• Transfer-to button optimization is not supported for call forwarding, call-park recall, call pickup on
hold, or call pickup at alert.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag
4. number number [secondary number] [no-reg [both | primary]]
5. trunk digit-string [timeout seconds] monitor-port port
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode to create a
directory number.
Example: • Configure this command in the default single line
Router(config)# ephone-dn 51 mode, without the dual-line keyword, when
configuring a simple key system trunk line.
Step 4 number number [secondary number] [no-reg [both | Configures a valid phone or extension number for this
primary]] directory number.

Example:
Router(config-ephone-dn)# number 801

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Command or Action Purpose

Step 5 trunk trunk-tag [timeout seconds] monitor-port Associates a directory number with a foreign exchange
port office (FXO) port.
• The monitor-port keyword is not supported before
Example: Cisco Unified CME 4.0.
Router(config-ephone-dn)# trunk 811 monitor-port
1/0/0 • The monitor-port keyword is not supported on
directory numbers for analog ports on the
Cisco VG 224 or Cisco ATA 180 Series.
Step 6 end Returns to privileged EXEC mode.

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

Examples

The following example shows the configuration for six instances of directory number 101, assigned to
the first six buttons of an IP phone, plus four PSTN line appearances that are assigned to buttons 7 to 10.
ephone-dn 10
number 101
no huntstop

ephone-dn 11
number 101
preference 1
no huntstop

ephone-dn 12
number 101
preference 2
no huntstop

ephone-dn 13
number 101
preference 3
no huntstop

ephone-dn 14
number 101
preference 4
no huntstop

ephone-dn 15
number 101
preference 5

ephone-dn 51
number 801
trunk 811 monitor-port 1/0/0

ephone-dn 52
number 802
trunk 812 monitor-port 1/0/1

ephone-dn 53
number 803
trunk 813 monitor-port 1/0/2

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ephone-dn 54
number 804
trunk 814 monitor-port 1/0/3

ephone 1
mac-address 0001.2345.6789
type 7931
button 1:11 2:12 3:13 4:14 5:15 6:16 7:51 8:52 9:53 10:54

voice-port 1/0/0
connection plar opx 801

voice-port 1/0/1
connection plar opx 802

voice-port 1/0/2
connection plar opx 803

voice-port 1/0/3
connection plar opx 804

dial-peer voice 811 pots

destination-pattern 811
port 1/0/0

dial-peer voice 812 pots

destination-pattern 812
port 1/0/1

dial-peer voice 813 pots

destination-pattern 813
port 1/0/2

dial-peer voice 814 pots

destination-pattern 814
port 1/0/3

What to Do Next

You are ready to configure each individual phone and assign button numbers, line characteristics, and
directory numbers to buttons on the phone. See the “SCCP: Configuring Individual IP Phones for Key
System” section on page 187.

SCCP: Configuring an Advanced Key System Phone Trunk Line Configuration

Perform the steps in this section to:
• Create directory numbers corresponding to each FXO line that allows phones to have shared or
private lines connected directly to the PSTN.
• Enable direct status monitoring of the FXO port on the line button of the IP phone. The line button
indicator, either a lamp or an icon depending on the phone, shows the in-use status of the FXO port
during the duration of the call.
• Allow transferred PSTN FXO line calls to be automatically recalled if the transfer target does not
answer after the specified number of seconds. The call is withdrawn from the transfer-to phone and
the call resumes ringing on the phone that initiated the transfer.

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Prerequisites

• FXO port for a private line automatic ringdown (PLAR) off-premises extension (OPX) connection
must be configured; for example:
voice-port 1/0/0
connection plar-opx 801 <<----Private number

• Dial peers for FXO port must be configured; for example:

dial-peer voice 111 pots
destination-pattern 811 <<----Trunk-tag
port 1/0/0

Restrictions

• An ephone-dn with a trunk line cannot be configured for call forward, busy, or no answer.
• 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 Unified CME router. Only the trunk tag is 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.
• FXO trunk lines do not support bulk speed dial.
• FXO port monitoring has the following restrictions:
– Not supported before Cisco Unified CME 4.0.
– Supported only for analog FXO loop-start and ground-start ports and T1/E1 FXO CAS ports.
FXS loop-start and ground-start ports and PRI/BRI PSTN trunks are not supported.
– Not supported for analog ports on the Cisco VG 224 or Cisco ATA 180 Series.
– T1 CAS DS0 group must be configured per time slot (cannot bundle more than one time slot
into a ds0-group).
• Transfer recall and transfer-to button optimization is supported on dual-line directory numbers only
in Cisco Unified CME 4.0 and later.
• Transfer-to button optimization is not supported for call forwarding, call-park recall, call pickup on
hold, or call pickup at alert.
• Transfer recall is not supported for analog ports on the Cisco VG 224 or Cisco ATA 180 Series.

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SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag dual-line
4. number number [secondary number] [no-reg [both | primary]]
5. trunk digit-string [timeout seconds] [transfer-timeout seconds] [monitor-port port]
6. huntstop [channel]
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag dual-line Enters ephone-dn configuration mode for the purposes of
creating and configuring a telephone or extension
number.
Example:
Router(config)# ephone-dn 51 dual-line • dual-line—Required when configuring an advanced
key system phone trunk line. Dual-line mode
provides a second call channel for the directory
number on which to place an outbound consultation
call during the call transfer attempt. This also allows
the phone to remain part of the call in order to
monitor the progress of the transfer attempt and if
the transfer is not answered, to pull the call back to
the phone on the original PSTN line button.
Step 4 number number [secondary number] [no-reg [both | Configures a valid telephone number or extension
primary]] number for this directory number.

Example:
Router(config-ephone-dn)# number 801

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Command or Action Purpose

Step 5 trunk digit-string [ timeout seconds] Associates this directory number with a foreign
[transfer-timeout seconds] [monitor-port port] exchange office (FXO) port.
• transfer-timeout seconds—For dual-line
Example: ephone-dns only. Range: 5 to 60000.
Router(config-ephone-dn)# trunk 811 Default: Disabled.
transfer-timeout 30 monitor-port 1/0/0
• The monitor-port keyword is not supported before
Cisco Unified CME 4.0.
• The monitor-port and transfer-timeout keywords
are not supported on directory numbers for analog
ports on the Cisco VG 224 or Cisco ATA 180 Series.
Step 6 huntstop [channel] Disables call hunting to the second channel of this
directory number if the first channel is busy or does not
answer.
Example:
Router(config-ephone-dn)# huntstop channel • channel—Required when configuring an advanced
key system phone trunk line. Reserves the second
channel created by configuring dual-line mode for
the ephone-dn command so that an outbound
consultation call can be placed during a call transfer
attempt.
Step 7 end Exits to privileged EXEC mode.

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

Examples

The following example shows the configuration for six instances of directory number 101, assigned to
the first six buttons of an IP phone, plus four PSTN line appearances that are assigned to buttons 7 to 10.
These four PSTN line appearances are configured as dual lines to provide a second call channel on which
to place an outbound consultation call during a call transfer attempt. This configuration allows the phone
to remain part of the call in order to monitor the progress of the transfer attempt, and if the transfer is
not answered, to pull the call back to the phone on the original PSTN line button.
ephone-dn 10
number 101
no huntstop

ephone-dn 11
number 101
preference 1
no huntstop

ephone-dn 12
number 101
preference 2
no huntstop

ephone-dn 13
number 101
preference 3
no huntstop

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ephone-dn 14
number 101
preference 4
no huntstop

ephone-dn 15
number 101
preference 5

ephone-dn 51 dual-line
number 801
trunk 811 transfer-timeout 30 monitor-port 1/0/0
huntstop channel

ephone-dn 52 dual-line
number 802
trunk 812 transfer-timeout 30 monitor-port 1/0/1
huntstop channel

ephone-dn 53 dual-line
number 803
trunk 813 transfer-timeout 30 monitor-port 1/0/2
huntstop channel

ephone-dn 54 dual-line
number 804
trunk 814 transfer-timeout 30 monitor-port 1/0/3
huntstop channel

ephone 1
mac-address 0001.2345.6789
type 7931
button 1:11 2:12 3:13 4:14 5:15 6:16 7:51 8:52 9:53 10:54

voice-port 1/0/0
connection plar opx 801

voice-port 1/0/1
connection plar opx 802

voice-port 1/0/2
connection plar opx 803

voice-port 1/0/3
connection plar opx 804

dial-peer voice 811 pots

destination-pattern 811
port 1/0/0

dial-peer voice 812 pots

destination-pattern 812
port 1/0/1

dial-peer voice 813 pots

destination-pattern 813
port 1/0/2

dial-peer voice 814 pots

destination-pattern 814
port 1/0/3

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SCCP: Configuring Individual IP Phones for Key System

To assign button numbers, line characteristics, and directory numbers to buttons on an individual phone
to operate as a key system phone, perform the following steps.

Restrictions
• Provisioning for Cisco Unified IP Phone 7931G is available only in Cisco Unified CME 4.0(2) and
later versions.
• Cisco Unified IP Phone 7931G can support only one call waiting overlaid per directory number.
• Cisco Unified IP Phone 7931G cannot support overlays that contain directory numbers configured
for dual-line mode.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. mac-address [mac-address]
5. type phone-type
6. button button-number{separator}dn-tag [,dn-tag...] [button-number{x}overlay-button-number]
[button-number...]
7. mwi-line line-number
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.

Example:
Router(config)# ephone 1
Step 4 mac-address [mac-address] Specifies the MAC address of the IP phone that is being
configured.
Example:
Router(config-ephone)# mac-address 0001.2345.6789

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Command or Action Purpose

Step 5 type phone-type Specifies the type of phone that is being configured.

Example:
Router(config-ephone)# type 7931
Step 6 button button-number{separator}dn-tag Associates a button number and line characteristics with
[,dn-tag...] an ephone-dn. Maximum number of buttons is
[button-number{x}overlay-button-number]
[button-number...]
determined by phone type.
Tip The line button layout for the Cisco Unified IP
Phone 7931G is a bottom-up array. Button 1 is at
Example:
the bottom right of the array and button 24 is at
Router(config-ephone)# button 1:11 2:12 3:13 4:14
5:15 6:16 7:51 8:52 9:53 10:54 the top left of the array.
Step 7 mwi-line line-number Selects a phone line to receive MWI treatment; when a
message is waiting for the selected line, the message
waiting indicator is activated.
Example:
Router(config-ephone)# mwi-line 3 • line-number—Range: 1 to 34. Default: 1.
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone)# end

What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see “SCCP: Selecting Button
Layout for a Cisco Unified IP Phone 7931G” on page 897 .
• If you are finished configuring phones to make basic calls, you are ready to generate configuration
files for the phones to be connected. See “SCCP: Generating Configuration Files for SCCP Phones”
on page 247.

How to Configure Cisco ATA, Analog Phone Support, Remote

Phones, and Cisco IP Communicator
This section contains the following tasks:
• Configuring Cisco ATA Support, page 189 (required)
• Verifying Cisco ATA Support, page 190 (optional)
• Using Call Pickup and Group Call Pickup with Cisco ATA, page 192 (optional)
• SCCP: Configuring Analog Phone Support, page 193 (required)
• SCCP: Verifying Analog Phone Support, page 196 (optional)
• SCCP: Enabling a Remote Phone, page 196 (required)
• SCCP: Verifying Remote Phones, page 198 (optional)

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• SCCP: Configuring Cisco IP Communicator Support, page 198 (required)

• SCCP: Troubleshooting Cisco IP Communicator Support, page 199 (optional)

Configuring Cisco ATA Support

To enable an analog phone that uses a Cisco ATA to register with Cisco Unified CME, perform the
following steps.

Restrictions
For a Cisco ATA that is registered to a Cisco Unified CME system to participate in fax calls, it must have
its ConnectMode parameter set to use the same RTP payload type as the Cisco voice gateway that is
performing the fax pass-through. Cisco voice gateways use standard payload type 0/8, which is selected
on Cisco ATAs by setting bit 2 of the ConnectMode parameter to 1. For more information, see the
“Parameters and Defaults” chapter in the Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor
Administrator's Guide for SCCP (version 3.0).

SUMMARY STEPS

1. Install Cisco ATA.

2. Configure Cisco ATA for SCCP.
3. Upgrade firmware.
4. Set network parameters on Cisco ATA.
5. Configure analog phones in Cisco Unified CME.

DETAILED STEPS

Step 1 Install the Cisco ATA. See the “Installing the Cisco ATA” chapter in the in Cisco ATA 186 and
Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide for SCCP (version 3.0).
Step 2 Configure the Cisco ATA. See the “Configuring the Cisco ATA for SCCP” chapter in the Cisco ATA 186
and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide for SCCP (version 3.0).
Step 3 Upgrade to the latest Cisco ATA image. If you are using either the v2.14 or v2.14ms Cisco ATA 186
image based on the 2.14 020315a build for H.323/SIP or the 2.14 020415a build for MGCP or SCCP,
you must upgrade to the latest version to install a security patch. This patch fixes a security hole in the
Cisco ATA Web server that allows users to bypass the user interface password.
For information about upgrading firmware, see “Installing and Upgrading Cisco Unified CME
Software” on page 69. Alternatively, you can use a manual method, as described in the “Upgrading the
Cisco ATA Signaling Image” chapter of the Cisco ATA 186 and Cisco ATA 188 Analog Telephone
Adaptor Administrator’s Guide for SCCP (version 3.0).
Step 4 Configure the Cisco ATA to set the following parameters:
– DHCP parameter to 1 (enabled).
– TFTP parameter to 1 (enabled).
– TFTPURL parameter to the IP address of the router running Cisco Unified CME.
– SID0 parameter to a period (.) or the MAC address of the Cisco ATA (to enable the first port).

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– SID1 parameter to a period (.) or a modified version the Cisco ATA’s MAC address, with the
first two hexadecimal numbers removed and 01 appended to the end, if you want to use the
second port. For example, if the MAC address of the Cisco ATA is 00012D01073D, set SID1 to
012D01073D01.
– Nprintf parameter to the IP address and port number of the host to which all Cisco ATA debug
messages are sent. The port number is usually set to 9001.
– To prevent tampering and unauthorized access to the Cisco ATA 186, you can disable the
web-based configuration. However, if you disable the web configuration page, you must use
either a TFTP server or the voice configuration menu to configure the Cisco ATA 186.
Step 5 Configure analog phones that use a Cisco ATA in the same way as a Cisco Unified IP phone. In the type
command, use the ata keyword. For information on how to provision phones, see the “SCCP: Creating
Directory Numbers” section on page 158.

What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see “SCCP: Selecting Button
Layout for a Cisco Unified IP Phone 7931G” on page 897 .
• If you are finished configuring phones to make basic calls, you are ready to generate configuration
files for the phones to be connected. See “SCCP: Generating Configuration Files for SCCP Phones”
on page 247 and “SIP: Generating Configuration Profiles for SIP Phones” on page 250.

Verifying Cisco ATA Support

Use the show ephone ata command to display SCCP phone configurations with the type ata command.
The following is sample output for a Cisco Unified CME configured for two analog phones using a
Cisco ATA with MAC address 000F.F758.E70E.

ephone-30 Mac:000F.F758.E70E TCP socket:[2] activeLine:0 REGISTERED in SCCP ver 1 and

Server in ver 1
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:7
IP:1.4.188.72 15325 ATA Phone keepalive 7 max_line 2 dual-line
button 1: dn 80 number 8080 CH1 IDLE CH2 IDLE

ephone-31 Mac:0FF7.58E7.0E01 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 1 and

Server in ver 1
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:3
IP:1.4.188.72 15400 ATA Phone keepalive 7 max_line 2 dual-line
button 1: dn 81 number 8081 CH1 IDLE CH2 IDLE

Troubleshooting Cisco ATA Support

Use the debug ephone detail command to diagnose problems with analog phones that use Cisco ATAs.
For more information, see the Cisco IOS Debug Command Reference for your Cisco IOS release.
The following is sample output for two analog phones using a Cisco ATA with MAC address
000F.F758.E70E. The sample shows the activities that take place when the phones register.

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Router# debug ephone detail mac-address 000F.F758.E70E

*Apr 5 02:50:11.966: New Skinny socket accepted [1] (33 active)

*Apr 5 02:50:11.970: sin_family 2, sin_port 15325, in_addr 1.4.188.72
*Apr 5 02:50:11.970: skinny_add_socket 1 1.4.188.72 15325
21:21:49: %IPPHONE-6-REG_ALARM: Name=ATA000FF758E70E Load=ATA030203SCCP051201A.zup
Last=Initialized
*Apr 5 02:50:11.974:
Skinny StationAlarmMessage on socket [2] 1.4.188.72 ATA000FF758E70E
*Apr 5 02:50:11.974: severityInformational p1=0 [0x0] p2=0 [0x0]
*Apr 5 02:50:11.974: Name=ATA000FF758E70E Load=ATA030203SCCP051201A.zup Last=Initialized
*Apr 5 02:50:12.066: ephone-(30)[2] StationRegisterMessage (29/31/48) from 1.4.188.72
*Apr 5 02:50:12.066: ephone-(30)[2] Register StationIdentifier DeviceName ATA000FF758E70E
*Apr 5 02:50:12.070: ephone-(30)[2] StationIdentifier Instance 1 deviceType 12
*Apr 5 02:50:12.070: ephone-30[-1]:stationIpAddr 1.4.188.72
*Apr 5 02:50:12.070: ephone-30[-1]:maxStreams 0
*Apr 5 02:50:12.070: ephone-30[-1]:protocol Ver 0x1
*Apr 5 02:50:12.070: ephone-30[-1]:phone-size 5392 dn-size 632
*Apr 5 02:50:12.070: ephone-(30) Allow any Skinny Server IP address 1.4.188.65
*Apr 5 02:50:12.070: ephone-30[-1]:Found entry 29 for 000FF758E70E
*Apr 5 02:50:12.070: ephone-30[-1]:socket change -1 to 2
*Apr 5 02:50:12.070: ephone-30[-1]:FAILED: CLOSED old socket -1
*Apr 5 02:50:12.074: ephone-30[2]:phone ATA000FF758E70E re-associate OK on socket [2]
21:21:49: %IPPHONE-6-REGISTER: ephone-30:ATA000FF758E70E IP:1.4.188.72 Socket:2
DeviceType:Phone has registered.
*Apr 5 02:50:12.074: Phone 29 socket 2
*Apr 5 02:50:12.074: Phone 29 socket 2: Running Bravo ??
*Apr 5 02:50:12.074: Skinny Local IP address = 1.4.188.65 on port 2000

*Apr 5 02:50:12.074: Skinny Phone IP address = 1.4.188.72 15325

*Apr 5 02:50:12.074: ephone-30[2]:Signal protocol ver 8 to phone with ver 1
*Apr 5 02:50:12.074: ephone-30[2]:Date Format M/D/Y
*Apr 5 02:50:12.078: ephone-30[2]:RegisterAck sent to ephone 2: keepalive period 30 use
sccp-version 1
*Apr 5 02:50:12.078: ephone-30[2]:CapabilitiesReq sent
*Apr 5 02:50:12.090: ephone-30[2]:VersionReq received
*Apr 5 02:50:12.090: ephone-30[2]:Version String not needed for ATA device. Part of XML
file
*Apr 5 02:50:12.090: ephone-30[2]:Version Message sent
*Apr 5 02:50:12.094: ephone-30[2]:CapabilitiesRes received
*Apr 5 02:50:12.098: ephone-30[2]:Caps list 7
G711Ulaw64k 60 ms
G711Alaw64k 60 ms
G729 60 ms
G729AnnexA 60 ms
G729AnnexB 60 ms
G729AnnexAwAnnexB 60 ms
Unrecognized Media Type 257 60 ms

*Apr 5 02:50:12.098: ephone-30[2]:ButtonTemplateReqMessage

*Apr 5 02:50:12.098: ephone-30[2]:StationButtonTemplateReqMessage set max presentation
to 2
*Apr 5 02:50:12.098: ephone-30[2]:CheckAutoReg
*Apr 5 02:50:12.102: ephone-30[2]:AutoReg is disabled
*Apr 5 02:50:12.102: ephone-30[2][ATA000FF758E70E]:Setting 1 lines 4 speed-dials on phone
(max_line 2)
*Apr 5 02:50:12.102: ephone-30[2]:First Speed Dial Button location is 2 (0)
*Apr 5 02:50:12.102: ephone-30[2]:Configured 4 speed dial buttons
*Apr 5 02:50:12.102: ephone-30[2]:ButtonTemplate lines=1 speed=4 buttons=5 offset=0
*Apr 5 02:50:12.102: ephone-30[2]:Skinny IP port 16384 set for socket [2]
*Apr 5 02:50:12.126: ephone-30[2]:StationSoftKeyTemplateReqMessage
*Apr 5 02:50:12.126: ephone-30[2]:StationSoftKeyTemplateResMessage
*Apr 5 02:50:12.206: ephone-30[2]:StationSoftKeySetReqMessage
*Apr 5 02:50:12.206: ephone-30[2]:StationSoftKeySetResMessage

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*Apr 5 02:50:12.307: ephone-30[2]:StationLineStatReqMessage from ephone line 1

*Apr 5 02:50:12.307: ephone-30[2]:StationLineStatReqMessage ephone line 1 DN 80 = 8080
desc = 8080 label =
*Apr 5 02:50:12.307: ephone-30[2][ATA000FF758E70E]:StationLineStatResMessage sent to
ephone (1 of 2)
*Apr 5 02:50:12.427: ephone-30[2]:StationSpeedDialStatReqMessage speed 9
*Apr 5 02:50:12.427: ephone-30[2]:No speed-dial set 9
*Apr 5 02:50:12.427: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:12.547: ephone-30[2]:StationSpeedDialStatReqMessage speed 8
*Apr 5 02:50:12.547: ephone-30[2]:No speed-dial set 8
*Apr 5 02:50:12.547: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:12.635: ephone-30[2]:StationSpeedDialStatReqMessage speed 7
*Apr 5 02:50:12.635: ephone-30[2]:No speed-dial set 7
*Apr 5 02:50:12.635: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:12.707: New Skinny socket accepted [1] (34 active)
*Apr 5 02:50:12.707: sin_family 2, sin_port 15400, in_addr 1.4.188.72
*Apr 5 02:50:12.711: skinny_add_socket 1 1.4.188.72 15400
*Apr 5 02:50:12.711: ephone-30[2]:StationSpeedDialStatReqMessage speed 6
*Apr 5 02:50:12.711: ephone-30[2]:No speed-dial set 6
*Apr 5 02:50:12.715: ephone-30[2]:StationSpeedDialStatMessage sent
21:21:50: %IPPHONE-6-REG_ALARM: Name=ATA0FF758E70E01 Load=ATA030203SCCP051201A.zup
Last=Initialized
*Apr 5 02:50:12.715:
Skinny StationAlarmMessage on socket [3] 1.4.188.72 ATA000FF758E70E
*Apr 5 02:50:12.715: severityInformational p1=0 [0x0] p2=0 [0x0]
*Apr 5 02:50:12.715: Name=ATA0FF758E70E01 Load=ATA030203SCCP051201A.zup Last=Initialized
*Apr 5 02:50:12.811: ephone-30[2]:StationSpeedDialStatReqMessage speed 5
*Apr 5 02:50:12.811: ephone-30[2]:No speed-dial set 5
*Apr 5 02:50:12.811: ephone-30[2]:StationSpeedDialStatMessage sent
21:21:50: %IPPHONE-6-REGISTER: ephone-31:ATA0FF758E70E01 IP:1.4.188.72 Socket:3
DeviceType:Phone has registered.
*Apr 5 02:50:12.908: ephone-30[2]:StationSpeedDialStatReqMessage speed 4
*Apr 5 02:50:12.908: ephone-30[2]:No speed-dial set 4
*Apr 5 02:50:12.908: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:13.008: ephone-30[2]:StationSpeedDialStatReqMessage speed 3
*Apr 5 02:50:13.008: ephone-30[2]:No speed-dial set 3
*Apr 5 02:50:13.008: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:13.108: ephone-30[2]:StationSpeedDialStatReqMessage speed 2
*Apr 5 02:50:13.108: ephone-30[2]:No speed-dial set 2
*Apr 5 02:50:13.108: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:13.208: ephone-30[2]:StationSpeedDialStatReqMessage speed 1
*Apr 5 02:50:13.208: ephone-30[2]:No speed-dial set 1
*Apr 5 02:50:13.208: ephone-30[2]:StationSpeedDialStatMessage sent
*Apr 5 02:50:14.626: New Skinny socket accepted [1] (33 active)
*Apr 5 02:50:14.626: sin_family 2, sin_port 15593, in_addr 1.4.188.72
*Apr 5 02:50:14.630: skinny_add_socket 1 1.4.188.72 15593
*Apr 5 02:50:15.628: New Skinny socket accepted [1] (34 active)
*Apr 5 02:50:15.628: sin_family 2, sin_port 15693, in_addr 1.4.188.72
*Apr 5 02:50:15.628: skinny_add_socket 1 1.4.188.72 15693
*Apr 5 02:50:21.538: ephone-30[2]:SkinnyCompleteRegistration

Using Call Pickup and Group Call Pickup with Cisco ATA
Most of the procedures for using Cisco ATAs with Cisco Unified CME are the same as those for using
Cisco ATAs with Cisco Unified Communications Manager, as described in the “How to Use Pre-Call
and Mid-Call Services” chapter of the Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor
Administrator’s Guide for SCCP (version 3.0). However, the call pickup and group call pickup
procedures are different when using Cisco ATAs with Cisco Unified CME, as described below:

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Call Pickup
When using Cisco ATAs with Cisco Unified CME:
• To pickup the last parked call, press **3*.
• To pickup a call on a specific extension, press **3 and enter the extension number.
• To pickup a call from a park slot, press **3 and enter the park slot number.

Group Call Pickup

When using Cisco ATAs with Cisco Unified CME:
• To answer a phone within your call pickup group, press **4*.
• To answer a phone outside of your call pickup group, press **4 and the group ID number.

Note If there is only one pickup group, you do not need to enter the group ID after the **4 to pickup a call.

SCCP: Configuring Analog Phone Support

Configuring Cisco Unified CME to support calls and features on analog endpoints is basically the same
as configuring any SCCP phone in Cisco Unified CME. This section describes only the steps that have
special meaning for SCCP analog phone support.

Prerequisites
• Cisco CME 3.2.2 or a later version for analog FXS ports on the Cisco VG 224 Voice Gateway.
• Cisco Unified CME 4.0 or a later version for analog FXS ports on the Cisco 2800 Series or
Cisco 3800 Series Integrated Services Routers.

Restrictions
• FXS ports on Cisco VG 248 Analog Phone Gateways are not supported by Cisco Unified CME.
• You must set the transfer-system command to full-blind or full-consult to enable call transfer on
analog endpoints.
• You must set the timeouts ringing command to infinity (default) on the analog ports to prevent this
timeout from expiring before the ringing no-answer timeout that is configured on
Cisco Unified CME with the timeouts ringing command in telephony-service mode.

Note In Cisco IOS Release 12.4(11)T and later the default value of the timeouts ringing
command is set to infinity for all SCCP-controlled analog ports. In releases earlier than
Cisco IOS Release 12.4(11)T, the default is 180 seconds.

SUMMARY STEPS

1. Set up ephone-dns for up to 24 analog endpoints on the Cisco IOS gateway.

2. Set the maximum number of ephones.
3. Assign ephone-dns to ephones.

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4. Set up feature parameters as desired.

5. Set up feature restrictions as desired.

DETAILED STEPS

Step 1 Set up ephone-dns for up to 24 endpoints on the Cisco IOS gateway.

Use the ephone-dn command:
ephone-dn 1 dual-line
number 1000
.
.
.
ephone-dn 24 dual-line
number 1024

Step 2 Set the maximum number of ephones.

Use the max ephones command to set a number equal to or greater than the total number of endpoints
that you intend to register on the Cisco Unified CME router, including both IP and analog endpoints. For
example, if you have 6 IP phones and 12 analog phones, set the max ephones command to 18 or greater.
Step 3 Assign ephone-dns to ephones.
Use the auto assign command to enable the automatic assignment of an available ephone-dn to each
phone as the phone contacts the Cisco Unified CME router to register. Note that the order of ephone-dn
assignment is not guaranteed. For example, if you have analog endpoints on ports 2/0 through 2/23 on
the Cisco IOS gateway, port 2/0 does not necessarily become ephone 1. Use one of the following
commands to enable automatic ephone-dn assignment.
• auto assign 1 to 24—You do not need to use the type keyword if you have only analog endpoints
to be assigned or if you want all endpoints to be automatically assigned.
• auto assign 1 to 24 type anl—Use the type keyword if you have other phone types in the system
and you want only the analog endpoints to be assigned to ephone-dns automatically.
An alternative to using the auto assign command is to manually assign ephone-dns to ephones (analog
phones on FXS ports). This method is more complicated, but you might need to use it if you want to
assign a specific extension number (ephone-dn) to a particular ephone. The reason that manual
assignment is more complicated is because a unique device ID is required for each registering ephone
and analog phones do not have unique MAC addresses like IP phones do. To create unique device IDs
for analog phones, the auto assign process uses a particular algorithm. When you make manual ephone
assignments, you have to use the same algorithm for each phone that receives a manual assignment. Note
that once you have assigned ephone-dns to all the ephones that you want to assign manually, you can use
the auto assign command to automatically assign the remaining ports.
The algorithm uses the single 12-digit SCCP local interface MAC address on the Cisco IOS gateway as
the base to create unique 12-digit device IDs for all the FXS ports on the Cisco IOS gateway. The
rightmost 9 digits of the SCCP local interface MAC address are shifted left three places and are used as
the leftmost 9 digits for all 24 individual device IDs. The remaining 3 digits are the hexadecimal
translation of the binary representation of the port’s slot number (3 digits), subunit number (2 digits),
and port number (7 digits). The following example shows the use of the algorithm to create a unique
device ID for one port:
1. The MAC address for the Cisco VG 224 SCCP local interface is 000C.8638.5EA6.
2. The FXS port has a slot number of 2 (010), a subunit number of 0 (00), and a port number of 1
(0000001). The binary digits are strung together to become 0100 0000 0001, which is then translated
to 401 in hexadecimal to create the final device ID for the port and ephone.

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3. The resulting unique device ID for this port is C863.85EA.6401.

When setting up an ephone manually in ephone configuration mode for an analog port, assign it just one
button because the port represents a single-line device. The button command can use the “:” (colon, for
normal), “o” (overlay) and “c” (call-waiting overlay) modes.
Step 4 Set up feature parameters as desired.
• Call transfer—To use call transfer from analog endpoints, the transfer-system command must be
configured for full-blind or full-consult in telephony-service configuration mode on the
Cisco CME router. This is the recommended setting for Cisco CME 3.0 and later versions, but it is
not the default.
• Call forwarding—Call forwarding destinations are specified for all, busy, and no-answer conditions
for each ephone-dn using the call-forward all, call-forward busy, and call-forward noan
commands in ephone-dn configuration mode.
• Call park—Call-park slots are created using the park-slot command in ephone-dn configuration
mode. Phone users must be instructed how to transfer calls to the call-park slots and use directed
pickup to retrieve the calls.
• Call pickup groups—Extensions are added to pickup groups using the pickup-group command in
ephone-dn configuration mode. Phone users must be told which phones are in which groups.
• Caller ID—Caller names are defined using the name command in ephone-dn configuration mode.
Caller numbers are defined using the number command in ephone-dn configuration mode.
• Speed dial—Numbers to be speed-dialed are stored with their associated speed-dial codes using the
speed-dial command in ephone configuration mode.
• Speed dial to voice mail—The voice-mail number is defined using the voicemail command in
telephony-service configuration mode.
Step 5 Set up feature restrictions as desired.
Features such as transfer, conference, park, pickup, group pickup (gpickup), and call forward all
(cfwdall) can be restricted from individual ephones using the Cisco Unified CME soft-key template
customization command, even though analog phones do not have soft keys. Simply create a template that
leaves out the soft key that represents the feature you want to restrict and apply the template to the
ephone for which you want the feature restricted. For more information about soft-key template
customization, see “Customizing Soft Keys” on page 829.

What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see “SCCP: Selecting Button
Layout for a Cisco Unified IP Phone 7931G” on page 897 .
• After configuring phones in Cisco Unified CME to make basic calls, you are ready to generate
configuration files for the phones to be connected. See “SCCP: Generating Configuration Files for
SCCP Phones” on page 247.

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SCCP: Verifying Analog Phone Support

Use the following show commands to display information about analog endpoints.
• show ephone anl—Displays MAC address, registration status, ephone-dn, and speed-dial numbers
for analog ephones.
• show telephony-service ephone-dn—Displays call forward, call waiting, pickup group, and more
information about ephone-dns.
• show running-config—Displays running configuration nondefault values.

SCCP: Enabling a Remote Phone

To enable IP phones or instances of Cisco IP Communicator to connect to a Cisco Unified CME system
over a WAN, perform the following steps.

Prerequisites
• The WAN link supporting remote teleworker phones should be configured with a Call Admission
Control (CAC) or Resource Reservation Protocol (RSVP) solution to prevent the oversubscription
of bandwidth, which can degrade the quality of all voice calls.
• If DSP farms will be used for transcoding, you must configure them separately. See “Configuring
Transcoding Resources” on page 303.
• A SCCP phone to be enabled as a remote phone is configured in Cisco Unified CME. For
configuration information, see the “SCCP: Creating Directory Numbers” section on page 158

Restrictions
• Because Cisco Unified CME is not designed for centralized call processing, remote phones are
supported only for fixed teleworker applications, such as working from a home office.
• Cisco Unified CME does not support 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 Unified CME does not support Emergency 911 (E911) calls from remote IP phones.
Teleworkers using remote phones connected to Cisco Unified CME over a WAN should be advised
not to use these phones for E911 emergency services because the local public safety answering point
(PSAP) will not be able to obtain valid calling-party information from them.
We recommend that you make all remote phone users aware of this issue. One way is to place a label
on all remote teleworker phones that reminds users not to place 911 emergency calls on remote IP
phones. Remote workers should place any emergency calls through locally configured hotel, office,
or home phones (normal land-line phones) whenever possible. Inform remote workers that if they
must use remote IP phones for emergency calls, they should be prepared to provide specific location
information to the answering PSAP personnel, including street address, city, state, and country.

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SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. mtp
5. codec {g711ulaw | g729r8 [dspfarm-assist]}
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: this ephone during configuration tasks.
Router(config)# ephone 36
Step 4 mtp Sends media packets to the Cisco Unified CME router.

Example:
Router(config-ephone)# mtp
Step 5 codec {g711ulaw | g729r8 [dspfarm-assist]} (Optional) Selects a preferred codec for setting up calls.
• g711ulaw—G.711 mu-law codec (default).
Example: • g729r8—G.729r8 codec.
Router(config-ephone)# codec g729r8
dspfarm-assist • dspfarm-assist—Attempts to use DSP-farm resources
for transcoding the segment between the phone and the
Cisco Unified CME router if G.711 is negotiated for
the call.
Note The dspfarm-assist keyword is ignored if the SCCP
endpoint type is ATA, VG224, or VG248.
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

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How to Configure Cisco ATA, Analog Phone Support, Remote Phones, and Cisco IP Communicator

What to Do Next
• If you have SIP and SCCP phones connected to the same Cisco Unified CME, see the “Configuring
Codec for Local Calling Between SIP and SCCP Phones” section on page 174.
• To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see “SCCP: Selecting Button
Layout for a Cisco Unified IP Phone 7931G” on page 897 .
• After configuring phones in Cisco Unified CME to make basic calls, you are ready to generate
configuration files for the phones to be connected. See “SCCP: Generating Configuration Files for
SCCP Phones” on page 247.

SCCP: Verifying Remote Phones

Step 1 Use the show running-config command or the show telephony-service ephone command to verify
parameter settings for remote ephones.

SCCP: Configuring Cisco IP Communicator Support

To enable support for Cisco IP Communicator, perform the following steps.

Prerequisites
• Cisco Unified CME 4.0 or a later version
• Cisco IP Communicator 2.0 or a later version
• IP address of the Cisco Unified CME TFTP server
• (Optional) Headsets with microphones for users

DETAILED STEPS

Step 1 Download the latest version of the Cisco IP Communicator software and install it on your PC.
The download website is at http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp.
Step 2 (Optional) Attach a headset with microphone to your PC.
Step 3 Start the Cisco IP Communicator application.
Step 4 Define the IP address of the Cisco Unified CME TFTP server.
a. Open the Network > User Preferences window.
b. Enter the IP address of the Cisco Unified CME TFTP server.
Step 5 Wait for the Cisco IP Communicator application to connect to Cisco Unified CME and register.
Step 6 Configure the extension numbers and line buttons for the Cisco IP Communicator.

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Use the normal phone provisioning commands described in the “SCCP: Creating Directory Numbers”
section on page 158. In the type command, use the CIPC keyword to identify this phone as a
Cisco IP Communicator.

SCCP: Verifying Cisco IP Communicator Support

Step 1 Use the show running-config command to display ephone-dn and ephone information associated with
this phone.
Step 2 After Cisco IP Communicator registers with Cisco Unified CME, it displays the phone extensions and
soft keys in its configuration. Verify that these are correct.
Step 3 Make a local call from the phone and have someone call you. Verify that you have a two-way voice path.

SCCP: Troubleshooting Cisco IP Communicator Support

Step 1 Use the debug ephone detail command to diagnose problems with calls. For more information, see the
Cisco Unified CME Command Reference.

Configuration Examples for Making Basic Calls

This section contains the following examples of the required Cisco Unified CME configurations with
some of the additional options that are discussed in other modules.
• Configuring SCCP Phones for Making Basic Calls: Example, page 199
• Configuring SIP Phones for Making Basic Calls: Example, page 204
• Disabling a Bulk Registration for a SIP Phone: Example, page 206
• Cisco ATA: Example, page 207
• SCCP Analog Phone: Example, page 207
• Remote Teleworker Phones: Example, page 208

Configuring SCCP Phones for Making Basic Calls: Example

Router# show running-config

version 12.4
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname CME40

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!
boot-start-marker
boot-end-marker
!
logging buffered 2000000 debugging
!
no aaa new-model
!
resource policy
!
clock timezone PST -8
clock summer-time PDT recurring
no network-clock-participate slot 2
voice-card 0
no dspfarm
dsp services dspfarm
!
voice-card 2
dspfarm
!
no ip source-route
ip cef
!
!
!
ip domain name cisco.com
ip multicast-routing
!
!
ftp-server enable
ftp-server topdir flash:
isdn switch-type primary-5ess
!
!
!
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
no supplementary-service h450.2
no supplementary-service h450.3
h323
call start slow
!
!
!
controller T1 2/0/0
framing esf
linecode b8zs
pri-group timeslots 1-24
!
controller T1 2/0/1
framing esf
linecode b8zs
!
!
interface GigabitEthernet0/0
ip address 192.168.1.1 255.255.255.0
ip pim dense-mode
duplex auto
speed auto
media-type rj45
negotiation auto
!
interface Service-Engine1/0

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ip unnumbered GigabitEthernet0/0
service-module ip address 192.168.1.2 255.255.255.0
service-module ip default-gateway 192.168.1.1
!
interface Serial2/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-5ess
isdn incoming-voice voice
isdn map address ^.* plan unknown type international
no cdp enable
!
!
ip route 0.0.0.0 0.0.0.0 192.168.1.254
ip route 192.168.1.2 255.255.255.255 Service-Engine1/0
ip route 192.168.2.253 255.255.255.255 10.2.0.1
ip route 192.168.3.254 255.255.255.255 10.2.0.1
!
!
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
!
!
!
!
tftp-server flash:P00307020300.loads
tftp-server flash:P00307020300.sb2
tftp-server flash:P00307020300.sbn
!
control-plane
!
!
!
voice-port 2/0/0:23
!
!
!
sccp local GigabitEthernet0/0
sccp ccm 192.168.1.1 identifier 1
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register MTP0013c49a0cd0
keepalive retries 5
!
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec gsmfr
codec g729r8
maximum sessions 90
associate application SCCP
!
!
dial-peer voice 9000 voip
mailbox-selection last-redirect-num
destination-pattern 78..
session protocol sipv2
session target ipv4:192.168.1.2

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dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 2 pots
incoming called-number .
direct-inward-dial
port 2/0/0:23
forward-digits all
!
dial-peer voice 1 pots
destination-pattern 9[2-9]......
port 2/0/0:23
forward-digits 8
!
dial-peer voice 3 pots
destination-pattern 91[2-9]..[2-9]......
port 2/0/0:23
forward-digits 12!
!
gateway
timer receive-rtp 1200
!
!
telephony-service
load 7960-7940 P00307020300
max-ephones 100
max-dn 300
ip source-address 192.168.1.1 port 2000
system message CCME 4.0
sdspfarm units 1
sdspfarm transcode sessions 128
sdspfarm tag 1 MTP0013c49a0cd0
voicemail 7800
max-conferences 24 gain -6
call-forward pattern .T
moh music-on-hold.au
multicast moh 239.1.1.1 port 2000
web admin system name admin password sjdfg
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn-template 1
!
!
ephone-template 1
keep-conference endcall local-only
codec g729r8 dspfarm-assist
!
!
ephone-template 2
!
!
ephone-dn 1
number 6001
call-forward busy 7800
call-forward noan 7800 timeout 10
!
!
ephone-dn 2
number 6002

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call-forward busy 7800

call-forward noan 7800 timeout 10
!
!
ephone-dn 10
number 6013
paging ip 239.1.1.1 port 2000
!
!
ephone-dn 20
number 8000....
mwi on
!
!
ephone-dn 21
number 8001....
mwi off
!
!
!
!
ephone 1
device-security-mode none
username "user1"
mac-address 002D.264E.54FA
codec g729r8 dspfarm-assist
type 7970
button 1:1
!
!
!
ephone 2
device-security-mode none
username "user2"
mac-address 001C.821C.ED23
type 7960
button 1:2
!
!
!
line con 0
stopbits 1
line aux 0
stopbits 1
line 66
no activation-character
no exec
transport preferred none
transport input all
transport output all
line 258
no activation-character
no exec
transport preferred none
transport input all
transport output all
line vty 0 4
exec-timeout 0 0
privilege level 15
password sgpxw
login
!
scheduler allocate 20000 1000
ntp server 192.168.224.18

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!
!
end

Configuring SIP Phones for Making Basic Calls: Example

The following is a configuration example for SIP phones running on Cisco Unified CME:
voice service voip
allow-connections sip to sip
sip
registrar server expires max 600 min 60

voice class codec 1

codec preference 1 g711ulaw

voice hunt-group 1 parallel

final 8000
list 2000,1000,2101
timeout 20
pilot 9000

voice hunt-group 2 sequential

final 1000
list 2000,2300
timeout 25
pilot 9100 secondary 9200

voice hunt-group 3 peer

final 2300
list 2100,2200,2101,2201
timeout 15
hops 3
pilot 9300
preference 5

voice hunt-group 4 longest-idle

final 2000
list 2300,2100,2201,2101,2200
timeout 15
hops 5
pilot 9400 secondary 9444
preference 5 secondary 9

voice register global

mode cme

external-ring bellcore-dr3

voice register dn 1
number 2300
mwi

voice register dn 2
number 2200
call-forward b2bua all 1000
call-forward b2bua mailbox 2200
mwi

voice register dn 3
number 2201
after-hour exempt

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voice register dn 4
number 2100
call-forward b2bua busy 2000
mwi

voice register dn 5
number 2101
mwi

voice register dn 76
number 2525
call-forward b2bua unreachable 2300
mwi
!
voice register template 1
!
voice register template 2
no conference enable
voicemail 7788 timeout 5
!

voice register pool 1

id mac 000D.ED22.EDFE
type 7960
number 1 dn 1
template 1
preference 1
no call-waiting
codec g711alaw
!
voice register pool 2
id mac 000D.ED23.CBA0
type 7960
number 1 dn 2
number 2 dn 2
template 1
preference 1

dtmf-relay rtp-nte
speed-dial 3 2001
speed-dial 4 2201
!
voice register pool 3
id mac 0030.94C3.053E
type 7960
number 1 dn 3
number 3 dn 3
template 2

!
voice register pool 5
id mac 0012.019B.3FD8
type ATA
number 1 dn 5
preference 1
dtmf-relay rtp-nte
codec g711alaw

voice register pool 6

id mac 0012.019B.3E88
type ATA
number 1 dn 6
number 2 dn 7

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template 2
dtmf-relay-rtp-nte
call-forward b2bua all 7778

voice register pool 7

voice register pool 8

id mac 0006.D737.CC42
type 7940
number 1 dn 8
template 2
preference 1
codec g711alaw

voice-port 1/0/0

voice-port 1/0/1

dial-peer voice 100 pots

destination-pattern 2000
port 1/0/0

dial-peer voice 101 pots

destination-pattern 2010
port 1/0/1

dial-peer voice 1001 voip

preference 1
destination-pattern 1...
session protocol sipv2
session target ipv4:10.15.6.13
codec g711ulaw

sip-ua
mwi-server ipv4:1.15.6.200 expires 3600 port 5060 transport udp

telephony-service
load 7960-7940 P0S3-07-2-00
max-ephones 24
max-dn 96
ip source-address 10.15.6.112 port 2000
create cnf-files version-stamp Aug 24 2004 00:00:00
max-conferences 8
after-hours block pattern 1 1...
after-hours day Mon 17:00 07:00

Disabling a Bulk Registration for a SIP Phone: Example

The following example shows the configuration for all phone numbers that match the pattern “408555..”
can register with the SIP proxy server (IP address 1.5.49.240) except directory number 1, number
“4085550101,” for which bulk registration is disabled
voice register global
mode cme
bulk 408555….

voice register dn 1
number 4085550101
no-reg
sip-ua
registrar ipv4:1.5.49.240

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Cisco ATA: Example

The following example shows the configuration for two analog phones using a single Cisco ATA with
MAC address 000F.F758.E70E. The analog phone attached to the first port uses the MAC address of the
Cisco ATA. The analog phone attached to the second port uses a modified version of the Cisco ATA’s
MAC address; the first two hexadecimal numbers are removed and 01 is appended to the end.
!
telephony-service
conference hardware
load ATA ATA030203SCCP051201A.zup
!
ephone-dn 80 dual-line
number 8080
!
ephone-dn 81 dual-line
number 8081
!
ephone 30
mac-address 000F.F758.E70E
type ata
button 1:80
!
ephone 31
mac-address 0FF7.58E7.0E01
type ata
button 1:81

SCCP Analog Phone: Example

The following excerpt from a Cisco Unified CME configuration sets transfer type to full-blind and sets
the voice-mail extension to 5200. Ephone-dn 10 has the extension 4443 and is assigned to Tommy; that
number and name will be used for caller-ID displays. The description field under ephone-dn is used to
indicate that this ephone-dn is on the Cisco VG 224 voice gateway at port 1/3. Extension 4443 is
assigned to ephone 7, which is an analog phone type with 10 speed-dial numbers.
CME_Router# show running-config
.
.
.
telephony-service
load 7910 P00403020214
load 7960-7940 P00305000301
load 7905 CP79050101SCCP030530B31
max-ephones 60
max-dn 60
ip source-address 10.8.1.2 port 2000
auto assign 1 to 60
create cnf-files version-stamp 7960 Sep 28 2004 17:23:02
voicemail 5200
mwi relay
mwi expires 99999
max-conferences 8 gain -6
web admin system name cisco password lab
web admin customer name ac2 password cisco
dn-webedit
time-webedit
transfer-system full-blind
transfer-pattern 6...
transfer-pattern 5...

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!
!
ephone-dn 10 dual-line
number 4443 secondary 9191114443
pickup-group 5
description vg224-1/3
name tommy
!
ephone 7
mac-address C863.9018.0402
speed-dial 1 4445
speed-dial 2 4445
speed-dial 3 4442
speed-dial 4 4441
speed-dial 5 6666
speed-dial 6 1111
speed-dial 7 1112
speed-dial 8 9191114441
speed-dial 9 9191114442
speed-dial 10 9191114442
type anl
button 1:10
!

Remote Teleworker Phones: Example

The following example shows the configuration for ephone 270, a remote teleworker phone with its
codec set to G.729r8. The dspfarm-assist keyword is used to ensure that calls from this phone will use
DSP resources to maintain the G.729r8 codec when calls would normally be switched to a G.711 codec.
ephone 270
button 1:36
mtp
codec g729r8 dspfarm-assist
description teleworker remote phone

Where to Go Next
To select a fixed-button layout for a Cisco Unified IP Phone 7931G, see “SCCP: Selecting Button Layout
for a Cisco Unified IP Phone 7931G” on page 897 .
After configuring phones in Cisco Unified CME to make basic calls, you are ready to generate
configuration files for the phones to be connected to your router. See “Generating Configuration Files
for Phones” on page 245.

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Configuring Phones to Make Basic Calls

Table 8 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 8 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 8 Feature Information for Basic Call Features

Cisco Unified CME

Feature Name Versions Feature Information
Analog Phones 4.0 • Support was introduced for fax pass-through mode
using SCCP and a Cisco VG 224 voice gateway or
Cisco ATA.
• Support was introduced for analog phones with SCCP
supplementary features using FXS ports on
Cisco Integrated Services Routers.
3.2.1 Support was introduced for analog phones with SCCP
supplementary features using FXS ports on a
Cisco VG 224 voice gateway.
3.0 Support was introduced for Cisco ATA 186 and
Cisco ATA 188.
1.0 Support was introduced for analog phones in H.323 mode
using FXS ports.
Cisco IP Communicator 4.0 Support for Cisco IP Communicator was introduced.
Dial Plans for SIP Phones 4.1 Dial plans for SIP phones was added.

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Table 8 Feature Information for Basic Call Features (continued)

Cisco Unified CME

Feature Name Versions Feature Information
Direct FXO Trunk Lines 4.0 Enhancements were added to improve the keyswitch
emulation behavior of PSTN lines in a Cisco Unified CME
system including the following:
• Status monitoring of the FXO port on the line button of
the IP phone.
• Transfer recall if a transfer-to phone does not answer
after a specified timeout.
• Transfer-to button optimization to free up the private
extension line on the transfer-to phone
• Directory numbers for FXO lines can be configured for
dual-line to support the FXO monitoring, transfer
recall, and transfer-to button optimization features.
3.2 Direct FXO trunk line capability was introduced.
KPML 4.1 KPML for SIP phones was added.
Remote Teleworker Phones 4.0 Support for teleworker remote phones was introduced.
Session Transport Protocol 4.1 Added selection for session-transport protocol for SIP
phones.

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Assigner

Last Updated: March 26, 2007

This chapter describes the Extension Assigner feature in Cisco Unified Communications Manager
Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support the feature documented in this module. For a list of
the versions in which this feature is supported, see the “Feature Information for Extension Assigner” section
on page 243.

Contents
• Prerequisites for Extension Assigner, page 213
• Restrictions for Extension Assigner, page 214
• Information About Extension Assigner, page 214
• How to Configure Extension Assigner, page 219
• Configuration Examples for Extension Assigner, page 239
• Additional References, page 242
• Feature Information for Extension Assigner, page 243

Prerequisites for Extension Assigner

• Cisco IOS Release 12.4(4)XC or a later release
• Cisco Unified CME 4.0(3) or a later version
• The following features must be enabled:
– Autoassign
– Auto-register-ephone (the default)
– DHCP

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Restrictions for Extension Assigner

• The number of phones that you install cannot exceed the maximum number of phones supported by
the router chassis. To find the maximum number of phones for a particular router and
Cisco Unified CME version, see the appropriate Cisco Unified CME Supported Firmware,
Platforms, Memory, and Voice Products for your Cisco IOS release.
• Do not edit the Tcl script that is provided with this feature.
• When loading the Tcl script for extension assigner, you must configure it to use English as the
language.

Information About Extension Assigner

To use extension assigner, you should understand the following concepts:
• Extension Assigner Overview, page 214
• Files Included in this Release, page 218

Extension Assigner Overview

This feature enables installation technicians to assign extension numbers to Cisco Unified CME phones
without administrative access to the server, typically during the installation of new phones or the
replacement of broken phones. However, before an installation technician can use this feature, the
system administrator must first configure Cisco Unified CME to allow specific extensions to be
assigned. The system administrator must also provide the installation technician with the information
necessary for assigning extension numbers to phones. The installation technician can then assign
extension numbers to phones with access to only the phones themselves and with no further intervention
from the administrator.
The documentation for this feature consists of two groups of procedures, one for installation technicians
and one for system administrators. This section describes both of these sets of procedures.

Procedures for System Administrators

Before an installation technician can assign new extension numbers to phones, you must complete these
procedures:
• Determine which extension numbers will be assigned to the new phones and plan your
configuration.
• Download the appropriate Tcl script and associated audio prompt files and place them in the correct
directory.
• Configure the Cisco Unified CME router to:
– Configure and load the appropriate Tcl script.
– Specify the extension that the installation technician calls to assign extension numbers.
– Optionally specify whether the extension used to assign extension numbers is dialed
automatically.
– Specify the password that the installation technician enters to assign extension numbers.

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– Configure the extension assigner feature.

– Configure ephone-dns with temporary extension numbers.
– Configure ephone-dns with the extension numbers that the installation technician can assign to
phones.
– Configure ephones with temporary MAC addresses for each phone that will be assigned an
extension number by the installation technician.
– Optionally configure the router to automatically save your configuration.
• Provide the installation technician with the information needed to assign extension numbers to the
new phones.
Before you can configure this feature, you must understand how the extension assigner application
works and what information the installation technician needs to assign extension numbers to phones.
You must also determine which extension numbers to assign to the new phones.
Other information you must provide to the installation technician involves the tasks that the installation
technician must perform. These tasks include:
• Dialing a configurable extension number to access the extension assigner application
• Entering a configurable password
• Entering a tag that references the extension number that will be assigned to the phone
Therefore, you must make the following decisions:
• Which extension number must be dialed to access the extension assigner application.
• Whether the number is dialed automatically when a phone goes off hook.
• What password the installation technician must enter to access the extension assigner application.
• What type of tag numbers to use to reference the extension number to assign to the phone.
• What specific tag numbers to use to reference the extension number to assign to the phone.
The first three decisions are straightforward, but the last two tag number decisions require some
knowledge of how the extension assigner feature works.
This feature is implemented using a Tcl script and audio files. To run this script, the installation
technician plugs in the phone, waits for a random extension number to be assigned by the autoassign
feature, and dials a specified extension number. Extension assigner requires that both the autoassign
feature and autoregister feature are enabled and configured as described in earlier versions.
After the phones have registered and received their temporary extension numbers, the installation
technician can access extension assigner and enter a tag number. This tag number is used to reference
the extension number and must match either an ephone tag or a similar new tag called the provision-tag.
You must decide on which tag you want to use before you configure your ephone and ephone-dn entries.
The advantage of using the provision-tag is that you can make it easier for the installation technician to
assign extension numbers because you can configure the tag to match the primary extension number or
some other unique identifier for the phone, such as a jack number.
The disadvantage is that you configure an additional keyword for each ephone entry, as shown in the
following example:
ephone 1
provision-tag 9001
mac-address 02EA.EAEA.0001
button 1:1

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If you decide to use the ephone tag, it will require less configuration. However, the installation technician
will enter an arbitrary tag number instead of the actual extension number when configuring a phone. This
restriction is because the number of ephone tags that you can configure is limited by your license. For
example, if you use the ephone tag and you have a 100-user license, the installation technician cannot
enter 9001 for the tag because you can configure only ephone 1 to ephone 100.
Note that each ephone entry that you configure must also include a temporary MAC address. As shown
in the above example, this address should begin with 02EA.EAEA and can end with any unique number.
We strongly recommend that you can configure this unique number to match the ephone tag.
You do not have to configure any ephone entries for the extension number that are randomly assigned.
The autoassign feature automatically creates an ephone entry for each new phone when it registers. The
autoassign feature then automatically assigns an ephone-dn entry if there is an available ephone-dn that
has one of the tag numbers specified by the auto assign command. The resulting ephone configurations
have the actual MAC address of the phone and a button with the first available ephone-dn designated for
the autoassign feature.
As shown in the following example, you configure at least one ephone-dn for a temporary extension and
specify which ephone-dns the autoassign feature will assign to the temporary ephone entries:
telephony-service
auto assign 101 to 105

ephone-dn 101
number 0001

When the installation technician assigns an extension number to a phone, the temporary MAC address
is replaced by the actual MAC address and the ephone entry created by the autoregister feature is deleted.
The number of ephone-dns that you configure for the autoassign feature determines how many phones
you can plug in at one time and get an automatically assigned extension. If you define four ephone-dns
for autoassign and you plug in five phones, one phone will not get a temporary extension number until
you assign an extension to one of the other four phones and reset the fifth phone. You are permitted to
set the max-ephone value higher than the number of purchased Cisco Unified CME phone seat licenses
for the purpose of enrolling licensed phones using extension assigner.
In addition to configuring one ephone-dn for each temporary extension number that is assigned
automatically, you also must configure an ephone-dn entry for each extension number that is assigned
by the installation technician.
Therefore, to complete the configuration, as shown in the following example, you must:
• Specify whether to use the ephone or the provision-tag number to reference the extension
number to assign to the phone. Set this when the feature is enabled with the new
extension-assigner tag-type command provided with this feature.
• Configure an ephone-dn for each temporary extension number that is assigned automatically.
• Configure an ephone-dn for each extension number that you want the installation technician to
assign to a phone.
• Configure an ephone with a temporary MAC address for each phone that is assigned an extension
number by the installation technician. Optionally, this ephone definition can include the new
provision-tag. For more information, see the “Configuring Ephones with Temporary MAC
Addresses” section on page 231.
telephony-service
extension-assigner tag-type provision-tag
auto assign 101 to 105

ephone-dn 1 dual-line
number 6001

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ephone-dn 101
number 0001
label Temp-Line-not assigned yet

ephone 1
provision-tag 6001
mac-address 02EA.EAEA.0001
button 1:1

Because you must configure two ephone-dns for each extension number that you want to assign, you may
exceed your max-dn setting. You are permitted to set the max-dn value higher than the number allowed
by your license for the purpose of enrolling licensed phones using extension assigner.
Assuming that your max-dn setting is set high enough, your max-ephone setting determines how many
phones you should plug in at one time. For example, if your max-ephone setting is ten more than the
number of phones to which you want to assign extension numbers, the you can plug in ten phones at a
time. If you plug in eleven phones, one phone will not register or get a temporary extension number until
you assign an extension to one of the first ten phones and reset the eleventh phone.
After you have configured your ephone and ephone-dn entries, you can complete your router
configuration by optionally configuring the router to automatically save your configuration. If the router
configuration is not saved, any extension assignments made by the installation technician will be lost
when the router is restarted. The alternative to this optional procedure is to have the installation
technician connect to the router and enter the write memory command to save the router configuration.
The final task of the system administrator is to document the information that the installation technician
needs to assign extension numbers to the new phones. You can also use this documentation as a guide
when you configure Cisco Unified CME to implement this feature. This information includes:
• How many phones the installation technician can plug in at one time
• Which extension number to dial to access the extension assigner application
• Whether the number is dialed automatically when a phone goes off hook
• What password to enter to access the application
• Which tag numbers to enter to assign en extension to each phone

Note Because this feature is implemented using a Tcl script and audio files, you must place the script and
associated audio prompt files in the correct directory. Do not edit this script; just configure
Cisco Unified CME to load the appropriate script.

Procedures for Installation Technicians

This feature is implemented using a Tcl script and audio prompt files that enable the installation
technician to assign an extension number to a new Cisco Unified CME phone by performing the
following procedure:

Step 1 Plug in a specified number of new phones.

Step 2 Wait for the phones to be assigned temporary, random extension numbers.
Step 3 Dial a specified number to access the extension assigner application.
Step 4 Enter a specified password.
Step 5 Enter a tag that references an extension number and enables the installation technician to perform one
of the following tasks:

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• Assign a new extension number to a phone.

• Unassign the current extension number.
• Reassign an extension number.

The system administrator provides the installation technician with all of the information needed to
perform this procedure.

Files Included in this Release

The app-cme-ea-2.0.0.0.tar or later archive file provided for the extension assigner feature includes a
readme file, a Tcl script, and several audio prompt files. If you want to replace the audio files with files
that use a language other than English, you must not change the name of the files. The Tcl script is
written to use only the following list of the filenames:
• app-cme-ea-2.0.0.0.tcl (script)
• en_cme_tag_assign_phone.au (audio file)
• en_cme_tag_assigned_to_phone.au (audio file)
• en_cme_tag_assigned_to_phone_idle.au (audio file)
• en_cme_tag_assigned_to_phone_inuse.au (audio file)
• en_cme_tag_assigned_to_phone_unreg.au (audio file)
• en_cme_tag_available.au (audio file)
• en_cme_tag_extension.au (audio file)
• en_cme_tag_invalid.au (audio file)
• en_cme_tag_unassign_phone.au (audio file)
• en_cme_tag_action_cancelled.au (audio file)
• en_cme_tag_assign_failed.au (audio file)
• en_cme_tag_assign_success.au (audio file)
• en_cme_tag_contact_admin.au (audio file)
• en_cme_tag_disconnect.au (audio file)
• en_cme_tag_ephone_tagid.au (audio file)
• en_cme_tag_invalid_password.au (audio file)
• en_cme_tag_invalidoption.au (audio file)
• en_cme_tag_noentry.au (audio file)
• en_cme_tag_password.au (audio file)
• en_cme_tag_unassign_failed.au (audio file)
• en_cme_tag_unassign_success.au (audio file)
• en_eight.au (audio file)
• en_five.au (audio file)
• en_four.au (audio file)
• en_nine.au (audio file)

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• en_one.au (audio file)

• en_seven.au (audio file)
• en_six.au (audio file)
• en_three.au (audio file)
• en_two.au (audio file)
• en_zero.au (audio file)
• readme.txt

How to Configure Extension Assigner

This section consists of tasks for both the system administrator and installation technician.
Before the installation technician can use extension assigner, the system administrator must perform the
following tasks:
• Determining Which Extension Numbers to Assign to the New Phones and Plan Your Configuration,
page 220 (required)
• Downloading the Tcl Script, page 220 (required)
• Configuring the Tcl Script, page 221 (required)
• Specifying the Extension That Installation Technicians Call to Assign Extension Numbers,
page 224 (required)
• Specifying Whether the Extension Used to Access Extension Assigner Is Dialed Automatically,
page 225 (optional)
• Configuring the Extension Assigner Feature, page 226 (optional)
• Configuring Temporary Extension Numbers for Phones That Use Extension Assigner, page 227
(required)
• Configuring Extension Numbers That Installation Technicians Can Assign to Phones, page 229
(required)
• Configuring Ephones with Temporary MAC Addresses, page 231 (required)
• Configuring the Router to Automatically Save Your Configuration, page 234 (optional)
• Provide the Installation Technician with the Needed Information, page 236 (required)
The installation technician can use extension assigner to perform following tasks:
• Assigning New Extension Numbers, page 236
• Unassigning the Current Extension Number, page 237
• Reassigning the Current Extension Number, page 237
To troubleshoot your installation, see the “Verifying Extension Assigner” section on page 238.

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Determining Which Extension Numbers to Assign to the New Phones and Plan
Your Configuration
After you determine which extension number to assign to each phone, you must make the following
decisions:
• Which extension number must be dialed to access the extension assigner application.
• Whether the number is dialed automatically when a phone goes off hook.
• What password the installation technician must enter to access the extension assigner application.
• Whether to use the ephone or the provision-tag number to reference the extension number to assign
to the phone.
• How many temporary extension numbers to configure. This will determine how many temporary
ephone-dns and temporary MAC addresses to configure.
• What specific tag numbers to use to reference the extension number to assign to the phone.

Downloading the Tcl Script

Perform this procedure to download the Tcl script and audio prompt files for the extension assigner
feature. As with all other Tcl scripts, you can download this script to a TFTP server or to the
Cisco Unified CME system’s flash memory. For more information about how to use Tcl scripts, see
Cisco IOS Tcl IVR and VoiceXML Application Guide for your Cisco IOS release.

Note Do not edit the Tcl script.

SUMMARY STEPS

1. Go to the Cisco Unified CME software download website at

http://www.cisco.com/pcgi-bin/tablebuild.pl/ip-iostsp.
2. Download the Cisco Unified CME extension assigner tar archive to a TFTP server or to the
Cisco Unified CME system’s flash memory.
3. enable
4. archive tar /xtract source-url destination-url

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DETAILED STEPS

Command or Action Purpose

Step 1 Go to the Cisco Unified CME software download • Gives you access to Cisco Unified CME software
website at downloads.
http://www.cisco.com/pcgi-bin/tablebuild.pl/ip-iosts. Note This Web site is only available to registered
Cisco.com users.
Step 2 Download the Cisco Unified CME extension assigner • Downloads the Cisco Unified CME extension assigner
tar archive to a TFTP server that is accessible to the tar archive called app-cme-ea-2.0.0.0.tar (or a later
Cisco Unified CME router. version) to a TFTP server that is accessible to the
Cisco Unified CME router.
This tar archive contains the extension assigner Tcl
script and the default audio files that you need for the
extension assigner service.
Step 3 enable Enters global configuration mode.

Example:
Router# enable
Step 4 archive tar /xtract source-url destination-url Uncompresses the files in the Cisco Unified CME extension
assigner app-cme-ea-2.0.0.0.tar or later archive file and
copies them to a location that is accessible by the
Example:
Router# archive tar /xtract
Cisco Unified CME router.
tftp://192.168.1.1/app-cme-ea-2.0.0.0.tar • source-url—URL of the source of the extension
flash:
assigner TAR file. Valid URLs can refer to TFTP or
HTTP servers or to flash memory.
• location—URL of the destination of the extension
assigner TAR file, including its Tcl script and audio
files. Valid URLs can refer to TFTP or HTTP servers or
to flash memory.

Configuring the Tcl Script

Perform this procedure to configure and load the Tcl script for the extension assigner feature. This
procedure also specifies the password that installation technicians enter to access the extension assigner
application. For more information about how to use Tcl scripts, see Cisco IOS Tcl IVR and VoiceXML
Application Guide for your Cisco IOS release.

Note For extension assigner, you must configure the Tcl script to use English as the language.

Tip To change the password, you must remove the existing extension assigner service and create a new
service that defines a new password.

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SUMMARY STEPS

1. enable
2. configure terminal
3. application
4. service service-name location
5. param ea-password password
6. paramspace english index number
7. paramspace english language en
8. paramspace english location location
9. paramspace english prefix en
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 application Enters application configuration mode to configure
packages and services.
Example:
Router(config)# application
Step 4 service service-name location Enters service parameter configuration mode to configure
parameters for the call-queue service.
Example: • service-name—Name of the extension assigner service.
Router(config-app)# service EA This arbitrary name is used to identify the service
tftp://10.1.1.100/app-cme-ea-2.0.0.0.tcl during configuration tasks.
• location—URL of the Tcl script for the extension
assigner service. Valid URLs can refer to TFTP or
HTTP servers or to flash memory.
Step 5 param ea-password password Sets the password that installation technicians enter to
access the extension assigner application.
Example: • password—Numerical password that installation
Router(config-app-param)# param ea-password technicians enter to access the extension assigner
1234 application. It can be 2 to 10 digits long.

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Command or Action Purpose

Step 6 paramspace english index number Defines the category of audio files that are used for dynamic
prompts by an IVR application.
Example: • language—Name of the language package. For the
Router(config-app-param)# paramspace english extension assigner, you must use English.
index 0
• number—Category group of the audio files (from
0 to 4). For example, audio files representing the days
and months can be category 1, audio files representing
units of currency can be category 2, and audio files
representing units of time—seconds, minutes, and
hours—can be category 3. Range is from 0 to 4; 0
means all categories.
Step 7 paramspace english language en Defines the language of audio files that are used for
dynamic prompts by an IVR application.
Example: • language—Name of the language package. For the
Router(config-app-param)# paramspace english extension assigner, you must use English.
language en
• prefix—Two-character code that identifies the language
associated with the audio files. For the extension
assigner, you must use en.
Step 8 paramspace english location location Defines the location of audio files that are used for dynamic
prompts by an IVR application.
Example: • language—Name of the language package. For the
Router(config-app-param)# paramspace english extension assigner, you must use English.
location
tftp://10.1.1.100/app-cme-ea-2.0.0.0.tcl • location—URL of the Tcl script for the extension
assigner service. Valid URLs can refer to TFTP or
HTTP servers or to flash memory.
Step 9 paramspace english prefix en Defines the prefix of audio files that are used for dynamic
prompts by an IVR application.
Example: • language—Name of the language package. For the
Router(config-app-param)# paramspace english extension assigner, you must use English.
prefix en
• prefix—Two-character code that identifies the language
associated with the audio files. For the extension
assigner, you must use en.
Step 10 end Returns to privileged EXEC mode.

Example:
Router(config-app-param)# end

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Specifying the Extension That Installation Technicians Call to Assign Extension

Numbers
Perform this procedure to specify the extension number that installation technicians call to access the
extension assigner application.

SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag voip
4. service service-name outbound
5. destination-pattern string
6. session target ipv4:destination-address
7. dtmf-relay h245-alphanumeric
8. codec g711ulaw
9. no vad
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 dial-peer voice tag voip Enters dial-peer configuration mode.
• tag—Number used during configuration tasks to
Example: identify this dial peer.
Router(config)# dial-peer voice 5999 voip
Step 4 service service-name outbound Loads and configures the extension assigner application on
a dial peer.
Example: • service-name—Name that identifies the voice
Router(config-dial-peer)# service EA out-bound application. This is a user-defined name and does not
have to match the script name. In this case, the name
must match the name that you used to load the
extension assigner Tcl script in the “Configuring the
Tcl Script” section on page 221.
• outbound—Indicates that this is an outbound dial peer.
It is required for extension assigner.

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Command or Action Purpose

Step 5 destination-pattern string Specifies either the prefix or the full E.164 telephone
number (depending on the dial plan) for a dial peer.
Example: • string—Series of digits that specify a pattern for the
Router(config-dial-peer)# destination pattern E.164 or private dialing plan telephone number. In this
5999 case, it is the extension number that the installation
technician calls when assigning an extension number to
a phone.
Step 6 session target ipv4:destination-address Designates a network-specific address to receive calls from
a VoIP dial peer.
Example: • destination—IP address of the dial peer to receive calls.
Router(config-dial-peer)# session target In this case, it must be the IP address for the
ipv4:172.16.200.200 Cisco Unified CME interface on the same router.
Step 7 dtmf-relay h245-alphanumeric Specifies how an H.323 or Session Initiation Protocol (SIP)
gateway relays dual tone multifrequency (DTMF) tones
between telephony interfaces and an IP network. Extension
Example:
Router(config-dial-peer)# dtmf-relay
assigner requires that you use h245-alphanumeric.
h245-alphanumeric • h245-alphanumeric—Forwards DTMF tones by using
the H.245 “alphanumeric” User Input Indication
method. Supports tones from 0 to 9, *, #, and from
A to D.
Step 8 codec codec Specifies the voice coder rate of speech for a dial peer.
• codec—Option that represents the correct voice
Example: decoder rate.
Router(config-dial-peer)# codec g711ulaw
Step 9 no vad Disables voice activity detection (VAD) for the calls using
a particular dial peer. Extension assigner requires this.
Example:
Router(config-dial-peer)# no vad
Step 10 end Returns to privileged EXEC mode.

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

Specifying Whether the Extension Used to Access Extension Assigner Is Dialed

Automatically
Perform this procedure to specify whether the extension number that installation technicians call to
access the extension assigner application is dialed automatically. Because this functionality should only
be available for the temporary extension numbers, you should configure it only for the ephone-dns for
those extensions.
This functionality is provided by the trunk command as described in the
Cisco Unified Communications Express Command Reference. To see an example of how to use the
trunk command with ephone-dns for the temporary extension numbers configured for extension
assigner, see the “Configuring Temporary Extension Numbers for Phones That Use Extension Assigner”
section on page 227.

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Configuring the Extension Assigner Feature

Perform this procedure to specify whether to use ephone tags (the tag numbers of your ephone
configurations) or provision-tags to reference the extension number. By default, the extension assigner
is enabled and configured to use the ephone tag.
The advantage of using the provision-tag is that you can make it easier for the installation technician to
assign extension numbers because you can configure the tag to match the primary extension number or
some other unique identifier for the phone, such as a jack number. The disadvantage is that you configure
an additional keyword for each ephone entry.
If you decide to use the ephone tag, it requires less configuration but the installation technician will have
to enter an arbitrary tag number instead of the actual extension number when configuring a phone. This
restriction is caused by the fact that the number of ephone tags that you can configure is limited by your
license. For example, if you use the ephone tag and you have a 100-user license, the installation
technician cannot enter 9001 for the tag because you can configure only ephone 1 to ephone 100.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. extension-assigner tag-type {ephone-tag | provision-tag}
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 4 extension-assigner tag-type {ephone-tag | Specifies whether the extension assigner feature uses
provision-tag} ephone tags (the tag numbers of your ephones
configurations) or provision-tags to reference the extension
Example: number.
Router(config-telephony)# extension-assigner • ephone-tag—Specifies that extension assigner uses the
tag-type provision-tag
ephone tag to reference the extension number that is
assigned to a phone. The installation technician enters
this number to assign an extension number to a phone.
• provision-tag—Specifies that extension assigner uses
the provision-tag to reference the extension number
that is assigned to a phone. The installation technician
enters this number to assign an extension number to a
phone.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Configuring Temporary Extension Numbers for Phones That Use Extension

Assigner
Perform this procedure to create ephone-dns to use as temporary extension numbers for any
Cisco Unified CME phones to which you want the installation technician to assign extension numbers.
These extensions numbers are automatically assigned to phones when they register and enable the
installation technician to then assign new extension numbers.
This procedure also configures the trunk command to enable each extension number to automatically
dial the extension assigner application.
The readme file that is included with the script contains some sample entries for this procedure that you
can edit to fit your needs.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. trunk digit-string [timeout seconds]
6. name name
7. exit
8. telephony-service
9. auto assign dn-tag to dn-tag
10. end

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Note Repeat steps 3 to 6 for each phone.

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 90 ephone-dn during configuration tasks.
• 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 We recommend that you use single-line mode for
your temporary extension numbers.
Step 4 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 9000 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 5 trunk digit-string [timeout seconds] (Optional) Configures the extension number to
automatically dial the extension assigner application.
Example: • digit-string—The number of the extension assigner
Router(config-ephone-dn)# trunk 9000 application. This number must match the number that
you configured in “Specifying the Extension That
Installation Technicians Call to Assign Extension
Numbers” section on page 224
• timeout seconds—(Optional) Interdigit timeout
between dialed digits, in seconds. Range is 3 to 30.
Default is 3.

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Command or Action Purpose

Step 6 name name (Optional) 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 hardware • 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 7 exit Exits ephone-dn configuration mode

Example:
Router(config-ephone-dn)# exit
Step 8 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 9 auto assign dn-tag to dn-tag Automatically assigns ephone-dn tags to Cisco Unified IP
phones as they register for service with a
Cisco Unified CME router. The ephone-dn tags that you
Example:
Router(config-telephony)# auto assign 90 to 99
specify in this command must match the tags that you
configured earlier in this procedure.
Step 10 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Configuring Extension Numbers That Installation Technicians Can Assign to

Phones
Perform this procedure to create ephone-dns for extensions numbers that installation technicians can
assign to phones.
The readme file provided with this feature contains some sample entries for this procedure that you can
edit to fit your needs.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. name name
6. end

Note Repeat steps 3 to 5 for each extension number that you want to assign.

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 20 ephone-dn during configuration tasks.
• 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.
Step 4 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 20 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 5 name name (Optional) 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 hardware • 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 6 end Returns to privileged EXEC mode.

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

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Configuring Ephones with Temporary MAC Addresses

Perform this procedure to create ephones with temporary MAC addresses for any Cisco Unified CME
phones to which you want the installation technician to assign extension numbers. When the installation
technician assigns a new extension number to a phone, this MAC address is overwritten by the actual
MAC address of the phone. The ephone entry that was created by the extension assigner for the
temporary extension number is then deleted.
You must also set the max-ephone value be at least one greater than the number of phones to which you
want to assign extension numbers. This will allow the autoregister feature to automatically create at least
one ephone for your temporarily extension numbers. Assuming that your max-dn setting is set high
enough, your max-ephone setting determines how many phones you should plug in at one time. For
example, if your max-ephone setting is ten more than the number of phones to which you want to assign
extension numbers, the you can plug in ten phones at a time. If you plug in eleven phones, one phone
will not register or get a temporary extension number until you assign an extension to one of the first ten
phones and reset the eleventh phone.

Note If you want to use Cisco VG224 analog voice gateways with extension assigner, you need a minimum of
24 temporary ephones available for each gateway because they will attempt to temporary register all 24
of their ports as ephones.

You are permitted to set the max-ephone value higher than the number of purchased CME phone seat
licenses is for the purpose of enrolling licensed phones using extension assigner.
The readme file provided with this feature contains some sample entries for this procedure that you can
edit to fit your needs.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. provision-tag number
5. mac-address 02EA.EAEA.number
6. type phone-type [addon 1 module-type [2 module-type]]
7. button
8. end

Note Repeat steps 3 to 7 for each phone.

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 20 number of ephones is version and platform-specific.
Type ? to display range.
If you use the ephone-tag keyword with the
extension-assigner tag-type command, this tag is used
to reference the extension number and must match the
number that the installation technician enters when
assigning an extension.
Step 4 provision-tag number (Optional) Specifies a unique sequence number that is used
by the extension assigner application only if you use the
provision-tag keyword with the extension-assigner
Example:
Router(config-ephone)# provision-tag 20
tag-type command.
• number—Unique sequence number that identifies
which ephone configuration and extension numbers to
assign to a phone. This number must match the number
that the installation technician enters when assigning an
extension.
Step 5 mac-address 02EA.EAEA.number Specifies a temporary MAC address number for this
ephone. For the extension assigner, this MAC address
should begin with 02EA.EAEA.
Example:
Router(config-ephone)# mac-address • number—We strongly recommends that you make this
02EA.EAEA.0020 number the same as the ephone number.

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Command or Action Purpose

Step 6 type phone-type [addon 1 module-type [2 Specifies the type of phone.
module-type]]
Note For Cisco Unified CME 4.0 and later versions, the
only types to which you can apply an add-on
Example: module are 7960, 7961, 7961GE, and 7970. For
Router(config-ephone)# type 7960 addon 1 7914 Cisco CME 3.4 and earlier versions, the only type to
which you can apply an add-on module is 7960.

• phone-type—Type ? to display valid phone types or see

the type command in the
Cisco Unified Communications Express Command
Reference.
• module-type—Valid entry is the following:
– 7914—Cisco Unified IP Phone 7914 Expansion
Module.
Step 7 button button-number{separator}dn-tag Associates a button number and line characteristics with an
extension (ephone-dn). Maximum number of buttons is
determined by phone type.
Example:
Router(config-ephone)# button 1:1 Note The Cisco Unified IP Phone 7910 has only one line
button, but can be given two ephone-dn tags.

• button-number—Number of a line button on an IP

phone, starting with 1 as the top button.
• dn-tag—Unique sequence number of the ephone-dn
that you want to appear on this button. For overlay lines
(separator is o or c), this argument can contain up to 25
ephone-dn tags, separated by commas
Step 8 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

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Configuring the Router to Automatically Save Your Configuration

Perform this procedure to configure the parameters for automatically saving your router configuration.
If the router configuration is not saved, any extension assignments made by the installation technician
will be lost when the router is restarted. We recommend that you configure your router to save your
configuration every 30 minutes.
The alternative to this optional procedure is to have the installation technician connect to the router and
manually issue the “write memory” command to save the router configuration.

SUMMARY STEPS

1. enable
2. configure terminal
3. kron policy-list list-name
4. cli write
5. exit
6. kron occurrence occurrence-name [user username] in [[numdays:]numhours:]nummin {oneshot |
recurring}
7. policy-list list-name
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 kron policy-list list-name Specifies a name for a new or existing Command Scheduler
policy list and enters kron-policy configuration mode.
Example: • If the value of the list-name argument is new, a new
Router(config)# kron policy-list save-config policy list structure is created.
• If the value of the list-name argument exists, the
existing policy list structure is accessed. No editor
function is available, and the policy list is run in the
order in which it was configured.
• Specifies a Command Scheduler policy list.
Step 4 cli write Specifies the fully-qualified EXEC command and
associated syntax to be added as an entry in the specified
Command Scheduler policy list. In this case, we want to
Example:
Router(config-kron-policy)# cli write
save the router configuration, so the command is write.

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Command or Action Purpose

Step 5 exit Returns to global configuration mode.

Example:
Router(config-kron-policy)# exit
Step 6 kron occurrence occurrence-name [user username] Specify schedule parameters for a Command Scheduler
[[in numdays:]numhours:]nummin {oneshot | occurrence and enters kron-occurrence configuration mode.
recurring}
We recommend that you configure your router to save your
configuration every 30 minutes.
Example: • occurrence-name—Specifies the name of the
Router(config)# kron occurrence backup in 30
occurrence. Length of occurrence-name is from 1 to 31
recurring
characters. If the occurrence-name is new, an
occurrence structure will be created. If the
occurrence-name is not new, the existing occurrence
will be edited.
• user—(Optional) Used to identify a particular user.
• username—Name of user.
• in—Identifies that the occurrence is to run after a
specified time interval. The timer starts when the
occurrence is configured.
• numdays:—(Optional) Number of days. If used, add a
colon after the number.
• numhours:—(Optional) Number of hours. If used, add
a colon after the number.
• nummin:—(Optional) Number of minutes.
• oneshot—Identifies that the occurrence is to run only
one time. After the occurrence has run, the
configuration is removed.
• recurring—Identifies that the occurrence is to run on a
recurring basis.
Step 7 policy-list list-name Specifies a Command Scheduler policy list.

Example:
Router(config-kron-occurrence)# policy-list
save-config
Step 8 end Returns to privileged EXEC mode.

Example:
Router(config-kron-occurrence)# end

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Provide the Installation Technician with the Needed Information

Before the installation technician can assign extension numbers to the new phones, you must provide the
following information:
• How many phones the installation technician can plug in at one time. This is determined by the
number of temporary MAC addresses that you configured.
• Which extension number to dial to access the extension assigner application.
• Whether the number is dialed automatically when a phone goes off hook.
• What password to enter to access the application.
• Which tag numbers to enter to assign an extension to each phone.

Assigning New Extension Numbers

Initially, when you install your phones, they are assigned a temporary, random extension number to
enable you to access extension assigner and assign new extension numbers.

Step 1 Get the information you need to use extension assigner from your system administrator. For a list of this
information, see the “Provide the Installation Technician with the Needed Information” section on
page 236.
Step 2 Dial the appropriate extension number to access the extension assigner system.
Step 3 Enter the password for the extension assigner and press #.
Step 4 Enter the ID number that represents this phone’s extension and press #.
Step 5 If the extension is not assigned to another phone, press 1 to confirm that you want to assign the extension
to your phone, then hang up. After the phone resets, the assignment is complete.
Step 6 If the extension is assigned to another phone that is idle:
a. Press 2 to confirm that you want to unassign the extension from the other phone.
b. Hang up.
c. Repeat this procedure beginning at Step 2.
Step 7 If the extension is assigned to another phone that is in use, either:
• Return to Step 5 to enter another extension number.
• Perform the procedures in the “Unassigning the Current Extension Number” section on page 237
and then repeat this procedure beginning at Step 2.

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Unassigning the Current Extension Number

After the new extension number is assigned, you may find that you assigned the wrong number or that
your original dial plan has changed. If this is the case, you can unassign the number so that it can be used
by an another phone.

Step 1 Get the information you need to use extension assigner from your system administrator. For a list of this
information, see the “Provide the Installation Technician with the Needed Information” section on
page 236.
Step 2 Dial the appropriate extension number to access the extension assigner system.
Step 3 Enter the password for the extension assigner and press #.
Step 4 Enter the ID number that represents this phone’s extension and press #.
Step 5 When you enter the ID number for the extension that is currently assigned to this phone, you are
prompted to press 2 to confirm that you want to unassign the extension from the phone.
Step 6 Hang up.

Reassigning the Current Extension Number

Use this procedure if you:
• Need to replace a broken phone
• Find that you assigned the wrong number
You can reassign a new extension number to a phone if that number:
• Is not assigned to another phone
• Is assigned to another phone but the phone is currently idle or you unassign the extension

Step 1 Get the information you need to use extension assigner from your system administrator. For a list of this
information, see the “Provide the Installation Technician with the Needed Information” section on
page 236.
Step 2 Dial the appropriate extension number to access the extension assigner system.
Step 3 Enter the password for the extension assigner and press #.
Step 4 Enter the ID number that represents this phone’s extension and press #.
Step 5 If the extension is not assigned to another phone, press 1 to confirm that you want to assign the extension
to your phone, then hang up. After the phone resets, the reassignment is complete.
Step 6 If the extension is assigned to another phone that is idle:
a. Press 2 to confirm that you want to unassign the extension from the other phone.
b. Hang up
c. Perform the procedure in the “Assigning New Extension Numbers” section on page 236.
Step 7 If the extension is assigned to another phone that is in use, either:
• Return to Step 5 to enter another extension number.

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• Perform the procedures in the “Unassigning the Current Extension Number” section on page 237
and “Assigning New Extension Numbers” section on page 236.

Verifying Extension Assigner

Step 1 Use the debug ephone extension-assigner command to display status messages produced by the
extension assigner application.
*Jun 9 19:08:10.627: ephone_query: inCallID=47, tag=4, ephone_tag=4
*Jun 9 19:08:10.627: extAssigner_IsEphoneMacPreset: ephone_tag = 4,
ipKeyswitch.max_ephones = 96
*Jun 9 19:08:10.627: extAssigner_IsEphoneMacPreset: ephone_ptr->mac_addr_str =
000B46BDE075, MAC_EXT_RESERVED_VALUE = 02EAEAEA0000
*Jun 9 19:08:10.627: SkinnyGetActivePhoneIndexFromCallid: callID = 47
*Jun 9 19:08:10.627: SkinnyGetActivePhoneIndexFromCallid: vdbPtr->physical_interface_type
(26); CV_VOICE_EFXS (26)
*Jun 9 19:08:10.627: SkinnyGetActivePhoneIndexFromCallid: vdbPtr->type (6);
CC_IF_TELEPHONY (6)
*Jun 9 19:08:10.627: SkinnyGetActivePhoneIndexFromCallid: htsp->sig_type (26);
CV_VOICE_EFXS (26)
*Jun 9 19:08:10.627: SkinnyGetActivePhoneIndexFromCallid: dn = 4, chan = 1
*Jun 9 19:08:10.627: ephone_query: EXTASSIGNER_RC_SLOT_ASSIGNED_TO_CALLING_PHONE
*Jun 9 19:08:22.763: ephone_unassign: inCallID=47, tag=4, ephone_tag=4
*Jun 9 19:08:22.763: extAssigner_IsEphoneMacPreset: ephone_tag = 4,
ipKeyswitch.max_ephones = 96
*Jun 9 19:08:22.763: extAssigner_IsEphoneMacPreset: ephone_ptr->mac_addr_str =
000B46BDE075, MAC_EXT_RESERVED_VALUE = 02EAEAEA000
*Jun 9 19:08:22.763: is_ephone_auto_assigned: button-1 dn_tag=4
*Jun 9 19:08:22.763: is_ephone_auto_assigned: NO
*Jun 9 19:08:22.763: SkinnyGetActivePhoneIndexFromCallid: callID = 47
*Jun 9 19:08:22.763: SkinnyGetActivePhoneIndexFromCallid: vdbPtr->physical_interface_type
(26); CV_VOICE_EFXS (26)
*Jun 9 19:08:22.767: SkinnyGetActivePhoneIndexFromCallid: vdbPtr->type (6);
CC_IF_TELEPHONY (6)
*Jun 9 19:08:22.767: SkinnyGetActivePhoneIndexFromCallid: htsp->sig_type (26);
CV_VOICE_EFXS (26)
*Jun 9 19:08:22.767: SkinnyGetActivePhoneIndexFromCallid: dn = 4, chan = 1
*Jun 9 19:08:29.795: ephone-4[8]:fStationOnHookMessage: Extension Assigner request
restart, cmd=2, new mac=02EAEAEA0004, ephone_tag=4
*Jun 9 19:08:30.063: %IPPHONE-6-UNREGISTER_NORMAL: ephone-4:SEP000B46BDE075 IP:5.5.0.1
Socket:8 DeviceType:Phone has unregistered normally.
*Jun 9 19:08:30.063: ephone-4[8][SEP000B46BDE075]:extAssigner_assign: new
mac=02EAEAEA0004, ephone-tag=4
*Jun 9 19:08:30.063: extAssigner_simple_assign: mac=02EAEAEA0004, tag=4
*Jun 9 19:08:30.063: ephone_updateCNF: update cnf_file ephone_tag=4
*Jun 9 19:08:30.063: extAssigner_assign: restart again (mac=02EAEAEA0004) ephone_tag=4
*Jun 9 19:08:30.131: %IPPHONE-6-REG_ALARM: 23: Name=SEP000B46BDE075 Load=8.0(2.0)
Last=Reset-Restart
*Jun 9 19:08:30.135: %IPPHONE-6-REGISTER_NEW: ephone-7:SEP000B46BDE075 IP:5.5.0.1
Socket:10 DeviceType:Phone has registered.
*Jun 9 19:08:30.503: %IPPHONE-6-UNREGISTER_NORMAL: ephone-7:SEP000B46BDE075 IP:5.5.0.1
Socket:10 DeviceType:Phone has unregistered normally.
*Jun 9 19:08:43.127: %IPPHONE-6-REG_ALARM: 22: Name=SEP000B46BDE075 Load=8.0(2.0)
Last=Reset-Reset
*Jun 9 19:08:43.131: %IPPHONE-6-REGISTER: ephone-7:SEP000B46BDE075 IP:5.5.0.1 Socket:13
DeviceType:Phone has registered.

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Step 2 Use the debug voip application script command to display status messages produced by the server as
it runs the assigner application Tcl script.
Jun 20 23:17:45.795: //22//TCL :/tcl_PutsObjCmd: TCL: ***** >>> app-cme-ea-2.0.0.0.tcl <<<
*****
Jun 20 23:17:45.799: //22//TCL :/tcl_PutsObjCmd: TCL: ***** >>> Cisco CME Extension
Assigner Application <<< ****
Jun 20 23:17:45.799: //22//TCL :/tcl_PutsObjCmd: >>> PROMPT: Enter password <<<
Jun 20 23:17:54.559: //22//TCL :/tcl_PutsObjCmd: >>> Collect Password Status = cd_005 <<<
Jun 20 23:17:54.563: //22//TCL :/tcl_PutsObjCmd: >>> INFO: Authentication Successful <<<
Jun 20 23:17:54.563: //22//TCL :/tcl_PutsObjCmd: >>> PROMPT: Please enter the phone tag
number followed by the # key. Press * to re-enter the tag number <<<
Jun 20 23:17:59.839: //22//TCL :/tcl_PutsObjCmd: >>> Ephone TAG Digit Collect Status =
cd_005 <<<
Jun 20 23:17:59.843: //22//TCL :/tcl_PutsObjCmd: >>> INFO: Phone Query result = 1 <<<
Jun 20 23:17:59.843: //22//TCL :/tcl_PutsObjCmd: >>> PROMPT: Ephone Tag 6 is available
<<<
Jun 20 23:17:59.843: //22//TCL :/tcl_PutsObjCmd: >>> PROMPT: To assign extension to Phone,
press 1 to confirm, 9 to cancel <<<
Jun 20 23:17:59.851: //22//TCL :/tcl_PutsObjCmd: >>> INFO: ephone 6 is available <<<
Jun 20 23:18:20.375: //22//TCL :/tcl_PutsObjCmd: >>> INFO: TAPS Status = cd_005 <<<
Jun 20 23:18:20.379: //22//TCL :/tcl_PutsObjCmd: >>> PROMPT: Extension assignment is
successful <<<
Jun 20 23:18:20.379: //22//TCL :/tcl_PutsObjCmd: >>> INFO: Ephone extension is assigned
successfully <<<
Jun 20 23:18:28.975: //22//TCL :/tcl_PutsObjCmd: **** >>> TCL: Closing Cisco CM

Step 3 Use the debug ephone state command as described in the Cisco IOS Debug Command Reference.

Configuration Examples for Extension Assigner

This example for extension assigner shows a router configuration that has these characteristics:
• The extension that the installation technician dials to access the extension assigner application is
0999.
• The password that the installation technician enters to access the extension assigner application is
1234.
• The auto assign command is configured to assign extensions 0001 to 0005.
• The installation technician can use extension assigner to assign extension numbers 6001 to 6005.
• The extension assigner uses the provision-tag to identify which ephone configuration and extension
numbers to assign to the phone.
• The auto-reg-ephone command is shown but is not required, as it is enabled by default.
• The kron command is used to automatically save the router configuration.
• The max-ephone and max-dn settings of 51 are high enough to allow the installation technician to
assign extensions to 50 phones, plugging them in one at a time. If the installation technician is
assigning extensions to 40 phones, eleven can be plugged in one at a time. There is an exception if
you use Cisco VG224 Analog Voice Gateways. Extension assigner creates 24 ephones for each
Cisco VG224 Analog Voice Gateway, one for each port.

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Router# show running-config

version 12.4
no service password-encryption
!
hostname Test-Router
!
boot-start-marker
boot system flash:c2800nm-ipvoice-mz.2006-05-31.GOPED_DEV
boot-end-marker
!
enable password ww
!
no aaa new-model
!
resource policy
!
ip cef
no ip dhcp use vrf connected
!
ip dhcp pool pool21
network 172.21.0.0 255.255.0.0
default-router 172.21.200.200
option 150 ip 172.30.1.60
!
no ip domain lookup
!
application
service EA flash:ea/app-cme-ea-2.0.0.0.tcl
paramspace english index 0
paramspace english language en
param ea-password 1234
paramspace english location flash:ea/
paramspace english prefix en
!
interface GigabitEthernet0/0
no ip address
duplex auto
speed 100
no keepalive
!
interface GigabitEthernet0/0.21
encapsulation dot1Q 21
ip address 172.21.200.200 255.255.0.0

ip http server
!
control-plane
!
dial-peer voice 999 voip
service EA out-bound
destination-pattern 0999
session target ipv4:172.21.200.200
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
telephony-service
extension-assigner tag-type provision-tag
max-ephones 51
max-dn 51
ip source-address 172.21.200.200 port 2000
auto-reg-ephone
auto assign 101 to 105

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system message Test-CME

create cnf-files version-stamp 7960 Jun 14 2006 05:37:34
!
ephone-dn 1 dual-line
number 6001
!
ephone-dn 2 dual-line
number 6002
!
ephone-dn 3 dual-line
number 6003
!
ephone-dn 4 dual-line
number 6004
!
ephone-dn 5 dual-line
number 6005
!
ephone-dn 101
number 0101
label Temp-Line-not assigned yet
!
ephone-dn 102
number 0102
label Temp-Line-not assigned yet
!
ephone-dn 103
number 0103
label Temp-Line-not assigned yet
!
ephone-dn 104
number 0104
label Temp-Line-not assigned yet
!
ephone-dn 105
number 0105
label Temp-Line-not assigned yet
!
ephone 1
provision-tag 101
mac-address 02EA.EAEA.0001
button 1:1
!
ephone 2
provision-tag 102
mac-address 02EA.EAEA.0002
button 1:2
!
ephone 3
provision-tag 103
mac-address 02EA.EAEA.0003
button 1:3
!
ephone 4
provision-tag 104
mac-address 02EA.EAEA.0004
button 1:4
!
ephone 5
provision-tag 105
mac-address 02EA.EAEA.0005
button 1:5
!
kron occurrence backup in 30 recurring

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Additional References

policy-list writeconfig
!
kron policy-list writeconfig
cli write
!
line con 0
line aux 0
line vty 0 4
logging synchronous
!
no scheduler max-task-time
scheduler allocate 20000 1000
!
end

Additional References
The following sections provide references related to extension assigner.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified Communications Express System Administrator
Guide
• Cisco Unified Communications Express Command Reference
Cisco IOS voice configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Voice Command Reference
• Cisco IOS Debug Command Reference
• Cisco IOS Tcl IVR and VoiceXML Application Guide
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Technical Support & Documentation http://www.cisco.com/techsupport
website contains thousands of pages of searchable
technical content, including links to products,
technologies, solutions, technical tips, and tools.
Registered Cisco.com users can log in from this page to
access even more content.

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Feature Information for Extension Assigner

Table 9 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 9 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 9 Feature Information for Extension Assigner

Cisco Unified CME

Feature Name Version Feature Information
Extension Assigner 4.0(3) Enables installation technicians to assign extension
numbers to Cisco Unified CME phones without accessing
the server.

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 Generating Configuration Files for Phones

Last Updated: March 26, 2007

This chapter describes how to generate configuration files for Cisco Unified IP phones that are
connected to a Cisco Unified Communications Manager Express (Cisco Unified CME) router.

Contents
• Information About Configuration Files, page 245
• How to Generate Configuration Files for Phones, page 247
• Where to Go Next, page 254
• Additional References, page 255

Information About Configuration Files

To generate configuration files for phones in Cisco Unified CME, you should understand the following
concepts:
• Configuration Files for Phones in Cisco Unified CME, page 245
• Per-Phone Configuration Files, page 246

Configuration Files for Phones in Cisco Unified CME

When a phone requests service from Cisco Unified CME, the registrar confirms the username, i.e. the
phone number for the phone. The phone accesses its configuration profile on the TFTP server, typically
the Cisco Unified CME router, and processes the information contained in the file, registers itself, and
puts the phone number on the phone console display.
Minimally, a configuration profile contains the MAC address, the type, and the number phone number
that is permitted by the registrar to handle the Register message for a particular Cisco Unified IP phone.
Any time you create or modify parameters for either an individual phone or a directory number, generate
a new phone configuration to properly propagate the parameters.

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By default, there is one shared XML configuration file located in system:/its/ for all Cisco Unified IP
phones that are running SCCP. For SIP phones directly connected to Cisco Unified CME, an individual
configuration profile is created for each phone and stored in system:/cme/sipphone/.
When an IP phone comes online or is rebooted, it automatically gets information about itself from the
appropriate configuration file.
The Cisco universal application loader for phone firmware files allows you to add additional phone
features across all protocols. To do this, a hunt algorithm searches for multiple configuration files. After
a phone is reset or restarted, the phone automatically selects protocol depending on which matching
configuration file is found first. To ensure that Cisco Unified IP phones download the appropriate
configuration for the desired protocol, SCCP or SIP, you must properly configure the IP phones before
connecting or rebooting the phones. The hunt algorithm searches for files in the following order:
1. CTLSEP<mac> file for a SCCP phone—For example, CTLSEP003094C25D2E.tlv
2. SEP <mac> file for a SCCP phone—For example, SEP003094C25D2E.cnf.xml
3. SIP <mac> file for a SIP phone—For example, SIP003094C25D2E.cnf or gk003069C25D2E
4. XML default file for SCCP phones—For example, SEPDefault.cnf.xmls
5. XML default file for SIP phones—For example, SIPDefault.cnf.
In Cisco Unified CME 4.0 and later for SCCP and in Cisco CME 3.4 and later for SIP, you can designate
one of the following locations in which to store configuration files:
• System (Default)—For SCCP phones, one configuration file is created, stored, and used for all
phones in the system. For SIP phones, an individual configuration profile is created for each phone.
• Flash or slot 0—When flash or slot 0 memory on the router is the storage location, you can create
additional configuration files to be applied per phone type or per individual phone, such as user or
network locales.
• TFTP—When an external TFTP server is the storage location, you can create additional
configuration files to be applied per phone type or per individual phone, which are required for
multiple user and network locales.

Per-Phone Configuration Files

If configurations files for SCCP phones are to be stored somewhere other than in the default location,
the following individual configuration files can be created for SCCP phones:
• Per phone type—Creates separate configuration files for each phone type and all phones of the same
type use the same configuration file. This method is not supported if the configuration files are to
stored in the system location.
• Per phone—Creates a separate configuration file for each phone, by MAC address. This method is
not supported if the configuration files are to be stored in the system location.
For configuration information, see the “SCCP: Defining Per-Phone Configuration Files and Alternate
Location” section on page 129.

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How to Generate Configuration Files for Phones

This section contains the following tasks:
• SCCP: Generating Configuration Files for SCCP Phones, page 247
• SCCP: Verifying Configuration Files for SCCP Phones, page 248
• SIP: Generating Configuration Profiles for SIP Phones, page 250
• SIP: Verifying Configuration Profiles for SIP Phones, page 251

SCCP: Generating Configuration Files for SCCP Phones

To generate the configuration profile files that are required by the SCCP phones in Cisco Unified CME
and write them to either system memory or to the location specified by the cnf-file location command,
follow the steps in this section.

Restrictions
• Externally stored and per-phone configuration files are not supported on the Cisco Unified IP Phone
7902G, 7910, 7910G, or 7920, or the Cisco Unified IP Conference Station 7935 and 7936.
• TFTP does not support file deletion. When configuration files are updated, they overwrite any
existing configuration files with the same name. If you change the configuration file location, files
are not deleted from the TFTP server.
• Generating configuration files on flash or slot 0 can take up to a minute, depending on the number
of files being generated.
• For smaller routers such as Cisco 2600 series routers, you must manually enter the squeeze
command to erase files after changing the configuration file location or entering any commands that
trigger the deletion of configuration files. Unless you use the squeeze command, the space used by
the moved or deleted configuration files is not usable by other files.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. create cnf-files
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 create cnf-files Builds the XML configuration files required for IP phones.

Example:
Router(config-telephony)# create cnf-files
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Examples
The following example selects flash memory as the configuration file storage location and per-phone as
the type of configuration files that the system generates.
telephony-service
cnf-file location flash:
cnf-file perphone

SCCP: Verifying Configuration Files for SCCP Phones

To verify the Cisco Unified CME phone configuration, perform the following steps.

SUMMARY STEPS

1. show telephony-service all

2. show telephony-service tftp-bindings

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DETAILED STEPS

Step 1 show telephony-service all

Use this command to verify the configuration for phones, directory numbers, voice ports, and dial peers
in Cisco Unified CME.
Router# show telephony-service all

CONFIG (Version=4.0(0))
=====================
Version 4.0(0)
Cisco Unified CallManager Express
For on-line documentation please see:
www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_support_series_home.html

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 2 show telephony-service tftp-bindings

Use this command to display the current configuration files accessible to IP phones.
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

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SIP: Generating Configuration Profiles for SIP Phones

To generate the configuration profile files that are required by the SIP phones in Cisco Unified CME and
write them to the location specified by the tftp-path (voice register global) command, follow the steps
in this section.
Any time you create or modify parameters under the voice register dn or voice register pool
configuration modes, generate a new configuration profile and properly propagate the parameters.

Caution If your Cisco Unified CME system supports SCCP and also SIP phones, do not connect your SIP phones
to the network until after you have verified the phone configuration profiles.

Prerequisites
• Cisco Unified CME 3.4 or a later version.
• The mode cme command must be enabled in Cisco Unified CME.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. file text
5. create profile
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 file text (Optional) Generates ASCII text files of the configuration
profiles generated for Cisco Unified IP Phone 7905s and
7905Gs, Cisco Unified IP Phone 7912s and 7912Gs,
Example:
Router(config-register-global)# file text
Cisco ATA-186, or Cisco ATA-188.
• Default—System generates binary files to save disk
space.
Step 5 create profile Generates configuration profile files required for SIP
phones and writes the files to the location specified with
tftp-path command.
Example:
Router(config-register-global;)# create profile
Step 6 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end

SIP: Verifying Configuration Profiles for SIP Phones

To verify the configuration profiles, perform the following steps. SIP phones to be connected to
Cisco Unified CME can register and minimally, have an assigned phone number, only if the
configuration is correct.

SUMMARY STEPS

1. show voice register tftp-bind

2. show voice register profile
3. more system

DETAILED STEPS

Step 1 show voice register tftp-bind

Use this command to display a list of configuration profiles that are accessible to SIP phones using TFTP.
The file name includes the MAC address for each SIP phone, such as SIP<mac-address>.cnf. Verify that
a configuration profile is available for each SIP phone in Cisco Unified CME.

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The following is sample output from this command:

Router(config)# show voice register tftp-bind

tftp-server SIPDefault.cnf url system:/cme/sipphone/SIPDefault.cnf

tftp-server syncinfo.xml url system:/cme/sipphone/syncinfo.xml
tftp-server SIP0009B7F7532E.cnf url system:/cme/sipphone/SIP0009B7F7532E.cnf
tftp-server SIP000ED7DF7932.cnf url system:/cme/sipphone/SIP000ED7DF7932.cnf
tftp-server SIP0012D9EDE0AA.cnf url system:/cme/sipphone/SIP0012D9EDE0AA.cnf
tftp-server gk123456789012 url system:/cme/sipphone/gk123456789012
tftp-server gk123456789012.txt url system:/cme/sipphone/gk123456789012.txt

Step 2 show voice register profile

Use this command to display the contents of the ASCII format configuration profile for a particular voice
register pool.

Note To generate ASCII text files of the configuration profiles for Cisco Unified IP Phone 7905s and 7905Gs,
Cisco Unified IP Phone 7912s and 7912Gs, Cisco ATA-186s, and Cisco ATA-188s, use the file text
command.

The following is sample output from this command displaying information in the configuration profile
for voice register pool 4.
Router# show voice register profile text 4

Pool Tag: 4
# txt
AutoLookUp:0
DirectoriesUrl:0
…
CallWaiting:1
CallForwardNumber:0
Conference:1
AttendedTransfer:1
BlindTransfer:1
…
SIPRegOn:1
UseTftp:1
UseLoginID:0
UIPassword:0
NTPIP:0.0.0.0
UID:2468
…

Step 3 more system

Use this command to display the contents of the configuration profile for a particular Cisco Unified IP
Phone 7940, Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7960, or Cisco Unified IP Phone
7960G.
The following is sample output from this command displaying information in two SIP configuration
profile files. The SIPDefault.cnf configuration profile is a shared file and SIP<MAC address>.cnf is the
SIP configuration profile for the SIP phone with the designated MAC address.
Router# more system:/cme/sipphone/SIPDefault.cnf

image_version: "P0S3-07-4-00";
proxy1_address: "10.1.18.100";
proxy2_address: "";
proxy3_address: "";
proxy4_address: "";

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proxy5_address: "";
proxy6_address: "";
proxy1_port: "5060";
proxy2_port: "";
proxy3_port: "";
proxy4_port: "";
proxy5_port: "";
proxy6_port: "";
proxy_register: "1";
time_zone: "EST";
dst_auto_adjust: "1";
dst_start_month: "April";
dst_start_day: "";
dst_start_day_of_week: "Sun";
dst_start_week_of_month: "1";
dst_start_time: "02:00";
dst_stop_month: "October";
dst_stop_day: "";
dst_stop_day_of_week: "Sun";
dst_stop_week_of_month: "8";
dst_stop_time: "02:00";
date_format: "M/D/Y";
time_format_24hr: "0";
local_cfwd_enable: "1";
directory_url: "";
messages_uri: "2000";
services_url: "";
logo_url: "";
stutter_msg_waiting: "0";
sync: "0000200155330856";
telnet_level: "1";
autocomplete: "1";
call_stats: "0";
Domain_Name: "";
dtmf_avt_payload: "101";
dtmf_db_level: "3";
dtmf_inband: "1";
dtmf_outofband: "avt";
dyn_dns_addr_1: "";
dyn_dns_addr_2: "";
dyn_tftp_addr: "";
end_media_port: "32766";
http_proxy_addr: "";
http_proxy_port: "80";
nat_address: "";
nat_enable: "0";
nat_received_processing: "0";
network_media_type: "Auto";
network_port2_type: "Hub/Switch";
outbound_proxy: "";
outbound_proxy_port: "5060";
proxy_backup: "";
proxy_backup_port: "5060";
proxy_emergency: "";
proxy_emergency_port: "5060";
remote_party_id: "0";
sip_invite_retx: "6";
sip_retx: "10";
sntp_mode: "directedbroadcast";
sntp_server: "0.0.0.0";
start_media_port: "16384";
tftp_cfg_dir: "";
timer_invite_expires: "180";
timer_register_delta: "5";

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timer_register_expires: "3600";
timer_t1: "500";
timer_t2: "4000";
tos_media: "5";
voip_control_port: "5060";

Router# more system:/cme/sipphone/SIP000CCE62BCED.cnf

image_version: "P0S3-07-4-00";
user_info: "phone";
line1_name: "1051";
line1_displayname: "";
line1_shortname: "";
line1_authname: "1051";
line1_password: "ww";
line2_name: "";
line2_displayname: "";
line2_shortname: "";
line2_authname: "";
line2_password: "";
auto_answer: "0";
speed_line1: "";
speed_label1: "";
speed_line2: "";
speed_label2: "";
speed_line3: "";
speed_label3: "";
speed_line4: "";
speed_label4: "";
speed_line5: "";
speed_label5: "";
call_hold_ringback: "0";
dnd_control: "0";
anonymous_call_block: "0";
callerid_blocking: "0";
enable_vad: "0";
semi_attended_transfer: "1";
call_waiting: "1";
cfwd_url: "";
cnf_join_enable: "1";
phone_label: "";
preferred_codec: "g711ulaw";

Where to Go Next
After you generate a configuration file for a Cisco Unified IP phone connected to the
Cisco Unified CME router, you are ready to download the file to the phone to be configured. See
“Resetting and Restarting Phones” on page 257.

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Last Updated: March 26, 2007

This chapter describes how to reset or restart Cisco Unified IP phones that are connected to
Cisco Unified Communications Manager Express (Cisco Unified CME).

Contents
• Information About Resetting and Restarting Phones, page 257
• How to Reset and Restart Phones, page 258
• Additional References, page 265

Information About Resetting and Restarting Phones

Before resetting and restarting IP phones in Cisco Unified CME, you should understand the following
concept:
• Differences between Resetting and Restarting IP Phones, page 257

Differences between Resetting and Restarting IP Phones

Cisco Unified IP phones must be rebooted after configuration changes in order for the changes to be
effective. Configurations for phones in Cisco Unified CME are downloaded when a phone is rebooted
or reset. You can reboot a single phone or you can reboot all phones in a Cisco Unified CME system.
The differences between reboot types are summarized in Table 10.

Note When rebooting multiple IP phones, it is possible for a conflict to occur if too many phones attempt to
access changed Cisco Unified CME configuration information via TFTP simultaneously.

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Table 10 reset and restart Command Differences

reset Command restart Command

Type of Reboot Similar to power-off, power-on reboot. Quick restart.
Phone Configurations Downloads configurations for IP phones. Downloads configurations for IP phones.
DHCP and TFTP Contacts DHCP and TFTP servers for updated Phones contact the TFTP server for updated
configuration information. configuration information and reregister without
Note This command was introduced for SIP contacting the DHCP server.
phones in Cisco CME 3.4. Note This command was introduced for SIP
phones in Cisco Unified CME 4.1.
Processing Time Takes longer to process when updating multiple Faster processing for multiple phones.
phones.
When Required • Date and time settings • Directory numbers
• Network locale • Phone buttons
• Phone firmware • Speed-dial numbers
• Source address
• TFTP path
• URL parameters
• User locale
• Voicemail access number
Can be used when updating the following:
• Directory numbers
• Phone buttons
• Speed-dial numbers

How to Reset and Restart Phones

Note If phones are not yet plugged in, resetting or restarting phones is not necessary. Instead, connect your IP
phones to your network to boot the phone and download the required configuration files.

This sections contains the following tasks:

• SCCP: Using the reset Command, page 259 (Required)
• SCCP: Using the restart Command, page 260 (Required)
• SIP: Using the reset Command, page 261 (Required)
• SIP: Using the restart Command, page 263 (Required)
• Verifying Basic Calling, page 264 (Optional)

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SCCP: Using the reset Command

To reboot and reregister one or more SCCP phones, including contacting the DHCP server for updated
information, perform the following steps.

Prerequisites
• Phones to be rebooted are connected to the Cisco Unified CME router.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
or
ephone phone-tag
4. reset {all [time-interval] | cancel | mac-address mac-address | sequence-all}
or
reset
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
or
or
ephone ephone-tag
Enters ephone configuration mode.

Example:
Router(config)# telephony-service
or
Router(config)# ephone 1

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Command or Action Purpose

Step 4 reset {all [time-interval] | cancel | Performs a complete reboot of the specified or all phones
mac-address mac-address | sequence-all} running SCCP, including contacting the DHCP and TFTP
or servers for the latest configuration information.
reset
or
Performs a complete reboot of the individual SCCP phone
Example: being configured.
Router(config-telephony)# reset all
or
Router(config-ephone)# reset
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end
or
Router(config-ephone)# end

SCCP: Using the restart Command

To fast reboot and reregister one or more SCCP phones, perform the following steps.

Prerequisites
• Phones to be rebooted are connected to the Cisco Unified CME router.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
or
ephone ephone-tag
4. restart {all [time-interval] | mac-address}
or
restart
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.
or
or
ephone ephone-tag
Enters ephone configuration mode.

Example:
Router(config)# telephony-service
or
Router(config)# ephone 1
Step 4 restart {all [time-interval] | mac-address} Performs a fast reboot of the specified phone or all phones
or running SCCP associated with this Cisco Unified CME
router. Does not contact the DHCP server for updated
restart
information.
or
Example:
Router(config-telephony)# restart all Performs a fast reboot of the individual SCCP phone being
or configured.
Router(config-ephone)# restart
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

SIP: Using the reset Command

To reboot and reregister one or more SIP phones, including contacting the DHCP server for updated
information, perform the following steps.

Prerequisites
• Cisco Unified CME 3.4 or later.
• The mode cme command must be enabled in Cisco Unified CME.
• Phones to be rebooted are connected to the Cisco Unified CME router.

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SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
or
voice register pool pool-tag
4. reset
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
or parameters for all supported SIP phones in
voice register pool pool-tag Cisco Unified CME.
or
Example: Enters voice register pool configuration mode to set
Router(config)# voice register global phone-specific parameters for SIP phones
or
Router(config)# voice register pool 1
Step 4 reset Performs a complete reboot of all phones connected to this
router that are running SIP, including contacting the DHCP
and TFTP servers for the latest configuration information.
Example:
Router(config-register-global)# reset or
or
Performs a complete reboot of the individual SIP phone
Router(config-register-pool)# reset being configured.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end
or
Router(config-register-pool)# end

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SIP: Using the restart Command

To fast reboot and reregister one or more SIP phones, perform the following steps.

Prerequisites
• Cisco Unified CME 4.1 or later.
• The mode cme command must be enabled in Cisco Unified CME.
• Phones to be rebooted are connected to the Cisco Unified CME router.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
or
voice register pool pool-tag
4. restart
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice register global Enters voice register global configuration mode to set
or parameters for all supported SIP phones in
voice register pool pool-tag Cisco Unified CME.
or
Example: Enters voice register pool configuration mode to set
Router(config)# voice register global phone-specific parameters for SIP phones
or
Router(config)# voice register pool 1
Step 4 restart Performs a fast reboot all SIP phones associated with this
Cisco Unified CME router. Does not contact the DHCP
server for updated information.
Example:
Router(config-register-global)# restart or
or
Performs a fast reboot of the individual SIP phone being
Router(config-register-pool)# restart configured.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end
or
Router(config-register-pool)# end

Verifying Basic Calling

To verify that Cisco IP phones in Cisco Unified CME can place and receive calls through the voice ports,
perform the following steps.

SUNNARY STEPS

1. Test local operation.

2. Test local calling area.
3. Test incoming calls.

DETAILED STEPS

Step 1 Test local phone operation. Make calls between phones on the Cisco Unified CME router.
Step 2 Place a call from a phone in Cisco Unified CME to a number in the local calling area.
Step 3 Place a call to a phone in Cisco Unified CME from a phone outside this Cisco Unified CME system.

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Last Updated: March 26, 2007

This chapter describes features that enable Cisco Unified Communications Manager Express
(Cisco Unified CME) to expand or manipulate internal extension numbers so that they conform to
numbering plans used by external systems.

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Dialing Plan Features”
section on page 286.

Contents
• Information About Dialing Plans, page 267
• How to Configure Dialing Plans, page 271
• Configuration Examples for Dialing Plan Features, page 284
• Additional References, page 285
• Feature Information for Dialing Plan Features, page 286

Information About Dialing Plans

To design and configure dialing plans, you should understand the following concepts:
• Phone Number Plan, page 268
• Dial-Plan Patterns, page 269
• Direct Inward Dialing Trunk Lines, page 270
• Voice Translation Rules and Profiles, page 270
• Secondary Dial Tone, page 270

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Phone Number Plan

If you install a Cisco Unified CME system to replace an older telephony system that had an established
telephone number plan, you can retain the old number plan. Cisco Unified CME supports flexible
extension number lengths and can provide automatic conversion between extension dialing and E.164
public telephone number dialing.
When a router receives a voice call, it selects an outbound dial peer by comparing the called number (the
full E.164 telephone number) in the call information with the number configured as the destination
pattern for the POTS dial peer. The router then strips out the left-justified numbers corresponding to the
destination pattern matching the called number. If you have configured a prefix, the prefix will be put
in front of the remaining numbers, creating a dial string, which the router will then dial. If all numbers
in the destination pattern are stripped-out, the user will receive (depending on the attached equipment)
a dial tone.
A successful Cisco Unified 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 Unified CME supports shared lines and 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 “Configuring
Phones to Make Basic Calls” on page 147.
If you are configuring more than one Cisco Unified CME site, you need to decide how calls between the
sites will be handled. Calls between Cisco Unified 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, 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 Unified CME system is assigned which number range.
• You can route calls using a local extension number plus a special prefix for each
Cisco Unified 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 Unified CME
sites. In this case, intersite callers use the PSTN area code and local prefix to route calls between
Cisco Unified CME systems.
If you choose to have a gatekeeper route calls among multiple Cisco Unified 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 Unified CME systems, but you might be
able to overcome this limitation. Cisco Unified 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.
Mapping of public telephone numbers to internal extension numbers is not restricted to simple truncation
of the digit string. Digit substitutions can be made by defining dial-plan patterns to be matched. For
information about dial plans, see the “Dial-Plan Patterns” section on page 269. More sophisticated
number manipulations can be managed with voice translation rules and voice translation profiles, which
are described in the “Voice Translation Rules and Profiles” section on page 270.

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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 Trunk Lines” section on page 270.

Dial-Plan Patterns
A dial-plan pattern enables abbreviated extensions to be expanded into fully qualified E.164 numbers.
Use dial-plan patterns when configuring a network with multiple Cisco Unified CMEs to ensure that the
appropriate calling number, extension or E.164 number, is provided to the target Cisco Unified CME,
and appears on the phone display of the called phone. In networks that have a single router, you do not
need to use dial-plan patterns.
.When you define a directory number for an SCCP phone, the Cisco Unified CME system automatically
creates a POTS dial peer with the ephone-dn endpoint as a destination. For SIP phones connected
directly into Cisco Unified CME, the dial peer is automatically created when the phone registers. By
default, Cisco Unified CME creates a single POTS dial peer for each directory number.
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

A dial-plan pattern builds additional dial peers for the expanded numbers it creates. If a dialplan pattern
is configured and it matches against a directory number, two POTS dial peers are created, one for the
abbreviated number and one for the complete E.164 direct-dial telephone number.
For example, if you then define a dial-plan pattern that 1001 will match, such as 40855500.., a second
dial peer is created so that calls to both the 0001 and 4085550001 numbers are completed. In this
example, the additional dial peer that is automatically created looks like the following:
dial-peer voice 20002 pots
destination-pattern 40855510001
voice-port 50/0/2

In networks with multiple routers, you may need to use dial-plan patterns 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
require E.164 numbers so that all numbers in the network are unique. Define dial-plan patterns to expand
extension numbers into unique E.164 numbers for registering with a gatekeeper.
If multiple dial-plan patterns are defined, the system matches extension numbers against the patterns in
sequential order, starting with the lowest numbered dial-plan pattern tag 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.

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Direct Inward Dialing Trunk Lines

Direct Inward Dialing (DID), is a one-way incoming trunking mechanism, that allows an external caller
to directly reach a specific extension without the call being served by an attendant or other intervention.
It is a service offered in which the last few (typically three or four) digits dialed by the caller are
forwarded to the called party on a special DID trunk. For example, all the phone numbers from 555-0000
to 555-0999 could be assigned to a company with 20 DID trunks. When a caller dials any number in this
range, the call is forwarded on any available trunk. If the caller dialed 555-0234, then the digits 2, 3, and
4 are forwarded. These DID trunks could be terminated on a PBX, so that the extension 234 gets the call
without operator assistance. This makes it look as though 555-0234 and the other 999 lines all have
direct outside lines, while only requiring 20 trunks to service the 1,000 telephone extensions. Using DID,
a company can offer its customers individual phone numbers for each person or workstation within the
company without requiring a physical line into the PBX for each possible connection. Compared to
regular PBX service, DID saves the cost of a switchboard operator. Calls go through faster, and callers
feel they are calling a person rather than a company.
Dial-plan patterns are required to enable calls to DID numbers. When the PSTN connects a DID call for
“4085550234” to the Cisco Unified CME system, it also forwards the extension digits “234” to allow
the system to route the call.

Voice Translation Rules and Profiles

Voice translation rules perform manipulations on numbers. Voice translation profiles allow you to group
voice translation rules together and associate them with the following:
• Called numbers
• Calling numbers
• Redirected called numbers
Voice translation rules have the ability to perform regular expression matches and replace substrings.
The Stream Editor (SED) utility is used to translate numbers. The translation rules replace a substring
of the input number if the number matches the match pattern, number plan, and type present in the rule.
The SED utility is used to check for a match based on the match pattern.
For examples of voice translation rules and profiles, see the “Voice Translation Rules” technical note
and the “Number Translation using Voice Translation Profiles” technical note.

Secondary Dial Tone

A secondary dial tone is available for Cisco Unified IP phones connected to Cisco Unified 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 a PSTN access
prefix, such as the number 9, is dialed to reach an outside line. For configuration information, see the
“Activating a Secondary Dial Tone” section on page 283.

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How to Configure Dialing Plans

This section contains the following tasks:

Dial-Plan Patterns
• SCCP: Configuring Dial-Plan Patterns, page 271 (required)
• SIP: Configuring Dial-Plan Patterns, page 272 (required)
• Verifying Dial-Plan Patterns, page 274 (optional)

Voice Translation Rules

• Defining Voice Translation Rules in Cisco CME 3.2 and Later Versions, page 275 (required)
• SCCP: Applying Voice Translation Rules in Cisco CME 3.2 and Later Versions, page 277
(required)
• SCCP: Applying Translation Rules Before Cisco CME 3.2, page 278 (required)
• SIP: Applying Voice Translation Rules in Cisco Unified CME 4.1 and Later, page 280 (required)
• SIP: Applying Voice Translation Rules before Cisco Unified CME 4.1, page 281 (required)
• Verifying Voice Translation Rules and Profiles, page 282 (optional)

Secondary Dial Tone

• Activating a Secondary Dial Tone, page 283 (optional)

SCCP: Configuring Dial-Plan Patterns

To define a dial-plan pattern, perform the following steps.

Tip In networks that have a single router, you do not need to define dial-plan patterns.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. dialplan-pattern tag pattern extension-length extension-length [extension-pattern
extension-pattern | no-reg]
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 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:
Router(config-telephony)# dialplan-pattern 1
4085550100 extension-length 3 extension-pattern 4..

Note This example maps all extension numbers 4xx to the

PSTN number 40855501xx, so that extension 412
corresponds to 4085550112.
Step 5 end Exits configuration mode and enters privileged
EXEC mode.
Example:
Router(config-telephony)# end

SIP: Configuring Dial-Plan Patterns

To create and apply a pattern for expanding individual abbreviated SIP extensions into fully qualified
E.164 numbers, follow the steps in this section. Dial-plan pattern expansion affects calling numbers and
for call forward using B2BUA, redirecting, including originating and last reroute, numbers for SIP
extensions in Cisco Unified CME.

Prerequisites
Cisco Unified CME 4.0 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global

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4. dialplan-pattern tag pattern extension-length extension-length [extension-pattern

extension-pattern] [no-reg]
5. call-forward system redirecting-expanded
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 dialplan-pattern tag pattern extension-length Defines pattern that is used to expand abbreviated extension
extension-length [extension-pattern numbers of SIP calling numbers in Cisco Unified CME into
extension-pattern | no-reg]
fully qualified E.164 numbers.

Example:
Router(config-register-global)#
dialplan-pattern 1 4085550... extension-length
5
Step 5 call-forward system redirecting-expanded Applies dial-plan pattern expansion globally to redirecting,
including originating and last reroute, numbers for SIP
extensions in Cisco Unified CME for call forward using
Example:
Router(config-register-global)# call-forward
B2BUA.
system redirecting-expanded
Step 6 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end

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Verifying Dial-Plan Patterns

To verify dial-plan pattern configurations, perform the following steps.

SUMMARY STEPS

1. show telephony-service
2. show telephony-service dial-peer
or
show dial-peer summary

DETAILED STEPS

Step 1 show telephony-service

Use this command to verify dial-plan patterns in the configuration.
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..

Step 2 SCCP: show telephony-service dial-peer

or
SIP: show dial-peer summary
Use the command to display dial peers that are automatically created by the dialplan-pattern command.
Use this command display the configuration for all VoIP and POTS dial peers configured for a router,
including dial peers created by using the dialplan-expansion (voice register) command.
The following example is output from the show dial-peer summary command displaying information
for four dial peers, one each for extensions 60001 and 60002 and because the dialplan-expansion
command is configured to expand 6.... to 4085555...., one each for 4085550001 and 4085550002. The
latter two dial peers will not appear in the running configuration.
Router# show dial-peer summary
AD PRE PASS OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STATT
20010 pots up up 60002$ 0 0
20011 pots up up 60001$ 0 9
20012 pots up up 5105555001$ 0 9
20013 pots up up 5105555002$ 0 0

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Defining Voice Translation Rules in Cisco CME 3.2 and Later Versions
To define voice translation rules and voice translation profiles, perform the following steps.

Note To configure translation rules for voice calls in Cisco CME 3.1 and earlier versions, see the Cisco IOS
Voice, Video, and FAX Configuration Guide.

Prerequisites
Cisco CME 3.2 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice translation-rule number
4. rule precedence /match-pattern/ /replace-pattern/
5. exit
6. voice translation-profile name
7. translate {called | calling | redirect-called} voice-translation-rule-tag
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice translation-rule number Defines a translation rule for voice calls and enters voice
translation-rule configuration mode.
Example: • number—Number that identifies the translation rule.
Router(config)# voice translation-rule 1 Range is 1 to 2147483647.

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Command or Action Purpose

Step 4 rule precedence /match-pattern/ Defines a translation rule.
/replace-pattern/
• precedence—Priority of the translation rule.
Range: 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 5 exit Exits voice translation-rule configuration mode.

Example:
Router(cfg-translation-rule)# exit
Step 6 voice translation-profile name Defines a translation profile for voice calls.
• name—Name of the translation profile. Maximum
Example: length of the voice translation profile name is
Router(config)# voice translation-profile name1 31 alphanumeric characters.
Step 7 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: numbers.
Router(cfg-translation-profile)# translate
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 1 to 2147483647.
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(cfg-translation-profile)# end

What to Do Next
• To apply voice translation profiles to SCCP phones connected to Cisco Unified CME 3.2 or a later
version, see the “SCCP: Applying Voice Translation Rules in Cisco CME 3.2 and Later Versions”
section on page 277.
• To apply voice translation profiles to SIP phones connected to Cisco Unified CME 4.1 or a later
version, see the “SIP: Applying Voice Translation Rules in Cisco Unified CME 4.1 and Later”
section on page 280.
• To apply voice translation profiles to SIP phones connected to Cisco Unified CME 3.4 or
Cisco Unified 4.0(x), see the “SIP: Applying Voice Translation Rules before Cisco Unified CME
4.1” section on page 281.

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SCCP: Applying Voice Translation Rules in Cisco CME 3.2 and Later Versions
To apply a voice translation profile to modify the number dialed by extensions on a SCCP phone,
perform the following steps.

Prerequisites
• Cisco CME 3.2 or a later version.
• Voice translation profile containing voice translation rules to be applied must be already configured.
For configuration information, see the “Defining Voice Translation Rules in Cisco CME 3.2 and
Later Versions” section on page 275.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn tag
4. translation-profile {incoming | outgoing} name
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn tag Enters ephone-dn configuration mode to create an
extension (ephone-dn) for a Cisco Unified 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
1 to the maximum number of ephone-dns allowed on
the router platform. See the CLI help for the maximum
value for this argument.

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Command or Action Purpose

Step 4 translation-profile {incoming | outgoing} name Assigns a translation profile for incoming or outgoing call
legs to or from Cisco Unified IP phones.
Example: • You can use an ephone-dn template to apply this
Router(config-ephone-dn)# translation-profile command to one or more directory numbers. If you use
outgoing name1 an ephone-dn template to apply a command to a
directory number and you also use the same command
in ephone-dn configuration mode for the same
directory number, the value that you set in ephone-dn
configuration mode has priority.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone-dn)# end

What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.

SCCP: Applying Translation Rules Before Cisco CME 3.2

To apply a translation rule to an individual directory number in Cisco CME 3.1 and earlier versions,
perform the following steps.

Prerequisites
Translation rule to be applied must be already configured by using the translation-rule and rule
commands. For configuration information, see the Cisco IOS Voice, Video, and FAX Configuration
Guide.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag
4. translate {called | calling} translation-rule-number
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn tag Enters ephone-dn configuration mode to create directory
number for a Cisco Unified 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 4 translate {called | calling} Specifies rule to be applied to the directory number being
translation-rule-tag configured.
• translation-rule-tag—Reference number of previously
Example: configured translation rule. Range: 1 to 2147483647.
Router(config-ephone-dn)# translate called 1
• You can use an ephone-dn template to apply this
command to one or more directory numbers. If you use
an ephone-dn template to apply a command to a
directory number and you also use the same command
in ephone-dn configuration mode for the same
directory number, the value that you set in ephone-dn
configuration mode has priority.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(cfg-translation-profile)# end

What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.

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SIP: Applying Voice Translation Rules in Cisco Unified CME 4.1 and Later
To apply a voice translation profile for incoming call legs to a directory number on a SIP phone, perform
the following steps.

Prerequisites
• Cisco Unified CME 4.1 or a later version.
• Voice translation profile containing voice translation rules to be applied must be already configured.
For configuration information, see the “Defining Voice Translation Rules in Cisco CME 3.2 and
Later Versions” section on page 275.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register dn dn-tag
4. translation-profile incoming name
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice
port, or a message-waiting indicator (MWI).
Example:
Router(config-register-dn)# ephone-dn 1
Step 4 translation-profile incoming name Assigns a translation profile for incoming call legs to this
directory number.
Example:
Router(config-register-dn)# translation-profile
incoming name1
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone-dn)# end

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What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “SIP: Generating Configuration Profiles for SIP Phones” on page 250.

SIP: Applying Voice Translation Rules before Cisco Unified CME 4.1
To apply an already-configured voice translation rule to modify the number dialed by extensions on a
SIP phone, perform the following steps.

Prerequisites
• Cisco CME 3.4 or a later version.
• Voice translation rule to be applied must be already configured. For configuration information, see
the “Defining Voice Translation Rules in Cisco CME 3.2 and Later Versions” section on page 275.

SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool tag
4. translate-outgoing {called | calling} rule-tag
5. end

DETAILED STEPS

Step 1 enable Enables privileged EXEC mode.

• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for SIP phones.
Example:
Router(config)# voice register pool 3

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Step 4 translate-outgoing {called | calling} rule-tag Specifies an already configured voice translation rule to be
applied to SIP phone being configured.
Example:
Router(config-register-pool)#
translate-outgoing called 1
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end

What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “SIP: Generating Configuration Profiles for SIP Phones” on page 250.

Verifying Voice Translation Rules and Profiles

To verify voice translation profiles, and rules, perform the following steps.

SUMMARY STEPS

1. show voice translation-profile

2. show voice translation-rule
3. test voice translation-rule

DETAILED STEPS

Step 1 show voice translation-profile [name]

This command displays the configuration of one or all translation profiles.
Router# show voice translation-profile profile-8415

Translation Profile: profile-8415

Rule for Calling number: 4
Rule for Called number: 1
Rule for Redirect number: 5
Rule for Redirect-target number: 2

Step 2 show voice translation-rule [number]

This command displays the configuration of one or all translation rules.
Router# show voice 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

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Step 3 test voice translation-rule number

This command enables you to test your translation rules.
Router(config)# voice translation-rule 5
Router(cfg-translation-rule)# rule 1 /201/ /102/
Router(cfg-translation-rule)# exit
Router(config)# 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

Activating a Secondary Dial Tone

To activate a secondary dial tone after a phone user dials the specified number string, perform the
following steps.

Prerequisite
• Cisco CME 3.0 or a later version.
• PSTN access prefix must be configured for outbound dial peer.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. secondary-dialtone digit-string
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 4 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. Typically, the
Router(config-telephony)# secondary-dialtone 9 digit-string is a predefined PSTN access prefix.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Configuration Examples for Dialing Plan Features

This section contains the following example:
• Secondary Dial Tone: Example, page 284

Secondary Dial Tone: Example

telephony-service
fxo hook-flash
load 7910 P00403020214
load 7960-7940 P00305000600
load 7914 S00103020002
load 7905 CP7905040000SCCP040701A
load 7912 CP7912040000SCCP040701A
max-ephones 100
max-dn 500
ip source-address 10.153.233.41 port 2000
max-redirect 20
no service directed-pickup
timeouts ringing 10
system message XYZ Company
voicemail 7189
max-conferences 8 gain -6
moh music-on-hold.au
web admin system name admin1 password admin1
dn-webedit
time-webedit
!
!
!
secondary-dialtone 9
!

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Dialing Plan Features

Feature Information for Dialing Plan Features

Table 11 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 11 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 11 Feature Information for Dialing Plan Features

Cisco Unified CME

Feature Name Versions Feature Information
Dial-Plan Pattern 4.0 Added support for dial-plan pattern expansion for call
forward and call transfer when the forward or transfer-to
target is an individual abbreviated SIP extension or an
extension that appear on a SIP phone.
2.1 Strips leading digit pattern from extension number when
expanding an extension to an E.164 telephone number. The
length of the extension pattern must equal the value
configured for the extension-length argument.
1.0 Adds a prefix to extensions to transform them into E.164
numbers.
Secondary Dial Tone 3.0 Support for secondary dial tone after dialing specified
number string.
Voice Translation Rules 4.1 Added support for voice translation profiles for incoming
call legs to a directory number on a SIP phone.
3.4 Added support for voice translation rules to modify the
number dialed by extensions on a SIP phone.
3.2 Adds, removes, or transforms digits for calls going to or
originating from specified ephone-dns.

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Last Updated: March 26, 2007

This chapter describes the localization support in Cisco Unified Communications Manager Express
(Cisco Unified CME) for languages other than English and network tones and cadences not specific to
the United States.

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Localization Support”
section on page 302.

Contents
• Information About Localization, page 287
• How to Configure Localization Support, page 289
• Configuration Examples for Localization, page 299
• Where to Go Next, page 301
• Additional References, page 301
• Feature Information for Localization Support, page 302

Information About Localization

To configure localization support, you should understand the following concepts:
• System-Defined Locales, page 288
• User-Defined Locales, page 288
• Multiple Locales, page 288

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Information About Localization

System-Defined Locales
Cisco Unified CME provides internal localization support for 12 languages including English and
16 countries including the United States. User locales specify the language to use for text displays;
network locales specify country-specific tones and cadences. View the list of system-defined locales and
their two-letter codes by typing ? after the user-locale and network-locale commands in
telephony-service configuration mode.
For the Cisco Unified IP Phone 7912, 7940, and 7960, the system-defined user locales and network
locales are preloaded into Cisco IOS software.
For the Cisco Unified IP Phone 7911, 7941, 7961, 7970, and 7971, you must download locale files to
support the system-defined locales and store the files in slot 0, flash memory, or on an external TFTP
server. See the “Installing System-Defined Locales for Cisco Unified IP Phone 7911, 7941, 7961, 7970,
and 7971” section on page 289.

User-Defined Locales
You may need to support locales other than the system-defined locales that are predefined in the system.
The user-defined locale feature allows you to specify user and network locales in addition to those that
are predefined in Cisco IOS software and apply them to individual ephones. For example, if your site
has phones that must use the language and tones for Traditional Chinese, which is not one of the
system-defined choices, you must install the locale files for Traditional Chinese.
In Cisco Unified CME 4.0 and later, you can download and install the files to support a particular user
and network locale in flash, slot 0, or an external TFTP server. You cannot install these files in the
system location. These user and network locales can then be assigned to all or some phones.
User-defined language codes for user locales are based on ISO 639 codes, which are available at the
Library of Congress website: http://www.loc.gov/standards/iso639-2/. User-defined country codes for
network locales are based on ISO 3166 codes.
For configuration information, see the “Installing User-Defined Locales” section on page 292.

Multiple Locales
In Cisco Unified CME 4.0 and later, you can specify up to five user and network locales and apply
different locales to individual ephones or groups of ephones using ephone templates. For example, you
can specify French for phones A, B, and C; German for phones D, E, and F; and English for phones G,
H, and I. Only one user and network locale can be applied to each phone.
Each of the five user and network locales that you can define in a multilocale system is identified by a
locale tag. The locale identified by tag 0 is always the default locale, although you can define this default
to be any supported locale. For example, if you define user locale 0 to be JP (Japanese), the default user
locale for all phones is JP. If you do not specify a locale for tag 0, the default is US (United States).
To apply alternative locales to different phones, you must use per-phone configuration files to build
individual configuration files for each phone. The configuration files automatically use the default
user-locale 0 and network-locale 0. You can override these defaults for individual phones by configuring
alternative locale codes and then creating ephone-templates to assign the locales to individual ephones.
For configuration information, see the “Configuring Multiple Locales” section on page 295.

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How to Configure Localization Support

This section contains the following tasks:
• Installing System-Defined Locales for Cisco Unified IP Phone 7911, 7941, 7961, 7970, and 7971,
page 289 (required)
• Installing User-Defined Locales, page 292 (optional)
• Verifying User-Defined Locales, page 294 (optional)
• Configuring Multiple Locales, page 295 (optional)
• Verifying Multiple Locales, page 298 (optional)

Installing System-Defined Locales for Cisco Unified IP Phone 7911, 7941, 7961,
7970, and 7971
To install locale files for system-defined locales, perform the following steps.

Prerequisites
• Cisco Unified CME 4.0(2) or a later version.
• You must create per-phone configuration files as described in the “SCCP: Defining Per-Phone
Configuration Files and Alternate Location” section on page 129.

Restrictions
• Cisco Unified IP Phone 7931G supports United States English only.

Step 1 Go to http://www.cisco.com/cgi-bin/tablebuild.pl/CME-Locale
You must have an account on Cisco.com to access the Software Download Center. 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.
Step 2 Select your version of Cisco Unified CME.
Step 3 Select the TAR file for the locale you want to install. Each TAR file contains locale files for a specific
language and country and uses the following naming convention:
CME-locale-language_country-CMEversion
For example, CME-locale-de_DE-4.0.2-2.0 is German for Germany for Cisco Unified CME 4.0(2).
Step 4 Download the TAR file to a TFTP server that is accessible to the Cisco Unified CME router. Each file
contains all the firmware required for all phone types supported by that version of Cisco Unified CME.
Step 5 Use the archive tar command to extract the files to flash, slot 0, or an external TFTP server.
Router# archive tar /xtract source-url flash:/file-url

For example, to extract the contents of CME-locale-de_DE-4.0.2-2.0.tar from TFTP server 192.168.1.1
to router flash memory, use this command:
Router# archive tar /xtract tftp://192.168.1.1/cme-locale-de_DE-4.0.2-2.0.tar flash:

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Step 6 See Table 12 and Table 13 for a description of the codes used in the filenames and the list of supported
directory names.
Each phone type has a JAR file that uses the following naming convention:
language-phone-sccp.jar
For example, de-td-sccp.jar is for German on the Cisco Unified IP Phone 7970.
Each TAR file also includes the file g3-tones.xml for country-specific network tones and cadences.

Table 12 Phone-Type Codes for Locale JAR Files

Phone Type Phone Code

7906/7911 tc
7941/7961 mk
7970/7971 td

Table 13 System-Defined User and Network Locales

Language User-Locale Country Network-Locale

Language Code Directory Name Code Directory Name
English en English_United_States1 US United_States
English_United_Kingdom UK United_Kingdom
CA Canada
Danish dk Danish_Denmark DK Denmark
Dutch nl Dutch_Netherlands NL Netherlands
French fr French_France FR France
CA Canada
German de German_Germany DE Germany
AT Austria
CH Switzerland
Italian it Italian_Italy IT Italy
2
Japanese jp Japanese_Japan JP Japan
Norwegian no Norwegian_Norway NO Norway
Portuguese pt Portuguese_Portugal PT Portugal
Russian ru Russian_Russia RU Russian_Federation
Spanish es Spanish_Spain ES Spain
Swedish se Swedish_Sweden SE Sweden
1. English for the United States is the default language. You do not need to install the JAR file for U.S. English unless you
assign a different language to a phone and then want to reassign English.
2. Katakana is supported by Cisco Unified IP Phone 7905, 7912, 7940, and 7960. Kanji is supported by Cisco Unified IP Phone
7911, 7941, 7961, 7970, and 7971.

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Step 7 If you store the locale files in flash or slot 0: on the Cisco Unified CME router, create a TFTP alias for
the user locale (text displays) and network locale (tones) using this format:
Router(config)# tftp-server flash:/jar_file alias directory_name/td-sccp.jar
Router(config)# tftp-server flash:/g3-tones.xml alias directory_name/g3-tones.xml

Use the appropriate directory name shown in Table 13 and remove the two-letter language code from
the JAR file name.
For example, the TFTP aliases for German and Germany for the Cisco Unified IP Phone 7970 are:
Router(config)# tftp-server flash:/de-td-sccp.jar alias German_Germany/td-sccp.jar
Router(config)# tftp-server flash:/g3-tones.xml alias Germany/g3-tones.xml

Note On Cisco 3800 series routers, you must include /its in the directory name (flash:/its or slot0:/its). For
example, the TFTP alias for German for the Cisco Unified IP Phone 7970 is:
Router# tftp-server flash:/its/de-td-sccp.jar alias German_Germany/td-sccp.jar

Step 8 If you store the locale files on an external TFTP server, create a directory under the TFTP root directory
for each user and network locale.
Use the appropriate directory name shown in Table 13 and remove the two-letter language code from
the JAR file name.
For example, the user-locale directory for German and the network-locale directory for Germany for the
Cisco Unified IP Phone 7970 are:
TFTP-Root/German_Germany/td-sccp.jar
TFTP-Root/Germany/g3-tones.xml
Step 9 For Russian and Japanese, you must copy the UTF8 dictionary file into flash to use special phrases.
• Only flash can be used for these locales. Copy russian_tags_utf8_phrases for Russian;
Japanese_tags_utf8_phrases for Japanese.
• Use the user-locale jp and user-locale ru command to load the UTF8 phrases into
Cisco Unified CME.
Step 10 Assign the locales to phones. To set a default locale for all phones, use the user-locale and
network-locale commands in telephony-service configuration mode.
Step 11 To support more than one user or network locale, see the “Configuring Multiple Locales” section on
page 295.
Step 12 Use the create cnf-files command to rebuild the configuration files.
Step 13 Use the reset command to reset the phones and see the localized displays.

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Installing User-Defined Locales

You must download XML files for locales that are not predefined in the system. To install up to five
user-defined locale files to use with phones, perform the following steps.

Prerequisites
• Cisco Unified CME 4.0(3) or a later version.
• You must create per-phone configuration files as described in the “SCCP: Defining Per-Phone
Configuration Files and Alternate Location” section on page 129.

Restrictions
• User-defined locales are not supported on the Cisco Unified IP Phone 7920 or 7936.
• User-defined locales are not supported if the configuration file location is system.
• When you use the setup tool from the telephony-service setup command to provision phones, you
can only choose a default user locale and network locale, and you are limited to selecting a locale
code that is supported in the system. You cannot use multiple locales or user-defined locales with
the setup tool.
• When using a user-defined locale, the phone normally displays text using the user-defined fonts,
except for any strings that are interpreted by Cisco Unified CME, such as “Cisco/Personal
Directory,” “Speed Dial/Fast Dial,” and so forth.

Step 1 Go to http://www.cisco.com/cgi-bin/tablebuild.pl/CME-Locale
Step 2 Select your version of Cisco Unified CME.
Step 3 Select the TAR file for the locale that you want to install. Each TAR file contains locale files for a
specific language and country and uses the following naming convention:
CME-locale-language_country-CMEversion-fileversion
For example, CME-locale-zh_CN-4.0.3-2.0 is Traditional Chinese for China for
Cisco Unified CME 4.0(3).
Step 4 Download the TAR file to a TFTP server that is accessible to the Cisco Unified CME router. Each file
contains all the firmware required for all phone types supported by that version of Cisco Unified CME.
Step 5 Use the archive tar command to extract the files to slot 0, flash, or an external TFTP server.
Router# archive tar /xtract source-url flash:/file-url

For example, to extract the contents of CME-locale-zh_CN-4.0.3-2.0.tar from TFTP server 192.168.1.1
to router flash memory, use this command:
Router# archive tar /xtract tftp://192.168.1.1/cme-locale-zh_CN-4.0.3-2.0.tar flash:

Step 6 For Cisco Unified IP Phones 7905, 7912, 7940, or 7960, go to Step 11.
For Cisco Unified IP Phones 7911, 7941, 7961, 7970, or 7971, go to Step 7.
Step 7 Each phone type has a JAR file that uses the following naming convention:
language-type-sccp.jar
For example, zh-td-sccp.jar is Traditional Chinese for the Cisco Unified IP Phone 7970.
See Table 14 and Table 15 for a description of the codes used in the filenames.

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Table 14 Phone-Type Codes for Locale Files

Phone Type Code

7906/7911 tc
7941/7961 mk
7970/7971 td

Table 15 Language Codes for User-Defined Locales

Language Language Code

Bulgarian bg
Chinese zh
Finnish fi
Hungarian hu
Korean ko
Polish pl

Step 8 If you store the locale files in flash or slot 0: on the Cisco Unified CME router, create a TFTP alias using
this format:
Router(config)# tftp-server flash:/jar_file alias directory_name/td-sccp.jar

Remove the two-letter language code from the JAR filename and use one of five supported directory
names with the following convention:
user_define_number, where number is 1 to 5
For example, the alias for Chinese on the Cisco Unified IP Phone 7970 is:
Router(config)# tftp-server flash:/zh-td-sccp.jar alias user_define_1/td-sccp.jar

Note On Cisco 3800 series routers, you must include /its in the directory name (flash:/its or slot0:/its). For
example, the TFTP alias for Chinese for the Cisco Unified IP Phone 7970 is:
Router(config)# tftp-server flash:/its/zh-td-sccp.jar alias user_define_1/td-sccp.jar

Step 9 If you store the locale files on an external TFTP server, create a directory under the TFTP root directory
for each locale.
Remove the two-letter language code from the JAR filename and use one of five supported directory
names with the following convention:
user_define_number, where number is 1 to 5
For example, for Chinese on the Cisco Unified IP Phone 7970, remove “zh” from the JAR filename and
create the “user_define_1” directory under TFTP-Root on the TFTP server:
TFTP-Root/user_define_1/td-sccp.jar
Step 10 Go to Step 14.
Step 11 Download one or more of the following XML files depending on your selected locale and phone type.
All required files are included in the JAR file.

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7905-dictionary.xml
7905-font.xml
7905-kate.xml
7920-dictionary.xml
7960-dictionary.xml
7960-font.xml
7960-kate.xml
7960-tones.xml
SCCP-dictionary.utf-8.xml
SCCP-dictionary.xml

Step 12 Rename these files and copy them to flash, slot 0, or an external TFTP server. Rename the files using
the format user_define_number_filename where number is 1 to 5. For example, use the following names
if you are setting up the first user-locale:
user_define_1_7905-dictionary.xml
user_define_1_7905-font.xml
user_define_1_7905-kate.xml
user_define_1_7920-dictionary.xml
user_define_1_7960-dictionary.xml
user_define_1_7960-font.xml
user_define_1_7960-kate.xml
user_define_1_7960-tones.xml
user_define_1_SCCP-dictionary.utf-8.xml
user_define_1_SCCP-dictionary.xml

Step 13 If you store the locale files in flash or slot 0: on the Cisco Unified CME router, create a TFTP alias, for
example:
tftp-server flash:user_define_1_7905-dictionary.xml
tftp-server flash:user_define_1_7905-font.xml
tftp-server flash:user_define_1_7905-kate.xml
tftp-server flash:user_define_1_7960-tones.xml
tftp-server flash:user_define_1_7960-dictionary.xml
tftp-server flash:user_define_1_7960-font.xml
tftp-server flash:user_define_1_7960-kate.xml
tftp-server flash:user_define_1_SCCP-dictionary.utf-8.xml
tftp-server flash:user_define_1_SCCP-dictionary.xml

Step 14 Copy the language_tags_file and language_utf8_tags_file to the location of the other locale files (flash,
slot 0, or TFTP server). Rename the files to user_define_number_tags_file and
user_define_number_utf8_tags_file respectively, where number is 1 to 5 and matches the user-defined
directory.
Step 15 Assign the locales to phones. See the “Configuring Multiple Locales” section on page 295.
Step 16 Use the create cnf-files command to rebuild the configuration files.
Step 17 Use the reset command to reset the phones and see the localized displays.

Verifying User-Defined Locales

See the “Verifying Multiple Locales” section on page 298.

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Configuring Multiple Locales

To define one or more alternatives to the default user and network locales, and apply them to individual
phones, perform the following steps.

Prerequisites
• Cisco Unified CME 4.0 or a later version.
• To specify alternative user and network locales for individual phones in a Cisco Unified CME
system, you must use per-phone configuration files. For more information, see the “SCCP: Defining
Per-Phone Configuration Files and Alternate Location” section on page 129.
• You can also use user-defined locale codes as alternative locales after you download the appropriate
XML files. See the“Installing User-Defined Locales” section on page 292.

Restrictions
• Multiple user and network locales are not supported on the Cisco Unified IP Phone 7902G, 7910,
7910G, or 7920, or the Cisco Unified IP Conference Station 7935 and 7936.
• When you use the setup tool from the telephony-service setup command to provision phones, you
can only choose a default user locale and network locale, and you must select a locale code that is
predefined in the system. You cannot use multiple or user-defined locales with the setup tool.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. user-locale user-locale-tag [user-defined-code] language-code
5. network-locale network-locale-tag [user-defined-code] country-code
6. create cnf-files
7. exit
8. ephone-template template-tag
9. user-locale user-locale-tag
10. network-locale network-locale-tag
11. exit
12. ephone phone-tag
13. ephone-template template-tag
14. exit
15. telephony service
16. reset {all [time-interval] | cancel | mac-address mac-address | sequence-all}
17. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 user-locale user-locale-tag [user-defined-code] Specifies a language for phone displays.
language-code
• user-locale-tag—Assigns a locale identifier to the
language code. Range is 0 to 4. This argument is
Example: required when using multiple locales; otherwise the
Router(config-telephony)# user-locale 1 U1 ZH specified language is the default applied to all phones.
• user-defined-code—(Optional) Assigns one of the
user-defined codes to the specified language code.
Valid codes are U1, U2, U3, U4, and U5.
• language-code—Type ? to display a list of
system-defined codes. United States (US) is the default.
You can assign any valid ISO 639 code to a
user-defined code (U1 to U5).
Step 5 network-locale network-locale-tag Specifies a country for tones and cadences.
[user-defined-code] country-code
• network-locale-tag—Assigns a locale identifier to the
country code. Range is 0 to 4. This argument is required
Example: when using multiple locales; otherwise the specified
Router(config-telephony)# network-locale 1 FR country is the default applied to all phones.
• user-defined-code—(Optional) Assigns one of the
user-defined codes to the specified country code. Valid
codes are U1, U2, U3, U4, and U5.
• country-code—Type ? to display a list of
system-defined codes. United States (US) is the default.
You can assign any valid ISO 3166 code to a
user-defined code (U1 to U5).
Step 6 create cnf-files Builds the required XML configuration files for IP phones.
Use this command after you update configuration file
parameters such as the user locale or network locale.
Example:
Router(config-telephony)# create cnf-files

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Command or Action Purpose

Step 7 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 8 ephone-template template-tag Enters ephone-template configuration mode.
• template-tag—Unique sequence number that identifies
Example: this template during configuration tasks.
Router(config)# ephone template 1
Step 9 user-locale user-locale-tag Assigns a user locale to this ephone template.
• user-locale-tag—A locale tag that was created in
Example: Step 4. Range is 0 to 4.
Router(config-ephone-template)# user-locale 2
Step 10 network-locale network-locale-tag Assigns a network locale to this ephone template.
• network-locale-tag—A locale tag that was created in
Example: Step 5. Range is 0 to 4.
Router(config-ephone-template)#
network-locale 2
Step 11 exit Exits ephone-template configuration mode.

Example:
Router(config-ephone-template)# exit
Step 12 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: this ephone during configuration tasks.
Router(config)# ephone 36
Step 13 ephone-template template-tag Applies an ephone template to an ephone.
• template-tag—Number of the template to apply to this
Example: ephone.
Router(config-ephone)# ephone-template 1
Step 14 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit
Step 15 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 16 reset {all [time-interval] | cancel | Performs a complete reboot of all phones or the specified
mac-address mac-address | sequence-all} phone, including contacting the DHCP and TFTP servers
for the latest configuration information.
Example: • all—All phones in the Cisco Unified CME system.
Router(config-telephony)# reset all
• time-interval—(Optional) Time interval, in seconds,
between each phone reset. Range is 0 to 60. Default
is 15.
• cancel—Interrupts a sequential reset cycle that was
started with a reset sequence-all command.
• mac-address mac-address—A specific phone.
• 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.
Step 17 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Verifying Multiple Locales

Step 1 Use the show telephony-service tftp-bindings command to display a list of configuration files that are
accessible to IP phones using TFTP, including the dictionary, language, and tone configuration files.
Router(config)# 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 2 Ensure that per-phone configuration files are defined with the cnf-file perphone command.
Step 3 Use the show telephony-service ephone-template command to check the user locale and network
locale settings in each ephone template.
Step 4 Use the show telephony-service ephone command to check that the correct templates are applied to
phones.

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Step 5 Use the debug tftp events command to see which files Cisco Unified CME is looking for and whether
the files are found and opened correctly. There are usually three states (“looking for x file” “opened x
file” and “finished x file”). The file is found when all three states are displayed. Use this command if
the configuration file location is not TFTP. For an external TFTP server you can use the logs from the
TFTP server.

Configuration Examples for Localization

This section contains the following examples:
• Multiple User and Network Locales: Example, page 299
• User-Defined Locales: Example, page 300

Multiple User and Network Locales: Example

The following example sets the default locale of 0 to Germany, which defines Germany as the default
user and network locale. Germany is used for all phones unless you apply a different locale to individual
phones using ephone templates.
telephony service
cnf-file location flash:
cnf-file perphone
user-locale 0 DE
network-locale 0 DE

After using the previous commands to define Germany as the default user and network locale, use the
following commands to return the default value of 0 to US:
telephony service
no user-locale 0 DE
no network-locale 0 DE

Another way to define Germany as the default user and network locale is to use the following commands:
telephony service
cnf-file location flash:
cnf-file perphone
user-locale DE
network-locale DE

After using the previous commands, use the following commands to return the default to US:
telephony service
no user-locale DE
no network-locale DE

The following example defines three alternative locales: JP (Japan), FR (France), and ES (Spain). The
default is US for all phones that do not have an alternative applied using ephone templates. In this
example, ephone 11 uses JP for its locales, ephone 12 uses FR, ephone 13 uses ES, and ephone 14 uses
the default, US.
telephony-service
cnf-file location flash:
cnf-file perphone
create cnf-files
user-locale 1 JP

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user-locale 2 FR
user-locale 3 ES
network-locale 1 JP
network-locale 2 FR
network-locale 3 ES
create cnf-files

ephone-template 1
user-locale 1
network-locale 1

ephone-template 2
user-locale 2
network-locale 2

ephone-template 3
user-locale 3
network-locale 3

ephone 11
button 1:25
ephone-template 1

ephone 12
button 1:26
ephone-template 2

ephone 13
button 1:27
ephone-template 3

ephone 14
button 1:28

User-Defined Locales: Example

The following example shows user-locale tag 1 assigned to code U1, which is defined as ZH for
Traditional Chinese. Traditional Chinese is not predefined in the system so you must download the
appropriate XML files to support this language.
In this example, ephone 11 uses Traditional Chinese (ZH) and ephone 12 uses the default, US English.
The default is US English for all phones that do not have an alternative applied using ephone templates.
telephony-service
cnf-file location flash:
cnf-file perphone
user-locale 1 U1 ZH
network-locale 1 U1 CN

ephone-template 2
user-locale 1
network-locale 1

ephone 11
button 1:25
ephone-template 2

ephone 12
button 1:26

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Where to Go Next
Ephone Templates
For more information about ephone templates, see “Creating Templates” on page 881.

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Localization Support

Table 16 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 16 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 16 Feature Information for Localization Support

Cisco Unified
Feature Name CME Version Feature Information
Multiple Locales 4.0 Support for multiple user and network locales was
introduced.
User-Defined Locales 4.0 User-defined locales were introduced.

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Last Updated: June 18, 2007

This chapter describes the transcoding support available in Cisco Unified Communications Manager
Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Transcoding
Resources” section on page 337.

Contents
• Prerequisites for Transcoding Resources, page 303
• Restrictions for Transcoding Resources, page 304
• Information About Transcoding Resources, page 304
• How to Configure Transcoding Resources, page 306
• Configuration Examples for Transcoding Resources, page 333
• Where to go Next, page 335
• Additional References, page 335
• Feature Information for Transcoding Resources, page 337

Prerequisites for Transcoding Resources

• Cisco Unified CME 3.2 or a later version.
• Cisco Unified CME routers and external voice routers on the same LAN must be configured with
digital signal processors (DSPs) that support transcoding.
• 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.

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Restrictions for Transcoding Resources

• Versions earlier than Cisco CME 3.2 support only G.729 for two-party voice calls.
• Transcoding between G.711 and G.729 does not support the following:
– Meet-me conferencing
– Multiple-party conferencing
– Transcoding security

Information About Transcoding Resources

To configure transcoding support, you should understand the following concepts:
• Transcoding Support, page 304
• Transcoding When a Remote Phone Uses G.729r8, page 305
• Secure DSP Farm Transcoding, page 306

Transcoding Support
Transcoding compresses and decompresses voice streams to match endpoint-device capabilities.
Transcoding is required when an incoming voice stream is digitized and compressed (by means of a
codec) to save bandwidth, and the local device does not support that type of compression.
Cisco CME 3.2 and later versions support transcoding between G.711 and G.729 codecs for the
following features:
• Ad hoc conferencing—One or more remote conferencing parties uses 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.
• Cisco Unity Express—An H.323 or SIP call using G.729 is forwarded to Cisco Unity Express.
Cisco Unity Express supports only G.711, so G.729 must be transcoded. See the
Cisco Unity Express documentation at
www.cisco.com/en/US/products/sw/voicesw/ps5520/tsd_products_support_series_home.html
• Music on hold (MOH)—The phone receiving MOH is part of a system that uses G.729. The G.711
MOH is transcoded into G.729 resulting in a poorer quality sound due to the lower compression of
G.729.
Figure 14 provides an example of each of the four call situations described.

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Figure 14 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

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Phone A calls phone B.
Phone B answers and places phone A on hold.

Transcoding When a Remote Phone Uses G.729r8

A situation in which transcoding resources may be used is when you use the codec command to select
the G.729r8 codec to help save network bandwidth for a remote IP phone. If a conference is initiated, all
phones in the conference switch to G.711 mu-law. To allow the phone to retain its G.729r8 codec setting
when joined to a conference, you can use the codec g729r8 dspfarm-assist command to specify that this
phone’s calls should use the resources of a DSP farm for transcoding. For example, there are two remote
phones (A and B) and a local phone (C) that initiates a conference with them. Both A and B are
configured to use the G.729r8 codec with the assistance of the DSP-farm transcoder. In the conference,
the call leg from C to the conference uses the G.711 mu-law codec, and the call legs from A and B to the
Cisco Unified CME router use the G.729r8 codec.
Consider your options carefully when deciding to use the codec g729r8 dspfarm-assist command. The
benefit is that it allows calls to use the G.729r8 codec on the call leg between the IP phone and the
Cisco Unified CME router, which saves network bandwidth. The disadvantage is that for situations
requiring G.711 codecs, such as conferencing and Cisco Unity Express, DSP resources that are possibly
scarce are used to transcode the call, and delay is introduced while voice is shuttled to and from the DSP.
In addition, the overuse of this feature can mask configuration errors in the codec selection mechanisms
involving dial peers and codec lists.
Therefore, we recommend using the codec g729r8 dspfarm-assist command sparingly and only when
absolutely required for bandwidth savings or when you know the phone will be participating very little,
if at all, in calls that require a G.711 codec.
Because of how Cisco Unified CME uses voice channels with Skinny Client Control Protocol (SCCP)
endpoints, you must configure at least two available transcoding sessions when establishing a call that
requires transcoding configured with the codec g729r8 dspfarm-assist command. Only one session is

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used after the voice path is established with transcoding. However, during the SCCP manipulations, a
temporary session may be allocated. If this temporary session cannot be allocated, the transcoding
request is not honored, and the call continues with the G.711 codec.
If the codec g729r8 dspfarm-assist command is configured for a phone and a DSP resource is not
available when needed for transcoding, a phone registered to the local Cisco Unified CME router will
use G.711 instead of G.729r8. This is not true for nonSCCP call legs; if DSP resources are not available
for the transcoding required for a conference, for example, the conference is not created.

Secure DSP Farm Transcoding

Cisco Unified CME uses the secure transcoding DSP farm capability only in the case described in the
“Transcoding When a Remote Phone Uses G.729r8” section on page 305. If a call using the codec
g729r8 dspfarm-assist command is secure, Cisco Unified CME looks for a secure transcoding resource.
If it cannot find one, transcoding is not done. If the call is not secure, Cisco Unified CME looks for a
nonsecure transcoding resource. If it cannot find one, Cisco Unified CME looks for a secure transcoding
resource. Even if Cisco Unified CME uses a secure transcoding resource, the call is not secure, and a
more expensive secure DSP Farm resource is not needed for a nonsecure call because
Cisco Unified CME cannot find a less expensive nonsecure transcoder.

How to Configure Transcoding Resources

This section contains the following tasks:
• Determining DSP Resources for Transcoding, page 306 (required)
• Provisioning NMs or NM Farms for Transcoding, page 309 (required)
• Configuring DSP Farms for NM-HDs and NM-HDV2s, page 310 (required)
• Configuring DSP Farms for NM-HDVs, page 314 (required)
• Modifying the Number of Transcoding Sessions for NM-HDVs, page 316 (optional)
• Configuring the Cisco Unified CME Router to Act as the DSP Farm Host, page 317 (optional)
• Registering the DSP Farm with Cisco Unified CME in Secure Mode, page 320 (optional)
• Verifying DSP Farm Operation, page 329 (optional)
• Tuning DSP Farm Performance, page 332 (optional)

Determining DSP Resources for Transcoding

Transcoding is facilitated through DSPs, which are located in network modules. All network modules
have single inline memory module (SIMM) sockets or packet voice/data modules (PVDM) slots that
each hold a Packet Voice DSP Module (PVDM). Each PVDM holds DSPs. A router can have multiple
network modules.
Figure 15 shows an NM-HDV with 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.

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Figure 15 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

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PVDM slots
or SIMM socket

Use DSP resources to provide voice termination of the digital voice trunk group or resources for a DSP
farm. DSP resources available for transcoding and not used for voice termination are referred to as a DSP
farm. Figure 16 shows a DSP farm managed by Cisco Unified CME.

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Figure 16 DSP Farm

DSP = Transcoding
DSP DSP DSP
DSP = Voice termination

DSP DSP DSP

DSP farm

DSP DSP DSP

DSP DSP DSP

DSP DSP DSP

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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.

To determine how many DSP voice resources are on your Cisco Unified 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. For more information about these commands, see the
Cisco Unified Communications Manager Express Command Reference.
For information on determining if your router has the correct DSP allocation for transcoding, see the
“Allocation of DSP Resources” section in the “Configuring Enhanced Conferencing and Transcoding for
Voice Gateway Routers” chapter of the Cisco Unified Communications Manager and
Cisco IOS Interoperability Guide.

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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.

Step 2 Determine the number of transcoding sessions that your router must support.
Step 3 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 Unified
Communications Manager and Cisco IOS Interoperability Guide, determine the number of DSPs that
are required to support the transcoding sessions. Note that Cisco Unified CME does not support
DSP-farm conferencing, so only the transcoding portion of this discussion applies to
Cisco Unified 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 4 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 5 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 6 Install hardware to prepare your system for DSP-farm configuration.
Install PVDMs, NMs, and NM farms as needed.

What to Do Next
Choose from one of the following options:
• To set up DSP farms for NM-HDVs, see the “Configuring DSP Farms for NM-HDVs” section on
page 314.
• To set up DSP farms on NM-HDs and NM-HDV2s, see the “Modifying the Number of Transcoding
Sessions for NM-HDVs” section on page 316.

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Configuring DSP Farms for NM-HDs and NM-HDV2s

To configure DSP farms for NM-HDs or NM-HDV2s and to configure secure transcoding profiles,
perform the following procedure.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice-card slot
4. dsp services dspfarm
5. exit
6. sccp local interface-type interface-number
7. sccp ccm ip-address identifier identifier-number
8. sccp
9. sccp ccm group group-number
10. bind interface interface-type interface-number
11. associate ccm identifier-number priority
12. associate profile profile-identifier register device-name
13. keepalive retries number
14. switchover method {graceful | immediate}
15. switchback method {graceful | guard timeout-guard-value | immediate | uptime
uptime-timeout-value}
16. switchback interval seconds
17. exit
18. dspfarm profile profile-identifier transcode [security]
19. trustpoint trustpoint-label
20. codec codec-type
21. maximum sessions number
22. associate application sccp
23. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 4 dsp services dspfarm Enables DSP-farm services on the NM-HDV or NM-HDV
farm.
Example:
Router(config-voicecard)# dsp services dspfarm
Step 5 exit Exits voice-card configuration mode.

Example:
Router(config-voicecard)# exit
Step 6 sccp local interface-type interface-number Selects the local interface that the SCCP applications
(transcoding and conferencing) should use to register with
Cisco Unified CME.
Example:
Router(config)# sccp local FastEthernet 0/0 • interface-type—Interface type that the SCCP
application uses to register with Cisco Unified 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 Unified CME.
Step 7 sccp ccm ip-address identifier Specifies the Cisco Unified CME address.
identifier-number
• ip-address—IP address of the Cisco Unified CME
server.
Example:
Router(config)# sccp ccm 10.10.10.1 priority 2
• identifier identifier-number—Identifier used to
associate the SCCP Cisco Unified CME IP address
with a Cisco Unified CME group. See the associate
ccm command in Step 11.
• Repeat this step to specify the address of a secondary
Cisco Unified CME server.
Step 8 sccp Enables SCCP and its associated transcoding and
conferencing applications.
Example:
Router(config)# sccp

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Command or Action Purpose

Step 9 sccp ccm group group-number Creates a Cisco Unified CME group and enters SCCP
configuration mode for Cisco Unified CME.
Example: • group-number—Number that identifies the
Router(config)# sccp ccm group 1 Cisco Unified CME group. Range is 1 to 65535. There
is no default value.
Note A Cisco Unified CME group is a naming device
under which data for the DSP farms is declared.
Only one group is required. For the
Cisco Unified 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.
Step 10 bind interface interface-type interface-number (Optional) Binds an interface to a Cisco Unified CME
group so that the selected interface is used for all calls that
belong to the profiles that are associated to this
Example:
Router(config-sccp-ccm)# bind interface
Cisco Unified CME group. This command is optional, but
FastEthernet 0/0 we recommend it if you have more than one profile or if you
are on different subnets, to ensure that the correct interface
is selected.
Step 11 associate ccm identifier-number priority Associates a Cisco Unified CME with a group and
establishes its priority within the group.
Example: • identifier-number—Number that identifies
Router(config-sccp-ccm)# associate ccm 1 Cisco Unified CME. Range is 1 to 65535. There is no
priority default value.
• priority—The priority of the Cisco Unified CME
router in the Cisco Unified CME group. The default
is 1 because only one Cisco Unified CME group is
possible.
• Repeat this step to associate a secondary
Cisco Unified CME server with a group.
Step 12 associate profile profile-identifier register Associates a DSP farm profile with a Cisco Unified CME
device-name group.
• profile-identifier—Number that identifies the DSP farm
Example: profile. Range is 1 to 65535. There is no default value.
Router(config-sccp-ccm)# associate profile 1
register mtp000a8eaca80 • register device-name—User-specified device name in
Cisco Unified CME. The device-name must use the
format of mtp mac-address, where the mac-address is
the burnt-in address of the physical interface that is
used to register as the SCCP device.
Step 13 keepalive retries number Sets the number of keepalive retries from SCCP to
Cisco Unified CME.
Example: • number—Number of keepalive attempts. Range is
Router(config-sccp-ccm)# keepalive retries 5 1 to 32. The default is 3.

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Command or Action Purpose

Step 14 switchover method [graceful | immediate] Sets the switchover method that the SCCP client uses when
its communication link to the active Cisco Unified CME
system goes down.
Example:
Router(config-sccp-ccm)# switchover method • graceful—Switchover happens only after all the active
immediate sessions have been terminated gracefully.
• immediate—Switches over to any one of the secondary
Cisco Unified CME systems immediately.
Step 15 switchback method {graceful | guard Sets the switch back method that the SCCP client uses when
timeout-guard-value | immediate | uptime the primary or higher priority Cisco Unified CME becomes
uptime-timeout-value}
available again.
• 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
60 to 172800. Default is 7200.
• immediate—Switches back to the higher order
Cisco Unified CME immediately when the timer
expires, whether there is an active connection or not.
• uptime uptime-timeout-value—Initiates the uptime
timer when the higher-order Cisco Unified CME
system comes alive. Timeout value is in seconds. Range
is 60 to 172800. Default is 7200.
Step 16 switchback interval seconds Sets the amount of time that the DSP farm waits before
polling the primary Cisco Unified CME system when the
current Cisco Unified 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 17 exit Exits SCCP configuration mode.

Example:
Router(config-sccp-ccm)# exit
Step 18 dspfarm profile profile-identifier transcode Enters DSP farm profile configuration mode and defines a
[security] profile for DSP farm services.
• profile-identifier—Number that uniquely identifies a
Example: profile. Range is 1 to 65535. There is no default.
Router(config)# dspfarm profile 1 transcode
security • transcode—Enables profile for transcoding.
• security—Enables profile for secure DSP farm
services.
Step 19 trustpoint trustpoint-label (Optional) Associates a trustpoint with a DSP farm profile.

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

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Command or Action Purpose

Step 20 codec codec-type Specifies the codecs supported by a DSP farm profile.
• codec-type—Specifies the preferred codec.
Example: • Use CLI help to locate a list of codecs.
Router(config-dspfarm-profile)# codec g711ulaw
• Repeat this step as necessary to specify all the
supported codecs.
Step 21 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.
Step 22 associate application sccp Associates SCCP with the DSP farm profile.

Example:
Router(config-dspfarm-profile)# associate
application sccp
Step 23 end Returns to privileged EXEC mode.

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

Configuring DSP Farms for NM-HDVs

To configure DSP farms for NM-HDVs, perform the following steps.

SUMMARY STEPS

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

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 4 dsp services dspfarm Enables DSP-farm services on the NM-HDV or NM-HDV
farm.
Example:
Router(config-voicecard)# dsp services dspfarm
Step 5 exit Returns to global configuration mode.

Example:
Router(config-voicecard)# exit
Step 6 sccp local interface-type interface-number Selects the local interface that the SCCP applications
(transcoding and conferencing) should use to register with
Cisco Unified CME.
Example:
Router(config)# sccp local FastEthernet 0/0 • interface-type—Interface type that the SCCP
application uses to register with Cisco Unified 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 Unified CME.
Step 7 sccp ccm ip-address priority priority-number Specifies the Cisco Unified CME address.
• ip-address—IP address of the Cisco Unified CME
Example: server.
Router(config)# sccp ccm 10.10.10.1 priority 1
• priority priority—Priority of the Cisco Unified CME
server relative to other connected servers. Range is
1 (highest) to 4 (lowest).
Step 8 sccp Enables SCCP and its associated transcoding and
conferencing applications.
Example:
Router(config)# sccp

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Command or Action Purpose

Step 9 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 10 dspfarm Enables the DSP farm.

Example:
Router(config)# dspfarm
Step 11 end Returns to privileged EXEC mode.

Example:
Router(config)# end

Modifying the Number of Transcoding Sessions for NM-HDVs

To modify the maximum number of transcoding sessions for NM-HDVs, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. no dspfarm
4. dspfarm transcoder maximum sessions number
5. dspfarm
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 no dspfarm Disables the DSP farm.

Example:
Router(config)# no dspfarm
Step 4 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 5 dspfarm Enables the DSP farm.

Example:
Router(config)# dspfarm
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config)# end

Configuring the Cisco Unified CME Router to Act as the DSP Farm Host
To configure the Cisco Unified CME router to act as the DSP farm host, perform the following tasks.
• Determining the Maximum Number of Transcoder Sessions, page 317
• Setting the Cisco Unified CME Router to Receive IP Phone Messages, page 318
• Configuring the Cisco Unified CME Router to Host a Secure DSP Farm, page 320

Determining the Maximum Number of Transcoder Sessions

To determine the maximum number of transcoder sessions that can occur at one time perform the
following steps.

SUMMARY STEPS

1. dspfarm transcoder maximum sessions

2. show sdspfarm sessions
3. show sdspfarm units
4. Determine maximum number of transcoder sessions based on values in steps 2 and 3.

DETAILED STEPS

Step 1 Use the dspfarm transcoder maximum sessions command to set the maximum number of transcoder
sessions you have configured.
Step 2 Use the show sdspfarm sessions command to display the number of transcoder sessions.
Step 3 Use the show sdspfarm units command to display the number of DSP farms.

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Step 4 Obtain the maximum number of transcoder sessions by multiplying the number of transcoder sessions
from Step 2 (configured in Step 1 using the dspfarm transcoder maximum sessions command) by the
number of DSP farms from Step 3.

Setting the Cisco Unified CME Router to Receive IP Phone Messages

To set the Cisco Unified CME router to receive IP phone messages and to set the maximum number of
DSP farms and transcoder sessions, perform the following steps.

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

Prerequisites

Identify the MAC address of the SCCP client interface.

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.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. ip source-address ip-address [port port] [any-match | strict-match]
5. sdspfarm units number
6. sdspfarm transcode sessions number
7. sdspfarm tag number device-number
8. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 ip source-address ip-address [port port] Enables a router to receive messages from Cisco Unified IP
[any-match | strict-match] phones through the router’s IP addresses and ports.
• address—The range is 0 to 5. The default is 0.
Example:
Router(config-telephony)# ip source address
• port port—(Optional) TCP/IP port used for SCCP. The
10.10.10.1 port 3000 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 5 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 to 5. The default is 0.
Router(config-telephony)# sdspfarm units 4
Step 6 sdspfarm transcode sessions number Specifies the maximum number of transcoder sessions for
G.729 allowed by the Cisco Unified CME router.
Example: • One transcoder session consists of two transcoding
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—Range is 0 to 128. Default is 0.
Note For the value of number, you can use the value
obtained in step 4 in the “Determining the
Maximum Number of Transcoder Sessions” section
on page 317.

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Command or Action Purpose

Step 7 sdspfarm tag number device-name Permits a DSP farm unit to be registered to
Cisco Unified CME and associates it with an SCCP client
interface’s MAC address.
Example:
Router(config-telephony)# sdspfarm tag 1 • number—The tag number. The range is 1 to 5.
mtp000a8eaca80
• device-name—The MAC address of the SCCP client
interface, with the “mtp” prefix added.
Step 8 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Configuring the Cisco Unified CME Router to Host a Secure DSP Farm
You must configure the Media Encryption Secure Real-Time Transport Protocol (SRTP) feature on the
Cisco Unified CME router, making it a secure Cisco Unified CME, before it can host a secure DSP farm.
See “Configuring Security” on page 387 for information on configuring a secure Cisco Unified CME.

Registering the DSP Farm with Cisco Unified CME in Secure Mode
The DSP farm can reside on the same router with the Cisco Unified CME or on a different router. Some
of the steps in the following tasks are optional depending the location of the DSP farm.
This section contains the following tasks:
• Obtaining a Digital Certificate from a CA Server, page 320
• Copying the CA Root Certificate of the DSP Farm Router to the Cisco Unified CME Router,
page 326
• Copying the CA Root Certificate of the Cisco Unified CME Router to the DSP farm Router,
page 327
• Configuring Cisco Unified CME to Allow the DSP Farm to Register, page 327
• Verifying DSP Farm Registration with Cisco Unified CME, page 328

Obtaining a Digital Certificate from a CA Server

The CA server can be the same router as the DSP farm. The DSP farm router can be configured as a CA
server. The configuration steps below show how to configure a CA server on the DSP farm router.
Additional configurations are required for configuring CA server on an external Cisco router or using a
different CA server by itself.
This section contains the following tasks:
• Configuring a CA Server, page 321 (Optional)
• Creating a Trustpoint, page 323
• Authenticating and Enrolling the Certificate with the CA Server, page 325

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Configuring a CA Server

Note Skip this procedure if the DSP farm resides on the same router as the Cisco Unified CME. Proceed to
the “Creating a Trustpoint” section on page 323.

The CA server automatically creates a trustpoint where the certificates are stored. The automatically
created trustpoint stores the CA root certificate.

Prerequisites

Cisco Unified CME 4.2 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. crypto pki server label
4. database level complete
5. grant auto
6. database url root-url
7. no shutdown
8. crypto pki trustpoint label
9. revocation-check crl
10. rsakeypair key-label

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 crypto pki server label Defines a label for the certificate server and enters
certificate-server configuration mode.
Example: • label—Name for CA certificate server.
Router(config)# crypto pki server dspcert

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Command or Action Purpose

Step 4 database level complete (Optional) Controls the type of data stored in the certificate
enrollment database. The default if this command is not
used is minimal.
Example:
Router(cs-server)# database level complete • complete—In addition to the information given in the
minimal and names levels, each issued certificate is
written to the database.
Note The complete keyword produces a large amount of
information; so specify an external TFTP server in
which to store the data using of the database url
command.
Step 5 grant auto (Optional) Allows an automatic certificate to be issued to
any requester. The recommended method and default if this
command is not used is manual enrollment.
Example:
Router(cs-server)# grant auto Tip Use this command only during enrollment when
testing and building simple networks. A security
best practice is to disable this functionality using
the no grant auto command after configuration so
that certificates cannot be continually granted.
Step 6 database url root-url (Optional) Specifies the location where all database entries
for the certificate server are to be written out. If this
command is not specified, all database entries are written to
Example:
Router(cs-server)# database url nvram:
NVRAM.
• root-url—Location where database entries will be
written out. The URL can be any URL that is supported
by the Cisco IOS file system.
Note If the CA is going to issue a large number of
certificates, select an appropriate storage location
like flash or other storage device to store the
certificates.

Note When the storage location chosen is flash and the

file system type on this device is Class B (LEFS),
make sure to check free space on the device
periodically and use the squeeze command to free
the space used up by deleted files. This process may
take several minutes and should be done during
scheduled maintenance periods or off-peak hours.
Step 7 no shutdown (Optional) Enables the CA.
Note You should use this command only after you have
Example: completely configured the CA.
Router(cs-server)# no shutdown
Step 8 exit Exits certificate-server configuration mode.

Example:
Router(cs-server)# exit

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Command or Action Purpose

Step 9 crypto pki trustpoint label (Optional) Declares a trustpoint and enters ca-trustpoint
configuration mode.
Example: • label—Name for the trustpoint. The label
Router(config)# crypto pki trustpoint dspcert
Note Use this command and the enrollment url
command if this CA is local to the
Cisco Unified CME router. These commands are
not needed for a CA running on an external router.

The label has to be the same as the label in Step 3.

Step 10 revocation-check crl (Optional) Checks the revocation status of a certificate and
specifies one or more methods to check the status. If a
second and third method are specified, each method is used
Example:
Router(ca-trustpoint)# revocation-check crl
only if the previous method returns an error, such as a server
being down.
• crl—Certificate checking is performed by a certificate
revocation list (CRL). This is the default behavior.
Step 11 rsakeypair key-label (Optional) Specifies an RSA key pair to use with a
certificate.
Example: • key-label—Name of the key pair, which is generated
Router(ca-trustpoint)# rsakeypair caserver during enrollment if it does not already exist or if the
auto-enroll regenerate command is used.
Note Multiple trustpoints can share the same key.

Creating a Trustpoint

The trustpoint stores the digital certificate for the DSP farm. To create a trustpoint, perform the following
procedure:

Prerequisites

Cisco Unified CME 4.2 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. crypto pki trustpoint label
4. enrollment url ca-url
5. serial-number none
6. fqdn none
7. ip-address none
8. subject-name [x.500-name]
9. revocation-check none
10. rsakeypair key-label

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 crypto pki trustpoint label Declares the trustpoint that your RA mode certificate server
should use and enters CA-trustpoint configuration mode.
Example: • label—Name for the trustpoint and RA.
Router(config)# crypto pki trustpoint dspcert
Step 4 enrollment url ca-url Specifies the enrollment URL of the issuing CA certificate
server (root certificate server).
Example: • ca-url—URL of the router on which the root CA is
Router(ca-trustpoint)# enrollment url installed.
http://10.3.105.40:80
Step 5 serial-number none Specifies whether the router serial number should be
included in the certificate request.
Example: • none—Specifies that a serial number will not be
Router(ca-trustpoint)# serial-number none included in the certificate request.
Step 6 fqdn none Specifies a fully qualified domain name (FQDN) that will
be included as “unstructuredName” in the certificate
request.
Example:
Router(ca-trustpoint)# fqdn none • none—Router FQDN will not be included in the
certificate request.
Step 7 ip-address none Specifies a dotted IP address or an interface that will be
included as “unstructuredAddress” in the certificate
request.
Example:
Router(ca-trustpoint)# ip-address none • none—Specifies that an IP address is not to be included
in the certificate request.
Step 8 subject-name [x.500-name] Specifies the subject name in the certificate request.
Note The example shows how to format the certificate
Example: subject name to be similar to that of an IP phone’s.
Router(ca-trustpoint)# subject-name cn=vg224,
ou=ABU, o=Cisco Systems Inc.

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Command or Action Purpose

Step 9 revocation-check none (Optional) Checks the revocation status of a certificate and
specifies one or more methods to check the status. If a
second and third method are specified, each method is used
Example:
Router(ca-trustpoint)# revocation-check none
only if the previous method returns an error, such as a server
being down.
• none—Certificate checking is not required.
Step 10 rsakeypair key-label (Optional) Specifies an RSA key pair to use with a
certificate.
Example: • key-label—Name of the key pair, which is generated
Router(ca-trustpoint)# rsakeypair dspcert during enrollment if it does not already exist or if the
auto-enroll regenerate command is used.
Note Multiple trustpoints can share the same key.

The key-label is the same as the label in Step 3.

Authenticating and Enrolling the Certificate with the CA Server

Prerequisites

Cisco Unified CME 4.2 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. crypto pki authenticate trustpoint-label
4. crypto pki enroll trustpoint-label

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 crypto pki authenticate trustpoint-label Retrieves the CA certificate and authenticates it. Checks the
certificate fingerprint if prompted.
Example: • trustpoint-label—Trustpoint label.
Router(config)# crypto pki authenticate dspcert
Note The trustpoint-label is the trustpoint label specified
in the “Creating a Trustpoint” section on page 323.
Step 4 crypto pki enroll trustpoint-label Enrolls with the CA and obtains the certificate for this
trustpoint.
Example: • trustpoint-label—Trustpoint label.
Router(config)# crypto pki enroll dspcert
Note The trustpoint-label is the trustpoint label specified
in the “Creating a Trustpoint” section on page 323.

Copying the CA Root Certificate of the DSP Farm Router to the Cisco Unified CME Router
The DSP farm router and Cisco Unified CME router exchanges certificates during the registration
process. These certificates are digitally signed by the CA server of the respective router. For the routers
to accept each others digital certificate, they should have the CA root certificate of each other. Manually
copy the CA root certificate of the DSP farm and Cisco Unified CME router to each other.

Prerequisites

Cisco Unified CME 4.2 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. crypto pki trustpoint name
4. enrollment terminal
5. crypto pki export trustpoint pem terminal
6. crypto pki authenticate trustpoint-label
7. You will be prompted to enter the CA certificate. Cut and paste the base 64 encoded certificate at
the command line, then press Enter, and type “quit.” The router prompts you to accept the certificate.
Enter “yes” to accept the certificate.

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 crypto pki trustpoint label Declares the trustpoint that your RA mode certificate server
should use and enters CA-trustpoint configuration mode.
Example: • label—Name for the trustpoint and RA.
Router(config)# crypto pki trustpoint dspcert
Note The label is the trustpoint label specified in the
“Creating a Trustpoint” section on page 323.
Step 4 enrollment terminal Specifies manual cut-and-paste certificate enrollment.

Example:
Router(ca-trustpoint)# enrollment terminal
Step 5 crypto pki export trustpoint pem terminal Exports certificates and RSA keys that are associated with
a trustpoint in a privacy-enhanced mail (PEM)-formatted
file.
Example:
Router(ca-trustpoint)# crypto pki export
dspcert pem terminal
Step 6 crypto pki authenticate trustpoint-label Retrieves the CA certificate and authenticates it. Checks the
certificate fingerprint if prompted.
Example: • trustpoint-label—Trustpoint label.
Router(config)# crypto pki authenticate vg224
Note This command is optional if the CA certificate is
already loaded into the configuration.
Step 7 You will be prompted to enter the CA certificate. Cut Completes the copying of the CA root certificate of the DSP
and paste the base 64 encoded certificate at the farm router to the Cisco Unified CME router.
command line, then press Enter, and type “quit.” The
router prompts you to accept the certificate. Enter
“yes” to accept the certificate.

Copying the CA Root Certificate of the Cisco Unified CME Router to the DSP farm Router
Repeat the steps in the “Copying the CA Root Certificate of the DSP Farm Router to the Cisco Unified
CME Router” section on page 326 in the opposite direction, that is, from Cisco Unified CME router to
the DSP farm router.

Prerequisites

Cisco Unified CME 4.2 or a later version.

Configuring Cisco Unified CME to Allow the DSP Farm to Register

Prerequisites

Cisco Unified CME 4.2 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service

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4. sdspfarm units number

5. sdspfarm transcode sessions number
6. sdspfarm tag number device-name
7. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 sdspfarm units number Specifies the maximum number of digital-signal-processor
(DSP) farms that are allowed to be registered to the Skinny
Client Control Protocol (SCCP) server.
Example:
Router(config-telephony)# sdspfarm units 1
Step 5 sdspfarm transcode sessions number Specifies the maximum number of transcoding sessions
allowed per Cisco Unified CME router.
Example: • number—Declares the number of DSP farm sessions.
Router(config-telephony)# sdspfarm transcode Valid values are numbers from 1 to 128.
sessions 30
Step 6 sdspfarm tag number device-name Permits a DSP farm to register to Cisco Unified CME and
associates it with a SCCP client interface's MAC address.
Example: Note The device-name in this step must be the same as the
Router(config-telephony)# sdspfarm tag 1 vg224 device-name in the associate profile command in
Step 17 of the “Configuring DSP Farms for
NM-HDs and NM-HDV2s” section on page 310.
Step 7 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Verifying DSP Farm Registration with Cisco Unified CME

Use the show sdspfarm units command to verify that the DSP farm is registering with
Cisco Unified CME. Use the show voice dsp group slot command to show the status of secure
conferencing.

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Prerequisites

Cisco Unified CME 4.2 or a later version.

show sdspfarm units: Example

Router# show sdspfarm units

mtp-2 Device:choc2851SecCFB1 TCP socket:[1] REGISTERED

actual_stream:8 max_stream 8 IP:10.1.0.20 37043 MTP YOKO keepalive 17391
Supported codec: G711Ulaw
G711Alaw
G729
G729a
G729ab
GSM FR

max-mtps:2, max-streams:60, alloc-streams:18, act-streams:0

show voice dsp: Example

Router# show voice dsp group slot 1
dsp 13:
State: UP, firmware: 4.4.706
Max signal/voice channel: 16/16
Max credits: 240
Group: FLEX_GROUP_VOICE, complexity: FLEX
Shared credits: 180, reserved credits: 0
Signaling channels allocated: 2
Voice channels allocated: 0
Credits used: 0
Group: FLEX_GROUP_XCODE, complexity: SECURE MEDIUM
Shared credits: 0, reserved credits: 60
Transcoding channels allocated: 0
Credits used: 0
dsp 14:
State: UP, firmware: 1.0.6
Max signal/voice channel: 16/16
Max credits: 240
Group: FLEX_GROUP_CONF, complexity: SECURE CONFERENCE
Shared credits: 0, reserved credits: 240
Conference session: 1
Credits used: 0

Verifying DSP Farm Operation

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

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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 Use the show sccp [statistics | connections] 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
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

Use the show sccp connections command to display information about the connections controlled by
the SCCP transcoding and conferencing applications. In the following example, the secure value of the
stype field indicates that the connection is encrypted:
Router# show sccp connections

sess_id conn_id stype mode codec ripaddr rport sport

16777222 16777409 secure-xcode sendrecv g729b 10.3.56.120 16772 19534

16777222 16777393 secure-xcode sendrecv g711u 10.3.56.50 17030 18464

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

Step 2 Use the show sdspfarm units 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 Use the show sdspfarm sessions 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

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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

Step 4 Use the show sdspfarm sessions summary 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

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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 Use the show sdspfarm sessions active 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

Step 6 Use the show sccp connections details 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 Use the debug sccp {all | errors | events | packets | parser} command to set debugging levels for SCCP
and its applications.
Step 8 Use the debug dspfarm {all | errors | events | packets} command to set debugging levels for DSP-farm
service
Step 9 Use the debug ephone mtp 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 DSP Farm Performance

To tune DSP farm performance, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal

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3. sccp ip precedence value

4. dspfarm rtp timeout seconds
5. dspfarm connection interval seconds
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 sccp ip precedence value (Optional) Sets the IP precedence value to increase the
priority of voice packets over connections controlled by
SCCP.
Example:
Router(config)# sccp ip precedence 5
Step 4 dspfarm rtp timeout seconds (Optional) Configures the Real-Time Transport Protocol
(RTP) timeout interval if the error condition “RTP port
unreachable” occurs.
Example:
Router(config)# dspfarm rtp timeout 60
Step 5 dspfarm connection interval seconds (Optional) Specifies how long to monitor RTP inactivity
before deleting an RTP stream.
Example:
Router(config)# dspfarm connection interval 60
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config)# end

Configuration Examples for Transcoding Resources

This section contains the following examples:
• DSP Farms for NM-HDVs: Example, page 334
• DSP Farms for NM-HDs and NM-HDV2s: Example, page 334
• Cisco Unified CME Router as the DSP Farm Host: Example, page 335

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DSP Farms for NM-HDVs: Example

The following 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

DSP Farms for NM-HDs and NM-HDV2s: Example

The following 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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Cisco Unified CME Router as the DSP Farm Host: Example

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

Where to go Next
Music on Hold
Music on hold can require transcoding resources. See “Configuring Music on Hold” on page 771.

Teleworker Remote Phones

Transcoding has benefits and disadvantages for remote teleworker phones. See the discussion in
“Configuring Phones to Make Basic Calls” on page 147.

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

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Additional References

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Transcoding Resources

Table 17 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 17 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 17 Feature Information for Transcoding Resources

Cisco Unified CME

Feature Name Version Feature Information
Transcoding Support 3.2 Transcoding between G.711 and G.729 was introduced.
Secure Transcoding 4.2 Secure transcoding for calls using the codec g729r8
dspfarm-assist command was introduced.

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Last Updated: June 18, 2007

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

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Enabling the GUI”
section on page 354.

Contents
• Prerequisites for Enabling the GUI, page 339
• Restrictions for Enabling the GUI, page 340
• Information About Enabling the GUI, page 340
• How to Enable the GUI, page 341
• Configuration Examples for Enabling the GUI, page 350
• Additional References, page 352
• Feature Information for Enabling the GUI, page 354

Prerequisites for Enabling the GUI

GUI files must be copied into flash memory on the router. For information about files, see “Installing
and Upgrading Cisco Unified CME Software” on page 69.

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Restrictions for Enabling the GUI

• Cisco Unified CME GUI files are version-specific; GUI files for one version of Cisco Unified CME
are not compatible with any other version of Cisco Unified CME. If you are downgrading or
upgrading your Cisco Unified CME version, you must downgrade or upgrade your GUI files. For
information, see “Installing and Upgrading Cisco Unified CME Software” on page 69.
• To access the GUI, you must use Microsoft Internet Explorer 5.5 or a later version. Other browsers
are not supported.
• If you use an XML configuration file to create a customer administrator login, the XML file can have
a maximum size of 4000 bytes.
• 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.

Information About Enabling the GUI

To enable GUI support, you should understand the following concepts:
• Cisco Unified CME GUI Support, page 340
• AAA Authentication, page 341

Cisco Unified CME GUI Support

The Cisco Unified CME GUI provides a web-based interface to manage most system-level and
phone-based features. In particular, the GUI facilitates the routine additions and changes associated with
employee turnover, allowing these changes to be performed by nontechnical staff. The GUI provides
three levels of access to support the following user classes:
• System administrator—Able to configure all system-level and phone-based features. This person is
familiar with Cisco IOS software and VoIP network configuration.
• Customer administrator—Able to perform routine phone additions and changes without having
access to system-level features. This person does not have to be familiar with Cisco IOS software.
• Phone user—Able to program a small set of features on his or her own phone and search the
Cisco Unified CME directory.
The Cisco Unified 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 accounts for customer administrators and phone users can be added from the
Cisco Unified CME router using CLI commands or from a PC using GUI screens.
Cisco Unified 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.

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AAA Authentication
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.
Using the ip http authentication command prevents unauthorized users from accessing the
Cisco Unified CME router. If this command is not used, the enable password for the router is the only
requirement to authenticate user access to the GUI. Instead, we recommend you use the local or
TACACS authentication options, configured as part of a global AAA framework. 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 Unified CME
router. If you select the AAA authentication method, you must also define an authentication method in
your AAA configuration.
For information on configuring AAA authentication, see the “Configuring Authentication” chapter of
the Cisco IOS Security Configuration Guide for your Cisco IOS release.

How to Enable the GUI

This section contains the following procedures:
• Enabling the HTTP Server, page 341 (required)
• Enabling GUI Access for the System Administrator, page 343 (required)
• Accessing the Cisco Unified CME GUI, page 344 (required)
• Creating a Customized XML File for Customer Administrator GUI, page 345 (optional)
• Enabling GUI Access for Customer Administrators, page 346 (optional)
• Enabling GUI Access for Phone Users, page 348 (optional)
• Troubleshooting the Cisco Unified CME GUI, page 349 (optional)

Enabling the HTTP Server

To enable the HTTP server, and specify the path to files for the GUI and a method of user authentication
for security, perform the following steps. The HTTP server on a router is disabled by default.

SUMMARY STEPS

1. enable
2. configure terminal
3. ip http server
4. ip http path flash:
5. ip http authentication {aaa | enable | local | tacacs}
6. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ip http server Enables the HTTP server on the Cisco Unified CME
router.
Example:
Router(config)# ip http server
Step 4 ip http path flash: Sets the location of the HTML files used by the HTTP
web server to flash memory on the router.
Example:
Router(config)# ip http path flash:
Step 5 ip http authentication {aaa | enable | local | Specifies the method of authentication for the HTTP
tacacs} 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 username, password, and
privilege level access combination specified in
the local system configuration (by the username
command).
• tacacs—Uses TACACS (or XTACACS) server.
Step 6 exit Returns to privileged EXEC mode.

Example:
Router(config)# exit

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Enabling GUI Access for the System Administrator

To define an initial username and password for a system administrator to access the GUI and enable the
GUI to be used to set the time and to add directory listings, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. web admin system name username {password string | secret {0 | 5} string}
5. dn-webedit
6. time-webedit
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 web admin system name username {password string | Defines username and password for a system
secret {0 | 5} string} administrator.
• name username—System administrator
Example: username. Default is Admin.
Router(config-telephony)# web admin system name pwa3
secret 0 wp78pw • password string—String to verify system
administrator’s identity. Default is empty string.
• secret {0 | 5} string—Digit specifies state of
encryption of the string that follows:
– 0—Password that follows is not 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.

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Command or Action Purpose

Step 5 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.
Step 6 time-webedit (Optional) Enables the ability to set the phone time
for the Cisco Unified CME system through the web
interface.
Example:
Router(config-telephony)# time-webedit Note We do not recommend this method for setting
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 7 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Accessing the Cisco Unified CME GUI

To access the Cisco Unified CME router through the GUI to make configuration changes, perform the
following steps.

Restrictions
The Cisco Unified CME GUI requires Microsoft Internet Explorer 5.5 or a later version. Other browsers
are not supported.

DETAILED STEPS

Step 1 Go to the following URL:

http://router_ipaddress/ccme.html

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

Step 2 Enter your username and password at the login screen.

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The Cisco Unified CME system evaluates your privilege level and presents the appropriate window.
Note that users with Cisco IOS software privilege level 15 also have system-administrator-level
privileges in the Cisco Unified CME GUI after being authenticated locally or remotely through AAA.
The ip http authentication command that is configured on the Cisco Unified CME router determines
where authentication occurs.
Step 3 After you login and are authenticated, the system displays one of the following home pages, based on
your user level:
• The system administrator home page.
• The customer administrator sees a reduced version of the options available on the system
administrator page, according to the XML configuration file that the system administrator created.
• The phone user home page.
After you log in successfully, online help is available from the Help menu.

Creating a Customized XML File for Customer Administrator GUI

The XML configuration file specifies the parameters and features that are available to customer
administrators and the parameters and features that are restricted. The file follows a template named
xml.template, which conforms to the Cisco XML Document Type Definition (DTD), as documented in
Cisco IP Phone Services Application Development Notes. This template is one of the
Cisco Unified CME files that you download from the Cisco Software Center during installation.
To edit and load the XML configuration file, perform the following steps.

SUMMARY STEPS

1. Copy 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 Unified CME router.
4. Copy your file to flash memory on the Cisco Unified CME router.
5. Load the XML file from router flash memory.

DETAILED STEPS

Step 1 Copy the XML template that you downloaded from the Cisco Software Center and open it in any text
editor (see the “XML Configuration File Template: Example” section on page 350). 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 Cisco CME GUI. For example,
“<AddExtension>” refers to the Add Extension capability, and “<Type>” refers to the Type field on the
Add Extension window. 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 in to the GUI. Delete the action that
you do not want and the vertical bar and brackets around the actions.

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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 351.
Step 3 Copy the file to a TFTP or FTP server that can be accessed by the Cisco Unified CME router.
Step 4 Copy your file to flash memory on the Cisco Unified CME router.
Router# copy tftp flash

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

Enabling GUI Access for Customer Administrators

Perform one of the following procedures to enable GUI access for a customer administrator, depending
on the method you want to use:
• Using the Cisco Unified CME GUI to Define a Customer Administrator Account, page 346
• Using the Cisco IOS CLI to Define a Customer Administrator Account, page 347

Prerequisites
• Enable a system administrator account for GUI access. See the “Enabling GUI Access for the
System Administrator” section on page 343.
• Create the XML configuration file for the customer administrator GUI. See the “Creating a
Customized XML File for Customer Administrator GUI” section on page 345.
• Reload the XML file using the web customize load command if you have made changes to the
customer administrator GUI.

Using the Cisco Unified CME GUI to Define a Customer Administrator Account
To allow the system administrator to use the GUI to create a customer administrator account, perform
the following steps.

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DETAILED STEPS

Step 1 From the Configure System Parameters menu, choose Administrator’s Login Account.
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.

Using the Cisco IOS CLI to Define a Customer Administrator Account

To allow the system administrator to create a customer administrator account by using the Cisco IOS
CLI, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. web admin customer name username password string
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 4 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 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Enabling GUI Access for Phone Users

Perform one of the following procedures to enable GUI access for a phone user, depending on the
method you want to use:
• Using the Cisco Unified CME GUI to Define a Phone User Account, page 348
• Using the Cisco IOS CLI to Define a Phone User Account, page 349

Prerequisites
• Enable a system administrator account for GUI access. See the “Enabling GUI Access for the
System Administrator” section on page 343.

Using the Cisco Unified CME GUI to Define a Phone User Account
To create a phone user account by using the Cisco Unified CME GUI, perform the following steps.

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. The Add Phone screen or the
Change Phone screen displays.
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.
Step 3 Click Change for your edits to become effective.

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Using the Cisco IOS CLI to Define a Phone User Account

To create a GUI account for a phone user by using the Cisco IOS CLI, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. username username password password
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.

Example:
Router(config)# ephone 2
Step 4 username username password password Assigns a phone user login account name and
password.
Example: • This allows the phone user to log in to the
Router(config-ephone)# username prx password pk59wq Cisco Unified CME GUI to change a limited
number of personal settings.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

Troubleshooting the Cisco Unified CME GUI

If you are having trouble starting the Cisco Unified CME GUI, try the following actions:

Step 1 Verify you are using Microsoft Internet Explorer 5.5 or a later version. No other browser is supported.
Step 2 Clear your browser cache or history.

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Step 3 Verify that the GUI files in router flash memory are the correct version for the version of
Cisco Unified CME that you have. Compare the filenames in flash memory with the list in the
Cisco Unified CME software archive that you downloaded. 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
ensure that you have the most recent files installed in flash memory. The latest version can be
downloaded from the Cisco Unified CME Software Download website at
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp.

Configuration Examples for Enabling the GUI

This section contains the following examples:
• HTTP and Account Configuration: Example, page 350
• XML Configuration File Template: Example, page 350
• XML Configuration File: Example, page 351

HTTP and Account Configuration: Example

The following example sets up the HTTP server and creates a system administrator account for pwa3, a
customer administrator account for user44, and a user account for prx.
ip http server
ip http path flash:
ip http authentication aaa

telephony-service
web admin system name pwa3 secret 0 wp78pw
web admin customer name user44 password pw10293847
dn-webedit
time-webedit

ephone 25
username prx password pswd

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>

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<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>

<Function>
<AddLineToPhone> [No | Yes] </AddLineToPhone>
<DeleteLineFromPhone> [No | Yes] </DeleteLineFromPhone>
<NewDnDpCheck> [No | Yes] </NewDnDpCheck>
<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>

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<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>
</Function>
</Presentation>

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Security Command Reference
• Cisco IOS Software Releases 12.4T Command References

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Related Topic Document Title

Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Security Configuration Guide
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Enabling the GUI

Table 18 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note The following table lists the Cisco Unified CME version that introduced support for a given feature.
Unless noted otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 18 Feature Information for Enabling the GUI

Cisco Unified CME

Feature Name Version Feature Information
Cisco Unified CME GUI 2.0 The Cisco Unified CME GUI was introduced.

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Last Updated: June 18, 2007

This chapter describes how to integrate your voice-mail system with Cisco Unified Communications
Manager Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Voice-Mail
Integration” section on page 386.

Contents
• Prerequisites, page 355
• Information About Voice-Mail Integration, page 355
• How to Configure Voice-Mail Integration, page 360
• Configuration Examples for Voice-Mail Integration, page 381
• Additional References, page 385
• Feature Information for Voice-Mail Integration, page 386

Prerequisites
• Voice mail must be installed and configured on your network.
• Calls can be successfully completed between phones on the same Cisco Unified CME router.

Information About Voice-Mail Integration

To enable voice-mail support, you should understand the following concepts:
• Cisco Unity Connection Integration, page 356
• Cisco Unity Express Integration, page 356
• Cisco Unity Integration, page 356

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• DTMF Integration for Legacy Voice-Mail Applications, page 357

• Mailbox Selection Policy, page 357
• RFC 2833 DTMF MTP Passthrough, page 357
• MWI Line Selection, page 358
• AMWI (Audible Message Waiting Indicator), page 358
• SIP MWI Prefix Specification, page 359
• SIP MWI - QSIG Translation, page 359

Cisco Unity Connection Integration

Cisco Unity Connection transparently integrates messaging and voice recognition components with your
data network to provide continuous global access to calls and messages. These advanced,
convergence-based communication services help you use voice commands to place calls or listen to
messages in “hands-free” mode and check voice messages from your desktop, either integrated into an
e-mail inbox or from a Web browser. Cisco Unity Connection also features robust automated-attendant
functions that include intelligent routing and easily customizable call-screening and
message-notification options.
For instructions on how to integrate Cisco Unified CME with Cisco Unity Connection, see the
Cisco CallManager Express 3.x Integration Guide for Cisco Unity Connection 1.1.

Cisco Unity Express Integration

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, see the administrator guides for
Cisco Unity Express.
For configuration information, see the “Enabling DTMF Integration Using SIP NOTIFY” section on
page 372.

Note Cisco Unified CME and Cisco Unity Express must both be configured before they can be integrated.

Cisco Unity Integration

Cisco Unity is a Microsoft Windows-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 Unified CME. This integration requires that you configure the Cisco Unified CME router and
Cisco Unity software to get voice-mail service.
For configuration instructions, see the “Enabling DTMF Integration Using RFC 2833” section on
page 369.

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DTMF Integration for Legacy Voice-Mail Applications

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 Unified CME router.
These patterns are required for DTMF integration of Cisco Unified CME with most voice-mail systems.
Voice-mail systems are designed to respond to DTMF after the system answers the incoming calls.
After configuring the DTMF integration patterns on the Cisco Unified CME router, you set up the
integration files on the third-party legacy voice-mail system by following 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 Unified CME router work with each other.
For configuration information, see the “Enabling DTMF Integration for Legacy Voice-Mail
Applications” section on page 367.

Mailbox Selection Policy

Typically a voice-mail system uses the number that a caller has dialed to determine the mailbox to which
a call should be sent. However, if a call has been diverted several times before reaching the voice-mail
system, the mailbox that is selected might vary for different types of voice-mail systems. For example,
Cisco Unity Express uses the last number to which the call was diverted before it was sent to voice mail
as the mailbox number. Cisco Unity and some legacy PBX systems use the originally called number as
the mailbox number.
The Mailbox Selection Policy feature allows you to provision the following options from the
Cisco Unified CME configuration.
• For Cisco Unity Express, you can select the originally dialed number.
• For PBX voice-mail systems, you can select the last number to which the call was diverted before it
was sent to voice mail. This option is configured on the outgoing dial peer for the voice-mail
system's pilot number.
• For Cisco Unity voice mail, you can select the last number to which the call was diverted before it
was sent to voice mail. This option is configured on the ephone-dn that is associated with the
voice-mail pilot number.
To enable Mailbox Selection Policy, see the “SCCP: Setting Mailbox Selection Policy for
Cisco Unity Express or PBX Voice-Mail Number” section on page 362 or the“SCCP: Setting Mailbox
Selection Policy for Cisco Unity” section on page 364.

RFC 2833 DTMF MTP Passthrough

In Cisco Unified CME 4.1, the RFC 2833 Dual-Tone Multifrequency (DTMF) Media Termination Point
(MTP) Passthrough feature provides the capability to pass DTMF tones transparently between SIP
endpoints that require transcoding or Resource Reservation Protocol (RSVP) agents.
This feature supports DTMF relay across SIP WAN devices that support RFC 2833, such as Cisco Unity
and SIP trunks. Devices registered to a Cisco Unified CME SIP back-to-back user agent (B2BUA) can
exchange RFC 2833 DTMF MTP with other devices that are not registered with the Cisco Unified CME
SIP B2BUA, or with devices that are registered in one of the following:
• Local or remote Cisco Unified CME
• Cisco Unified Communications Manager

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• Third party proxy

By default, the RFC 2833 DTMF MTP Passthrough feature uses payload type 101 on MTP, and MTP
accepts all the other dynamic payload types if it is indicated by Cisco Unified CME. For configuration
information, see the “Enabling DTMF Integration Using RFC 2833” section on page 369.

MWI Line Selection

Message waiting indicator (MWI) line selection allows you to choose the phone line that is monitored
for voice-mail messages and that lights an indicator when messages are present.
Before Cisco Unified CME 4.0, the MWI lamp on a phone running SCCP could be associated only with
the primary line of the phone.
In Cisco Unified CME 4.0 and later versions, you can designate a phone line other than the primary line
to be associated with the MWI lamp. Lines other than the one associated with the MWI lamp display an
envelope icon when a message is waiting. A logical phone “line” is not the same as a phone button. A
button with one or more directory numbers is considered one line. A button with no directory number
assigned does not count as a line.
In Cisco Unified CME 4.0 and later versions, a SIP directory number that is used for call forward all,
presence BLF status, and MWI features must be configured by using the dn keyword in the number
command; direct line numbers are not supported.
For configuration information, see the“SCCP: Configuring a Voice Mailbox Pilot Number” section on
page 361 or “SIP: Configuring a Directory Number for MWI” section on page 377.

AMWI (Audible Message Waiting Indicator)

The AMWI (Audible Message Line Indicator) feature provides a special stutter dial tone to indicate
message waiting. This is an accessibility feature for vision-impaired phone users. The stutter dial tone
is defined as 10 ms ON, 100 ms OFF, repeat 10 times, then steady on.
In Cisco Unified CME 4.0(3), you can configure the AMWI feature on the Cisco Unified IP Phone 7911
and Cisco Unified IP Phone 7931G to receive audible, visual, or audible and visual MWI notification
from an external voice-messaging system. AMWI cannot be enabled unless the the number command
is already configured for the IP phone to be configured. Cisco Unified CME applies the following logic
based on the capabilities of the IP phone and how MWI is configured:
• If the phone supports (visual) MWI and MWI is configured for the phone, turn on the Message
Waiting light.
• If the phone supports (visual) MWI only, turn on the Message Waiting light regardless of the
configuration.
• If the phone supports AMWI and AMWI is configured for the phone, send the stutter dial tone to
the phone when it goes off-hook.
• If the phone supports AMWI only and AMWI is configured, send the stutter dial tone to the phone
when it goes off-hook regardless of the configuration.
If a phone supports (visual) MWI and AMWI and both options are configured for the phone, turn on the
Message Waiting light and send the stutter dial tone to the phone when it goes off-hook.
For configuration informations, see the “SCCP: Configuring Audible MWI” section on page 375.

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SIP MWI Prefix Specification

Central voice-messaging servers that provide mailboxes for several Cisco Unified CME sites may use
site codes or prefixes to distinguish among similarly numbered ranges of extensions at different sites. In
Cisco Unified CME 4.0 and later versions, you can specify that your Cisco Unified CME system should
accept unsolicited SIP Notify messages for MWI that include a prefix string as a site identifier.
For example, an MWI message might indicate that the central mailbox number 555-0123 has a voice
message. In this example, the digits 555 are set as the prefix string or site identifier using the mwi prefix
command. The local Cisco Unified CME system is able to convert 555-0123 to 0123 and deliver the
MWI to the correct phone. Without this prefix string manipulation, the system would reject an MWI for
555-0123 as not matching the local Cisco Unified CME extension 0123.
To enable SIP MWI Prefix Specification, see the “Enabling SIP MWI Prefix Specification” section on
page 380.

SIP MWI - QSIG Translation

In Cisco Unified CME 4.1 and later, the SIP MWI - QSIG Translation feature extends MWI
functionality for SIP MWI and QSIG MWI interoperation to enable sending and receiving MWI over
QSIG to a PBX.
When the SIP Unsolicited NOTIFY is received from voice mail, the Cisco router translates this event to
activate QSIG MWI to the PBX, via PSTN. The PBX will switch on, or off, the MWI lamp on the
corresponding IP phone. This feature supports only Unsolicited NOTIFY. Subscribe NOTIFY is not
supported by this feature.
In Figure 17, the Cisco router receives the SIP Unsolicited NOTIFY, performs the protocol translation,
and initiates the QSIG MWI call to the PBX, where it is routed to the appropriate phone.

Figure 17 SIP MWI to ISDN QSIG When Voice Mail and Cisco Router are On the Same LAN

SIP MWI NOTIFY message QSIG MWI message

Cisco Unified CME PBX

U
146430

LAN

It makes no difference if the SIP Unsolicited NOTIFY is received via LAN or WAN as long as the PBX
is connected to the Cisco router, and not to the remote voice mail server.

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In Figure 18, a voice mail server and Cisco Unified CME are connected to the same LAN and a remote
Cisco Unified CME is connected across the WAN. In this scenario, the protocol translation is performed
at the remote Cisco router and the QSIG MWI message is sent to the PBX.

Figure 18 SIP MWI to ISDN QSIG When PBX is Connected to a Remote Cisco Router

WAN

SIP MWI NOTIFY message QSIG MWI message

U SIP MWI NOTIFY message

146570
PBX
Cisco Unified CME Cisco Unified CME
LAN

How to Configure Voice-Mail Integration

This section contains the following tasks:
• SCCP: Configuring a Voice Mailbox Pilot Number, page 361 (required)
• SCCP: Configuring a Mailbox Selection Policy, page 362 (optional)
• SIP: Configuring a Voice Mailbox Pilot Number, page 365 (required)
• Enabling DTMF Integration, page 367 (required)
• SCCP: Configuring a Phone Line for MWI Outcall, page 374 (required)
• SCCP: Configuring Audible MWI, page 375 (optional)
• SIP: Enabling MWI at System-Level, page 376 (required)
• SIP: Configuring a Directory Number for MWI, page 377 (required)
• Enabling SIP MWI Prefix Specification, page 380 (optional)
• Verifying Voice-Mail Integration, page 381 (optional)

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SCCP: Configuring a Voice Mailbox Pilot Number

To configure the telephone number that is speed-dialed when the Message button on a SCCP phone is
pressed, perform the following steps.

Note The same telephone number is configured for voice messaging for all SCCP phones in
Cisco Unified CME.

Prerequisites
• Voicemail phone number must be a valid number; directory number and number for voicemail
phone number must be configured. For configuration information, see “Configuring Phones to Make
Basic Calls” on page 147.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. voicemail phone-number
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters voice register global configuration mode to set
parameters for all supported phones in Cisco Unified CME.
Example:
Router(config)# telephony-service
Step 4 voicemail phone-number Defines the telephone number that is speed-dialed when the
Messages button on a Cisco Unified IP phone is pressed.
Example: • phone-number—Same phone number is configured for
Router(config-telephony)# voice mail 0123 voice messaging for all SCCP phones in a
Cisco Unified CME.

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Command or Action Purpose

Step 5 end Exits to privileged EXEC mode.

Example:
Router(config-telephony)# end

What to Do Next
• (Cisco Unified CME 4.0 or a later version only) To set up a mailbox selection policy, see the
“SCCP: Configuring a Mailbox Selection Policy” section on page 362.
• To set up DTMF integration patterns for connecting to analog voice-mail applications, see the
“Enabling DTMF Integration for Legacy Voice-Mail Applications” section on page 367.
• To connect to a remote SIP-based IVR or Cisco Unity, or to connect to a remote SIP-PSTN that goes
through the PSTN to a voice-mail or IVR application, see the “Enabling DTMF Integration Using
RFC 2833” section on page 369.
• To connect to a Cisco Unity Express system, configure a nonstandard SIP NOTIFY format. See the
“Enabling DTMF Integration Using SIP NOTIFY” section on page 372.

SCCP: Configuring a Mailbox Selection Policy

Perform one of the following tasks, depending on which voice-mail application is used:
• SCCP: Setting Mailbox Selection Policy for Cisco Unity Express or PBX Voice-Mail Number,
page 362
• SCCP: Setting Mailbox Selection Policy for Cisco Unity, page 364

SCCP: Setting Mailbox Selection Policy for Cisco Unity Express or PBX Voice-Mail Number
To set a policy for selecting a mailbox for calls from a Cisco Unified CME system that are diverted
before being sent to a Cisco Unity Express or PBX voice-mail pilot number, perform the following steps.

Prerequisites

Cisco Unified CME 4.0 or a later version.

Restrictions

In the following scenarios, the mailbox selection policy can fail to work properly:
• The last redirecting endpoint is not hosted on Cisco Unified CME. This may rarely occur with a
PBX.
• A call is forwarded across several SIP trunks. Multiple SIP Diversion Headers (stacking hierarchy)
are not supported in Cisco IOS software.
• A call is forwarded across non-Cisco voice gateways that do not support the optional H450.3
originalCalledNr field.

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SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag voip
or
dial-peer voice tag pots
4. mailbox-selection [last-redirect-num | orig-called-num]
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 dial-peer voice tag voip Enters dial-peer configuration mode.
or
• tag—Identifies the dial peer. Valid entries are from 1 to
dial-peer voice tag pots 2147483647.
Note This command should be used on the outbound dial
Example: peer associated with the pilot number of the
Router(config)# dial-peer voice 7000 voip voice-mail system. For systems using
or Cisco Unity Express, this is a VoIP dial peer. For
Router(config)# dial-peer voice 35 pots systems using PBX-based voice mail, this is a POTS
dial peer.
Step 4 mailbox-selection [last-redirect-num | Sets a policy for selecting a mailbox for calls that are
orig-called-num] diverted before being sent to a voice-mail line.
• last-redirect-num—(PBX voice mail only) The
Example: mailbox number to which the call will be sent is the last
Router(config-dial-peer)# mailbox-selection number to divert the call (the number that sends the call
orig-called-num
to the voice-mail pilot number).
• orig-called-num—(Cisco Unity Express only) The
mailbox number to which the call will be sent is the
number that was originally dialed before the call was
diverted.
Step 5 end Returns to privileged EXEC mode.

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

What to Do Next.

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• To use voice mail on a SIP network that connects to a Cisco Unity Express system, configure a
nonstandard SIP NOTIFY format. See the “Enabling DTMF Integration Using SIP NOTIFY”
section on page 372.

SCCP: Setting Mailbox Selection Policy for Cisco Unity

To set a policy for selecting a mailbox for calls that are diverted before being sent to a Cisco Unity
voice-mail pilot number, perform the following steps.

Prerequisites

• Cisco Unified CME 4.0 or a later version.

• Director number to be configured is associated with a voice mailbox.

Restrictions

This feature might not work properly in certain network topologies, including the following cases:
• When the last redirecting endpoint is not hosted on Cisco Unified CME. This may rarely occur with
a PBX.
• When a call is forwarded across several SIP trunks. Multiple SIP Diversion Headers (stacking
hierarchy) are not supported in Cisco IOS software.
• When a call is forwarded across other voice gateways that do not support the optional H450.3
originalCalledNr field.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag
4. mailbox-selection last-redirect-num
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 exit Exits dial-peer configuration mode.

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

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Command or Action Purpose

Step 4 ephone-dn Enters ephone-dn configuration mode.

Example:
Router(config)# ephone-dn 752
Step 5 mailbox-selection [last-redirect-num] Sets a policy for selecting a mailbox for calls that are
diverted before being sent to a Cisco Unity voice-mail pilot
number.
Example:
Router(config-ephone-dn)# mailbox-selection
last-redirect-num
Step 6 end Returns to privileged EXEC mode.

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

What to Do Next

• To use a remote SIP-based IVR or Cisco Unity, or to connect Cisco Unified CME to a remote
SIP-PSTN that goes through the PSTN to a voice-mail or IVR application, see the “Enabling DTMF
Integration Using RFC 2833” section on page 369.

SIP: Configuring a Voice Mailbox Pilot Number

To configure the telephone number that is speed-dialed when the Message button on a SIP phone is
pressed, follow the steps in this section.

Note The same telephone number is configured for voice messaging for all SIP phones in Cisco Unified CME.
The call forward b2bua command enables call forwarding and designates that calls that are forwarded
to a busy or no-answer extension be sent to a voicemail box.

Prerequisites
• Directory number and number for voicemail phone number must be configured. For configuration
information, see “Configuring Phones to Make Basic Calls” on page 147.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. voicemail phone-number
5. exit
6. voice register dn dn-tag
7. call-forward b2bua busy directory-number
8. call-forward b2bua mailbox directory-number

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9. call-forward b2bua noan directory-number

10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 voicemail phone-number Defines the telephone number that is speed-dialed when the
Messages button on a Cisco Unified IP phone is pressed.
Example: • phone-number—Same phone number is configured for
Router(config-register-global)# voice mail 1111 voice messaging for all SIP phones in a
Cisco Unified CME.
Step 5 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-register-global)# exit
Step 6 voice register dn dn-tag Enters voice register dn mode to define a directory number
for a SIP phone, intercom line, voice port, or an MWI.
Example:
Router(config-register-global)# voice register
dn 2
Step 7 call-forward b2bua busy directory-number Enables call forwarding for a SIP back-to-back user agent
so that incoming calls to an extension that is busy will be
forwarded to the designated directory number.
Example:
Router(config-register-dn)# call-forward b2bua
busy 1000
Step 8 call-forward b2bua mailbox directory-number Designates voice mailbox to use at the end of a chain of call
forwards. Incoming calls have been forwarded to a busy or
no-answer extension will be forwarded to the
Example:
Router(config-register-dn)# call-forward b2bua
directory-number specified.
mailbox 2200

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Command or Action Purpose

Step 9 call-forward b2bua noan directory-number Enables call forwarding for a SIP back-to-back user agent
timeout seconds so that incoming calls to an extension that does not answer
will be forwarded to the designated directory number.
Example: • timeout seconds—Range:3 to 60000. Default: 20.
Router(config-register-dn)# call-forward b2bua
noan 2201 timeout 15
Step 10 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-dn)# end

What to Do Next
• To set up DTMF integration patterns for connecting to analog voice-mail applications, see the
“Enabling DTMF Integration for Legacy Voice-Mail Applications” section on page 367.
• To use a remote SIP-based IVR or Cisco Unity, or to connect to a remote SIP-PSTN that goes
through the PSTN to a voice-mail or IVR application, see the “Enabling DTMF Integration Using
RFC 2833” section on page 369.
• To connect to a Cisco Unity Express system, configure a nonstandard SIP NOTIFY format, see the
“Enabling DTMF Integration Using SIP NOTIFY” section on page 372.

Enabling DTMF Integration

Perform one of the following tasks, depending on which DTMF relay method is required:
• Enabling DTMF Integration for Legacy Voice-Mail Applications, page 367—To set up DTMF
integration patterns for connecting to analog voice-mail applications.
• Enabling DTMF Integration Using RFC 2833, page 369—To connect to a remote SIP-based IVR or
voice-mail application such as Cisco Unity or when SIP is used to connect Cisco Unified CME to a
remote SIP-PSTN voice gateway that goes through the PSTN to a voice-mail or IVR application.
• Enabling DTMF Integration Using SIP NOTIFY, page 372—To configure a SIP dial peer to point
to Cisco Unity Express.

Enabling DTMF Integration for Legacy Voice-Mail Applications

To set up DTMF integration patterns for analog voice-mail applications, perform the following steps.

Note You can configure multiple tags and tokens for each pattern, depending on the voice-mail system and
type of access.

SUMMARY STEPS

1. enable
2. configure terminal
3. vm-integration

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4. pattern direct tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
5. pattern ext-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
6. pattern ext-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
7. pattern trunk-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
8. pattern trunk-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 4 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 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).

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Command or Action Purpose

Step 5 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 when 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 6 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 when 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 *
Step 7 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 when 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 8 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 9 end Exits configuration mode and enters privileged EXEC mode.

Example:
Router(config-vm-integration)# exit

What to Do Next

After integrating configuring DTMF relay, you are ready to configure Message Waiting Indicator (MWI)
notification for either the MWI outcall, unsolicited notify, or subscribe/notify mechanism. See the
“SCCP: Configuring a Phone Line for MWI Outcall” section on page 374.

Enabling DTMF Integration Using RFC 2833

To configure a SIP dial peer to point to Cisco Unity and enable SIP dual-tone multifrequency (DTMF)
relay using RFC 2833, use the commands in this section on both the originating and terminating
gateways.
This DTMF relay method is required in the following situations:
• When SIP is used to connect Cisco Unified CME to a remote SIP-based IVR or voice-mail
application such as Cisco Unity.

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• When SIP is used to connect Cisco Unified CME to a remote SIP-PSTN voice gateway that goes
through the PSTN to a voice-mail or IVR application.

Note If the T38 Fax Relay feature is also configured on this IP network, we recommend that you either
configure the voice gateways to use a payload type other than PT96 or PT97 for fax relay negotiation,
or depending on whether the SIP endpoints support different payload types, configure Cisco Unified
CME to use a payload type other than PT96 or PT97 for DTMF.

Prerequisites

• Configure the codec or voice-class codec command for transcoding between G.711 and G.729. See
“Configuring Phones to Make Basic Calls” on page 147.

SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag voip
4. description string
5. destination-pattern string
6. session protocol sipv2
7. session target {dns:address | ipv4:destination-address}
8. dtmf-relay rtp-nte
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 dial-peer voice tag voip Enters dial-peer configuration mode to define a VoIP dial
peer for the voice-mail system.
Example: • tag—Defines the dial peer being configured. Range is
Router (config)# dial-peer voice 123 voip from 1 to 2147483647.

Step 4 description string (Optional) Associates a description with the dial peer being
configured. Enter a string of up to 64 characters.
Example:
Router (config-voice-dial-peer)# description CU
pilot

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Command or Action Purpose

Step 5 destination-pattern string Specifies the pattern of the numbers that the user must dial
to place a call.
Example: • string—Prefix or full E.164 number.
Router (config-voice-dial-peer)#
destination-pattern 20
Step 6 session protocol sipv2 Specifies that Internet Engineering Task Force (IETF)
Session Initiation Protocol (SIP) is protocol to be used for
calls between local and remote routers using the packet
Example:
Router (config-voice-dial-peer)# session
network.
protocol sipv2
Step 7 session target {dns:address | Designates a network-specific address to receive calls from
ipv4:destination-address} the dial peer being configured.
• dns:address—Specifies the DNS address of the
Example: voice-mail system.
Router (config-voice-dial-peer)# session target
ipv4:10.8.17.42 • ipv4:destination- address—Specifies the IP address of
voice-mail system.
Step 8 dtmf-relay rtp-nte Sets DTMF relay method for the voice dial peer being
configured.
Example: • rtp-nte— Provides conversion from the out-of-band
Router (config-voice-dial-peer)# dtmf-relay SCCP indication to the SIP standard for DTMF relay
rtp-nte (RFC2833). Forwards DTMF tones by using Real-Time
Transport Protocol (RTP) with the Named Telephone
Event (NTE) payload type.
• This command can also be configured in
voice-register-pool configuration mode. For individual
phones, the phone-level configuration for this
command overrides the system-level configuration for
this command.
Note The need to use out-of-band conversion is limited to
SCCP phones. SIP phones natively support in-band.
Step 9 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-voice-dial-peer)# end

What to Do Next

After integrating configuring DTMF relay, you are ready to configure Message Waiting Indicator (MWI)
notification for either the MWI outcall, unsolicited notify, or subscribe/notify mechanism. See the
“SCCP: Configuring a Phone Line for MWI Outcall” section on page 374.

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Enabling DTMF Integration Using SIP NOTIFY

To configure a SIP dial peer to point to Cisco Unity Express and enable SIP dual-tone multifrequency
(DTMF) relay using SIP NOTIFY format, follow the steps in this task.

SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag voip
4. description string
5. destination-pattern string
6. b2bua
7. session protocol sipv2
8. session target {dns:address | ipv4:destination-address}
9. dtmf-relay sip-notify
10. codec g711ulaw
11. no vad
12. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal#
Step 3 dial-peer voice tag voip Enters dial-peer configuration mode to define a VoIP dial
peer for the voice-mail system.
Example: • tag—Defines the dial peer being configured. Range is
Router (config)# dial-peer voice 2 voip from 1 to 2147483647.
Step 4 description string (Optional) Associates a description with the dial peer being
configured. Enter a string of up to 64 characters.
Example:
Router (config-voice-dial-peer)# description
cue pilot
Step 5 destination-pattern string Specifies the pattern of the numbers that the user must dial
to place a call.
Example: • string—Prefix or full E.164 number.
Router (config-voice-dial-peer)#
destination-pattern 20

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Command or Action Purpose

Step 6 b2bua (Optional) Includes the Cisco Unified CME address as part
of contact in 3XX response to point to Cisco Unity Express
and enables SIP-to-SCCP call forward.
Example:
Router (config-voice-dial-peer)# b2bua
Step 7 session protocol sipv2 Specifies that Internet Engineering Task Force (IETF)
Session Initiation Protocol (SIP) is protocol to be used for
calls between local and remote routers using the packet
Example:
Router (config-voice-dial-peer)# session
network.
protocol sipv2
Step 8 session target {dns:address | Designates a network-specific address to receive calls from
ipv4:destination-address} the dial peer being configured.
• dns:address—Specifies the DNS address of voice-mail
Example: system.
Router (config-voice-dial-peer)# session target
ipv4:10.5.49.80 • ipv4:destination- address—Specifies the IP address of
voice-mail system.
Step 9 dtmf-relay sip-notify Sets DTMF relay method for the voice dial peer being
configured.
Example: • sip-notify— Forwards DTMF tones using SIP NOTIFY
Router (config-voice-dial-peer)# dtmf-relay messages.
sip-notify
• This command can also be configured in
voice-register-pool configuration mode. For individual
phones, the phone-level configuration for this
command overrides the system-level configuration for
this command.
Step 10 codec g711ulaw Specifies the voice coder rate of speech for a dial peer being
configured.
Example:
Router (config-voice-dial-peer)# codec g711ulaw
Step 11 no vad Disable voice activity detection (VAD) for the calls using
the dial peer being configured.
Example:
Router (config-voice-dial-peer)# no vad
Step 12 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-voice-dial-peer)# end

What to Do Next

After configuring DTMF relay, you are ready to configure Message Waiting Indicator (MWI). See the
“SCCP: Configuring a Phone Line for MWI Outcall” section on page 374.

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SCCP: Configuring a Phone Line for MWI Outcall

To designate a phone line on an individual SCCP phone to be monitored for voice-mail messages,
perform the following steps.

Prerequisites
• Directory number and number for MWI line must be configured. For configuration information, see
“Configuring Phones to Make Basic Calls” on page 147.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. mwi-line line-number
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.

Example:
Router(config)# ephone 36
Step 4 mwi-line line-number Selects a phone line to be monitored for voice-mail messages.
• line-number—Number of phone line to receive MWI
Example: notification. Range: 1 to 34. Default: 1.
Router(config-ephone)# mwi-line 3
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

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SCCP: Configuring Audible MWI

To enable audible MWI, perform the following steps.

Prerequisites
• Directory number and number for MWI line must be configured. For configuration information, see
“Configuring Phones to Make Basic Calls” on page 147.

Restrictions
• Audible MWI is supported only in Cisco Unified CME 4.0(2) and later versions.
• Audible MWI is supported only on Cisco Unified IP Phone 7931G and Cisco Unified IP Phone
7911.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. mwi-line line-number
5. ephone-dn dn-tag
6. mwi-type {visual | audio | both}
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.

Example:
Router(config)# ephone 36
Step 4 mwi-line line-number Selects a phone line to receive MWI treatment.
• line-number—Number of phone line to receive MWI
Example: notification. Range: 1 to 34. Default: 1.
Router(config-ephone)# mwi-line 3

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Command or Action Purpose

Step 5 ephone-dn dn-tag Enters ephone-dn configuration mode to create a directory
number.
Example:
Router(config)# ephone-dn 11
Step 6 mwi-type {visual | audio | both} Specifies which type of MWI notification to be received.
Note This command can also be configured in
Example: ephone-dn-template configuration mode. The value that
Router(config-ephone-dn)# mwi-type you set in ephone-dn configuration mode has priority over
audible the value set in ephone-dn-template mode. For
configuration information, see “SCCP: Enabling
Ephone-dn Templates” on page 884.
Step 7 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

SIP: Enabling MWI at System-Level

To enable message waiting indicator (MWI) at a system-level, perform the following steps.

Prerequisites
• Cisco CME 3.4 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. mwi reg-e164
5. mwi stutter
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 mwi reg-e164 Registers full E.164 number to the MWI server in
Cisco Unified CME and enables MWI.
Example:
Router(config-register-global)# mwi reg-e164
Step 5 mwi stutter Enables Cisco Unified CME router at the central site to
relay MWI notification to remote SIP phones.
Example:
Router(config-register-global)# mwi stutter
Step 6 end Exits to privileged EXEC mode.

Example:
Router(config-register-global)# end

SIP: Configuring a Directory Number for MWI

Perform one of the following tasks, depending on whether you want to configure MWI outcall or MWI
notify (unsolicited notify or subscribe/notify) for SIP endpoints in Cisco Unified CME.
• SIP: Defining Pilot Call Back Number for MWI Outcall, page 377
• SIP: Configuring a Directory Number for MWI NOTIFY, page 378

SIP: Defining Pilot Call Back Number for MWI Outcall

To designate a phone line on an individual SIP directory number to be monitored for voice-mail
messages, perform the following steps.

Prerequisites

• Cisco CME 3.4 or a later version.

• Directory number and number for receiving MWI must be configured. For configuration
information, see “Configuring Phones to Make Basic Calls” on page 147.

Restrictions

• For Cisco Unified CME 4.1 and later versions, the Call Forward All, Presence, and
Message-Waiting Indication (MWI) features require that SIP phones must be configured with a
directory number by using the number command with the dn keyword; direct line numbers are not
supported.

SUMMARY STEPS

1. enable
2. configure terminal

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3. voice register dn dn-tag

4. mwi
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or an MWI.
Example:
Router(config-register-global)# voice register
dn 1
Step 4 mwi Enables a specific directory number to receive MWI
notification.
Example:
Router(config-register-dn)# mwi
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone-dn)# end

SIP: Configuring a Directory Number for MWI NOTIFY

To identify the MWI server and specify a directory number for receiving MWI Subscribe/NOTIFY or
MWI Unsolicited NOTIFY, follow the steps in this section.

Note We recommend using the Subscribe/NOTIFY method rather than an Unsolicited NOTIFY when
possible.

Prerequisites

• Cisco CME 3.4 or a later version.

• For Cisco Unified CME 4.0 and later, QSIQ supplementary services must be configured on the
Cisco router. For information, see “Enabling H.450.7 and QSIG Supplementary Services at a
System-Level” on page 535 or “Enabling H.450.7 and QSIG Supplementary Services on a Dial
Peer” section on page 536.
• Directory number and number for receiving MWI must be configured. For configuration
information, see “Configuring Phones to Make Basic Calls” on page 147.

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Restrictions

• For Cisco Unified CME 4.1 and later versions, the Call Forward All, Presence, and
Message-Waiting Indication (MWI) features require that SIP phones must be configured with a
directory number by using the number command with the dn keyword; direct line numbers are not
supported.
• The SIP MWI - QSIG Translation feature in Cisco Unified CME 4.1 does not support Subscribe
NOTIFY.
• Cisco Unified IP Phone 7960, 7940, 7905, and 7911 support only Unsolicited NOTIFY for MWI.

SUMMARY STEPS

1. enable
2. configure terminal
3. sip-ua
4. mwi-server {ipv4:destination-address | dns:host-name} [unsolicited]
5. exit
6. voice register dn dn-tag
7. mwi
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 sip-ua Enters Session Initiation Protocol (SIP) user agent (ua)
configuration mode for configuring the user agent.
Example:
Router(config)# sip-ua
Step 4 mwi-server {ipv4:destination-address | Specifies voice-mail server settings on a voice gateway or
dns:host-name} [unsolicited] UA.
Note The sip-server and mwi expires commands under
Example: the telephony-service configuration mode have
Router(config-sip-ua)# mwi-server been migrated to mwi-server to support DNS
ipv4:1.5.49.200
format of the SIP server.
or
Router(config-sip-ua)# mwi-server
dns:server.yourcompany.com unsolicited

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Command or Action Purpose

Step 5 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-sip-ua)# exit
Step 6 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or an MWI.
Example:
Router(config-register-global)# voice register
dn 1
Step 7 mwi Enables a specific directory number to receive MWI
notification.
Example:
Router(config-register-dn)# mwi
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-dn)# end

Enabling SIP MWI Prefix Specification

To accept unsolicited SIP Notify messages for MWI that include a prefix string as a site identifier,
perform the following steps.

Prerequisites
• Cisco Unified CME 4.0 or a later version.
• Directory number for receiving MWI Unsolicited NOTIFY must be configured. For information, see
“SIP: Configuring a Directory Number for MWI NOTIFY” section on page 378.

SUMMARY STEPS

1. enable
2. telephony-service
3. mwi prefix prefix-string
4. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 3 mwi prefix prefix-string Specifies a string of digits that, if present before a known
Cisco Unified CME extension number, should be
recognized as a prefix.
Example:
Router(config-telephony)# mwi prefix 555 • prefix-string—Digit string. The maximum prefix length
is 32 digits.
Step 4 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Verifying Voice-Mail Integration

• Press the Messages button on a local phone in Cisco Unified CME and listen for the voice mail
greeting.
• Dial an unattended local phone and listen for the voice mail greeting.
• Leave a test message.
• Go to the phone that you called. Verify that the [Message] indicator is lit.
• Press the Messages button on this phone and retrieve the voice mail message.

Configuration Examples for Voice-Mail Integration

This section contains the following examples:
• Enabling DTMF Integration for Legacy Voice-Mail Applications: Example, page 382
• Enabling Mailbox Selection Policy for SCCP Phones: Example, page 382
• Enabling DTMF Integration Using RFC 2833: Example, page 382
• Enabling DTMF Integration Using SIP Notify: Example, page 383
• Configuring a SCCP Phone Line for MWI: Example, page 383
• Enabling SIP MWI Prefix Specification: Example, page 384
• Configuring SIP Directory Number for MWI Outcall: Example, page 384
• Configuring a SIP Directory Number for MWI Unsolicited Notify: Example, page 384
• Configuring a SIP Directory Number for MWI Subscribe/NOTIFY: Example, page 385

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Configuration Examples for Voice-Mail Integration

Enabling DTMF Integration for Legacy Voice-Mail Applications: Example

The following example sets up DTMF integration for an analog voice-mail system.
vm-integration
pattern direct 2 CGN *
pattern ext-to-ext busy 7 FDN * CGN *
pattern ext-to-ext no-answer 5 FDN * CGN *
pattern trunk-to-ext busy 6 FDN * CGN *
pattern trunk-to-ext no-answer 4 FDN * CGN *

Enabling Mailbox Selection Policy for SCCP Phones: Example

The following example sets a policy to select the mailbox of the originally called number when a call is
diverted to a Cisco Unity Express or PBX voice-mail system with the pilot number 7000.
dial-peer voice 7000 voip
destination-pattern 7000
session target ipv4:10.3.34.211
codec g711ulaw
no vad
mailbox-selection orig-called-num

The following example sets a policy to select the mailbox of the last number that the call was diverted
to before being diverted to a Cisco Unity voice-mail system with the pilot number 8000.
ephone-dn 825
number 8000
mailbox-selection last-redirect-num

Configuring a Voice Mailbox: Example

The following example shows how to configure the call forward b2bua mailbox for SIP endpoints:
voice register global
voicemail 1234
!
voice register dn 2
number 2200
call-forward b2bua all 1000
call-forward b2bua mailbox 2200
call-forward b2bua noan 2201 timeout 15
mwi

Enabling DTMF Integration Using RFC 2833: Example

The following example shows the configuration for a DTMF Relay:
dial-peer voice 1 voip
destination-pattern 4…
session target ipv4:10.8.17.42
session protocol sipv2
dtmf-relay sip-notify rtp-nte

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Enabling DTMF Integration Using SIP Notify: Example

The following example shows the configuration for a DTMF relay:
dial-peer voice 1 voip
destination-pattern 4…
session target ipv4:10.5.49.80
session protocol sipv2
dtmf-relay sip-notify
b2bua

Configuring a SCCP Phone Line for MWI: Example

The following example enables MWI on ephone 18 for line 2 (button 2), which has overlaid ephone-dns.
Only a message waiting for the first ephone-dn (2021) on this line will activate the MWI lamp. Button 4
is unused. The line numbers in this example are as follows:
• Line 1—Button 1—Extension 2020
• Line 2—Button 2—Extension 2021, 2022, 2023, 2024
• Line 3—Button 3—Extension 2021, 2022, 2023, 2024 (rollover line)
• Button 4—Unused
• Line 4—Button 5—Extension 2025
ephone-dn 20
number 2020

ephone-dn 21
number 2021

ephone-dn 22
number 2022

ephone-dn 23
number 2023

ephone-dn 24
number 2024

ephone-dn 25
number 2025

ephone 18
button 1:20 2o21,22,23,24,25 3x2 5:26
mwi-line 2

The following example enables MWI on ephone 17 for line 3 (extension 609). In this example, the button
numbers do not match the line numbers because buttons 2 and 4 are not used. The line numbers in this
example are as follows:
• Line 1—Button 1—Extension 607
• Button 2—Unused
• Line 2—Button 3—Extension 608
• Button 4—Unused
• Line 3—Button 5—Extension 609

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ephone-dn 17
number 607

ephone-dn 18
number 608

ephone-dn 19
number 609

ephone 25
button 1:17 3:18 5:19
mwi-line 3

Enabling SIP MWI Prefix Specification: Example

The following example identifies the SIP server for MWI notification at the IP address 172.16.14.22. It
states that the Cisco Unified CME system will accept unsolicited SIP Notify messages for known
mailbox numbers using the prefix 555.
sip-ua
mwi-server 172.16.14.22 unsolicited

telephony-service
mwi prefix 555

Configuring SIP Directory Number for MWI Outcall: Example

The following example shows an MWI callback pilot number:
voice register dn
number 9000….
mwi on-off

Configuring a SIP Directory Number for MWI Unsolicited Notify: Example

The following example shows how to specify voice-mail server settings on a UA. The example includes
the unsolicited keyword, enabling the voice-mail server to send a SIP notification message to the UA if
the mailbox status changes and specifies that voice dn 1, number 1234 on the SIP phone in
Cisco Unified CME will receive the MWI notification:
sip-ua
mwi-server dns:server.yourcompany.com expires 60 port 5060 transport udp unsolicited

voice register dn 1
number 1234
mwi

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Additional References

Configuring a SIP Directory Number for MWI Subscribe/NOTIFY: Example

The following example shows how to define an MWI server and specify that directory number 1, number
1234 on a SIP phone in Cisco Unified CME is to receive the MWI notification:
sip-ua
mwi-server ipv4:1.5.49.200

voice register dn 1
number 1234
mwi

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Voice-Mail Integration

Feature Information for Voice-Mail Integration

Table 19 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 19 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 19 Feature Information for Voice-Mail Integration

Cisco Unified CME

Feature Name Version Feature Information
Audible MWI 4.0(2) Provides support for selecting audible, visual, or audible
and visual Message Waiting Indictor (MWI) on supported
Cisco Unified IP phones.
DTMF Integration 3.4 Added support for voice messaging systems connected via
a SIP trunk or SIP user agent.
The standard Subscribe/NOTIFY method is preferred over
an Unsolicited NOTIFY.
2.0 DTMF integration patterns were introduced.
Mailbox Selection Policy 4.0 Mailbox selection policy was introduced.
MWI 4.0 MWI line selection of phone line other than the primary
line on a SCCP phone was introduced.
3.4 Voice messaging systems (including Cisco Unity)
connected via a SIP trunk or SIP user agent can pass a
Message Waiting Indicator (MWI) that will be received and
understood by a SIP phone directly connected to
Cisco Unified CME.
SIP MWI Prefix Specification 4.0 SIP MWI prefix specification was introduced.
SIP MWI - QSIG Translation 4.1 Extends message waiting indicator (MWI) functionality for
SIP MWI and QSIG MWI interoperation to enable sending
and receiving of MWI over QSIG to PBX.

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Last Updated: June 18, 2007

This chapter describes the phone authentication support in Cisco Unified Communications Manager
Express (Cisco Unified CME) and the Media Encryption (SRTP) on Cisco Unified CME feature which
provide the following secure voice call capabilities:
• Secure call control signaling and media streams in Cisco Unified CME networks using Secure
Real-Time Transport Protocol (SRTP) and H.323 protocols.
• Secure supplementary services for Cisco Unified CME networks using H.323 trunks.
• Secure Cisco VG224 Analog Phone Gateway endpoints.

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Security” section
on page 460.

Contents
• Prerequisites for Security, page 388
• Restrictions for Security, page 388
• Information About Security, page 389
• How to Configure Security, page 402
• Configuration Examples for Security, page 438
• Where to Go Next, page 458
• Additional References, page 459
• Feature Information for Security, page 460

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Prerequisites for Security

Prerequisites for Security

Phone Authentication
• Cisco Unified CME phone authentication requires the Cisco IOS feature set Advanced Enterprise
Services (adventerprisek9) or Advanced IP Services (advipservicesk9) on supported platforms.
• Set the system clock using one of these methods:
– Configure Network Time Protocol (NTP).
– Manually set the software clock using the clock set command.
Both methods are explained in the “Performing Basic System Management” chapter of the
Cisco IOS Network Management Configuration Guide for your Cisco IOS release.

Media Encryption
• Cisco Unity 4.2 or later version
• Cisco IOS Release 12.5 or a later release on the Cisco VG224 Analog Phone Gateway
• Cisco Unified CME 4.2 or a later version

Restrictions for Security

Phone Authentication
• Cisco Unified CME phone authentication is not supported on the Cisco IAD 2400 series or the
Cisco 1700 series.

Media Encryption
• Secure three-way software conference is not supported. A secure call beginning with SRTP will
always fall back to nonsecure Real-Time Transport Protocol (RTP) when it is joined to a conference.
• If a party drops from a three-party conference, the call between the remaining two parties returns to
secure if the two parties are SRTP-capable local Skinny Client Control Protocol (SCCP) endpoints
to a single Cisco Unified CME and the conference creator is one of the remaining parties. If either
of the two remaining parties are only RTP-capable, the call remains nonsecure. If the two remaining
parties are connected through FXS, PSTN, or VoIP, the call remains nonsecure.
• Calls to Cisco Unity Express are not secure.
• Music on Hold (MOH) is not secure.
• Video calls are not secure.
• Modem relay and T.3 fax relay calls are not secure.
• Media flow-around is not supported for call transfer and call forward.
• Conversion between inband tone and RFC 2833 DTMF is not supported. RFC 2833 DTMF handling
is supported when encryption keys are sent to secure DSP farm devices but is not supported for
codec passthrough.
• Secure Cisco Unified CME does not support SIP trunks; only H.323 trunks are supported.

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Information About Security

Table 20 lists supported gateways, network modules, and codecs for Media Encryption (SRTP) on
Cisco Unified CME.

Table 20 Supported Gateways, Network Modules, and IP Phones for Media Encryption (SRTP) on
Cisco Unified CME

Supported Gateways Supported Network Modules Supported SCCP Endpoints

• Cisco 2801 • AIM-VOICE-30 • Cisco IP Phone 7931
• Cisco 2811 • AIM-ATM-VOICE-30 • Cisco IP Phone 7940
• Cisco 2821 • NM-HDA-4FXS • Cisco IP Phone 7941
• Cisco 2851 • NM-HDV • Cisco IP Phone 7941GE
• Cisco 3725 • NM-HDV2 • Cisco IP Phone 7960
• Cisco 3745 • NM-HDV2-1T1/E1 • Cisco IP Phone 7961
• Cisco 3825 • NM-HDV2-2T1/E1 • Cisco IP Phone 7961GE
• Cisco 3845 • NM-HD-1V • Cisco IP Phone 7970
• NM-HD-2V • Cisco IP Phone 7971
• NM-HD-2VE • Cisco IP Phone 7911
• PVDM2 • Cisco IP Phone 7921
• Cisco VG224 Analog Phone Gateway

Information About Security

To enable security, you should understand the following concepts:

Phone Authentication
• Phone Authentication Overview, page 390
• Public Key Infrastructure, page 391
• Phone Authentication Components, page 391
• Phone Authentication Process, page 394
• Startup Messages, page 395
• Configuration File Maintenance, page 395
• CTL File Maintenance, page 396
• CTL Client and Provider, page 396
• Manually Importing MIC Root Certificate, page 397

Media Encryption
• Feature Design of Media Encryption, page 397
• Secure Cisco Unified CME, page 398
• Secure Supplementary Services, page 399
• Secure Transcoding for Remote Phones with DSP Farm Transcoding Configured, page 400

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Information About Security

• Secure Cisco Unified CME with Cisco Unity Express, page 401
• Secure Cisco Unified CME with Cisco Unity, page 401

Phone Authentication Overview

Phone authentication is a security infrastructure for providing secure SCCP signaling between
Cisco Unified CME and IP phones. The goal of Cisco Unified CME phone authentication is to create a
secure environment for a Cisco Unified CME IP telephony system.
Phone authentication addresses the following security needs:
• Establishing the identity of each endpoint in the system
• Authenticating devices
• Providing signaling-session privacy
• Providing protection for configuration files
Cisco Unified CME phone authentication implements authentication and encryption to prevent identity
theft of the phone or Cisco Unified CME system, data tampering, call-signaling tampering, or
media-stream tampering. To prevent these threats, the Cisco Unified IP telephony network establishes
and maintains authenticated communication streams, digitally signs files before they are transferred to
phones, and encrypts call signaling between Cisco Unified IP phones.
Cisco Unified CME phone authentication depends upon the following processes:
• Phone Authentication, page 390
• File Authentication, page 390
• Signaling Authentication, page 391

Phone Authentication
The phone authentication process occurs between the Cisco Unified CME router and a supported device
when each entity accepts the certificate of the other entity; only then does a secure connection between
the entities occur. Phone authentication relies on the creation of a Certificate Trust List (CTL) file, which
is a list of known, trusted certificates and tokens. Phones communicate with Cisco Unified CME using
a secure transport-layer-session (TLS) connection, which requires that the following criteria be met:
• A certificate must exist on the phone.
• A phone configuration file must exist on the phone, and the Cisco Unified CME entry and certificate
must exist in the file.

File Authentication
The file authentication process validates digitally signed files that a phone downloads from a Trivial File
Transfer Protocol (TFTP) server—for example, configuration files, ring list files, locale files, and CTL
files. When the phone receives these types of files from the TFTP server, the phone validates the file
signatures to verify that file tampering did not occur after the files were created.

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Signaling Authentication
The signaling authentication process, also known as signaling integrity, uses the TLS protocol to
validate that signaling packets have not been tampered with during transmission. Signaling
authentication relies on the creation of the CTL file.

Public Key Infrastructure

Cisco Unified CME phone authentication uses the public-key-infrastructure (PKI) capabilities in
Cisco IOS software for certificate-based authentication of IP phones. PKI provides customers with a
scalable, secure mechanism for distributing, managing, and revoking encryption and identity
information in a secured data network. Every entity (a person or a device) participating in the secured
communication is enrolled in the PKI using a process in which the entity generates a
Rivest-Shamir-Adleman (RSA) key pair (one private key and one public key) and has its identity
validated by a trusted entity (also known as a certification authority [CA] or trustpoint).
After each entity enrolls in a PKI, every peer (also known as an end host) in a PKI is granted a digital
certificate that has been issued by a CA.
When peers must negotiate a secured communication session, they exchange digital certificates. Based
on the information in the certificate, a peer can validate the identity of another peer and establish an
encrypted session with the public keys contained in the certificate.
For more information about PKI, see the “Implementing and Managing a PKI” section of the
Cisco IOS Security Configuration Guide for your Cisco IOS release.

Phone Authentication Components

A variety of components work together to ensure secure communications in a Cisco Unified CME
system. Table 21 describes the Cisco Unified CME phone authentication components.

Table 21 Cisco Unified CME Phone Authentication Components

Component Definition
certificate An electronic document that binds a user's or device's name to its
public key. Certificates are commonly used to validate digital
signatures. Certificates are needed for authentication during secure
communication. An entity obtains a certificate by enrolling with the
CA.
signature An assurance from an entity that the transaction it accompanies is
authentic. The entity’s private key is used to sign transactions and the
corresponding public key is used for decryption.

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Table 21 Cisco Unified CME Phone Authentication Components (continued)

Component Definition
RSA key pair RSA is a public key cryptographic system developed by Ron Rivest,
Adi Shamir, and Leonard Adleman.
An RSA key pair consists of a public key and a private key. The public
key is included in a certificate so that peers can use it to encrypt data
that is sent to the router. The private key is kept on the router and used
both to decrypt the data sent by peers and to digitally sign transactions
when negotiating with peers.
You can configure multiple RSA key pairs to match policy
requirements, such as key length, key lifetime, and type of keys, for
different certificate authorities or for different certificates.
certificate server A certificate server generates and issues certificates on receipt of
trustpoint legitimate requests. A trustpoint with the same name as the certificate
server stores the certificates. Each trustpoint has one certificate plus
a copy of the CA certificate.
certification authority (CA) The root certificate server. It is responsible for managing certificate
requests and issuing certificates to participating network devices.
This service provides centralized key management for participating
devices and is explicitly trusted by the receiver to validate identities
and to create digital certificates. The CA can be a Cisco IOS CA on
the Cisco Unified CME router, a Cisco IOS CA on another router, or
a third-party CA.
registration authority (RA) Records or verifies some or all of the data required for the CA to issue
certificates. It is required when the CA is a third-party CA or
Cisco IOS CA is not on the Cisco Unified CME router.
certificate trust list (CTL) file A mandatory structure that contains the public key information
(server identities) of all the servers with which the IP phone needs to
CTL client
interact (for example, the Cisco Unified CME server, TFTP server,
CTL provider and CAPF server). The CTL file is digitally signed by the system
administrator security token (SAST).
After you configure the CTL client, it creates the CTL file and makes
it available in the TFTP directory. The CTL file is signed using the
SAST certificate’s corresponding private key. An IP phone is then
able to download this CTL file from the TFTP directory. The filename
format for each phone’s CTL file is CTLSEP<mac-addr>.tlv.
When the CTL client is run on a router in the network that is not a
Cisco Unified CME router, you must configure a CTL provider on
each Cisco Unified CME router in the network. Similarly, if a CTL
client is running on one of two Cisco Unified CME routers in a
network, a CTL provider must be configured on the other
Cisco Unified CME router. The CTL protocol transfers information to
and from the CTL provider that allows the second Cisco Unified CME
router to be trusted by phones and vice versa.
certificate revocation list File that contains certificate expiration dates and used to determine
(CRL) whether a certificate that is presented is valid or revoked.

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Table 21 Cisco Unified CME Phone Authentication Components (continued)

Component Definition
system administrator security Part of the CTL client that is responsible for signing the CTL file. The
token (SAST) Cisco Unified CME certificate and its associated key pair are used for
the SAST function. There are actually two SAST records pertaining
to two different certificates in the CTL file for security reasons. They
are known as SAST1 and SAST2. If one of the certificates is lost or
compromised, then the CTL client regenerates the CTL file using the
other certificate. When a phone downloads the new CTL file, it
verifies with only one of the two original public keys that was
installed earlier. This mechanism is to prevent IP phones from
accepting CTL files from unknown sources.
certificate authority proxy Entity that issues certificates (LSCs) to phones that request them. The
function (CAPF) CAPF is a proxy for the phones, which are unable to directly
communicate with the CA. The CAPF can also perform the following
certificate-management tasks:
• Upgrade existing locally significant certificates on the phones.
• Retrieve phone certificates for viewing and troubleshooting.
• Delete locally significant certificates on the phone.
manufacture-installed Phones need certificates to engage in secure communications. Many
certificate (MIC) phones come from the factory with MICs, but MICs may expire or
locally significant certificate become lost or compromised. Some phones do not come with MICs.
(LSC) LSCs are certificates that are issued locally to the phones using the
CAPF server.
transport Layer Security (TLS) IETF standard (RFC 2246) protocol, based on Netscape Secure
protocol Socket Layer (SSL) protocol. TLS sessions are established using a
handshake protocol to provide privacy and data integrity.
The TLS record layer fragments and defragments, compresses and
decompresses, and performs encryption and decryption of application
data and other TLS information, including handshake messages.

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Figure 19 shows the components in a Cisco Unified CME phone authentication environment.

Figure 19 Cisco Unified CME Phone Authentication

Secondary
External CA Cisco Unified
server CME router
(optional) (optional)
CA CTL provider

Primary
Cisco Unified CME router

CTL
protocol
Cisco IOS CA Cisco IOS RA

CTL file
CTL client

Signed
configuration
Secure Telephony files
CAPF server TFTP store
SCCP server services module

Signaling Certificate

TFTP server
TLS TLS

Note This illustration shows both an external

Port 2443 Port 3804 CA server and a Cisco IOS CA on the
Cisco Unified CME router. In practice,
you would have only one or the other.

146624
IP

Phone Authentication Process

The following is a high-level summary of the phone-authentication process.
To enable Cisco Unified CME phone authentication:
1. Certificates are issued.
The CA issues certificates to Cisco Unified CME, SAST, CAPF, and TFTP functions.
2. The CTL file is created, signed and published.
a. The CTL file is created by the CTL client, which is configuration driven. Its goal is to create a
CTLfile.tlv for each phone and deposit it in the TFTP directory. To complete its task, the CTL
client needs the certificates and public key information of the CAPF server, Cisco Unified CME
server, TFTP server, and SASTs.
b. The CTL file is signed by the SAST credentials. There are two SAST records pertaining to two
different certificates in the CTL file for security reasons. If one of the certificates is lost or
compromised, then the CTL client regenerates the CTL file using the other certificate. When a

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phone downloads the new CTL file, it verifies the download with only one of the two original
public keys that was installed earlier. This mechanism prevents IP phones from accepting CTL
files from unknown sources.
c. The CTL file is published on the TFTP server. Because an external TFTP server is not supported
in secure mode, the configuration files are generated by the Cisco Unified CME system itself
and are digitally signed by the TFTP server’s credentials. The TFTP server credentials can be
the same as the Cisco Unified CME credentials. If desired, a separate certificate can be
generated for the TFTP function if the appropriate trustpoint is configured under the CTL-client
interface.
3. The telephony service module signs phone configuration files and each phone requests its file.
4. When an IP phone boots up, it requests the CTL file (CTLfile.tlv) from the TFTP server and
downloads its digitally signed configuration file, which has the filename format of
SEP<mac-address>.cnf.xml.sgn.
5. The phone then reads the CAPF configuration status from the configuration file. If a certificate
operation is needed, the phone initiates a TLS session with the CAPF server on TCP port 3804 and
begins the CAPF protocol dialogue. The certificate operation can be an upgrade, delete, or fetch
operation. If an upgrade operation is needed, the CAPF server makes a request on behalf of the
phone for a certificate from the CA. The CAPF server uses the CAPF protocol to obtain the
information it needs from the phone, such as the public key and phone ID. After the phone
successfully receives a certificate from the server, the phone stores it in its flash memory.
6. With the certificate in its flash, the phone initiates a TLS connection with the secure
Cisco Unified CME server on a well-known TCP port (2443), if the device security mode settings
in the .cnf.xml file are set to authenticated or encrypted. This TLS session is mutually authenticated
by both parties. The IP phone knows the Cisco Unified CME server’s certificate from the CTL file,
which it initially downloaded from the TFTP server. The phone’s LSC is a trusted party for the
Cisco Unified CME server, because the issuing CA certificate is present in the router.

Startup Messages
If the certificate server is part of your startup configuration, you may see the following messages during
the boot procedure:
% Failed to find Certificate Server's trustpoint at startup
% Failed to find Certificate Server's cert.

These messages are informational messages that show a temporary inability to configure the certificate
server because the startup configuration has not been fully parsed yet. The messages are useful for
debugging, if the startup configuration has been corrupted.

Configuration File Maintenance

In a secure environment, several types of configuration files must be digitally signed before they can be
hosted and used. The filenames of all signed files have a .sgn suffix.
The Cisco Unified CME telephony service module creates phone configuration files (.cnf.xml suffix)
and hosts them on a Cisco IOS TFTP server. These files are signed by the TFTP server’s credentials.
In addition to the phone configuration files, other Cisco Unified CME configuration files such as the
network and user-locale files must be signed. These files are internally generated by
Cisco Unified CME, and the signed versions are automatically created in the current code path whenever
the unsigned versions are updated or created.

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Other configuration files that are not generated by Cisco Unified CME, such as ringlist.xml,
distinctiveringlist.xml, audio files, and so forth, are often used for Cisco Unified CME features. Signed
versions of these configuration files are not automatically created. Whenever a new configuration file
that has not been generated by Cisco Unified CME is imported into Cisco Unified CME, use the
load-cfg-file command, which does all of the following:
• Hosts the unsigned version of the file on the TFTP server.
• Creates a signed version of the file.
• Hosts the signed version of the file on the TFTP server.
You can also use the load-cfg-file command instead of the tftp-server command when only the
unsigned version of a file needs to be hosted on the TFTP server.

CTL File Maintenance

The CTL file contains the SAST records and other records. (A maximum of two SAST records may
exist.) The CTL file is digitally signed by one of the SAST credentials that are listed in the CTL file
before the CTL file is downloaded by the phone and saved in its flash. After receiving the CTL file, a
phone trusts a newer or changed CTL file only if it is signed by one of the SAST credentials that is
present in the original CTL file.
For this reason, you should take care to regenerate the CTL file only with one of the original SAST
credentials. If both SAST credentials are compromised and a CTL file must be generated with a new
credential, you must reset the phone to its factory defaults.

CTL Client and Provider

The CTL client generates the CTL file. The CTL client must be provided with the names of the
trustpoints it needs for the CTL file. It can run on the same router as Cisco Unified CME or on another,
standalone router. When the CTL client runs on a standalone router (not a Cisco Unified CME router),
you must configure a CTL provider on each Cisco Unified CME router. The CTL provider securely
communicates the credentials of the Cisco Unified CME server functions to the CTL client that is
running on another router.
When the CTL client is running on either a primary or secondary Cisco Unified CME router, you must
configure a CTL provider on each Cisco Unified CME router on which the CTL client is not running.
The CTL protocol is used to communicate between the CTL client and a CTL provider. Using the CTL
protocol ensures that the credentials of all Cisco Unified CME routers are present in the CTL file and
that all Cisco Unified CME routers have access to the phone certificates that were issued by the CA.
Both elements are prerequisites to secure communications.
To enable CTL clients and providers, see the “Configuring the CTL Client” section on page 419 and the
“Configuring the CTL Provider” section on page 424.

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Manually Importing MIC Root Certificate

When a phone uses a MIC for authentication during the TLS handshake with the CAPF server, the CAPF
server must have a copy of the MIC in order to verify it. Different certificates are used for different types
of IP phones.
A phone uses a MIC for authentication when it has a MIC but no LSC. For example, you have a
Cisco Unified IP Phone 7970 that has a MIC by default but no LSC. When you schedule a certificate
upgrade with the authentication mode set to MIC for this phone, the phone presents its MIC to the
Cisco Unified CME CAPF server for authentication. The CAPF server must have a copy of the MIC's
root certificate to verify the phone's MIC. Without this copy, the CAPF upgrade operation fails.
To ensure that the CAPF server has copies of the MICs it needs, you must manually import certificates
to the CAPF server. The number of certificates that you must import depends on your network
configuration. Manual enrollment refers to copy-and-paste or TFTP transfer methods.
For more information on certificate enrollment, see the “Configuring Cut-and-Paste Certificate
Enrollment” section of the “Configuring Certificate Enrollment for a PKI” chapter in the Cisco IOS
Security Configuration Guide for your Cisco IOS release.
To manually import the MIC root certificate, see the “Manually Importing MIC Root Certificate” section
on page 412.

Feature Design of Media Encryption

Companion voice security Cisco IOS features provide an overall architecture for secure end-to-end IP
telephony calls on supported network devices that enable the following:
• SRTP capable Cisco Unified CME networks with secure interoperability
• Secure Cisco IP phone calls
• Secure Cisco VG224 Analog Phone Gateway endpoints
• Secure supplementary services
We implement these features using media and signaling authentication and encryption in Cisco IOS
H.323 networks. H.323, the ITU-T standard that describes packet-based video, audio, and data
conferencing, refers to a set of other standards, including H.450, to describe its actual protocols. H.323
allows dissimilar communication devices to communicate with each other by using a standard
communication protocol, and defines a common set of codecs, call setup and negotiating procedures, and
basic data transport methods. H.450, a component of the H.323 standard, defines signaling and
procedures that are used to provide telephony-like supplementary services. We use H.450 messages in
H.323 networks to implement secure supplementary service support, and also empty capability set (ECS)
messaging for media capability negotiation.

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Secure Cisco Unified CME

The secure Cisco Unified CME solution includes secure-capable voice ports, SCCP endpoints, and a
secure H.323 trunk between Cisco Unified CME and Cisco Unified Communications Manager for audio
media. SIP trunks are not supported. Figure 20 shows the components of a secure Cisco Unified CME
system.

Figure 20 Secure Cisco Unified CME System

H.323 gateway

VoIP
Cisco VG224 analog V IP
V phone gateway Nonsecure
H.323 secure phone D
DSP farm

Cisco Unified CME 2

U
Cisco Unity
V
Cisco Unified CME 1
Cisco Unity Express Cisco 800 series router

WAN
IP IP

170910
Local phone A Local phone B IP
Remote phone C

Secure Cisco Unified CME implements call control signaling using Transport Layer Security (TLS) or
IPsec (IP Security) for the secure channel, and uses SRTP for media encryption. Secure
Cisco Unified CME manages the SRTP keys to endpoints and to gateways.
The Media Encryption (SRTP) on Cisco Unified CME feature supports the following features:
• Secure voice calls using SRTP for SCCP endpoints
• Secure voice calls in a mixed shared line environment that allows both RTP and SRTP capable
endpoints; shared line media security depends on the endpoint configuration.
• Secure supplementary services using H.450 including:
– Call forward
– Call transfer
– Call hold and resume
– Call park and call pickup
– Nonsecure software conference

Note SRTP conference calls over H.323 may experience a 0 to 2 second noise interval when the call is joined
to the conference.

• Secure calls in a nonH.450 environment

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• Secure Cisco Unified CME interaction with secure Cisco Unity

• Secure Cisco Unified CME interaction with Cisco Unity Express (interaction is supported and calls
are downgraded to nonsecure mode)
• Secure transcoding for remote phones with DSP farm transcoding configured
These features are discussed in the following sections.

Secure Supplementary Services

The Media Encryption (SRTP) feature supports secure supplementary services in both H.450 and
nonH.450 Cisco Unified CME networks. A secure Cisco Unified CME network should be either H.450
or nonH.450, not a hybrid.

Secure Cisco Unified CME in an H.450 Environment

Signaling and media encryption among secure endpoints is supported, enabling supplementary services
such as call transfer (H.450.2) and call forward (H.450.3) between secure endpoints. Call park and pick
up use H.450 messages. Secure Cisco Unified CME is H.450-enabled by default; however, secure music
on hold (MOH) and secure conferences (three-way calling) are not supported. For example, when
supplementary services are initiated as shown in Figure 21, ECS and Terminal Capabilities Set (TCS)
are used to negotiate the initially secure call between A and B down to RTP so A can hear MOH. When
B resumes the call to A, the call goes back to SRTP. Similarly, when a transfer is initiated, the party
being transferred is put on hold, and the call is negotiated down to RTP. When the call is transferred, it
goes back to SRTP if the other end is SRTP capable.

Figure 21 Music on Hold in an H.450 Environment

1 A calls B 2 B initiates supplementary

(starts as services by putting
secure call) A on hold, then calling C
IP IP IP

231361
Phone A Phone B Phone C
SRST-capable SRST-capable

3 A hears music
on hold

Secure Cisco Unified CME in a NonH.450 Environment

Security for supplementary services requires midcall key negotiation or midcall media renegotiation. In
an H.323 network where there are no H.450 messages, media renegotiation is implemented using ECS
for scenarios such as mismatched codecs and secure calls. If you disable H.450 on the router globally,
the configuration is applied to RTP and SRTP calls. The signaling path is hairpin on XOR for
Cisco Unified CME and Cisco Unified Communications Manager. For example, in Figure 22 the
signaling path goes from A through B, the supplementary services initiator, to C. When deploying voice
security in this scenario, consider that the media security keys will pass through XOR, that is, through
B, the endpoint that issued the transfer request. To avoid the man-in-the-middle attack, the XOR must
be a trusted entity.

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Figure 22 Transfer in a NonH.45 Environment

2 B initiates supplementary
1 A calls B services by calling
C to transfer the call
IP IP IP

231360
Phone A Phone B Phone C

The media path is optional. The default media path for Cisco Unified CME is hairpin. However,
whenever possible media flow around can be configured on Cisco Unified CME. When configuring
media flow through, which is the default, remember that chaining multiple XOR gateways in the media
path introduces more delay and thus reduces voice quality. Router resources and voice quality limit the
number of XOR gateways that can be chained. The requirement is platform dependent, and may vary
between signaling and media. The practical chaining level is three.
A transcoder is inserted when there is a codec mismatch and ECS and TCS negotiation fails. For
example, if Phone A and Phone B are SRTP capable, but Phone A uses the G.711 codec and Phone B
uses the G.729 codec, a transcoder is inserted if Phone B has one. However, the call is negotiated down
to RTP to fulfill the codec requirement, so the call is not secure.

Secure Transcoding for Remote Phones with DSP Farm Transcoding

Configured
Transcoding is supported for remote phones that have the dspfarm-assist keyword of the codec
command configured. A remote phone is a phone that is registered to a Cisco Unified CME and that is
residing on a remote location across the WAN. To save bandwidth across the WAN connection, calls to
such a phone can be made to use the G.729r8 codec by configuring the codec g729r8 dspfarm assist
command for the ephone. The g729r8 keyword forces calls to such a phone to use the G.729 codec. The
dspfarm-assist keyword enables using available DSP resources, if an H.323 call to the phone needs to
be transcoded.

Note Transcoding is enabled only if an H.323 call with a different codec from the remote phone tries to make
a call to the remote phone. If a local phone on the same Cisco Unified CME as the remote phone makes
a call to remote phone, the local phone is forced to change its codec to G.729 instead of using
transcoding.

Secure transcoding for point-to-point SRTP calls can only occur when both the SCCP phone which is to
be serviced by Cisco Unified CME transcoding and its peer in the call are SRTP-capable and have
successfully negotiated the SRTP keys. Secure transcoding for point-to-point SRTP calls cannot occur
when only one of the peers in the call is SRTP-capable.
If Cisco Unified CME transcoding is to be performed on a secure call, the Media Encryption (SRTP) on
Cisco Unified CME feature allows Cisco Unified CME to provide the DSP farm with the encryption
keys for the secure call as additional parameters, so that Cisco Unified CME transcoding can be
performed successfully. Without the encryption keys, the DSP farm would not be able to read the
encrypted voice data in order to transcode it.

Note The secure transcoding described here does not apply to IP-IP gateway transcoding.

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Cisco Unified CME transcoding is different from IP-to-IP gateway transcoding because it is invoked for
an SCCP endpoint only, instead of for bridging VoIP call legs. Cisco Unified CME transcoding and
IP-to-IP gateway transcoding are mutually exclusive, that is, only one type of transcoding can be
invoked for a call. If no DSP farm capable of SRTP transcoding is available, Cisco Unified CME secure
transcoding is not performed and the call goes through using G.711.

Secure Cisco Unified CME with Cisco Unity Express

Note Cisco Unity Express does not support secure signaling and media encryption. Secure
Cisco Unified CME interoperates with Cisco Unity Express, but calls between Cisco Unified CME and
Cisco Unity Express are not secure.

In a typical Cisco Unity Express deployment with Cisco Unified CME in a secure H.323 network,
Session Initiation Protocol (SIP) is used for signaling, and the media path is G.711 with RTP. For Call
Forward No Answer (CFNA) and Call Forward All (CFA), before the media path is established,
signaling messages are sent to negotiate an RTP media path. If codec negotiation fails, a transcoder is
inserted. The Media Encryption (SRTP) on Cisco Unified CME feature’s H.323 service provider
interface (SPI) supports fast start calls. In general, calls transferred or forwarded back to
Cisco Unified CME from Cisco Unity Express fall into existing call flows and are treated as regular SIP
and RTP calls.
The Media Encryption (SRTP) on Cisco Unified CME feature supports blind transfer back to
Cisco Unified CME only. When midcall media renegotiation is configured, the secure capability for the
endpoint is renegotiated regardless of which transfer mechanism, H.450.2 or Empty Capability Set
(ECS), was used.

Secure Cisco Unified CME with Cisco Unity

The Media Encryption (SRTP) on Cisco Unified CME feature supports Cisco Unity 4.2 or a later version
and Cisco Unity Connection 1.1 or a later version using SCCP. Secure Cisco Unity for
Cisco Unified CME acts like a secure SCCP phone. Some provisioning is required before secure
signaling can be established. Cisco Unity receives Cisco Unified CME device certificates from the
Certificate Trust List (CTL) and Cisco Unity certificates are inserted into Cisco Unified CME manually.
Cisco Unity with SIP is not supported.
The certificate for the Cisco Unity Connection is in the Cisco Unity administration web application
under the “port group settings.”

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How to Configure Security

This section contains the following tasks:

Phone Authentication
• Configuring the Cisco IOS Certification Authority, page 402 (required)
• Verifying the Cisco IOS Certification Authority, page 406 (optional)
• Configuring the Registration Authority, page 406 (optional)
• Verifying the Registration Authority, page 409 (optional)
• Authenticating Certificates for Server Functions, page 409 (required)
• Verifying Certificates for Server Functions, page 412 (optional)
• Manually Importing MIC Root Certificate, page 412 (optional)
• Configuring Telephony-Service Security Parameters, page 414 (required)
• Verifying Telephony-Service Security Parameters, page 419 (optional)
• Configuring the CTL Client, page 419 (required)
• Verifying the CTL Client, page 424 (optional)
• Configuring the CTL Provider, page 424 (optional)
• Verifying the CTL Provider, page 426 (optional)
• Configuring the CAPF Server, page 426 (required)
• Verifying the CAPF Server, page 429 (optional)
• Entering the Authentication String on the Phone, page 430 (optional)
• Verifying the Authentication String on the Phone, page 431 (optional)

Media Encryption
• Configuring Secure Calls Between Cisco Unified CMEs Across an H.323 Trunk, page 431
(required)
• Configuring Cisco Unified CME SRTP Fallback for H.323 Dial Peers, page 433 (optional)
• Configuring Cisco Unity for Secure Cisco Unified CME Operation, page 435 (optional)

Configuring the Cisco IOS Certification Authority

To configure a root certificate server, also called a certification authority (CA), on a Cisco IOS router,
perform the following steps. The router can be the Cisco Unified CME router or an external router.
Setting up a Cisco IOS CA is a standard PKI task. The basic steps are included here for ease of use. For
more information, see the “Configuring and Managing a Cisco IOS Certificate Server for PKI
Deployment” section in “Part 5: Implementing and Managing a PKI” in the Cisco IOS Security
Configuration Guide for your Cisco IOS release.

Note If you use a third-party CA, follow the provider’s instructions instead of performing these steps.

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SUMMARY STEPS

1. enable
2. configure terminal
3. ip http server
4. crypto pki server label
5. database level {minimal | names | complete}
6. database url root-url
7. lifetime certificate time
8. issuer-name CN=label
9. exit
10. crypto pki trustpoint label
11. enrollment url ca-url
12. exit
13. crypto pki server label
14. grant auto
15. no shutdown
16. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
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 local
Cisco Unified CME router.
Example:
Router(config)# ip http server
Step 4 crypto pki server label Defines a label for the certificate server and enters
certificate-server configuration mode.
Example: • label—Name for CA certificate server.
Router(config)# crypto pki server sanjose1

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Command or Action Purpose

Step 5 database level {minimal | names | complete} (Optional) Controls the type of data stored in the certificate
enrollment database.
Example: • minimal—Enough information is stored only to
Router(config-cs-server)# database level continue issuing new certificates without conflict. This
complete is the default value.
• names—In addition to the minimal information given,
the serial number and subject name of each certificate.
• complete—In addition to the information given in the
minimal and names levels, each issued certificate is
written to the database.
Note The complete keyword produces a large amount of
information; so specify an external TFTP server in
which to store the data by using the database url
command.
Step 6 database url root-url (Optional) Specifies the location where all database entries
for the certificate server are to be written out. If this
command is not specified, all database entries are written to
Example:
Router(config-cs-server)# database url nvram:
NVRAM.
• root-url—Location where database entries will be
written out. The URL can be any URL that is supported
by the Cisco IOS file system.
Note If the CA is going to issue a large number of
certificates, select an appropriate storage location
like flash or other storage device to store the
certificates.

Note When the storage location chosen is flash and the

file system type on this device is Class B (LEFS),
make sure to check free space on the device
periodically and use the squeeze command to free
the space used up by deleted files. This process may
take several minutes and should be done during
scheduled maintenance periods or off-peak hours.
Step 7 lifetime certificate time (Optional) Specifies the lifetime, in days, of certificates
issued by this CA server.
Example: • time—Number of days until a certificate expires. Range
Router(config-cs-server) lifetime certificate is 1 to 1825. Default is 1 year. The maximum certificate
888 lifetime is 1 month less than the lifetime of the CA
certificate.
Note If you want to use this command is used, use it
before the server is enabled with the no shutdown
command.
Step 8 issuer-name CN=name (Optional) Specifies a distinguished name (DN) as the
certification-authority (CA) issuer name for the certificate
server.
Example:
Router(config-cs-server)# issuer-name If the issuer name is not configured, CN = CA label.
CN=sanjose1

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Command or Action Purpose

Step 9 exit Exits certificate-server configuration mode.

Example:
Router(config-cs-server)# exit
Step 10 crypto pki trustpoint label (Optional) Declares a trustpoint and enters ca-trustpoint
configuration mode.
Example: • label—Name for the trustpoint.
Router(config)# crypto pki trustpoint sanjose1
• A trustpoint for the CA is automatically generated by
the router when the CA is started. If you must use a
specific RSA key for the CA, you can create your own
trustpoint by using the same label used in the crypto
pki server command in Step 13. If the router sees a
configured trustpoint with the same label as that of the
“crypto pki server,” it uses this trustpoint and does not
automatically create a trustpoint.
Note Use this command and the enrollment url
command if this CA is local to the
Cisco Unified CME router. These commands are
not needed for a CA on an external router.
Step 11 enrollment url ca-url Specifies the enrollment URL of the issuing CA certificate
server (root certificate server).
Example: • ca-url—URL of the router on which the root CA is
Router(config-ca-trustpoint)# enrollment url installed.
http://ca-server.company.com
Step 12 exit Exits ca-trustpoint configuration mode.

Example:
Router(config-ca-trustpoint)# exit
Step 13 crypto pki server label Enters certificate-server configuration mode.
• label—Name for CA certificate server.
Example:
Router(config)# crypto pki server sanjose1
Step 14 grant auto (Optional) Allows certificates to be issued automatically to
any requester.
Example: • Default and recommended method is manual
Router(config-cs-server)# grant auto enrollment.
Tip Use this command only when testing and building
simple networks. Use the no grant auto command
after configuration is complete to prevent
certificates from being automatically granted.

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Command or Action Purpose

Step 15 no shutdown (Optional) Enables the CA.
Note You should use this command only after you have
Example: completely configured the CA.
Router(config-cs-server)# no shutdown
Step 16 end Returns to privileged EXEC mode.

Example:
Router(config-cs-server)# end

Verifying the Cisco IOS Certification Authority

Step 1 Use the show crypto pki server command to display the status of the certificate server.
Step 2 Use the show running-config command to display the running configuration, including the
certificate-server configuration.
The following example defines a CA named authority1 running locally on the Cisco Unified CME
router:
ip http server

crypto pki server authority1

database level complete
database url nvram:

crypto pki trustpoint authority1

enrollment url http://ca-server.company.com

crypto pki server authority1

no grant auto
no shutdown

Configuring the Registration Authority

This task is required if the CA is a third-party CA or if the CA is on a Cisco IOS router external to the
Cisco Unified CME router. In these cases, the CAPF server requires an RA to issue certificates to
phones.
The RA is the authority charged with recording or verifying some or all of the data required for the CA
to issue certificates. In many cases the CA undertakes all of the RA functions itself, but where a CA
operates over a wide geographical area or when there is security concern over exposing the CA at the
edge of the network, it may be advisable to delegate some of the tasks to an RA and let the CA
concentrate on its primary tasks of signing certificates.
You can configure a Cisco IOS certificate server to run in RA mode. When the RA receives a manual or
Simple Certificate Enrollment Protocol (SCEP) enrollment request, the administrator can either reject
or grant it on the basis of local policy. If the request is granted, it is forwarded to the issuing CA, and the
CA automatically generates the certificate and returns it to the RA. The client can later retrieve the
granted certificate from the RA.

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To configure an RA, perform the following steps on the Cisco Unified CME router.

SUMMARY STEPS

1. enable
2. configure terminal
3. crypto pki trustpoint label
4. enrollment url ca-url
5. revocation-check method1 [method2 [method3]]
6. serial-number [none]
7. rsakeypair key-label [key-size [encryption-key-size]]
8. exit
9. crypto pki server label
10. mode ra
11. lifetime certificate time
12. grant auto
13. no shutdown
14. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 crypto pki trustpoint label Declares the trustpoint that your RA mode certificate server
should use and enters CA-trustpoint configuration mode.
Example: • label—Name for the trustpoint and RA. The
Router(config)# crypto pki trustpoint ra12 certificate-server label that you use here is also used in
the crypto pki server command in Step 9.
Note This name is also specified in the
cert-enroll-trustpoint command when you set up
the CA proxy as described in the “Configuring the
CAPF Server” section on page 426.
Step 4 enrollment url ca-url Specifies the enrollment URL of the issuing CA certificate
server (root certificate server).
Example: • ca-url—URL of the router on which the root CA has
Router(config-ca-trustpoint)# enrollment url been installed.
http://ca-server.company.com

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Command or Action Purpose

Step 5 revocation-check method1 [method2 [method3]] (Optional) Checks the revocation status of a certificate and
specifies one or more methods to check the status. If a
second and third method are specified, each method is used
Example:
Router(config-ca-trustpoint)# revocation-check
only if the previous method returns an error, such as a server
none being down.
Valid values for methodn are as follows:
• crl—Certificate checking is performed by a certificate
revocation list (CRL). This is the default behavior.
• none—Certificate checking is not required.
• ocsp—Certificate checking is performed by an Online
Certificate Status Protocol (OCSP) server.
Step 6 serial-number [none] (Optional) Specifies whether the router serial number
should be included in the certificate request. When this
command is not used, you are prompted for the serial
Example:
Router(config-ca-trustpoint)# serial-number
number during certificate enrollment.
• none—(Optional) A serial number is not included in
the certificate request.
Step 7 rsakeypair key-label [key-size (Optional) Specifies an RSA key pair to use with a
[encryption-key-size]] certificate.
• key-label—Name of the key pair, which is generated
Example: during enrollment if it does not already exist or if the
Router(config-ca-trustpoint)# rsakeypair auto-enroll regenerate command is used.
exampleCAkeys 1024 1024
• key-size—(Optional) Size of the desired RSA key. If
not specified, the existing key size is used.
• encryption-key-size—(Optional) Size of the second
key, which is used to request separate encryption,
signature keys, and certificates.
Note Multiple trustpoints can share the same key.
Step 8 exit Exits ca-trustpoint configuration mode.

Example:
Router(config-ca-trustpoint)# exit
Step 9 crypto pki server label Defines a label for the certificate server and enters
certificate-server configuration mode.
Example: • label—Name for the trustpoint and RA. The
Router(config)# crypto pki server ra12 certificate-server label must have the same name as the
trustpoint that was created in Step 3.
Step 10 mode ra Places the PKI server into certificate-server mode for the
RA.
Example:
Router(config-cs-server)# mode ra

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Command or Action Purpose

Step 11 lifetime certificate time (Optional) Specifies the lifetime, in days, of a certificate.
• time—Number of days until the certificate expires.
Example: Range is 1 to 1825. Default is 1 year. The maximum
Router(config-cs-server)# lifetime certificate certificate lifetime is 1 month less than the lifetime of
1800 the CA certificate.
Note If this command is used, it must be used before the
server is enabled with the no shutdown command.
Step 12 grant auto Allows a certificate to be issued automatically to any
requester.
Example: Note Use this command only during enrollment when
Router(config-cs-server)# grant auto testing and building simple networks. As a security
best practice, disable this functionality after
configuration using the no grant auto command so
that certificates are not continually granted.
Step 13 no shutdown (Optional) Enables the certificate server.
You are prompted to provide input regarding acceptance of
Example: the CA certificate, the router certificate, the challenge
Router(config-cs-server)# no shutdown password, and a password for protecting the private key.
Note Use this command only after you have completely
configured your certificate server.
Step 14 end Returns to privileged EXEC mode.

Example:
Router(config-cs-server)# end

Verifying the Registration Authority

Step 1 Use the show crypto pki server command to display the status of the certificate server.
Step 2 Use the show crypto pki certificates command to display certificate information.
Step 3 Use the show running-config command to display the running configuration.

Authenticating Certificates for Server Functions

The Cisco Unified CME router needs certificates for the following server functions:
• Secure SCCP server (Cisco Unified CME)—Requires a certificate for TLS sessions with phones.
• TFTP server credentials—Requires a key pair and certificate for signing configuration files.
• CAPF server—Requires a certificate for TLS sessions with phones.
• Security tokens—Required for signing the CTL file. We recommend creating two certificates, one
for primary use and the other for backup.

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To obtain a certificate for each of these functions, perform the following steps for each server function.

SUMMARY STEPS

1. enable
2. configure terminal
3. crypto pki trustpoint trustpoint-label
4. enrollment url url
5. revocation-check method1 [method2 [method3]]
6. rsakeypair key-label [key-size [encryption-key-size]]
7. exit
8. crypto pki authenticate trustpoint-label
9. crypto pki enroll trustpoint-label
10. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 crypto pki trustpoint trustpoint-label Declares the trustpoint that the Cisco Unified CME
certificate server should use and enters ca-trustpoint
configuration mode.
Example:
Router(config)# crypto pki trustpoint capf • trustpoint-label—Label for the trustpoint.
Step 4 enrollment url url Specifies the enrollment URL of the issuing CA certificate
server (root certificate server).
Example: • url—URL of the router on which the root CA is
Router(config-ca-trustpoint)# enrollment url installed.
http://ca-server.company.com

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Command or Action Purpose

Step 5 revocation-check method1 [method2 [method3]] (Optional) Checks the revocation status of a certificate.
• method—Method used by the router to check the
Example: revocation status of the certificate. If a second and third
Router(config-ca-trustpoint)# revocation-check method are specified, each method is used only if the
none previous method returns an error, such as a server being
down.
– crl—Certificate checking is performed by a
certificate revocation list (CRL). This is the default
behavior.
– none—Certificate checking is not required.
– ocsp—Certificate checking is performed by an
Online Certificate Status Protocol (OCSP) server.
Step 6 rsakeypair key-label [key-size (Optional) Specifies a key pair to use with a certificate.
[encryption-key-size]]
• key-label—Name of the key pair, which is generated
during enrollment if it does not already exist or if the
Example: auto-enroll regenerate command is configured.
Router(config-ca-trustpoint)# rsakeypair capf
1024 1024 • key-size—(Optional) Size of the desired RSA key. If
not specified, the existing key size is used.
• encryption-key-size—(Optional) Size of the second
key, which is used to request separate encryption,
signature keys, and certificates.
Note Multiple trustpoints can share the same key.
Step 7 exit Exits CA trustpoint configuration mode.

Example:
Router(config-ca-trustpoint)# exit
Step 8 crypto pki authenticate trustpoint-label Retrieves the CA certificate and authenticates it. Checks the
certificate fingerprint if prompted.
Example: • trustpoint-label—Trustpoint label.
Router(config)# crypto pki authenticate capf
Note This command is optional if the CA certificate is
already loaded into the configuration.
Step 9 crypto pki enroll trustpoint-label Enrolls with the CA and obtains the certificate for this
trustpoint.
Example: • trustpoint-label—Trustpoint label.
Router(config)# crypto pki enroll capf
Step 10 exit Returns to privileged EXEC mode.

Example:
Router(config)# exit

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Verifying Certificates for Server Functions

Step 1 Use the show crypto pki certificates command to display information about the certificates.
Step 2 Use the show running-config command to display the running configuration.

Manually Importing MIC Root Certificate

The MIC root certificate must be present in the Cisco Unified CME router to allow Cisco Unified CME
to authenticate the MIC that is presented to it. To manually import the MIC root certificate on the
Cisco Unified CME router, perform the following steps for each type of phone that requires a MIC for
authentication.

Prerequisites
One of the following situations must be true before you perform this task:
• You choose to use MIC as the method for phone authentication during CAPF certificate operation
• You plan to establish the TLS session for SCCP signaling using the phone’s MIC instead of an LSC

SUMMARY STEPS

1. enable
2. configure terminal
3. crypto pki trustpoint name
4. revocation-check method1
5. enrollment terminal
6. exit
7. crypto pki authenticate name
8. Open the MIC root file and copy the certificate.
9. When prompted, paste the certificate, press Enter, and type quit.
10. Enter y to accept the certificate.
11. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 crypto pki trustpoint name Declares the CA that your router should use and enters
CA-trustpoint configuration mode.
Example: • name—CA trustpoint name.
Router(config)# crypto pki trustpoint sanjose1
Step 4 revocation-check method1 Checks the revocation status of a certificate.
• method1—The method used by the router to check the
Example: revocation status of the certificate. For this task, the
Router(ca-trustpoint)# revocation-check none only available method is none. The keyword none is
required for this task and means that a revocation check
is not performed and the certificate is always accepted.
Step 5 enrollment terminal Specifies manual (copy-and-paste) certificate enrollment.

Example:
Router(ca-trustpoint)# enrollment terminal
Step 6 exit Exits CA-trustpoint configuration mode.

Example:
Router(ca-trustpoint)# exit
Step 7 crypto pki authenticate name Authenticates the CA (by getting the certificate from the
CA).
Example: • name—Name of the CA.
Router(config)# crypto pki authenticate
sanjose1
Step 8 Open the MIC root file and copy the certificate. The MIC root file is a file with name a*.0, located in the
directory C:\Program Files\Cisco\Certificates
Copy to a buffer or temporary location all of the contents
that appear between “-----BEGIN CERTIFICATE-----” and
“-----END CERTIFICATE-----”.
Step 9 When prompted, paste the certificate, press Enter, and Paste the text from the a*.0 file, press Enter after pasting the
type quit. certificate, and type quit on a line by itself.

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Command or Action Purpose

Step 10 Enter y to accept the certificate. The system responds to the pasted certificate text by
providing the MD5 and SHA1 fingerprints, and asks
whether you accept the certificate.
Enter y to accept the certificate or n to reject it.
Step 11 exit Returns to privileged EXEC mode.

Example:
Router(config)# exit

Configuring Telephony-Service Security Parameters

To enable telephony-service security parameters, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. secure-signaling trustpoint label
5. tftp-server-credentials trustpoint label
6. device-security-mode {authenticated | none | encrypted}
7. cnf-file perphone
8. load-cfg-file file-url alias file-alias [sign] [create]
9. server-security-mode {secure | non-secure}
10. exit
11. ephone phone-tag
12. device-security-mode {authenticated | none | encrypted}
13. codec {g711ulaw | g729r8 [dspfarm-assist]}
14. capf-auth-str digit-string
15. cert-oper {delete | fetch | upgrade} auth-mode {auth-string | LSC | MIC | null-string}
16. reset
17. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 secure-signaling trustpoint label Specifies the name of the PKI trustpoint that has the valid
certificate to be used for TLS handshakes with IP phones on
TCP port 2443.
Example:
Router(config-telephony)# secure-signaling • label—Name of a configured PKI trustpoint with a
trustpoint cme-sccp valid certificate.
Step 5 tftp-server-credentials trustpoint label Specifies the name of the PKI trustpoint to be used to sign
the phone configuration files. This can be the CAPF-server
trustpoint that was used in the previous step or any
Example:
Router(config-telephony)#
trustpoint with a valid certificate.
tftp-server-credentials trustpoint cme-tftp • label—Name of a configured PKI trustpoint with a
valid certificate.
Step 6 device-security-mode {authenticated | none | Enables security mode for all security-capable phones in the
encrypted} system.
• authenticated—SCCP signaling between a device and
Example: Cisco Unified CME takes place through the secure TLS
Router(config-telephony)# device-security-mode connection on TCP port 2443.
authenticated
• none—SCCP signaling is not secure. This is the
default.
• encrypted—SCCP signaling between a device and
Cisco Unified CME takes place through the secure TLS
connection on TCP port 2443, and the media uses
Secure Real-Time Transport Protocol (SRTP). Use the
encrypted keyword to enable Secure
Cisco Unified CME functionality.
Note You can override the setting you make in this
command for individual ephones by using the
device-security-mode command in ephone
configuration mode.

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Command or Action Purpose

Step 7 cnf-file perphone Specifies the generation of a separate configuration file for
each individual phone. Separate configuration files for each
endpoint are required for security.
Example:
Router(config-telephony)# cnf-file perphone
Step 8 load-cfg-file file-url alias file-alias [sign] (Optional) Signs configuration files that are not created by
[create] Cisco Unified CME. Also loads the signed and unsigned
versions of a file on the TFTP server. To serve an already
Example: signed file on the TFTP server, use this command without
Router(config-telephony)# load-cfg-file the sign and create keywords.
slot0:Ringlist.xml alias Ringlist.xml sign
• file-url—Complete path of a configuration file in a
create
local directory.
• alias file-alias—Alias name of the file to be served on
the TFTP server.
• sign—(Optional) The file needs to be digitally signed
and served on the TFTP server.
• create—(Optional) Creates the signed file in the local
directory.
Note The first time that you use this command for each
file, use the create keyword in addition to the sign
keyword. The create keyword is not maintained in
the running configuration to prevent signed files
from being recreated during every reload.
Step 9 server-security-mode {secure | non-secure} (Optional) Changes the security mode of the server.
• secure—Secure mode.
Example: • non-secure—Nonsecure mode.
Router(config-telephony)# server-security-mode
secure Note This command has no impact until the CTL file is
initially generated by the CTL client. When the CTL
file is generated, the CTL client automatically sets
server security mode to secure.

Note This command must be followed by the regenerate

command in CTL-client configuration mode.
Step 10 exit Exits telephony-service configuration mode.

Example:
Router(config)# exit
Step 11 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Identifier of the ephone to be configured.
Example:
Router(config)# ephone 24

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Command or Action Purpose

Step 12 device-security-mode {authenticated | none | (Optional) Sets the security mode for SCCP signaling for an
encrypted} ephone communicating with the Cisco Unified CME router.
• authenticated—SCCP signaling between a device and
Example: Cisco Unified CME takes place through the secure TLS
Router(config-ephone)# device-security-mode connection on TCP port 2443.
authenticated
• none—SCCP signaling is not secure.
• encrypted—SCCP signaling between a device and
Cisco Unified CME takes place through the secure TLS
connection on TCP port 2443, and the media uses
Secure Real-Time Transport Protocol (SRTP). Use the
encrypted keyword to enable Secure
Cisco Unified CME functionality.
Note You can set this value globally using the
device-security-mode command in
telephony-service configuration mode. A per-phone
setting in ephone configuration mode overrides the
global setting for that phone.
Step 13 codec {g711ulaw | g729r8 [dspfarm-assist]} (Optional) Sets the security mode for SCCP signaling for a
phone communicating with the Cisco Unified CME router.
Example: • dspfarm-assist—Attempts to use DSP-farm resources
Router(config-ephone)# codec g711ulaw for transcoding the segment between the phone and the
dspfarm-assist Cisco Unified CME router if G.711 is negotiated for
the call. The dspfarm-assist keyword is ignored if the
SCCP endpoint type is ATA, VG224, or VG248.
Note The dspfarm-assist keyword is required for secure
transcoding with Cisco Unified CME to work.
Step 14 capf-auth-str digit-string (Optional) Defines a string to use as a personal
identification number (PIN) for CAPF authentication. Use
the show capf-server auth-string command to display
Example:
Router(config-ephone)# capf-auth-str 2734
configured strings. For instructions on how to enter the
string from the phone, see the “Entering the Authentication
String on the Phone” section on page 430.
• digit-string—String of digits that the phone user must
dial for CAPF authentication. The string can be from
4 to 10 digits in length.
Note You can set this value globally using this command
or per ephone using the auth-string command in
CAPF-server configuration mode.

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Command or Action Purpose

Step 15 cert-oper {delete | fetch | upgrade} auth-mode (Optional) Initiates the indicated certificate operation on
{auth-string | LSC | MIC | null-string} this ephone.
• delete—Removes the phone certificate.
Example:
Router(config-ephone)# cert-oper upgrade
• fetch—Retrieves the phone certificate for
auth-mode auth-string troubleshooting.
• upgrade—Upgrades the phone certificate.
• auth-mode—Type of authentication to use during
CAPF sessions to verify endpoints that request
certificates.
• auth-string—Phone user enters a special
authentication string at the phone. The string is set with
the capf-auth-str command and is provided to the
phone user by the system administrator. See the
“Entering the Authentication String on the Phone”
section on page 430.
• LSC—Phone provides its phone certificate for
authentication. Precedence is given to an LSC if one
exists.
• MIC—Phone provides its phone certificate for
authentication. Precedence is given to an MIC if one
exists. If this option is chosen, the MIC’s issuer
certificate must be imported into a PKI trustpoint. See
the “Manually Importing MIC Root Certificate” section
on page 412.
• null-string—No authentication.
Note You can initiate certificate operations globally using
the cert-oper command in CAPF-server
configuration mode. You can set authentication
mode globally using the auth-mode command in
CAPF-server configuration mode.
Step 16 reset Performs a complete reboot of the phone.

Example:
Router(config-ephone)# reset
Step 17 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

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Verifying Telephony-Service Security Parameters

Step 1 show telephony-service security-info
Use this command to display the security-related information that is configured in telephony-service
configuration mode.
Router# show telephony-service security-info

Skinny Server Trustpoint for TLS: cme-sccp

TFTP Credentials Trustpoint: cme-tftp
Server Security Mode: Secure
Global Device Security Mode: Authenticated

Step 2 show capf-server auth-string

Use this command to display authentication strings for phones.
Router# show capf-server auth-string

Authentication Strings for configured Ephones

Mac-Addr Auth-String
-------- -----------
000CCE3A817C 2734
001121116BDD 922
000D299D50DF 9182
000ED7B10DAC 3114
000F90485077 3328
0013C352E7F1 0678

Step 3 show running-config

Use this command to display the running configuration to verify telephony and per-phone security
configuration.
Router# show running-config

telephony-service
secure-signaling trustpoint cme-sccp
server-security-mode secure
device-security-mode authenticated
tftp-server-credentials trustpoint cme-tftp
.
.
.

Configuring the CTL Client

The tasks to configure the CTL client differ slightly depending on whether the CTL client is running on
the same router as Cisco Unified CME. Choose the appropriate procedure based on your network:
• Configuring the CTL Client on a Cisco Unified CME Router, page 420
• Configuring the CTL Client on a Router Other Than a Cisco Unified CME Router, page 422

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Configuring the CTL Client on a Cisco Unified CME Router

The credentials of various functions are included in the CTL file, which is created and hosted on the
TFTP server. To configure a CTL client on a Cisco Unified CME router, perform the following steps.
If you have primary and secondary Cisco Unified CME routers, you can configure the CTL client on
either one of them.

SUMMARY STEPS

1. enable
2. configure terminal
3. ctl-client
4. sast1 trustpoint trustpoint-label
5. sast2 trustpoint trustpoint-label
6. server {capf | cme | cme-tftp | tftp} ip-address trustpoint trustpoint-label
7. server cme ip-address username string password 0 string
8. regenerate
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ctl-client Enters CTL-client configuration mode.

Example:
Router(config)# ctl-client
Step 4 sast1 trustpoint label Configures credentials for the primary SAST.
• label—SAST1 trustpoint name.
Example: Note SAST1 and SAST2 certificates must be different
Router(config-ctl-client)# sast1 trustpoint
sast1tp
from each other. The CTL file is always signed by
SAST1. The SAST2 credentials are included in the
CTL file, so that if the SAST1 certificate is
compromised, the file can be signed by SAST2 to
prevent phones from being reset to the factory
default.

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Command or Action Purpose

Step 5 sast2 trustpoint label Configures credentials for the secondary SAST.
• label—SAST2 trustpoint name.
Example: Note SAST1 and SAST2 certificates must be different
Router(config-ctl-client)# sast2 trustpoint
from each other. The CTL file is always signed by
SAST1. The SAST2 credentials are included in the
CTL file, so that if the SAST1 certificate is
compromised, the file can be signed by SAST2 to
prevent phones from being reset to the factory
default.
Step 6 server {capf | cme | cme-tftp | tftp} Configures a trustpoint for each server function that is
ip-address trustpoint trustpoint-label running locally on the Cisco Unified CME router.
Note Repeat this command with the appropriate keyword
Example: for each function that is running locally on the
Router(config-ctl-client)# server capf 10.2.2.2 Cisco Unified CME router.
trustpoint capftp
• capf—CAPF server.
• cme—Cisco Unified CME router.
• cme-tftp—Combined Cisco Unified CME router and
TFTP server.
• tftp—TFTP server.
• ip-address—IP address of the Cisco Unified CME
router. If there are multiple network interfaces, use the
interface address in the local LAN to which the phones
are connected.
• trustpoint trustpoint-label—Name of the PKI
trustpoint for the entity.
Step 7 server cme ip-address username name-string (Optional) Provides information about another
password {0 | 1} password-string Cisco Unified CME router (primary or secondary) in the
network, if one exists.
Example: • ip-address—IP address of the other
Router(config-ctl-client)# server cme 10.2.2.2 Cisco Unified CME router.
username user3 password 0 38h2KL
• username name-string—Username that is configured
on the CTL provider.
• password—Encryption status of the password string.
– 0—Not encrypted.
– 1—Encrypted using Message Digest 5 (MD5).
Note This option refers to the way that you want the
password to appear in show command output and
not to the way that you enter the password.

• password-string—Administrative password of the CTL

provider running on the remote Cisco Unified CME
router.

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Command or Action Purpose

Step 8 regenerate Creates a new CTLFile.tlv after you make changes to the
CTL client configuration.
Example:
Router(config-ctl-client)# regenerate
Step 9 end Returns to privileged EXEC mode.

Example:
Router(config-ctl-client)# end

What to do Next
When you have more than one Cisco Unified CME router in your network, you must configure a CTL
provider on each Cisco Unified CME router that is not running the CTL client. See the “Configuring the
CTL Provider” section on page 424.

Configuring the CTL Client on a Router Other Than a Cisco Unified CME Router
To configure a CTL client on an external router that is not a Cisco Unified CME router, perform the
following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. ctl-client
4. sast1 trustpoint trustpoint-label
5. sast2 trustpoint trustpoint-label
6. server cme ip-address username name-string password {0 | 1} password-string
7. regenerate
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 ctl-client Enters CTL-client configuration mode.

Example:
Router(config)# ctl-client
Step 4 sast1 trustpoint label Configures credentials for the primary SAST.
• label—SAST1 trustpoint name.
Example: Note SAST1 and SAST2 certificates must be different
Router(config-ctl-client)# sast1 trustpoint
sast1tp
from each other, but either of them may use the
same certificate as the Cisco Unified CME router to
conserve memory. The CTL file is always signed by
SAST1. The SAST2 credentials are included in the
CTL file, so that if the SAST1 certificate is
compromised, the file can be signed by SAST2 to
prevent phones from being reset to the factory
default.
Step 5 sast2 trustpoint label Configures credentials for the secondary SAST.
• label—SAST2 trustpoint name.
Example: Note SAST1 and SAST2 certificates must be different
Router(config-ctl-client)# sast2 trustpoint
from each other, but either of them may use the
same certificate as the Cisco Unified CME router to
conserve memory. The CTL file is always signed by
SAST1. The SAST2 credentials are included in the
CTL file, so that if the SAST1 certificate is
compromised, the file can be signed by SAST2 to
prevent phones from being reset to the factory
default.
Step 6 server cme ip-address username name-string (Optional) Provides information about another
password {0 | 1} password-string Cisco Unified CME router (primary or secondary) in the
network, if one exists.
Example: • ip-address—IP address of the other
Router(config-ctl-client)# server cme 10.2.2.2 Cisco Unified CME router.
username user3 password 0 38h2KL
• username name-string—Username that is configured
on the CTL provider.
• password—Encryption status of the password string.
– 0—Not encrypted.
– 1—Encrypted using Message Digest 5 (MD5).
Note This option refers to the way that you want the
password to appear in show command output and
not to the way that you enter the password in this
command.

• password-string—Administrative password of the CTL

provider running on the remote Cisco Unified CME
router.

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Command or Action Purpose

Step 7 regenerate Creates a new CTLFile.tlv after you make changes to the
CTL client configuration.
Example:
Router(config-ctl-client)# regenerate
Step 8 end Returns to privileged EXEC mode.

Example:
Router(config-ctl-client)# end

What to do Next
You must configure a CTL provider on each Cisco Unified CME router. See the “Configuring the CTL
Provider” section on page 424.

Verifying the CTL Client

Step 1 Use the show ctl-client command to display the CTL client configuration.
The following sample output from the show ctl-client command displays the trustpoints in the system.
Router# show ctl-client

CTL Client Information

-----------------------------
SAST 1 Certificate Trustpoint: cmeserver
SAST 1 Certificate Trustpoint: sast2
List of Trusted Servers in the CTL
CME 10.1.1.1 cmeserver
TFTP 10.1.1.1 cmeserver
CAPF 10.1.1.1 cmeserver

Configuring the CTL Provider

If you have more than one Cisco Unified CME router in your network, perform the following steps to
configure a CTL provider on each Cisco Unified CME router on which the CTL client is not running.

SUMMARY STEPS

1. enable
2. configure terminal
3. credentials
4. ip source-address ip-address port port-number
5. trustpoint trustpoint-label
6. ctl-service admin username secret {0 | 1} password-string
7. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 credentials Enters credentials-interface mode to configure a CTL
provider.
Example:
Router(config)# credentials
Step 4 ip source-address [ip-address [port Identifies the local router on which this CTL provider is
[port-number]]] being configured.
• ip-address—Router IP address, typically one of the
Example: addresses of the Ethernet port of the router.
Router(config-credentials)# ip source-address
172.19.245.1 port 2444 • port port-number—TCP port for credentials service
communication. Default is 2444. You should use 2444.
Step 5 trustpoint trustpoint-label Configures the trustpoint to be used for TLS sessions with
the CTL client.
Example: • trustpoint-label—CTL provider trustpoint label.
Router(config-credentials)# trustpoint ctlpv
Step 6 ctl-service admin username secret {0 | 1} Specifies a username and password to authenticate the CTL
password-string client when it connects to retrieve the credentials during the
CTL protocol. You must use this command before you
Example: enable the CTL provider.
Router(config-credentials)# ctl-service admin • username—Name that will be used to authenticate the
user4 secret 0 c89L8o
client.
• secret—Character string for login authentication and
whether the string should be encrypted when it is stored
in the running configuration.
– 0—Not encrypted.
– 1—Encrypted using Message Digest 5 (MD5).
• password-string—Character string for login
authentication.
Step 7 end Returns to privileged EXEC mode.

Example:
Router(config-credentials)# end

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Verifying the CTL Provider

Step 1 show credentials
Use this command to display credentials settings.
Router# show credentials

Credentials IP: 172.19.245.1

Credentials PORT: 2444
Trustpoint: ctlpv

Configuring the CAPF Server

A certificate must be obtained for the CAPF server so that it can establish a TLS session with the phone
during certificate operation. The CAPF server can install, fetch, or delete locally significant certificates
(LSCs) on security-enabled phones. To enable the CAPF server on the Cisco Unified CME router,
perform the following steps.

Tip When you use the CAPF server to install phone certificates, arrange to do so during a scheduled period
of maintenance. Generating many certificates at the same time may cause call-processing interruptions.

SUMMARY STEPS

1. enable
2. configure terminal
3. capf-server
4. trustpoint-label label
5. cert-enroll-trustpoint label password {0 | 1} password-string
6. source-addr ip-address
7. port tcp-port
8. auth-mode {auth-string | LSC | MIC | none | null-string}
9. auth-string {delete | generate} {all | ephone-tag} [auth-string]
10. phone-key-size {512 | 1024 | 2048}
11. keygen-retry number
12. keygen-timeout minutes
13. cert-oper {delete all | fetch all | upgrade all}
14. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 capf-server Enters CAPF-server configuration mode.

Example:
Router(config)# capf-server
Step 4 trustpoint-label label Specifies the label of the trustpoint whose certificate is to be
used for TLS connection between the CAPF server and the
phone.
Example:
Router(config-capf-server)# trustpoint-label • label—Trustpoint name.
tp1
Step 5 cert-enroll-trustpoint trustpoint-label Enrolls the CAPF with the CA (or RA if the CA is not local
password {0 | 1} password-string to the Cisco Unified CME router).
• trustpoint-label—PKI trustpoint label for the CA or
Example: RA.
Router(config-capf-server)#
cert-enroll-trustpoint ra1 password 0 x8oWiet • password—Encryption status of the password string.
• password-string—Password to use for certificate
enrollment. This password is the revocation password
that is sent along with the certificate request to the CA.
Step 6 source-addr ip-address Defines the IP address of the CAPF server on the
Cisco Unified CME router.
Example: • ip-address—IP address of the CAPF server.
Router(config-capf-server)# source addr
10.10.10.1
Step 7 port tcp-port (Optional) Defines the TCP port number on which the
CAPF server listens for socket connections from the
phones.
Example:
Router(config-capf-server)# port 3804 • tcp-port—TCP port number. Range is 2000 to 9999.
Default is 3804.

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Command or Action Purpose

Step 8 auth-mode {auth-string | LSC | MIC | none | Specifies the type of authentication to use during CAPF
null-string} sessions to verify endpoints that request certificates.
• auth-string—The phone user enters a special
Example: authentication string at the phone. The string is
Router(config-capf-server)# auth-mode provided to the user by the system administrator and is
auth-string
configured using the auth-string generate command.
• LSC—The phone provides its LSC for authentication,
if one exists.
• MIC—The phone provides its MIC for authentication,
if one exists. If this option is chosen, the MIC’s issuer
certificate must be imported into a PKI trustpoint. See
the “Manually Importing MIC Root Certificate” section
on page 412.
• none—No certificate upgrade is initiated. This is the
default.
• null-string—No authentication.
Step 9 auth-string {delete | generate} {all | (Optional) Creates or removes authentication strings for all
ephone-tag} [digit-string] the secure ephones or for specified secure ephones. Use this
command if the auth-string keyword is specified in the
Example: auth-mode command. Strings become part of the ephone
Router(config-capf-server)# auth-string configuration. Use the show capf-server auth-string
generate all command to view authentication strings.
• delete—Remove authentication strings for the
specified secure devices.
• generate—Create authentication strings for the
specified secure devices.
• all—All phones.
• ephone-tag—Identifier for the ephone to receive the
authentication string.
• digit-string—String of digits that the phone user must
dial for CAPF authentication. The string can be 4 to 10
digits. If this value is not specified, a random string is
generated for each phone. For instructions on how to
enter the string from the phone, see the “Entering the
Authentication String on the Phone” section on
page 430.
Note You can also define an authentication string for an
individual ephone using the capf-auth-str
command.
Step 10 phone-key-size {512 | 1024 | 2048} (Optional) Specifies the size of the RSA key pair that is
generated on the phone for the phone’s certificate, in bits.
Example: • 512—512.
Router(config-capf-server)# phone-key-size 2048
• 1024—1024. This is the default.
• 2048—2048.

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Command or Action Purpose

Step 11 keygen-retry number (Optional) Specifies the number of times that the server
sends a key generation request.
Example: • number—Number of retries. Range is 0 to 100. Default
Router(config-capf-server)# keygen-retry 5 is 3.
Step 12 keygen-timeout minutes (Optional) Specifies the amount of time that the server waits
for a key generation response from the phone, in minutes.
Example: • minutes—Number of minutes before the generation
Router(config-capf-server)# keygen-timeout 45 process times out. Range is 1 to 120. Default is 30.
Step 13 cert-oper {delete all | fetch all | upgrade Initiates the indicated certificate operation on all configured
all} endpoints in the system.
• delete all—Remove all phone certificates.
Example:
Router(config-capf-server)# cert-oper upgrade
• fetch all—Retrieve all phone certificates for
all troubleshooting.
• upgrade all—Upgrade all phone certificates.
Note You can use the cert-oper command in ephone
configuration mode for certificate operations on
individual ephones. See the “Configuring
Telephony-Service Security Parameters” section on
page 414.
Step 14 end Returns to privileged EXEC mode.

Example:
Router(config-capf-server)# end

What to Do Next
If you select the authentication-string method of authentication in the auth-mode command, you must
also enter an authentication string on each phone that is receiving an updated LSC. For instructions on
this task, see the “Entering the Authentication String on the Phone” section on page 430.

Verifying the CAPF Server

Step 1 show capf-server summary
Use this command to display CAPF-server configuration information.
Router# show capf-server summary

CAPF Server Configuration Details

Trustpoint for TLS With Phone: tp1
Trustpoint for CA operation: ra1
Source Address: 10.10.10.1
Listening Port: 3804
Phone Key Size: 1024
Phone KeyGen Retries: 3
Phone KeyGen Timeout: 30 minutes

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Step 2 show capf-server auth-string

Use this command to display configured strings (PINs) that users enter at the phone to establish CAPF
authentication:
Router# show capf-server auth-string

Authentication Strings for configured Ephones

Mac-Addr Auth-String
-------- -----------
000CCE3A817C 7012
001121116BDD 922
000D299D50DF 9182
000ED7B10DAC 3114
000F90485077 3328
0013C352E7F1 0678

Entering the Authentication String on the Phone

This procedure is required only for the one-time installation of an LSC on a phone and only if you
specify the authentication string method of authentication.
If an authentication string is defined using the auth-string command in CAPF-server configuration
mode or the capf-auth-str command in ephone configuration mode, the authentication string must be
communicated to the phone user so that it can be entered on the phone before the LSC is installed.
The phone user can perform the following procedure to install the certificate. The authentication string
applies for one-time use only.

Note You can list authentication strings for phones by using the show capf-server auth-string command.

Prerequisites
• The CAPF certificate exists in the CTL file.
• A signed image exists on the phone; see the Cisco Unified IP phone administration documentation
that supports your phone model.
• The device has registered.
• The device security mode is nonsecure.

DETAILED STEPS

Step 1 Press the Settings button.

On the Cisco Unified IP Phone 7921, use the down arrow key to access the Settings menu.
Step 2 If the configuration is locked, press **# (asterisk, asterisk, pound sign) to unlock it.
Step 3 Scroll down the Settings menu. Highlight Security Configuration and press the Select soft key.
Step 4 Scroll down the Security Configuration menu. Highlight LSC and press the Update soft key.
On the Cisco Unified IP Phone 7921, press **# to unlock the Security Configuration menu.

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Step 5 When prompted for the authentication string, enter the string provided by the system administrator and
press the Submit soft key.
The phone installs, updates, deletes, or fetches the certificate, depending on the CAPF configuration.
You can monitor the progress of the certificate operation by viewing the messages that display on the
phone. After you press Submit, the message “Pending” displays under the LSC option. The phone
generates the public and private key pair and displays the information on the phone. When the phone
successfully completes the process, the phone displays a successful message. If the phone displays a
failure message, you entered the wrong authentication string or did not enable the phone for upgrade.
You can stop the process by choosing the Stop option at any time.

Verifying the Authentication String on the Phone

Step 1 Verify that the certificate was installed on the phone by choosing Settings > Model Information and
viewing the LSC setting, which indicates Installed or Not Installed.

Configuring Secure Calls Between Cisco Unified CMEs Across an H.323 Trunk
To configure the network for secure calls between Cisco Unified CME systems across an H.323 trunk,
perform the following steps on the Cisco Unified CME router.

Prerequisites
To make secure H.323 calls, telephony-service security parameters must be configured. See the
“Configuring Telephony-Service Security Parameters” section on page 414.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice service voip
4. supplementary-service media-renegotiate
5. srtp fallback
6. h323
7. emptycapability
8. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip Enters voice-service configuration mode.
• The voip keyword specifies VoIP encapsulation.
Example:
Router(config)# voice service voip
Step 4 supplementary-service media-renegotiate Enables midcall renegotiation of SRTP cryptographic keys.

Example:
Router(conf-voi-serv)# supplementary-service
media-renegotiate
Step 5 srtp fallback Enables security policies.
• The srtp command enables secure calls using SRTP for
Example: media encryption and authentication and disables
Router(conf-voi-serv)# srtp fallback fallback.
• The fallback keyword enables call fallback to
nonsecure (RTP) mode, allowing the user to make calls
that are not secure.
• SRTP-to-RTP fallback must be configured for
supplementary services such as ringback tone and
MOH to function. Without SRTP-to-RTP fallback
configured, MOH causes secure calls to be dropped.
Note This security policy applies to all calls going
through the gateway and is not configurable on a
per-call basis.

If fallback is not configured it will drop all calls that are not
secure so only secure phones can call you.
This step configures fallback globally. To configure
fallback for individual dial peers, see the “Configuring
Cisco Unified CME SRTP Fallback for H.323 Dial Peers”
section on page 433. Skip this step if you are going to
configure fallback on individual dial peers.
Step 6 h323 Enters H.323 voice-service configuration mode.

Example:
Router(conf-voi-serv)# h323

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Command or Action Purpose

Step 7 emptycapability Eliminates the need for identical codec capabilities for all
dial peers in the rotary group.
Example:
Router(conf-serv-h323)# emptycapability
Step 8 exit Exits H.323 voice-service configuration mode.

Example:
Router(conf-serv-h323)# exit

Configuring Cisco Unified CME SRTP Fallback for H.323 Dial Peers
To configure SRTP fallback for an individual dial peer, perform the following steps on the
Cisco Unified CME router.

Note SRTP-to-RTP fallback must be configured for supplementary services such as ringback tone and MOH
to function. Without SRTP-to-RTP fallback configured, MOH causes secure calls to be dropped.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice class codec tag
4. codec preference value codec-type
5. exit
6. dial-peer voice tag voip
7. srtp fallback
8. voice-class codec tag
9. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice class codec tag Enters voice-class configuration mode and assigns an
identification tag number for a codec voice class.
Example:
Router(config)# voice class codec 1
Step 4 codec preference value codec-type Specifies a list of preferred codecs to use on a dial peer.
• Repeat this step to build a list of preferred codecs.
Example: • Use the same preference order for the codec list on both
Router(config-voice-class)# codec preference 1
g711alaw
Cisco Unified CMEs on either side of the H.323 trunk.
Step 5 exit Exits voice-class configuration mode.

Example:
Router(config-voice-class)# exit
Step 6 dial-peer voice tag voip Enters dial peer voice configuration mode.

Example:
Router(config)# dial-peer voice 101 voip
Step 7 srtp fallback Enables secure calls that use SRTP for media encryption
and authentication and specifies fallback capability. Using
the no srtp command disables security and causes the dial
Example:
Router(config-dial-peer)# srtp fallback
peer to fall back to RTP mode.
• The srtp command enables secure calls.
• The fallback keyword enables fallback to nonsecure
mode (RTP) on an individual dial peer. The no form of
this command disables fallback and disables SRTP.
Note This dial-peer configuration command takes
precedence over the globally configured srtp
command enabled in voice service voip
configuration mode shown in the “Configuring
Secure Calls Between Cisco Unified CMEs Across
an H.323 Trunk” section on page 431.
Step 8 voice-class codec tag Assigns a previously configured codec selection preference
list (codec voice class) to a Voice over IP (VoIP) dial peer.
Example: • The tag argument in this step is the same as the tag in
Router(config-dial-peer)# voice-class codec 1 Step 3.
Step 9 exit Exits dial-peer voice configuration mode.

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

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Configuring Cisco Unity for Secure Cisco Unified CME Operation

This section contains the following tasks:
• Configuring Integration Between Cisco Unified CME and Cisco Unity, page 435
• Importing the Cisco Unity Root Certificate to Cisco Unified CME, page 436
• Configuring Cisco Unity Ports for Secure Registration, page 437
• Verifying that Cisco Unity are Registering Securely, page 437

Configuring Integration Between Cisco Unified CME and Cisco Unity

To change the settings for the integration between Cisco Unified CME and Cisco Unity, perform the
following steps on the Cisco Unity server:

Step 1 If Cisco Unity Telephony Integration Manager (UTIM) is not already open, on the Cisco Unity server,
on the Windows Start menu, click Programs > Cisco Unity > Manage Integrations. The UTIM
window appears.
Step 2 In the left pane, double-click Cisco Unity Server. The existing integrations appear.
Step 3 Click the Cisco Unified Communications Manager integration.
Step 4 In the right pane, click the cluster for the integration.
Step 5 Click the Servers tab.
Step 6 In the Cisco Unified Communications Manager Cluster Security Mode field, click the applicable setting.
Step 7 If you clicked the Non-secure setting, click Save and skip the remaining steps in this procedure.
If you clicked the Authenticated or the Encrypted settings, the Security tab and the Add TFTP Server
dialog box appear. In the Add TFTP Server dialog box, in the IP Address or Host Name field, enter the
IP address (or DNS name) of the primary TFTP server for the Cisco Unified Communications Manager
cluster, and click OK.
Step 8 If there are more TFTP servers that Cisco Unity will use to download the Cisco Unified Communications
Manager certificates, click Add. The Add TFTP Server dialog box appears.
Step 9 In the IP Address or Host Name field, enter the IP address (or DNS name) of the secondary TFTP server
for the Cisco Unified Communications Manager cluster, and click OK.
Step 10 Click Save.
Cisco Unity creates the voice messaging port device certificates, exports the Cisco Unity server root
certificate, and displays the Export Cisco Unity Root Certificate dialog box.
Step 11 Note the file name of the exported Cisco Unity server root certificate and click OK.
Step 12 On the Cisco Unity server, navigate to the CommServer\SkinnyCerts directory.
Step 13 Locate the Cisco Unity server root certificate file that you exported in Step 11.
Step 14 Right-click the file and click Rename.
Step 15 Change the file extension from .0 to .pem. For example, change the filename “12345.0” to “12345.pem”
for the exported Cisco Unity server root certificate file.
Step 16 Copy this file to a PC from which you can access the Cisco Unified CME router.

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Importing the Cisco Unity Root Certificate to Cisco Unified CME

To import the Cisco Unity root certificate to Cisco Unified CME, perform the following steps on the
Cisco Unified CME router:

SUMMARY STEPS

1. enable
2. configure terminal
3. crypto pki trustpoint name
4. revocation-check none
5. enrollment terminal
6. exit
7. crypto pki authenticate trustpoint-label
8. Open the root certificate file that you copied from the Cisco Unity Server in Step 16.
9. You will be prompted to enter the CA certificate. Cut and paste the entire contents of the base 64
encoded certificate between “BEGIN CERTIFICATE” and “END CERTIFICATE” at the command
line. Press Enter, and type “quit.” The router prompts you to accept the certificate. Enter “yes” to
accept the certificate.

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 crypto pki trustpoint name Declares the trustpoint that your RA mode certificate server
should use and enters CA-trustpoint configuration mode.
Example: • label—Name for the trustpoint and RA.
Router(config)# crypto pki trustpoint PEM
Step 4 revocation-check none (Optional) Checks the revocation status of a certificate and
specifies one or more methods to check the status. If a
second and third method are specified, each method is used
Example:
Router(ca-trustpoint)# revocation-check none
only if the previous method returns an error, such as a server
being down.
• none—Certificate checking is not required.
Step 5 enrollment terminal Specifies manual cut-and-paste certificate enrollment.

Example:
Router(ca-trustpoint)# enrollment terminal

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Command or Action Purpose

Step 6 exit Exits CA-trustpoint configuration mode.

Example:
Router(ca-trustpoint)# exit
Step 7 crypto pki authenticate trustpoint-label Retrieves the CA certificate and authenticates it. Checks the
certificate fingerprint if prompted.
Example: • trustpoint-label—Trustpoint label.
Router(config)# crypto pki authenticate pem
Note The trustpoint-label must be the same as the name
in step 3.
Step 8 You will be prompted to enter the CA certificate. Cut Completes the copying of the Cisco Unity root certificate to
and paste the entire contents of the base 64 encoded the Cisco Unified CME router.
certificate between “BEGIN CERTIFICATE” and
“END CERTIFICATE” at the command line. Press
Enter, and type “quit.” The router prompts you to
accept the certificate. Enter “yes” to accept the
certificate.

Configuring Cisco Unity Ports for Secure Registration

To configure Cisco Unity ports for registration in secure mode, perform the following steps:

Step 1 Choose the Cisco voice-mail port that you want to update.
Step 2 In the Device Security Mode field, choose Encrypted from the drop-down list box.
Step 3 Click Update.

Verifying that Cisco Unity are Registering Securely

Use the show sccp connections command to verify that Cisco Unity ports are registered securely with
Cisco Unified CME.

show sccp connection: Example

In the following example, the secure value of the stype field shows that the connections are secure.
Router# show sccp connections

sess_id conn_id stype mode codec ripaddr rport sport

16777222 16777409 secure-xcode sendrecv g729b 10.3.56.120 16772 19534

16777222 16777393 secure-xcode sendrecv g711u 10.3.56.50 17030 18464

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

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Configuration Examples for Security

This section contains the following examples:

Phone Authentication
• Cisco IOS CA Server: Example, page 438
• Enabling a Registration Authority: Example, page 438
• Manually Importing MIC Root Certificate on the Cisco Unified CME Router: Example, page 439
• Obtaining a Certificate for Cisco Unified CME Server Functions: Example, page 442
• CTL Client Running on Cisco Unified CME Router: Example, page 442
• CTL Client Running on Another Router: Example, page 442
• Telephony-Service Security Parameters: Example, page 442
• CAPF Server: Example, page 443

Media Encryption
• Secure Cisco Voice Gateway with Media Encryption (SRTP) on Cisco Unified CME: Example,
page 445
• Secure Cisco Unified CME: Example, page 451

Cisco IOS CA Server: Example

!
crypto pki server iosca
grant auto
database url flash:
!
crypto pki trustpoint iosca
revocation-check none
rsakeypair iosca
!
crypto pki certificate chain iosca
certificate ca 01
308201F9 30820162 ...

Enabling a Registration Authority: Example

The following example sets up an RA and trustpoint named ra12:
Router(config)# crypto pki trustpoint ra12
Router(config-ca-trustpoint)# enrollment url http://ca-server.company.com
Router(config-ca-trustpoint)# revocation-check none
Router(config-ca-trustpoint)# rsakeypair exampleCAkeys 1024 1024
Router(config-ca-trustpoint)# exit
Router(config)# crypto pki server ra12
Router(config-cs-server)# mode ra
Router(config-cs-server)# lifetime certificate 1800
Router(config-cs-server)# no grant auto
Router(config-cs-server)# no shutdown
Router(config-cs-server)# exit

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The following example sets up a trustpoint named sast2 that periodically generates a CRL instead of
having it generated manually. Third-party CAs may require this functionality.
Router(config)# crypto pki trustpoint sast2
Router(config-ca-trustpoint)# enrollment url http://NTP-ab11:80
Router(config-ca-trustpoint)# serial-number
Router(config-ca-trustpoint)# revocation-check crl
Router(config-ca-trustpoint)# rsakeypair sast2

Manually Importing MIC Root Certificate on the Cisco Unified CME Router:
Example
The following example shows three certificates imported to the router (7970, 7960, PEM).
Router(config)# crypto pki trustpoint 7970
Router(ca-trustpoint)# revocation-check none
Router(ca-trustpoint)# enrollment terminal
Router(ca-trustpoint)# exit
Router(config)# crypto pki authenticate 7970

Enter the base 64 encoded CA certificate.

End with a blank line or the word "quit" on a line by itself
MIIDqDCCApCgAwIBAgIQNT+yS9cPFKNGwfOprHJWdTANBgkqhkiG9w0BAQUFADAu
MRYwFAYDVQQKEw1DaXNjbyBTeXN0ZW1zMRQwEgYDVQQDEwtDQVAtUlRQLTAwMjAe
Fw0wMzEwMTAyMDE4NDlaFw0yMzEwMTAyMDI3MzdaMC4xFjAUBgNVBAoTDUNpc2Nv
IFN5c3RlbXMxFDASBgNVBAMTC0NBUC1SVFAtMDAyMIIBIDANBgkqhkiG9w0BAQEF
AAOCAQ0AMIIBCAKCAQEAxCZlBK19w/2NZVVvpjCPrpW1cCY7V1q9lhzI85RZZdnQ
2M4CufgIzNa3zYxGJIAYeFfcRECnMB3f5A+x7xNiEuzE87UPvK+7S80uWCY0Uhtl
AVVf5NQgZ3YDNoNXg5MmONb8lT86F55EZyVac0XGne77TSIbIdejrTgYQXGP2MJx
Qhg+ZQlGFDRzbHfM84Duv2Msez+l+SqmqO80kIckqE9Nr3/XCSj1hXZNNVg8D+mv
Hth2P6KZqAKXAAStGRLSZX3jNbS8tveJ3Gi5+sj9+F6KKK2PD0iDwHcRKkcUHb7g
lI++U/5nswjUDIAph715Ds2rn9ehkMGipGLF8kpuCwIBA6OBwzCBwDALBgNVHQ8E
BAMCAYYwDwYDVR0TAQH/BAUwAwEB/zAdBgNVHQ4EFgQUUpIr4ojuLgmKTn5wLFal
mrTUm5YwbwYDVR0fBGgwZjBkoGKgYIYtaHR0cDovL2NhcC1ydHAtMDAyL0NlcnRF
bnJvbGwvQ0FQLVJUUC0wMDIuY3Jshi9maWxlOi8vXFxjYXAtcnRwLTAwMlxDZXJ0
RW5yb2xsXENBUC1SVFAtMDAyLmNybDAQBgkrBgEEAYI3FQEEAwIBADANBgkqhkiG
9w0BAQUFAAOCAQEAVoOM78TaOtHqj7sVL/5u5VChlyvU168f0piJLNWip2vDRihm
E+DlXdwMS5JaqUtuaSd/m/xzxpcRJm4ZRRwPq6VeaiiQGkjFuZEe5jSKiSAK7eHg
tup4HP/ZfKSwPA40DlsGSYsKNMm3OmVOCQUMH02lPkS/eEQ9sIw6QS7uuHN4y4CJ
NPnRbpFRLw06hnStCZHtGpKEHnY213QOy3h/EWhbnp0MZ+hdr20FujSI6G1+L39l
aRjeD708f2fYoz9wnEpZbtn2Kzse3uhU1Ygq1D1x9yuPq388C18HWdmCj4OVTXux
V6Y47H1yv/GJM8FvdgvKlExbGTFnlHpPiaG9tQ==
quit
Certificate has the following attributes:
Fingerprint MD5: F7E150EA 5E6E3AC5 615FC696 66415C9F
Fingerprint SHA1: 1BE2B503 DC72EE28 0C0F6B18 798236D8 D3B18BE6
% Do you accept this certificate? [yes/no]: y
Trustpoint CA certificate accepted.
% Certificate successfully imported
Router(config)# crypto pki trustpoint 7960
Router(ca-trustpoint)# revocation-check none
Router(ca-trustpoint)# enrollment terminal
Router(ca-trustpoint)# exit
Router(config)# crypto pki authenticate 7960

Enter the base 64 encoded CA certificate.

End with a blank line or the word "quit" on a line by itself

MIICKDCCAZGgAwIBAgIC8wEwDQYJKoZIhvcNAQEFBQAwQDELMAkGA1UEBhMCVVMx
GjAYBgNVBAoTEUNpc2NvIFN5c3RlbXMgSW5jMRUwEwYDVQQDEwxDQVBGLTdEN0Qw
QzAwHhcNMDQwNzE1MjIzODMyWhcNMTkwNzEyMjIzODMxWjBAMQswCQYDVQQGEwJV

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UzEaMBgGA1UEChMRQ2lzY28gU3lzdGVtcyBJbmMxFTATBgNVBAMTDENBUEYtN0Q3
RDBDMDCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEA0hvMOZZ9ENYWme11YGY1
it2rvE3Nk/eqhnv8P9eqB1iqt+fFBeAG0WZ5bO5FetdU+BCmPnddvAeSpsfr3Z+h
x+r58fOEIBRHQLgnDZ+nwYH39uwXcRWWqWwlW147YHjV7M5c/R8T6daCx4B5NBo6
kdQdQNOrV3IP7kQaCShdM/kCAwEAAaMxMC8wDgYDVR0PAQH/BAQDAgKEMB0GA1Ud
JQQWMBQGCCsGAQUFBwMBBggrBgEFBQcDBTANBgkqhkiG9w0BAQUFAAOBgQCaNi6x
sL6M5NlDezpSBO3QmUVyXMfrONV2ysrSwcXzHu0gJ9MSJ8TwiQmVaJ47hSTlF5a8
YVYJ0IdifXbXRo+/EEO7kkmFE8MZta5rM7UWj8bAeR42iqA3RzQaDwuJgNWT9Fhh
GgfuNAlo5h1AikxsvxivmDlLdZyCMoqJJd7B2Q==
quit
Certificate has the following attributes:
Fingerprint MD5: 4B9636DF 0F3BA6B7 5F54BE72 24762DBC
Fingerprint SHA1: A9917775 F86BB37A 5C130ED2 3E528BB8 286E8C2D
% Do you accept this certificate? [yes/no]: y
Trustpoint CA certificate accepted.
% Certificate successfully imported

Router(config)# crypto pki trustpoint PEM

Router(ca-trustpoint)# revocation-check none
Router(ca-trustpoint)# enrollment terminal
Router(ca-trustpoint)# exit
Router(config)# crypto pki authenticate PEM

Enter the base 64 encoded CA certificate.

End with a blank line or the word "quit" on a line by itself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quit

Certificate has the following attributes:

Fingerprint MD5: 233C8E33 8632EA4E 76D79FEB FFB061C6
Fingerprint SHA1: F7B40B94 5831D2AB 447AB8F2 25990732 227631BE
% Do you accept this certificate? [yes/no]: y
Trustpoint CA certificate accepted.
% Certificate successfully imported

Use the show crypto pki trustpoint status command to show that enrollment has succeeded and that
five CA certificates were granted. The five certificates include the three certificates just entered and the
CA server certificate and the router certificate.
Router# show crypto pki trustpoint status

Trustpoint 7970:
Issuing CA certificate configured:
Subject Name:

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cn=CAP-RTP-002,o=Cisco Systems
Fingerprint MD5: F7E150EA 5E6E3AC5 615FC696 66415C9F
Fingerprint SHA1: 1BE2B503 DC72EE28 0C0F6B18 798236D8 D3B18BE6
State:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... None
Trustpoint 7960:
Issuing CA certificate configured:
Subject Name:
cn=CAPF-508A3754,o=Cisco Systems Inc,c=US
Fingerprint MD5: 6BAE18C2 0BCE391E DAE2FE4C 5810F576
Fingerprint SHA1: B7735A2E 3A5C274F C311D7F1 3BE89942 355102DE
State:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... None
Trustpoint PEM:
Issuing CA certificate configured:
Subject Name:
cn=CAP-RTP-001,o=Cisco Systems
Fingerprint MD5: 233C8E33 8632EA4E 76D79FEB FFB061C6
Fingerprint SHA1: F7B40B94 5831D2AB 447AB8F2 25990732 227631BE
State:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... None
Trustpoint srstcaserver:
Issuing CA certificate configured:
Subject Name:
cn=srstcaserver
Fingerprint MD5: 6AF5B084 79C93F2B 76CC8FE6 8781AF5E
Fingerprint SHA1: 47D30503 38FF1524 711448B4 9763FAF6 3A8E7DCF
State:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... None

Trustpoint srstca:
Issuing CA certificate configured:
Subject Name:
cn=srstcaserver
Fingerprint MD5: 6AF5B084 79C93F2B 76CC8FE6 8781AF5E
Fingerprint SHA1: 47D30503 38FF1524 711448B4 9763FAF6 3A8E7DCF
Router General Purpose certificate configured:
Subject Name:
serialNumber=F3246544+hostname=c2611XM-sSRST.cisco.com
Fingerprint: 35471295 1C907EC1 45B347BC 7A9C4B86
State:
Keys generated ............. Yes (General Purpose)
Issuing CA authenticated ....... Yes
Certificate request(s) ..... Yes

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Obtaining a Certificate for Cisco Unified CME Server Functions: Example

The following example establishes a trustpoint for the CAPF server called capf.
Router(config)# crypto pki trustpoint capf
Router(config-ca-trustpoint)# enrollment url http://ca-server.company.com
Router(config-ca-trustpoint)# revocation-check none
Router(config-ca-trustpoint)# rsakeypair capf 1024 1024
Router(config-ca-trustpoint)# exit
Router(config)# crypto pki authenticate capf
Router(config)# crypto pki enroll capf

Telephony-Service Security Parameters: Example

The following example shows Cisco Unified CME security parameters.
telephony-service
device-security-mode authenticated
secure-signaling trustpoint cme-sccp
tftp-server-credentials trustpoint cme-tftp
load-cfg-file slot0:Ringlist.xml alias Ringlist.xml sign create

ephone 24
device-security-mode authenticated
capf-auth-str 2734
cert-oper upgrade auth-mode auth-string

CTL Client Running on Cisco Unified CME Router: Example

ctl-client
server capf 10.1.1.1 trustpoint cmeserver
server cme 10.1.1.1 trustpoint cmeserver
server tftp 10.1.1.1 trustpoint cmeserver
sast1 trustpoint cmeserver
sast2 trustpoint sast2

CTL Client Running on Another Router: Example

ctl-client
server cme 10.1.1.100 trustpoint cmeserver
server cme 10.1.1.1 username cisco password 1 0822455D0A16544541
sast1 trustpoint cmeserver
sast2 trustpoint sast1

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CAPF Server: Example

!
ip dhcp pool cme-pool
network 10.1.1.0 255.255.255.0
option 150 ip 10.1.1.1
default-router 10.1.1.1
!
capf-server
port 3804
auth-mode null-string
cert-enroll-trustpoint iosra password 1 00071A1507545A545C
trustpoint-label cmeserver
source-addr 10.1.1.1
!
crypto pki server iosra
grant auto
mode ra
database url slot0:
!
crypto pki trustpoint cmeserver
enrollment url http://10.1.1.100:80
serial-number
revocation-check none
rsakeypair cmeserver
!
crypto pki trustpoint sast2
enrollment url http://10.1.1.100:80
serial-number
revocation-check none
rsakeypair sast2
!
!
crypto pki trustpoint iosra
enrollment url http://10.1.1.200:80
revocation-check none
rsakeypair iosra
!
!
crypto pki certificate chain cmeserver
certificate 1B
30820207 30820170 A0030201 0202011B 300D0609 2A864886 F70D0101 04050030
....
quit
certificate ca 01
3082026B 308201D4 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
...
quit
crypto pki certificate chain sast2
certificate 1C
30820207 30820170 A0030201 0202011C 300D0609 2A864886 F70D0101 04050030
....
quit
certificate ca 01
3082026B 308201D4 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
.....
quit
crypto pki certificate chain capf-tp
crypto pki certificate chain iosra
certificate 04
30820201 3082016A A0030201 02020104 300D0609 2A864886 F70D0101 04050030
......
certificate ca 01

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308201F9 30820162 A0030201 02020101 300D0609 2A864886 F70D0101 04050030

....
quit
!
!
credentials
ctl-service admin cisco secret 1 094F471A1A0A464058
ip source-address 10.1.1.1 port 2444
trustpoint cmeserver
!
!
telephony-service
no auto-reg-ephone
load 7960-7940 P00307010200
load 7914 S00104000100
load 7941GE TERM41.7-0-0-129DEV
load 7970 TERM70.7-0-0-77DEV
max-ephones 20
max-dn 10
ip source-address 10.1.1.1 port 2000 secondary 10.1.1.100
secure-signaling trustpoint cmeserver
cnf-file location flash:
cnf-file perphone
dialplan-pattern 1 2... extension-length 4
max-conferences 8 gain -6
transfer-pattern ....
tftp-server-credentials trustpoint cmeserver
server-security-mode secure
device-security-mode encrypted
load-cfg-file slot0:Ringlist.xml alias Ringlist.xml sign
load-cfg-file slot0:P00307010200.bin alias P00307010200.bin
load-cfg-file slot0:P00307010200.loads alias P00307010200.loads
load-cfg-file slot0:P00307010200.sb2 alias P00307010200.sb2
load-cfg-file slot0:P00307010200.sbn alias P00307010200.sbn
load-cfg-file slot0:cnu41.2-7-4-116dev.sbn alias cnu41.2-7-4-116dev.sbn
load-cfg-file slot0:Jar41.2-9-0-101dev.sbn alias Jar41.2-9-0-101dev.sbn
load-cfg-file slot0:CVM41.2-0-0-96dev.sbn alias CVM41.2-0-0-96dev.sbn
load-cfg-file slot0:TERM41.DEFAULT.loads alias TERM41.DEFAULT.loads
load-cfg-file slot0:TERM70.DEFAULT.loads alias TERM70.DEFAULT.loads
load-cfg-file slot0:Jar70.2-9-0-54dev.sbn alias Jar70.2-9-0-54dev.sbn
load-cfg-file slot0:cnu70.2-7-4-58dev.sbn alias cnu70.2-7-4-58dev.sbn
load-cfg-file slot0:CVM70.2-0-0-49dev.sbn alias CVM70.2-0-0-49dev.sbn
load-cfg-file slot0:DistinctiveRingList.xml alias DistinctiveRingList.xml sign
load-cfg-file slot0:Piano1.raw alias Piano1.raw sign
load-cfg-file slot0:S00104000100.sbn alias S00104000100.sbn
create cnf-files version-stamp 7960 Aug 13 2005 12:39:24
!
!
ephone 1
device-security-mode encrypted
cert-oper upgrade auth-mode null-string
mac-address 000C.CE3A.817C
type 7960 addon 1 7914
button 1:2 8:8
!
!
ephone 2
device-security-mode encrypted
capf-auth-str 2476
cert-oper upgrade auth-mode null-string
mac-address 0011.2111.6BDD
type 7970
button 1:1
!

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!
ephone 3
device-security-mode encrypted
capf-auth-str 5425
cert-oper upgrade auth-mode null-string
mac-address 000D.299D.50DF
type 7970
button 1:3
!
!
ephone 4
device-security-mode encrypted
capf-auth-str 7176
cert-oper upgrade auth-mode null-string
mac-address 000E.D7B1.0DAC
type 7960
button 1:4
!
!
ephone 5
device-security-mode encrypted
mac-address 000F.9048.5077
type 7960
button 1:5
!
!
ephone 6
device-security-mode encrypted
mac-address 0013.C352.E7F1
type 7941GE
button 1:6
!

Secure Cisco Voice Gateway with Media Encryption (SRTP) on

Cisco Unified CME: Example
Router# show running-config

Building configuration...

Current configuration : 8906 bytes

!
! Last configuration change at 15:41:09 PDT Mon Oct 23 2006
!
version 12.4
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname akash
!
boot-start-marker
boot-end-marker
!
logging buffered 400000 debugging
no logging console
enable password lab
!
no aaa new-model
!

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resource policy
!
clock timezone PST -8
clock summer-time PDT recurring
no ip domain lookup
!
!
!
! The following lines show STCAPP security enabled at the system level:
stcapp ccm-group 1
stcapp security trustpoint analog
stcapp security mode encrypted
stcapp
!
voice-card 0
dsp services dspfarm
!
crypto pki trustpoint analog
enrollment url http://10.4.177.51:80
serial-number
revocation-check none
certificate ca 01
30820201 3082016A A0030201 02020101 300D0609 2A864886 F70D0101 04050030
14311230 10060355 04031309 756E6974 69746573 74301E17 0D303630 35303132
33303130 335A170D 30393034 33303233 30313033 5A301431 12301006 03550403
1309756E 69746974 65737430 819F300D 06092A86 4886F70D 01010105 0003818D
00308189 02818100 C2D07857 B8DF7F55 3C2365B3 2E1524CF EE898D1F D7A04075
D36F0229 392803DF B45246B4 A447506F A3FCDD00 9FC93CD7 5B5573E0 7BFD25E1
AB2F24E2 740D5765 7F628B6E 0FD39BEE 940D80FF 3B9F9F17 7ACA8F82 1A9E3179
458781E8 87C95E1B 17E6A61C 7D138AC1 D8E30F3C 88BFAFEE A94D5F8C E433DF71
F076E96C 9BB5327F 02030100 01A36330 61300F06 03551D13 0101FF04 05300301
01FF300E 0603551D 0F0101FF 04040302 0186301F 0603551D 23041830 168014B5
418287D0 61FE277C 9A1862B3 673BF7F7 0E47DD30 1D060355 1D0E0416 0414B541
8287D061 FE277C9A 1862B367 3BF7F70E 47DD300D 06092A86 4886F70D 01010405
00038181 002BB76E 22A59D73 6DBB62BA BAC3D5B4 2F739A26 D5FFF911 EDEB9BDC
7B29FECC E0B68E0F 22A3C0D0 8BA64592 30C6B628 5EFA3905 1B13BFE7 7CEB1456
55214435 07F752A6 73D5646A 4BB7B3C2 61E2C185 3A638FCA AE5AC6A1 3DB3590B
C3C6C924 D1E1E365 FE041B07 F3E2AF24 3701B664 A7879229 AFDF163A 00AA12AA
85866101 53
quit
crypto pki certificate chain analog
certificate 0A
308201BF 30820128 A0030201 0202010A 300D0609 2A864886 F70D0101 04050030
14311230 10060355 04031309 756E6974 69746573 74301E17 0D303630 35333032
31313630 345A170D 30373035 33303231 31363034 5A302A31 28301206 03550405
130B4648 4B303930 37463050 47301206 092A8648 86F70D01 09021605 616B6173
68305C30 0D06092A 864886F7 0D010101 0500034B 00304802 4100A6AD 0A376A6C
9EB668CC D0DF2A17 180E6CA2 FA5F243B 861EAA29 BE5FC488 A22AD4E8 5DFC22AC
13B43337 2F9FBA64 14E838EA 888E79DE 93AB63E4 4B4E2ECD 256D0203 010001A3
4F304D30 0B060355 1D0F0404 030205A0 301F0603 551D2304 18301680 14B54182
87D061FE 277C9A18 62B3673B F7F70E47 DD301D06 03551D0E 04160414 34D2D41C
274AB6E3 71A3A32C EC19D533 D3C0A020 300D0609 2A864886 F70D0101 04050003
818100A2 3947B1D0 FC5E9B79 0C1A28E7 BCB34C6C BB68C5F6 356F3F61 7525053E
0AED7325 9F286888 887810A6 B62FBAF3 BDC81542 C9828BBF 6A9FE936 AD3ED33B
D4F5AD22 E703C8E0 C3DDEAC8 2097A209 542551F7 6340A2A4 55A25A99 6A87367F
A0CBD9B6 E38D5E40 6479EB71 EFA644B3 93222D6F 235039AE BB9AA7B7 B1D07B3C FC6339
quit
certificate ca 01
30820201 3082016A A0030201 02020101 300D0609 2A864886 F70D0101 04050030
14311230 10060355 04031309 756E6974 69746573 74301E17 0D303630 35303132
33303130 335A170D 30393034 33303233 30313033 5A301431 12301006 03550403
1309756E 69746974 65737430 819F300D 06092A86 4886F70D 01010105 0003818D
00308189 02818100 C2D07857 B8DF7F55 3C2365B3 2E1524CF EE898D1F D7A04075
D36F0229 392803DF B45246B4 A447506F A3FCDD00 9FC93CD7 5B5573E0 7BFD25E1

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Configuration Examples for Security

AB2F24E2 740D5765 7F628B6E 0FD39BEE 940D80FF 3B9F9F17 7ACA8F82 1A9E3179

458781E8 87C95E1B 17E6A61C 7D138AC1 D8E30F3C 88BFAFEE A94D5F8C E433DF71
F076E96C 9BB5327F 02030100 01A36330 61300F06 03551D13 0101FF04 05300301
01FF300E 0603551D 0F0101FF 04040302 0186301F 0603551D 23041830 168014B5
418287D0 61FE277C 9A1862B3 673BF7F7 0E47DD30 1D060355 1D0E0416 0414B541
8287D061 FE277C9A 1862B367 3BF7F70E 47DD300D 06092A86 4886F70D 01010405
00038181 002BB76E 22A59D73 6DBB62BA BAC3D5B4 2F739A26 D5FFF911 EDEB9BDC
7B29FECC E0B68E0F 22A3C0D0 8BA64592 30C6B628 5EFA3905 1B13BFE7 7CEB1456
55214435 07F752A6 73D5646A 4BB7B3C2 61E2C185 3A638FCA AE5AC6A1 3DB3590B
C3C6C924 D1E1E365 FE041B07 F3E2AF24 3701B664 A7879229 AFDF163A 00AA12AA
85866101 53
quit
!
!
voice service voip
!
!
interface FastEthernet0/0
ip address 10.4.177.53 255.255.0.0
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip route 0.0.0.0 0.0.0.0 1.4.0.1
!
ip http server
no ip http secure-server
!
no cdp advertise-v2
!
!
control-plane
!
!
voice-port 2/0
!
voice-port 2/1
!
voice-port 2/2
!
voice-port 2/3
!
voice-port 2/4
!
voice-port 2/5
!
voice-port 2/6
!
voice-port 2/7
!
voice-port 2/8
!
voice-port 2/9
!
voice-port 2/10
!
voice-port 2/11
!
voice-port 2/12

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Configuration Examples for Security

!
voice-port 2/13
!
voice-port 2/14
!
voice-port 2/15
!
voice-port 2/16
!
voice-port 2/17
!
voice-port 2/18
!
voice-port 2/19
!
voice-port 2/20
!
voice-port 2/21
!
voice-port 2/22
!
voice-port 2/23
description choctaw
!
!
!
sccp local FastEthernet0/0
sccp ccm 10.4.177.51 identifier 1 version 4.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1

!
dial-peer voice 5001 pots
service stcapp
port 2/0
!
dial-peer voice 5002 pots
service stcapp
! The following line shows the security mode configured on the dial peer.
security mode authenticated
port 2/1
!
dial-peer voice 5003 pots
service stcapp
security mode none
port 2/2
!
dial-peer voice 2000 voip
destination-pattern 7...
session target ipv4:10.4.177.100
incoming called-number 7000
codec g711ulaw
!
dial-peer voice 1 pots
!
dial-peer voice 5004 pots
service stcapp
shutdown
port 2/3
!
dial-peer voice 5005 pots
shutdown

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Configuration Examples for Security

destination-pattern 3001
port 2/4
!
dial-peer voice 5006 pots
service stcapp
shutdown
port 2/5
!
dial-peer voice 5007 pots
service stcapp
shutdown
port 2/6
!
dial-peer voice 5008 pots
service stcapp
shutdown
port 2/7
!
dial-peer voice 5009 pots
service stcapp
shutdown
port 2/8
!
dial-peer voice 5010 pots
service stcapp
shutdown
port 2/9
!
dial-peer voice 5011 pots
service stcapp
shutdown
port 2/10
!
dial-peer voice 5012 pots
service stcapp
shutdown
port 2/11
!
dial-peer voice 5013 pots
service stcapp
shutdown
port 2/12
!
dial-peer voice 5014 pots
service stcapp
shutdown
port 2/13
!
dial-peer voice 5015 pots
service stcapp
shutdown
port 2/14
!
dial-peer voice 5016 pots
service stcapp
shutdown
port 2/15
!
dial-peer voice 5017 pots
service stcapp
shutdown
port 2/16
!

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Configuration Examples for Security

dial-peer voice 5018 pots

service stcapp
shutdown
port 2/17
!
dial-peer voice 2001 pots
destination-pattern 2001
port 2/18
!
dial-peer voice 1000 voip
destination-pattern 1...
session target ipv4:10.3.105.5
!
dial-peer voice 5900 voip
destination-pattern 59..
session target ipv4:10.3.105.5
!
dial-peer voice 500 voip
destination-pattern 5...
session target ipv4:10.4.177.51
!
dial-peer voice 5019 pots
service stcapp
shutdown
port 2/18
!
dial-peer voice 5020 pots
service stcapp
shutdown
port 2/19
!
dial-peer voice 5021 pots
service stcapp
shutdown
port 2/20
!
dial-peer voice 5022 pots
service stcapp
shutdown
port 2/21
!
dial-peer voice 5023 pots
service stcapp
shutdown
port 2/22
!
dial-peer voice 5024 pots
service stcapp
shutdown
port 2/23
!
!
!
!
line con 0
transport output all
line aux 0
transport output all
line vty 0 4
password lab
login
transport input all
transport output all
!

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Configuration Examples for Security

ntp clock-period 17179541

ntp server 10.4.177.51
end

Secure Cisco Unified CME: Example

2851-CiscoUnifiedCME# show running-config

Building configuration...

Current configuration : 12735 bytes

!
! No configuration change since last restart
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
service internal
!
hostname 2851-CiscoUnifiedCME
!
boot-start-marker
boot-end-marker
!
card type e1 1 1
logging queue-limit 10000
logging buffered 9999999 debugging
logging rate-limit 10000
no logging console
!
aaa new-model
!
!
aaa accounting connection h323 start-stop group radius
!
aaa session-id common
!
resource policy
!
clock timezone IST 5
no network-clock-participate slot 1
!
!
ip cef
!
!
isdn switch-type primary-net5
!
voice-card 0
no dspfarm
!
voice-card 1
no dspfarm
!
!
ctl-client
server capf 10.13.32.11 trustpoint mytrustpoint1
server tftp 10.13.32.11 trustpoint mytrustpoint1
server cme 10.13.32.11 trustpoint mytrustpoint1
sast1 trustpoint mytrustpoint1
sast2 trustpoint sast2

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Configuration Examples for Security

!
capf-server
port 3084
auth-mode null-string
cert-enroll-trustpoint iosra password 1 mypassword
trustpoint-label mytrustpoint1
source-addr 10.13.32.11
phone-key-size 512
!
voice call debug full-guid
!
voice service voip
srtp fallback
allow-connections h323 to h323
no supplementary-service h450.2
no supplementary-service h450.3
no supplementary-service h450.7
supplementary-service media-renegotiate
h323
emptycapability
ras rrq ttl 4000
!
!
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
!
voice class codec 3
codec preference 1 g729r8
codec preference 8 g711alaw
codec preference 9 g711ulaw
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g728
codec preference 3 g723ar63
codec preference 4 g711ulaw
!
!
voice iec syslog
voice statistics type iec
voice statistics time-range since-reset
!
!
!
crypto pki server myra
database level complete
grant auto
lifetime certificate 1800
!
crypto pki trustpoint myra
enrollment url http://10.13.32.11:80
revocation-check none
rsakeypair iosra
!
crypto pki trustpoint mytrustpoint1
enrollment url http://10.13.32.11:80
revocation-check none
rsakeypair mytrustpoint1
!
crypto pki trustpoint sast2
enrollment url http://10.13.32.11:80
revocation-check none
rsakeypair sast2

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Configuration Examples for Security

!
!
crypto pki certificate chain myra
certificate ca 01
308201F9 30820162 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
10310E30 0C060355 04031305 696F7372 61301E17 0D303630 37303730 35343031
375A170D 30393037 30363035 34303137 5A301031 0E300C06 03550403 1305696F
73726130 819F300D 06092A86 4886F70D 01010105 0003818D 00308189 02818100
D8CE29F9 C9FDB1DD 0E1517E3 6CB4AAF7 52B83DE2 1C017ACA DFC4AF42 F9D10D08
E74BF95B 29378902 B49E32C4 85907384 84CAE4B2 7759BB84 8AB1F578 580793C4
B11A2DBE B2ED02CC DA0C3824 A5FCC377 18CE87EA C0C297BA BE54530F E62247D8
1483CD14 9FD89EFE 05DFBB37 E03FD3F8 B2B1C0B8 A1931BCC B1174A9E 6566F8F5
02030100 01A36330 61300F06 03551D13 0101FF04 05300301 01FF300E 0603551D
0F0101FF 04040302 0186301F 0603551D 23041830 168014B7 16F6FD67 29666C90
D0C62515 E14265A9 EB256230 1D060355 1D0E0416 0414B716 F6FD6729 666C90D0
C62515E1 4265A9EB 2562300D 06092A86 4886F70D 01010405 00038181 002B7F41
64535A66 D20D888E 661B9584 5E3A28DF 4E5A95B9 97E57CAE B07A7C38 7F3B60EE
75C7E5DE 6DF19B06 5F755FB5 190BABFC EF272CEF 865FE01B 1CE80F98 F320A569
CAFFA5D9 3DB3E7D8 8A86C66C F227FF81 6C4449F2 AF8015D9 8129C909 81AFDC01
180B61E8 85E19873 96DB3AE3 E6B70726 9BF93521 CA2FA906 99194ECA 8F
quit
crypto pki certificate chain mytrustpoint1
certificate 02
308201AB 30820114 A0030201 02020102 300D0609 2A864886 F70D0101 04050030
10310E30 0C060355 04031305 696F7372 61301E17 0D303630 37303730 35343233
385A170D 30393037 30363035 34303137 5A301A31 18301606 092A8648 86F70D01
09021609 32383531 2D434D45 32305C30 0D06092A 864886F7 0D010101 0500034B
00304802 4100B3ED A902646C 3851B7F6 CF94887F 0EC437E3 3B6FEDB2 2B4B45A6
3611C243 5A0759EA 1E8D96D1 60ABE028 ED6A3F2A E95DCE45 BE0921AF 82E53E57
17CC12F0 C1270203 010001A3 4F304D30 0B060355 1D0F0404 030205A0 301F0603
551D2304 18301680 14B716F6 FD672966 6C90D0C6 2515E142 65A9EB25 62301D06
03551D0E 04160414 4EE1943C EA817A9E 7010D5B8 0467E9B0 6BA76746 300D0609
2A864886 F70D0101 04050003 81810003 564A6DA1 868B2669 7C096F9A 41173CFC
E49246EE C645E30B A0753E3B E1A265D1 6EA5A829 F10CD0E8 3F2E3AD4 39D8DFE8
83525F2B D19F5E15 F27D6262 62852D1F 43629B68 86D91B5F 7B2E2C25 3BD2CCC3
00EF4028 714339B2 6A7E0B2F 131D2D9E 0BE08853 5CCAE47C 4F74953C 19305A20
B2C97808 D6E01351 48366421 A1D407
quit
certificate ca 01
308201F9 30820162 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
10310E30 0C060355 04031305 696F7372 61301E17 0D303630 37303730 35343031
375A170D 30393037 30363035 34303137 5A301031 0E300C06 03550403 1305696F
73726130 819F300D 06092A86 4886F70D 01010105 0003818D 00308189 02818100
D8CE29F9 C9FDB1DD 0E1517E3 6CB4AAF7 52B83DE2 1C017ACA DFC4AF42 F9D10D08
E74BF95B 29378902 B49E32C4 85907384 84CAE4B2 7759BB84 8AB1F578 580793C4
B11A2DBE B2ED02CC DA0C3824 A5FCC377 18CE87EA C0C297BA BE54530F E62247D8
1483CD14 9FD89EFE 05DFBB37 E03FD3F8 B2B1C0B8 A1931BCC B1174A9E 6566F8F5
02030100 01A36330 61300F06 03551D13 0101FF04 05300301 01FF300E 0603551D
0F0101FF 04040302 0186301F 0603551D 23041830 168014B7 16F6FD67 29666C90
D0C62515 E14265A9 EB256230 1D060355 1D0E0416 0414B716 F6FD6729 666C90D0
C62515E1 4265A9EB 2562300D 06092A86 4886F70D 01010405 00038181 002B7F41
64535A66 D20D888E 661B9584 5E3A28DF 4E5A95B9 97E57CAE B07A7C38 7F3B60EE
75C7E5DE 6DF19B06 5F755FB5 190BABFC EF272CEF 865FE01B 1CE80F98 F320A569
CAFFA5D9 3DB3E7D8 8A86C66C F227FF81 6C4449F2 AF8015D9 8129C909 81AFDC01
180B61E8 85E19873 96DB3AE3 E6B70726 9BF93521 CA2FA906 99194ECA 8F
quit
crypto pki certificate chain sast2
certificate 03
308201AB 30820114 A0030201 02020103 300D0609 2A864886 F70D0101 04050030
10310E30 0C060355 04031305 696F7372 61301E17 0D303630 37303730 35343331
375A170D 30393037 30363035 34303137 5A301A31 18301606 092A8648 86F70D01
09021609 32383531 2D434D45 32305C30 0D06092A 864886F7 0D010101 0500034B
00304802 4100C703 840B11A7 81FCE5AE A14FE593 5114D3C2 5473F488 B8FB4CC5
41EAFA3A D99381D8 21AE6AA9 BA83A84E 9DF3E8C6 54978787 5EF6CC35 C334D55E

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Configuration Examples for Security

A3051372 17D30203 010001A3 4F304D30 0B060355 1D0F0404 030205A0 301F0603

551D2304 18301680 14B716F6 FD672966 6C90D0C6 2515E142 65A9EB25 62301D06
03551D0E 04160414 EB2146B4 EE24AA61 8B5D2F8D 2AD3B786 CBADC8F2 300D0609
2A864886 F70D0101 04050003 81810057 BA0053E9 8FD54B25 72D85A4C CAB47F26
8316F494 E94DFFB9 8E9D065C 9748465C F54719CA C7724F50 67FBCAFF BC332109
DC2FB93D 5AD86583 EDC3E648 39274CE8 D4A5F002 5F21ED3C 6D524AB7 7F5B1876
51867027 9BD2FFED 06984558 C903064E 5552015F 289BA9BB 308D327A DFE0A3B9
78CF2B02 2DD4C208 80CDC0A8 43A26A
quit
certificate ca 01
308201F9 30820162 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
10310E30 0C060355 04031305 696F7372 61301E17 0D303630 37303730 35343031
375A170D 30393037 30363035 34303137 5A301031 0E300C06 03550403 1305696F
73726130 819F300D 06092A86 4886F70D 01010105 0003818D 00308189 02818100
D8CE29F9 C9FDB1DD 0E1517E3 6CB4AAF7 52B83DE2 1C017ACA DFC4AF42 F9D10D08
E74BF95B 29378902 B49E32C4 85907384 84CAE4B2 7759BB84 8AB1F578 580793C4
B11A2DBE B2ED02CC DA0C3824 A5FCC377 18CE87EA C0C297BA BE54530F E62247D8
1483CD14 9FD89EFE 05DFBB37 E03FD3F8 B2B1C0B8 A1931BCC B1174A9E 6566F8F5
02030100 01A36330 61300F06 03551D13 0101FF04 05300301 01FF300E 0603551D
0F0101FF 04040302 0186301F 0603551D 23041830 168014B7 16F6FD67 29666C90
D0C62515 E14265A9 EB256230 1D060355 1D0E0416 0414B716 F6FD6729 666C90D0
C62515E1 4265A9EB 2562300D 06092A86 4886F70D 01010405 00038181 002B7F41
64535A66 D20D888E 661B9584 5E3A28DF 4E5A95B9 97E57CAE B07A7C38 7F3B60EE
75C7E5DE 6DF19B06 5F755FB5 190BABFC EF272CEF 865FE01B 1CE80F98 F320A569
CAFFA5D9 3DB3E7D8 8A86C66C F227FF81 6C4449F2 AF8015D9 8129C909 81AFDC01
180B61E8 85E19873 96DB3AE3 E6B70726 9BF93521 CA2FA906 99194ECA 8F
quit
!
!
username admin password 0 mypassword2
username cisco password 0 mypassword2
!
!
controller E1 1/0
pri-group timeslots 1-31
!
controller E1 1/1
pri-group timeslots 1-31
gw-accounting aaa
!
!
!
!
!
!
!
interface GigabitEthernet0/0
ip address 10.13.32.11 255.255.255.0
duplex auto
speed auto
fair-queue 64 256 32
h323-gateway voip interface
h323-gateway voip id GK1 ipaddr 10.13.32.13 1719
h323-gateway voip id GK2 ipaddr 10.13.32.16 1719
h323-gateway voip h323-id 2851-CiscoUnifiedCME
h323-gateway voip tech-prefix 1#
ip rsvp bandwidth 1000 100
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!

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Configuration Examples for Security

interface Serial1/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
!
interface Serial1/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
!
ip route 0.0.0.0 0.0.0.0 10.13.32.1
!
!
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
!
!
!
!
!
!
tftp-server flash:music-on-hold.au
tftp-server flash:TERM70.DEFAULT.loads
tftp-server flash:TERM71.DEFAULT.loads
tftp-server flash:P00308000300.bin
tftp-server flash:P00308000300.loads
tftp-server flash:P00308000300.sb2
tftp-server flash:P00308000300.sbn
tftp-server flash:SCCP70.8-0-3S.loads
tftp-server flash:cvm70sccp.8-0-2-25.sbn
tftp-server flash:apps70.1-1-2-26.sbn
tftp-server flash:dsp70.1-1-2-26.sbn
tftp-server flash:cnu70.3-1-2-26.sbn
tftp-server flash:jar70sccp.8-0-2-25.sbn
radius-server host 10.13.32.241 auth-port 1645 acct-port 1646
radius-server timeout 40
radius-server deadtime 2
radius-server key cisco
radius-server vsa send accounting
!
control-plane
!
no call rsvp-sync
!
!
voice-port 1/0/0
!
voice-port 1/0/1
!
voice-port 1/0:15
!
voice-port 1/1:15
!
!
!
!

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Configuration Examples for Security

!
dial-peer voice 1 voip
destination-pattern ........
voice-class codec 2
session target ras
incoming called-number 9362....
dtmf-relay h245-alphanumeric
req-qos controlled-load audio
!
dial-peer voice 2 pots
destination-pattern 93621101
!
dial-peer voice 3 pots
destination-pattern 93621102
!
dial-peer voice 10 voip
destination-pattern 2668....
voice-class codec 1
session target ipv4:10.13.46.200
!
dial-peer voice 101 voip
shutdown
destination-pattern 5694....
voice-class codec 1
session target ipv4:10.13.32.10
incoming called-number 9362....
!
dial-peer voice 102 voip
shutdown
destination-pattern 2558....
voice-class codec 1
session target ipv4:10.13.32.12
incoming called-number 9362....
!
dial-peer voice 103 voip
shutdown
destination-pattern 9845....
voice-class codec 1
session target ipv4:10.13.32.14
incoming called-number 9362....
!
dial-peer voice 104 voip
shutdown
destination-pattern 9844....
voice-class codec 1
session target ipv4:10.13.32.15
incoming called-number 9362....
!
dial-peer voice 201 pots
destination-pattern 93625...
no digit-strip
direct-inward-dial
port 1/0:15
!
dial-peer voice 202 pots
destination-pattern 93625...
no digit-strip
direct-inward-dial
port 1/1:15
!
!
gateway
timer receive-rtp 1200
!

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Configuration Examples for Security

!
!
telephony-service
load 7960-7940 P00308000300
max-ephones 4
max-dn 4
ip source-address 10.13.32.11 port 2000
auto assign 1 to 4
secure-signaling trustpoint mytrustpoint1
cnf-file location flash:
cnf-file perphone
voicemail 25589000
max-conferences 4 gain -6
call-forward pattern .T
moh flash:music-on-hold.au
web admin system name admin password mypassword2
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern ........
tftp-server-credentials trustpoint mytrustpoint1
server-security-mode secure
device-security-mode encrypted
create cnf-files version-stamp 7960 Oct 25 2006 07:19:39
!
!
ephone-dn 1
number 93621000
name 2851-PH1
call-forward noan 25581101 timeout 10
!
!
ephone-dn 2
number 93621001
name 2851-PH2
call-forward noan 98441000 timeout 10
!
!
ephone-dn 3
number 93621002
name 2851-PH3
!
!
ephone-dn 4
number 93621003
name 2851-PH4
!
!
ephone 1
no multicast-moh
device-security-mode encrypted
mac-address 0012.4302.A7CC
type 7970
button 1:1
!
!
!
ephone 2
no multicast-moh
device-security-mode encrypted
mac-address 0017.94CA.9CCD
type 7960
button 1:2
!

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Where to Go Next

!
!
ephone 3
no multicast-moh
device-security-mode encrypted
mac-address 0017.94CA.9833
type 7960
button 1:3
!
!
!
ephone 4
no multicast-moh
device-security-mode none
mac-address 0017.94CA.A141
type 7960
button 1:4
!
!
!
line con 0
logging synchronous level all limit 20480000
line aux 0
line vty 0 4
!
scheduler allocate 20000 1000
ntp clock-period 17179791
ntp server 10.13.32.12
!
webvpn context Default_context
ssl authenticate verify all
!
no inservice
!
!
end

Where to Go Next
PKI Management
Cisco IOS public key infrastructure (PKI) provides certificate management to support security protocols
such as IP Security (IPsec), secure shell (SSH), and secure socket layer (SSL). For more information,
see the following documents:
• “Part 5: Implementing and Managing a PKI” in the Cisco IOS Security Configuration Guide for your
Cisco IOS release.
• Cisco IOS Security Command Reference for your Cisco IOS release.

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • “Implementing and Managing a PKI” section in the
Cisco IOS Security Configuration Guide.
• Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Security

Feature Information for Security

Table 22 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 22 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 22 Feature Information for Security

Cisco Unified CME

Feature Name Version Feature Information
Media Encryption (SRTP) on 4.2 Media encryption on Cisco Unified CME was introduced.
Cisco Unified CME
Phone Authentication 4.0 Phone authentication for Cisco Unified CME phones was
introduced.

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 Configuring Automatic Line Selection

Last Updated: March 26, 2007

This chapter describes automatic line selection features in Cisco Unified Communications Manager
Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Automatic Line
Selection” section on page 466.

Contents
• Information About Automatic Line Selection, page 461
• How to Configure Automatic Line Selection, page 462
• Configuration Examples for Automatic Line Selection, page 464
• Additional References, page 465
• Feature Information for Automatic Line Selection, page 466

Information About Automatic Line Selection

To enable automatic line selection, you should understand the following concept:
• Automatic Line Selection for Incoming and Outgoing Calls, page 461

Automatic Line Selection for Incoming and Outgoing Calls

On multiline IP phones, lifting the handset automatically selects the first ringing line on the phone or, if
no line is ringing, selects the first available idle line for outgoing calls. This is the default behavior for
all multiline IP phones.
Under some circumstances, however, you might want to require that a line button be explicitly pressed
to select an outgoing line or to answer an incoming call. In Cisco CME 3.0 and later, you have the
flexibility to assign the type of line selection that each IP phone uses.

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How to Configure Automatic Line Selection

The Automatic Line Selection feature allows you to specify, on a per-phone basis, the line that is selected
when you pick up a phone handset.
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. This is
the default.
• 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 does not 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 for an outgoing call selects
the line associated with the button-number argument. If a button number is 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. You must press an available line button to make an
outgoing call. Incoming calls must be answered by pressing the Answer soft key or pressing a
ringing line button. Use the auto-line command with the button-number argument.
• Automatic line selection for incoming and outgoing calls—Pressing the Answer soft key or picking
up the handset answers an incoming call on the line associated with the specified button. Picking up
the handset for outgoing calls selects the line associated with the specified button. Use the auto-line
command with the button-number argument and answer-incoming keyword.

How to Configure Automatic Line Selection

This section contains the following tasks:
• Enabling Automatic Line Selection, page 462 (required)
• Verifying Automatic Line Selection, page 464 (optional)

Enabling Automatic Line Selection

To enable automatic line selection for answering incoming calls or making outgoing calls, perform the
following steps:

Restrictions
Automatic line selection is bypassed if it is configured for a trunk directory number and the line is seized
by pressing the Park or Callfwd soft keys. The first available directory number is seized.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. auto-line [button-number [answer-incoming] | incoming]

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5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 4 auto-line [button-number [answer-incoming] | Assigns a type of line selection behavior to this phone.
incoming]
• auto-line—Picking up the handset answers the first
ringing line or, if no line is ringing, selects the first idle
Example:
Router(config-ephone)# auto-line 5 line. This is the default.
answer-incoming • auto-line button-number—Picking up the handset for
an outgoing call selects the line associated with the
specified button. The default if this argument is not
used is the topmost available line.
• auto-line button-number answer-incoming—Picking
up the handset answers the incoming call on the line
associated with the specified button.
• 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.
• no auto-line—Disables 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.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

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Configuration Examples for Automatic Line Selection

Verifying Automatic Line Selection

Step 1 Use the show running-config command to verify your configuration. Automatic line selection is listed
in the ephone portion of the output.
Router# show running-config

ephone 2
headset auto-answer line 1
headset auto-answer line 4
ephone-template 1
mac-address 011F.9010.1790
paging-dn 48
type 7960
no dnd feature-ring
no auto-line

Step 2 Use the show telephony-service ephone command to display only ephone configuration information.
Router# show telephony-service ephone

ephone 4
device-security-mode none
username "Accounting"
mac-address FF0E.4857.5E91
button 1c34,35
no auto-line

Configuration Examples for Automatic Line Selection

This section contains the following example:
• Automatic Line Selection: Example, page 464

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

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Additional References

The following example enables automatic selection of line button 1 when the handset is lifted to answer
incoming calls or to make outgoing calls.
ephone 1
mac-address 0001.0002.0003
type 7960
auto-line 1 answer-incoming
button 1:1 2:2 3:3

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Automatic Line Selection

Feature Information for Automatic Line Selection

Table 23 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 23 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

.
Table 23 Feature Information for Automatic Line Selection

Cisco Unified CME

Feature Name Version Feature Information
Automatic Line Selection 4.0 The answer-incoming keyword was added to the auto-line
command.
3.1 The button-number argument was added to the auto-line
command.
3.0 Automatic line selection was introduced.

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 Configuring Call Blocking

Last Updated: March 26, 2007

This chapter describes Call Blocking features in Cisco Unified Communications Manager Express
(Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Call Blocking” section
on page 483.

Contents
• Information About Call Blocking, page 467
• How to Configure Call Blocking, page 469
• Configuration Examples for Call Blocking, page 479
• Where to Go Next, page 481
• Additional References, page 481
• Feature Information for Call Blocking, page 483

Information About Call Blocking

To configure call blocking features, you should understand the following concepts:
• Call Blocking Based on Date and Time (After-Hours Toll Bar), page 468
• Call Blocking Override, page 468
• Class of Restriction, page 469

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Information About Call Blocking

Call Blocking Based on Date and Time (After-Hours Toll Bar)

Call blocking to prevent unauthorized use of phones is implemented by matching dialed numbers against
a pattern of specified digits and matching the time against the time of day and day of week or date that
has been specified for call blocking. 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 Cisco Unified CME, although individual IP phones can be
exempted from all call blocking.
In Cisco CME 3.4 and later versions, the same time-based call-blocking mechanism that is provided for
SCCP phones is expanded to SIP endpoints. Call blocking to prevent unauthorized use of Cisco Unified
IP phones is implemented by matching a pattern of specified digits during a particular time of the day
and day of the week or date.You can specify up to 32 patterns of digits for blocking.
Prior to Cisco CME 3.4, call blocking is supported on IP phones and on analog phones connected to
SCCP-controlled analog telephone adaptors (Cisco ATA) or SCCP-controlled foreign exchange station
(FXS) ports. This feature supports incoming SIP and analog FXS calls. In Cisco CME 3.4 and later,
call-blocking configuration applies to all SCCP, H.323, SIP and POTS calls that go through the
Cisco Unified CME router.
The Cisco Unified CME session application accesses the current after-hours configuration and applies
it to calls originated by SIP phones that are registered to the Cisco Unified CME router. The after-hours
commands are the same as for SCCP phones in Cisco Unified CME.
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, the call is immediately terminated and the
caller will hear a fast busy signal.
For configuration information, see the “Configuring Call Blocking Based on Date and Time” section on
page 474.

Call Blocking Override

The after-hours configuration applies globally to all dial peers in Cisco Unified CME. You can disable
the feature on phones using one of three mechanisms:
• directory number—To configure an exception for an individual directory number.
• phone-level—To configure an exception for all directory numbers associated to a Cisco Unified
IP phone regardless of any configuration for an individual directory number.
• dial peer—To configure an exception for a particular dial peer.
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 IP
phones that support soft keys, such as the Cisco Unified IP Phone 7940G and the Cisco Unified 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.
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 in effect 7 days a week, 24 hours a day, and they
cannot be overridden by using a PIN.

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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.
For configuration information, see the following sections:
• “Configuring Call Blocking Exemption for a Dial Peer” section on page 475.
• “SCCP: Configuring Call Blocking Exemption for an Individual Phone” section on page 476.
• “SIP: Configuring Call Blocking Exemption for an Individual Phone or Directory Number” section
on page 477.

Class of Restriction
Class of restriction (COR) is the capability to deny certain call attempts based on the incoming and
outgoing class of restrictions provisioned on the dial peers. COR specifies 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.
COR functionality provides flexibility in network design by allowing users to block calls (for example,
calls to 900 numbers) and allowing different restrictions to call attempts from different originators.
For configuration information, see the “SCCP: Applying Class of Restriction to a Directory Number”
section on page 470.

How to Configure Call Blocking

This section contains the following tasks:
• SCCP: Applying Class of Restriction to a Directory Number, page 470
• SIP: Applying Class of Restriction to Directory Number, page 471
• Verifying Class of Restriction, page 472
• Configuring Call Blocking Based on Date and Time, page 474
• Configuring Call Blocking Exemption for a Dial Peer, page 475
• SCCP: Configuring Call Blocking Exemption for an Individual Phone, page 476
• SIP: Configuring Call Blocking Exemption for an Individual Phone or Directory Number, page 477
• Verifying Call Blocking Based on Date and Time, page 478

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How to Configure Call Blocking

SCCP: Applying Class of Restriction to a Directory Number

To apply a class of restriction to a directory number, perform the following steps.

Prerequisites
• COR lists must be created in dial peers. For information, see the “Class of Restrictions” section in
the “Dial Peer Configuration on Voice Gateway Routers” document in the Cisco IOS Voice
Configuration Library.
• Directory number to which COR is to be applied must be configured in Cisco Unified CME. For
configuration information, see “SCCP: Creating Directory Numbers” on page 158.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag
4. corlist {incoming | outgoing} cor-list-name
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode.

Example:
Router(config)# ephone-dn 12
Step 4 corlist {incoming | outgoing} Configures a COR on the dial peers associated with an ephone-dn.
cor-list-name

Example:
Router(config-ephone-dn)# corlist
outgoing localcor
Step 5 end Returns to privileged EXEC mode.

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

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SIP: Applying Class of Restriction to Directory Number

To apply a class of restriction to virtual dial peers for directory numbers associated with a SIP IP phone
connected to Cisco Unified CME, perform the following steps.

Prerequisites
• Cisco unified CME 3.4 or a later version.
• COR lists must be created in dial peers. For information, see the “Class of Restrictions” section in
the “Dial Peer Configuration on Voice Gateway Routers” document in the Cisco IOS Voice
Configuration Library.
• Individual phones to which COR is to be applied must be configured in Cisco Unified CME. For
configuration information, see “SIP: Creating Directory Numbers” on page 162.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register pool pool-tag
4. cor {incoming | outgoing} cor-list-name {cor-list-number starting-number [- ending-number] |
default}
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone in
Cisco Unified CME.
Example:
Router(config)# voice register pool 3
Step 4 cor { incoming | outgoing} cor-list-name Configures a class of restriction (COR) for the dynamically
{cor-list-number starting-number [- created VoIP dial peers associated with directory numbers
ending-number] | default}
and specifies which incoming dial peer can use which
outgoing dial peer to make a call.
Example: • Each dial peer can be provisioned with an incoming and
Router(config-register-pool)# cor incoming
an outgoing COR list.
call91 1 91011

Step 5 end Exits configuration mode and enters privileged EXEC

mode.
Example:
Router(config-register-pool)# end

Verifying Class of Restriction

Step 1 Use the show running-config command or the show telephony-service ephone-dn command to verify
whether the COR lists have been applied to the appropriate ephone-dns.
Router# show running-config

ephone-dn 23
number 2835
corlist outgoing 5x
Step 2 Use the show dialplan dialpeer command to determine which outbound dial peer is matched for an
incoming call, based on the COR criteria and the dialed number specified in the command line. Use the
timeout keyword to enable matching variable-length destination patters associated with dial peers. This
can increase your chances of finding a match for the dial peer number you specify.
Router# show dialplan dialpeer 300 number 1900111

VoiceOverIpPeer900
information type = voice,
description = `',
tag = 900, destination-pattern = `1900',
answer-address = `', preference=0,
numbering Type = `unknown'
group = 900, Admin state is up, Operation state is up,

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incoming called-number = `', connections/maximum = 0/unlimited,

DTMF Relay = disabled,
modem passthrough = system,
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:to900
type = voip, session-target = `ipv4:1.8.50.7',
technology prefix:
settle-call = disabled
...
Time elapsed since last clearing of voice call statistics never
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.
Matched: 19001111 Digits: 4
Target: ipv4:1.8.50.7

Step 3 Use the show dial-peer voice command to display the attributes associated with a particular dial peer.
Router# show dial-peer voice 100

VoiceEncapPeer100
information type = voice,
description = `',
tag = 100, destination-pattern = `',
answer-address = `', preference=0,
numbering Type = `unknown'
group = 100, Admin state is up, Operation state is up,
Outbound state is up,
incoming called-number = `555....', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
huntstop = disabled,
in bound application associated: 'vxml_inb_app'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
type = pots, prefix = `',
forward-digits default
session-target = `', voice-port = `',
direct-inward-dial = disabled,
digit_strip = enabled,
register E.164 number with GK = TRUE

Connect Time = 0, Charged Units = 0,

Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.

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Configuring Call Blocking Based on Date and Time

To define dial patterns and time periods during which calls to those dial patterns are blocked, perform
the following steps.

Restrictions
• Before Cisco CME 3.3, call blocking is not supported on analog phones connected to Cisco ATAs
or FXS ports in H.323 mode.
• Before Cisco CME 3.4, call blocking is not supported on SIP IP phones connected directly in
Cisco Unified CME.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. after-hours block pattern tag pattern [7-24]
5. after-hours day day start-time stop-time
6. after-hours date month date start-time stop-time
7. login [timeout [minutes]] [clear time]
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony service Enters telephony service configuration mode.

Example:
Router(config)# telephony service
Step 4 after-hours block pattern pattern-tag pattern Defines pattern to be matched for blocking calls from IP
[7-24] phones.
• pattern-tag—Unique number pattern for call blocking.
Example: Define up to 32 call-blocking patterns in separate
Router(config-telephony)# after-hours block commands. Range is 1 to 32.
pattern 2 91

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Command or Action Purpose

Step 5 after-hours date month date start-time Defines a recurring period based on date of month during
stop-time which outgoing calls that match defined block patterns are
blocked on IP phones.
Example: • Enter beginning and ending times for call blocking in
Router(config-telephony)# after-hours date jan an HH:MM format using a 24-hour clock. The stop-
1 0:00 23:59
time must be greater than the start-time. The value
24:00 is not valid. If you enter 00:00as a stop time, it is
changed to 23:59. If you enter 00:00 for both start time
and stop time, calls are blocked for the entire 24-hour
period on the specified date.
Step 6 after-hours day day start-time stop-time Defines a recurring period based on day of the week during
which outgoing calls that match defined block patterns are
blocked on IP phones
Example:
Router(config-telephony)# after-hours day sun • Enter beginning and ending times for call blocking, in
0:00 23:59 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 you enter 00:00 as a stop time, it
is changed to 23:59. If you enter 00:00 for both start
time and stop time, calls are blocked for the entire
24-hour period on the specified day.
Step 7 login [timeout [minutes]] [clear time] Deactivates all user logins at a specific time or after a
designated period of idle time on a phone.
Example: • Only for IP phones running SCCP.
Router(config-telephony)# login timeout 120
clear 23:00
• minutes—(Optional) Number from 5 to 1440. Default
is 60.
Step 8 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Configuring Call Blocking Exemption for a Dial Peer

To allow H.323 and SIP trunk calls to utilize the voice gateway in spite of the the after-hours
configuration in Cisco Unified CME, follow the steps in this section.

SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag {pots | voatm | vofr | voip}
4. paramspace callsetup after-hours-exempt true
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 dial-peer voice tag {pots | voatm | vofr | Defines a particular dial peer, specifies the method of voice
voip} encapsulation, and enters dial-peer configuration mode.

Example:
Router(config)# dial peer voice 501 voip
Step 4 paramspace callsetup after-hours-exempt true Exempts a dial peer from call blocking configuration.

Example:
Router(config-dialpeer)# paramspace callsetup
after-hours-exempt true
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-dialpeer)# end
or
Router(config-register-dn)# end

SCCP: Configuring Call Blocking Exemption for an Individual Phone

To exempt all directory numbers associated with an individual SCCP phone from the call blocking
configuration, follow the steps in this section.

Restrictions
Call blocking override is supported only on phones that support soft-key display.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. after-hour exempt
5. pin pin-number
6. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 4 after-hour exempt Specifies that this phone is exempt from call blocking.
Phones exempted in this manner are not restricted from any
call-blocking patterns and no authentication of the phone
Example:
Router(config-ephone)# after-hour exempt
user is required.
Step 5 pin pin-number Declares a personal identification number (PIN) that is used
to log into an ephone.
Example: • pin-number—Number from four to eight digits in
Router(config-ephone)# pin 5555 length.
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

SIP: Configuring Call Blocking Exemption for an Individual Phone or Directory

Number
To exempt all extensions associated with an individual SIP phone or an individual directory number
from the call blocking configuration, follow the steps in this section.

Restrictions

The Login toll-bar override is not supported on SIP IP phones; there is no pin to bypass blocking on IP
phones that are connected to Cisco Unified CME and running SIP.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register pool pool-tag
or
voice register dn dn-tag

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4. after-hour exempt
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
or parameters for specified SIP phone.
voice register dn dn-tag or
Enters voice register dn mode to define a directory number
Example: for a SIP phone, intercom line, voice port, or an MWI.
Router(config)# voice register pool 1
or
Router(config)# voice register dn 1
Step 4 after-hour exempt Exempts all numbers on a SIP phone from call blocking.
or
Example: Exempts an individual directory number from call blocking.
Router(config-register-pool)# after-hour exempt
or
Router(config-register-dn)# after-hour exempt
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-dn)# end
or
Router(config-register-dn)# end

Verifying Call Blocking Based on Date and Time

Step 1 Use the show running-config command to display an entire configuration, including call blocking
number patterns and time periods and the phones that are marked as exempt from call blocking.
telephony-service
fxo hook-flash
load 7960-7940 P00305000600
load 7914 S00103020002
max-ephones 100
max-dn 500
ip source-address 10.115.43.121 port 2000
timeouts ringing 10

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voicemail 7189
max-conferences 8 gain -6
moh music-on-hold.au
web admin system name sys3 password sys3
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
after-hours block pattern 1 91900 7-24
after-hours block pattern 2 9976 7-24
after-hours block pattern 3 9011 7-24
after-hours block pattern 4 91...976.... 7-24
!
create cnf-files version-stamp 7960 Jul 13 2004 03:39:28

Step 2 Use the show ephone login command to display the login status of all phones.
Router# show ephone login

ephone 1 Pin enabled:TRUE Logged-in:FALSE

ephone 2 Pin enabled:FALSE
ephone 3 Pin enabled:FALSE
Step 3 The show voice register dial-peer command displays all the dial peers created dynamically by
SIP phones that have registered, along with configurations for after hours blocking.

Configuration Examples for Call Blocking

This section contains the following examples:
• Call Blocking: Example, page 479
• Class of Restriction: Example, page 480

Call Blocking: 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 tag
number 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 day mon 19:00 07:00
after-hours day tue 19:00 07:00
after-hours day wed 19:00 07:00
after-hours day thu 19:00 07:00
after-hours day fri 19:00 07:00
after-hours day sat 13:00 12:00
after-hours day sun 12:00 07:00
!
ephone 23
mac 00e0.8646.9242
button 1:33

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after-hour exempt
!
ephone 24
mac 2234.1543.6352
button 1:34

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

Class of Restriction: Example

The following example shows 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
!
dial-peer voice 3 pots
corlist outgoing call-911
destination-pattern 9911
port 2/0/0
!

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ephone-dn 1
corlist incoming user1
corlist outgoing user1
!
ephone-dn 2
corlist incoming user2
corlist outgoing user2

Where to Go Next
After modifying a configuration for a Cisco Unified IP phone connected to Cisco Unified CME, you
must reboot the phone to make the changes take effect. For more information, see “Resetting and
Restarting Phones” on page 257.

Soft Key Control

To move or remove the Login soft key on one or more phones, create and apply an ephone template that
contains the appropriate softkeys commands.
For more information, see “Customizing Soft Keys” on page 829.

Ephone-dn Templates
The corlist command can be included in an ephone-dn template that is applied to one or more
ephone-dns. For more information, see “Creating Templates” on page 881.

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

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Additional References

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Call Blocking

Table 24 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 24 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 24 Feature Information for Call Blocking

Cisco Unified CME

Feature Name Version Feature Information
Call Blocking 3.4 Added support for call blocking on SIP IP phones
connected directly in Cisco Unified CME.
3.3 Added support for call blocking on analog phones
connected to Cisco ATAs or FXS ports in H.323 mode.
3.0 • Call blocking based on date and time was introduced.
• Override of call blocking was introduced.
Class of Restriction 3.4 Added support for COR on SIP IP Phones connected
directly in Cisco Unified CME.
2.0 Class of restriction was introduced.

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Last Updated: March 26, 2007

This chapter describes the call park feature in Cisco Unified Communications Manager Express
(Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Call Park” section on
page 498.

Contents
• Information About Call Park, page 485
• How to Configure Call Park, page 490
• Configuration Examples for Call Park, page 495
• Where to Go Next, page 496
• Additional References, page 497
• Feature Information for Call Park, page 498

Information About Call Park

To enable call park, you should understand the following concepts:
• Basic Call Park, page 486
• Dedicated Call-Park Slots, page 488
• Call-Park Blocking, page 489
• Call-Park Redirect, page 489

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Basic 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.
After at least one call-park slot has been defined and the Cisco Unified CME 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 caller who is parked in a park slot hears the music-on-hold (MOH) audio stream if the call uses the
G.711 codec or if the call uses G.729 with transcoding, a feature that is available in Cisco CME 3.2 and
later versions; otherwise, callers hear a tone on hold.
A phone user who has parked a call can retrieve the 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 from phones to
call-park slot numbers. For versions before Cisco Unified CME 4.0, callers can directly dial call-park
slot numbers to be placed in park. If another call is already parked in the slot, the caller hears a busy tone.
In Cisco Unified CME 4.0 and later versions, a direct call to a call-park slot is interpreted as an attempt
to pick up a call that is parked there; if no call is parked in the slot, the caller hears a busy tone.
The ability to directly dial a park slot to retrieve a call is useful in the following scenario. An attendant
connected to a remote Cisco Unified CME system can perform a directed call park (transfer-to-park)
into a park slot on the local Cisco Unified CME router by simply transferring the call to the telephone
number associated with the local Cisco Unified CME park slot. The remote attendant can then inform
local phone users of the existence of the parked call by dialing (across VoIP) into a paging number on
the local Cisco Unified CME system, or the parked call may simply be visible to one or more local users
whose phones are configured to monitor the park-slot. Then, when a local IP phone user directly dials
the extension number of the park slot, the system assumes that the user is requesting retrieval (pickup)
of the call in the park slot. If there is no call in the park slot, the Cisco Unified CME system returns a
busy tone to the local user.
A caller who is parked in a park slot hears the music-on-hold (MOH) audio stream if the call uses a G.711
codec or if it uses G.729 with transcoding, a feature that is available in Cisco CME 3.2 and later versions;
otherwise, callers hear a tone on hold.
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, provided the total number of park slots and
normal extensions does not exceed the maximum number of ephone-dns that was defined with the
max-dn command. After an administrator defines at least one call-park slot and restarts phones, the Park
soft key is displayed on all 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 Unified CME system. In
Cisco CME 3.2.1 and 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. A call that is parked on the monitored call-park slot can be picked up by pressing
the assigned monitor button.

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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.
If 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 the configuration of the
preference and huntstop commands, all calls that are parked after a second call has been parked will generate
a busy signal. The caller who is being transferred to park will hear a busy signal, while the phone user who
parked the call will receive no indication that the call was lost.
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.
You can define both the length of the timeout interval for calls parked at a call-park slot and the number
of timeout intervals that should occur before the call is either recalled or transferred. If you specify a
transfer target in the park-slot command, the call is transferred to the specified target after the timeout
intervals expire rather than to the primary number of the parking phone.
If a name has been specified for the call-park slot using the name command, that name will be displayed
on a recall or transfer rather than an extension number.
You can also specify an alternate target extension to which to transfer a parked call if the recall or
transfer target is in use. In use is defined as either ringing or connected. For example, a call is parked at
the private park slot for the phone with the primary extension of 2001, as shown in Figure 23. After the
timeouts expire, the system attempts to recall the call to extension 2001, but that line is connected to
another call. The system then transfers the call to the alternate target, extension 3784.

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Dedicated Call-Park Slots

A dedicated, private call-park slot can be configured for an ephone using the reserved-for keyword in
the park-slot command. The dedicated call-park slot is associated with the primary extension of the
ephone. All extensions on this phone can park calls in the dedicated park slot. The extensions on this
phone are the only extensions that can park a call in the dedicated park slot. Only one call at a time can
be parked in a park slot; a busy tone is returned to any attempt to park a call in a slot that is already in use.
Calls can be parked in dedicated call-park slots using any of the following methods (the extension doing
the parking must be on a phone whose primary extension is associated with a dedicated park slot).
• With an active call, an IP phone user presses the Park soft key.
• With an active call, an IP phone user presses the Transfer soft key and a standard or custom FAC
(feature access code) for the call-park feature. The standard FAC for call park is **6.
• With an active call, an analog phone user presses hookflash and the standard or custom FAC for the
call park feature.
Calls can be retrieved from dedicated call-park slots using any of the following methods:
• An IP phone user presses the Pickup soft key and dials the park-slot number.
• An IP phone user presses the New Call soft key and dials the park-slot number.
• An analog phone user lifts the handset, presses the standard or custom FAC for directed call pickup,
and dials the park-slot number. The standard FAC for directed pickup is **5.
If no dedicated park slot is found anywhere in the Cisco Unified CME system for an ephone-dn that is
attempting to park a call, the system uses the standard call-park procedure; that is, the system searches
for a preferred park slot (one with an ephone-dn number that matches the last two digits of the ephone-dn
attempting to park the call) and if none is found, uses any available call-park slot.
Figure 23 shows an example of a dedicated call-park slot.
If the configuration specifies that a call should be recalled to the parking phone after the timeout
intervals expire, the call is always returned to the phone’s primary extension number, regardless of
which extension on the phone did the parking. Figure 23 shows an ephone that is configured with the
extension numbers 2001, 2002, and 2003, and a private call-park slot at extension 3333. The private park
slot has been set up to recall calls to the parking phone when the parked call’s timeouts expire. In the
example, extension 2003 parks a call using the Park soft key. When the timeout intervals expire, the call
rings back on extension 2001.
The configuration in Figure 23 specifies that the call will recall or transfer from the park slot after 3
times the 60-second timeout, or after 180 seconds. Also, before the exhaustion of the 3 timeouts the
phone will receive reminder notifications that a parked call is waiting. The reminders are sent after each
60-second timeout interval expires (that is, at 60 seconds and at 120 seconds). You may want to set the
timeout command with a limit of 1 instead, so that the call simply parks and recalls or transfers without
sending a reminder ring.

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Figure 23 Dedicated Call Park Example

2001 2 3754
2002
2003 1

Dedicated
Call-Park Slot 3
3333
1. A user on extension 2003
ephone-dn 1 parks a call using the Park
number 2001
soft key.
ephone-dn 2 2. After three intervals of 60
number 2002
seconds, the call is recalled to the
ephone-dn 3 phone’s primary number, 2001.
number 2003 3. If 2001 is busy, the call is
ephone-dn 4 transferred to 3754.
number 3333
name Park 2001

135130
park-slot reserved-for 2001 timeout 60 limit 3 recall alternate 3754
ephone 2
button 1:1 2:2 3:3

Call-Park Blocking
In Cisco Unified CME 4.0 and later versions, individual ephones can be prevented from making transfers
to call-park slots by using the transfer-park blocked command. This command prevents transfers to
park that use the Transfer soft key and a call-park slot number, while allowing call-parks that use only
the Park soft key. (To prevent use of the Park soft key, use an ephone template to remove it from the
phone. See “Customizing Soft Keys” on page 829.)
An exception is made for phones with reserved, or dedicated, park slots. If the transfer-park blocked
command is used on an ephone that has a dedicated park slot, the phone is blocked from parking calls at
park slots other than the phone’s dedicated park slot but can still park calls at its own dedicated park slot.

Call-Park Redirect
By default, H.323 and SIP calls that use the call-park feature use hairpin call forwarding or transfer to
park calls and to pick up calls from park. The call-park system redirect command allows you to specify
that these calls should use H.450 or the SIP Refer method of call forwarding or transfer. The no form of
the command returns the system to the default behavior.

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How to Configure Call Park

This section contains the following tasks:
• Enabling Call Park, page 490
• Verifying Call Park, page 494
• Troubleshooting Call Park, page 495

Enabling Call Park

To enable call-park slots, optional call-park blocking, or call-park redirect, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. park-slot [reserved-for extension-number] [timeout seconds limit count] [notify
extension-number [only]] [recall] [transfer extension-number] [alternate extension-number]
[retry seconds limit count]
6. exit
7. ephone phone-tag
8. transfer-park blocked
9. exit
10. telephony-service
11. call-park system redirect
12. restart all
13. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 20 ephone-dn during configuration tasks. The maximum number
of ephone-dns for a particular Cisco Unified CME system is
version- and platform-specific.
• 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.
Step 4 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.

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Command or Action Purpose

Step 5 park-slot [reserved-for extension-number] Creates a floating extension (ephone-dn) at which calls can be
[timeout seconds limit count] [notify temporarily held (parked).
extension-number [only]] [recall]
[transfer extension-number] [alternate • reserved-for extension-number—(Optional) Indicates that
extension-number] [retry seconds limit this slot is a private park slot for the phone with the specified
count]
extension number as its primary line. All lines on that phone
can use this park slot.
Example: • timeout seconds—(Optional) Sets the call-park reminder
Router(config-ephone-dn)# park-slot timeout interval, in seconds. Range is 0 to 65535. When the
reserved-for 2458 timeout 60 limit 3
recall alternate 3754
interval expires, the call-park reminder sends a 1-second ring
and displays a message on the LCD panel of the
Cisco Unified IP phone that parked the call and that of any
extension that may be specified with the notify keyword.
Default is that the reminder ring is sent only to the phone that
parked the call.
• limit count—(Optional, applies to timeout keyword) Sets a
limit for the number of time-out intervals for a parked call.
For example, a limit of 3 sends 2 reminder rings (interval 1,
ring 1, interval 2, ring 2, interval 3). A call parked at this slot
is disconnected after the limit has been reached unless
another action has been specified. Range is 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.
• recall—(Optional) Returns the call to the phone that parked
it after the timeout limits expire.
• transfer extension-number—(Optional) Returns the call to
the specified number after timeout limits expire.
• alternate extension-number—(Optional) Returns the call to
the specified second target number if the recall or transfer
target phone is in use on any of its extensions (ringing or in
conversation).
• retry seconds—(Optional) Sets the delay before another
attempt to recall or transfer a parked call, in seconds. Range
is 0 to 65535. Number of attempts is set by the limit
keyword.
• limit count—(Optional, applies to retry keyword) Sets a
limit for the number of retries. When a limit is set, a call
parked at this slot is disconnected after the limit has been
reached. Range is 1 to 65535.

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Command or Action Purpose

Step 6 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 7 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies this
Example: ephone during configuration tasks.
Router(config)# ephone 25
Step 8 transfer-park blocked (Optional) Prevents extensions on this ephone from transferring
calls to call-park slots.
Example: Note This command prevents the use of the Transfer soft key
Router(config-ephone)# transfer-park and slot number to transfer calls to park slots. It does not
blocked prevent use of the Park soft key.
Step 9 exit Exits ephone configuration mode.

Example:
Router(config-telephony)# exit
Step 10 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 11 call-park system redirect Specifies that within the call-park feature, H.323 and SIP calls
will use H.450 or the SIP Refer method of call forwarding or
transfer to park calls and to pick up calls from park.
Example:
Router(config)# call-park system redirect The no form of the command returns to the default behavior,
which is to use hairpin call forwarding or transfer to park calls and
pick up calls from park.
Step 12 restart all Performs a fast reboot of all phones associated with this
Cisco Unified CME router. Does not contact the DHCP server.
Example: Note The first time that call-park slots are defined, IP phones
Router(config)# restart all must be rebooted before the Park soft key is displayed on
phones. This command is not required after subsequent
call-park slot definitions.
Step 13 exit Returns to privileged EXEC mode.

Example:
Router(config)# exit

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Verifying Call Park

Step 1 Use the show running-config command to verify your configuration. Call-park slots are listed in the
ephone-dn portion of the output.
Router# show running-config

!
ephone-dn 23
number 853
park-slot timeout 10 limit 1 recall
description park slot for Sales
!
!
ephone-dn 24
number 8126
park-slot reserved-for 126 timeout 10 limit 1 transfer 8145
!
!
ephone-dn 25
number 8121 secondary 121
park-slot reserved-for 121 timeout 30 limit 1 transfer 8145
!
!
ephone-dn 26
number 8136 secondary 136
park-slot reserved-for 136 timeout 10 limit 1 recall
!
!
ephone-dn 30 dual-line
number 451 secondary 501
preference 10
huntstop channel
!
!
ephone-dn 31 dual-line
number 452 secondary 502
preference 10
huntstop channel
!

Step 2 Use the show telephony-service ephone-dn command to display call park configuration information.
Router# show telephony-service ephone-dn

ephone-dn 26
number 8136 secondary 136
park-slot reserved-for 136 timeout 10 limit 1 recall

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Troubleshooting Call Park

Step 1 show ephone-dn park
Use this command to display configured call-park slots and their status.
Router# show ephone-dn park

DN 50 (1560) park-slot state IDLE

Notify to () timeout 30 limit 10

Step 2 Use the debug ephone commands to observe messages and states associated with an ephone. For more
information, see the Cisco Unified CME Command Reference.

Configuration Examples for Call Park

This section contains the following examples:
• Basic Call Park: Example, page 495
• Phone Blocked From Using Call Park: Example, page 495
• Call-Park Redirect: Example, page 496

Basic Call Park: 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

Phone Blocked From Using Call Park: Example

The following example prevents ephone 25 and extensions 234, 235, and 236 from parking calls at any
call-park slots.
ephone-dn 11
number 234

ephone-dn 12
number 235

ephone-dn 13
number 236

ephone 25
button 1:11 2:12 3:13
transfer-park blocked

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The following example sets up a dedicated park slot for the extensions on ephone 6 and blocks transfers
to call park from extensions 2977, 2978, and 2979 on that phone. Those extensions can still park calls
at the phone’s dedicated park slot by using the Park soft key or the Transfer soft key and the FAC for call
park.
ephone-dn 3
number 2558
name Park 2977
park-slot reserved-for 2977 timeout 60 limit 3 recall alternate 3754

ephone-dn 4
number 2977

ephone-dn 5
number 2978

ephone-dn 6
number 2979

ephone 6
button 1:4 2:5 3:6
transfer-park blocked

Call-Park Redirect: Example

The following example specifies that H.323 and SIP calls that are parked should use H.450 or the SIP
Refer method to when they are parked or picked up.
telephony-service
call-park system redirect

Where to Go Next
Controlling Use of the Park Soft Key
To block the functioning of the call park (Park) soft key without removing the key display, create and
apply an ephone template that contains the features blocked command. For more information, see
“Customizing Soft Keys” on page 829.
To remove the call park (Park) soft key from one or more phones, create and apply an ephone template
that contains the appropriate softkeys command. For more information, see “Customizing Soft Keys”
on page 829.

Ephone Templates
The transfer-park blocked command, which blocks transfers to call-park slots, can be included in
ephone templates that are applied to individual ephones.
The Park soft key can be removed from the display of one or more phones by including the appropriate
softkeys command in an ephone template and applying the template to individual ephones.
For more information, see “Creating Templates” on page 881.

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Additional References

Feature Access Codes

You can park calls using a feature access code (FAC) instead of a soft key on the phone if standard or
custom FACs have been enabled for your system. The call-park FAC is considered a transfer to a
call-park slot and therefore is valid only after the Trnsfer soft key (IP phones) or hookflash (analog
phones) has been used to initiate a transfer. The following are the standard FACs for call park:
• Dedicated park slot—Standard FAC is **6.
• Any available park slot—Standard FAC is **6 plus optional park-slot number.
For more information about FACs, see “Customizing Soft Keys” on page 829.

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Call Park

Table 25 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 25 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 25 Feature Information for Call Park

Cisco Unified CME

Feature Name Version Feature Information
Call Park 4.0 Dedicated call-park slots, alternative recall locations, and
call-park blocking were introduced. Direct calls to park
slots are now interpreted as attempts to pick up parked calls
rather than attempts to be parked at the slot.
3.2.1 Monitoring of call-park slots was introduced.
3.1 Call park was introduced.

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Last Updated: May 23, 2007

This chapter describes call transfer and forwarding features in Cisco Unified Communications Manager
Express (Cisco Unified CME) to enable interworking with various network requirements.

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Call Transfer and
Forwarding” section on page 561.

Contents
• Information About Call Transfer and Forwarding, page 499
• How to Configure Call Transfer and Forwarding, page 518
• Configuration Examples for Call Transfer and Forwarding, page 552
• Where to Go Next, page 559
• Additional References, page 560
• Feature Information for Call Transfer and Forwarding, page 561

Information About Call Transfer and Forwarding

To configure transfer and forwarding features, you should understand the following concepts:
• Call Forwarding, page 500
• B2BUA Call Forwarding for SIP Devices, page 501
• Call Forward All Synchronization for SIP Phones, page 501
• Call Transfer, page 502
• H.450.2 and H.450.3 Support, page 503
• Transfer Method Recommendations by Cisco Unified CME Version, page 506
• H.450.12 Support, page 507
• Hairpin Call Routing, page 507

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• H.450 Tandem Gateways, page 510

• Dial Peers, page 512
• QSIG Supplementary Services, page 512
• Disabling SIP Supplementary Services for Call Forward and Call Transfer, page 513
• Typical Network Scenarios for Call Transfer and Call Forwarding, page 514

Call Forwarding
Call forwarding diverts calls to a specified number under one or more of the following conditions:
• All calls—When all-call call forwarding is activated by a phone user, all incoming calls are diverted.
The target destination for diverted calls can be specified in the router configuration or by the phone
user with a soft key or feature access code. The most recently entered destination is recognized by
Cisco Unified CME, regardless of how it was entered.
• No answer—Incoming calls are diverted when the extension does not answer before the timeout
expires. The target destination for diverted calls is specified in the router configuration.
• Busy—Incoming calls are diverted when the extension is busy and call waiting is not active. The
target destination for diverted calls is specified in the router configuration.
• Night service—All incoming calls are automatically diverted during night-service hours. The target
destination for diverted calls is specified in the router configuration.
A directory number can have all four types of call forwarding defined at the same time with a different
forwarding destination defined for each type of call forwarding. If more than one type of call forwarding
is active at one time, the order for evaluating the different types is as follows:
1. Call forward night-service
2. Call forward all
3. Call forward busy and call forward no-answer
H.450.3 capabilities are enabled globally on the router by default, and can be disabled either globally or
for individual dial peers. You can configure incoming patterns for using the H.450.3 standard.
Calling-party numbers that do not match the patterns defined with this command are forwarded using
Cisco-proprietary call forwarding for backward compatibility. For information about configuring
H.450.3 on a Cisco Unified CME system, see the “SCCP: Enabling Call Forwarding for a Directory
Number” section on page 523.

Selective Call Forwarding

You can apply call forwarding to a busy or no-answer directory number based on the number that is
dialed to reach the directory number: the primary number, the secondary number, or either of those
numbers expanded by a dial-plan pattern.
Cisco Unified CME automatically creates one POTS dial peer for each ephone-dn when it is assigned a
primary number. If the ephone-dn is assigned a secondary number, it creates a second POTS dial peer.
If the dialplan-pattern command is used to expand the primary and secondary numbers for ephone-dns,
it creates two more dial peers, resulting in the creation of the following four dial peers for the ephone-dn:
• A POTS dial peer for the primary number
• A POTS dial peer for the secondary number
• A POTS dial peer for the primary number as expanded by the dialplan-pattern command

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• A POTS dial peer for the secondary number as expanded by the dialplan-pattern command
Call forwarding is normally applied to all dial peers created for an ephone-dn. Selective call forwarding
allows you to apply call forwarding for busy or no-answer calls only for the dial peers you have specified,
based on the called number that was used to route the call to the ephone-dn.
For example, the following commands set up a single ephone-dn (ephone-dn 5) with four dial peers:
telephony-service
dialplan-pattern 1 40855501.. extension-length 4 extension-pattern 50..

ephone-dn 5
number 5066 secondary 5067

In this example, selective call forwarding can be applied so that calls are forwarded when:
• callers dial the primary number 5066.
• when callers dial the secondary number 5067.
• when callers dial the expanded numbers 4085550166 or 4085550167.
For configuration information, see the “SCCP: Enabling Call Forwarding for a Directory Number”
section on page 523.

B2BUA Call Forwarding for SIP Devices

Cisco Unified CME 3.4 an d later versions acts as both UA server and UA client; that is, as a B2BUA.
Calls into a SIP phone can be forwarded to other SIP or SCCP devices (including Cisco Unity or
Cisco Unity Express, third-party voice mail systems, an auto attendant or an IVR system, such as
Cisco Unified IPCC and Cisco Unified IPCC Express). In addition, SCCP phones can be forwarded to
SIP phones.
Cisco Unity or other voice-messaging systems connected by a SIP trunk or SIP user agent are able to
pass an MWI to a SIP phone when a call is forwarded. The SIP phone then displays the MWI when
indicated by the voice-messaging system.
The call-forward busy response is triggered when a call is sent to a SIP phone using a VoIP dial peer and
a busy response is received back from the phone. SIP-to-SIP call forwarding is invoked only if the phone
is dialed directly. Call forwarding is not invoked when the phone number is called through a sequential,
longest-idle, or peer hunt group.
You can configure call forwarding for an individual directory number, or for every number on a SIP
phone. If the information is configured in both, the information under voice register dn takes precedence
over the information configured under voice register pool.
For configuration information, see the “SIP: Configuring SIP-to-SIP Phone Call Forwarding” section on
page 546.

Call Forward All Synchronization for SIP Phones

The Call Forward All feature allows users to forward all incoming calls to a phone number that they
specify. This feature is supported on all SIP phones and can be provisioned from either
Cisco Unified CME or the individual SIP phone. Before Cisco Unified CME 4.1, there was no method
for exchanging the Call Forward All configuration between Cisco Unified CME and the SIP phone. If
Call Forward All was enabled on the phone, the configuration in Cisco Unified CME was not updated;
conversely, the configuration in Cisco Unified CME was not sent to the phone.

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In Cisco Unified CME 4.1 and later, the following enhancements are supported for the
Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE to keep the
configuration consistent between Cisco Unified CME and the SIP phone:
• When Call Forward All is configured on Cisco Unified CME with the call-forward b2bua all
command, the configuration is sent to the phone which updates the CfwdAll soft key to indicate that
Call forward All is enabled. Because Call Forward All is configured on a per line basis, the CfwdAll
soft key is updated only when Call Forward All is enabled for the primary line.
• When a user enables Call Forward All on a phone using the CfwdAll soft key, the uniform resource
identifier (URI) for the service (defined with the call-feature-uri command) and the call forward
number (unless Call Forward All is disabled) is sent to Cisco Unified CME. It updates its voice
register pool and voice register dn configuration with the call-forward b2bua all command to be
consistent with the phone configuration.
• Call Forward All supports KPML so that a user does not need to press the Dial or # key, or wait for
the interdigit timeout, to configure the Call Forward All number. Cisco Unified CME collects the
Call Forward All digits until it finds a match in the dial peers.
For configuration information, see the “SIP: Configuring Call-Forwarding-All Soft Key URI” section
on page 548.

Call Transfer
When you are connected to another party, call transfer allows you to shift the connection of the other
party to a different number. Call transfer methods must interoperate with systems in the other networks
with which you interface. Cisco CME 3.2 and later versions provide full call-transfer and
call-forwarding interoperability with call processing systems 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.
Call transfers can be blind or consultative. A blind transfer is one in which the transferring extension
connects the caller to a destination extension before ringback begins. A consultative transfer is one in
which the transferring party either connects the caller to a ringing phone (ringback heard) or speaks with
the third party before connecting the caller to the third party.
You can configure blind or consultative transfer on a systemwide basis or for individual extensions. For
example, in a system that is set up for consultative transfer, a specific extension with an auto-attendant
that automatically transfers incoming calls to specific extension numbers can be set to use blind transfer,
because auto-attendants do not use consultative transfer.

Consultative Transfer With Direct Station Select

Direct Station Select (DSS) is a feature that allows a multibutton phone user to transfer calls to an idle
monitored line by pressing the Transfer key and the appropriate monitored line button. A monitored line
is one that appears on two phones; one phone can use the line to make and receive calls and the other
phone simply monitors whether the line is in use. For Cisco CME 3.2 and later versions, consultative
transfers can occur during Direct Station Select (transferring calls to idle monitored lines).
If the person sharing the monitored 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.

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Direct Station Select consultative transfer is enabled with the transfer-system full-consult dss
command, which defines the call transfer method for all lines served by the router. The transfer-system
full-consult dss command supports the keep-conference command. See “Configuring Conferencing”
on page 647.

Call Transfer Blocking

Transfers to all numbers except those on local phones are automatically blocked by default. During
configuration, you can allow transfers to nonlocal numbers. In Cisco Unified CME 4.0 and later
versions, you can prevent individual phones from transferring calls to numbers that are globally enabled
for transfer. This ensures that individual phones do not incur toll charges by transferring calls outside
the Cisco Unified CME system. Call transfer blocking can be configured for individual phones or
configured as part of a template that is applied to a set of phones.
Another way to eliminate toll charges on call transfers is to limit the number of digits that phone users
can dial when transferring calls. For example, if you specify a maximum of eight digits in the
configuration, users who are transferring calls can dial one digit for external access and seven digits
more, which is generally enough for a local number but not a long-distance number. In most locations,
this plan will limit transfers to nontoll destinations. Long-distance calls, which typically require ten
digits or more, will not be allowed. This configuration is only necessary when global transfer to numbers
outside the Cisco Unified CME system has been enabled using the transfer-pattern
(telephony-service) command. Transfers to numbers outside the Cisco Unified CME system are not
permitted by default.

H.450.2 and H.450.3 Support

H.450.2 is a standard protocol for exchanging call-transfer information across a network, and H.450.3
is a standard protocol for exchanging call-forwarding information across a network. Cisco CME 3.0 and
later versions support the H.450.2 call-transfer standards and the H.450.3 call-forwarding standards that
were introduced in Cisco ITS V2.1. Using the H.450.2 and H.450.3 standards to manage call transfer
and forwarding in a VoIP network 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 regardless of whether the transfer-to or forward-to target is on the same
Cisco Unified 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 is calls
that can be reoriginated through hairpin call routing or through the use of an H.450 tandem gateway.
• 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, you must specify a pattern that matches the calling-party numbers of the calls that
you want to be able to forward.

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• Cisco Unified CME supports all SIP Refer method call transfer scenarios, but you must ensure that
call transfer is enabled using H.450.2 standards.
• H.450 standards are not supported by Cisco Unified Communications Manager, 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 24 on
page 504 shows A calling B. Figure 25 on page 504 shows B consulting with C and putting A on hold.
Figure 26 on page 505 shows that B has connected A and C, and Figure 27 on page 505 shows A and C
directly connected, with B no longer involved in the call.

Figure 24 Call Transfer Using H.450.2: A Calls B

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

Cisco Unified CME 2

146629
IP
Phone B

Figure 25 Call Transfer Using H.450.2: B Consults with C

H.323 V
Non-H.450
Cisco Unified CME 1 gateway

IP
Cisco Unified CME 2 Phone C
Phone A

H.450.2 connection H.450.2 connection

146634

IP
Phone B

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Figure 26 Call Transfer Using H.450.2: B Transfers A to C

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

146633
IP
Phone B

Figure 27 Call Transfer Using H.450.2: A and C Are Connected

H.323 V
Non-H.450
Cisco Unified CME 1 gateway

IP
Cisco Unified CME 2 Phone C
Phone A

H.450.2 connection H.450.2 connection

146634
IP
Phone B

Tips for Using H.450 Standards

Use H.450 standards when a network meets the following conditions:
• The router that you are configuring uses Cisco CME 3.0 or a later version, 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 or later systems, if some of the endpoints do not support
H.450 standards (for example, Cisco Unified Communications Manager, 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. For configuration information, see the “Enabling Call Transfer and
Forwarding at System-Level” section on page 519.

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Transfer Method Recommendations by Cisco Unified CME Version

You must specify the method to use for call transfers: H.450.2 standard signaling or Cisco proprietary
signaling, and whether transfers should be blind or allow consultation. Table 26 summarizes transfer
method recommendations for all Cisco Unified CME versions.

Table 26 Transfer Method Recommendations

Cisco Unified CME transfer-system transfer-system

Version Command Default Keyword to Use Transfer Method Recommendation
4.0 and later full-consult full-consult Use H.450.2 for call transfer, which is the default for this
or version. You do not need to use the transfer-system command
full-blind unless you want to use the full-blind or dss keyword.
Optionally, you can use the proprietary Cisco method by using
the transfer-system command with the blind or local-consult
keyword.
Use H.450.7 for call transfer using QSIG supplementary
services
3.0 to 3.3 blind full-consult Use H.450.2 for call transfer. You must explicitly configure the
or transfer-system command with the full-consult or full-blind
full-blind keyword because H.450.2 is not the default for this version.
Optionally, you can use the proprietary Cisco method by using
the transfer-system command with the blind or local-consult
keyword.
2.1 blind blind Use the Cisco proprietary method, which is the default for this
or version. You do not need to use the transfer-system command
local-consult unless you want to use the local-consult keyword.
Optionally, you can use the transfer-system command with the
full-consult or full-blind keyword. You must also configure the
router with a Tcl script that is contained in the
app-h450-transfer.x.x.x.x.zip file. This file is available from the
Cisco Unified CME software download website at
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp. For
configuration information, see the Cisco IOS Telephony
Services Version 2.1 guide.
Earlier than 2.1 blind blind Use the Cisco proprietary method, which is the default for this
version. You do not need to use the transfer-system command
unless you want to use the local-consult keyword.

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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 on a call-by-call basis. When 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.
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 other VoIP endpoints involved in the call. If a
positive H.450.12 indication is 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.
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.
• Not all gateway endpoints support H.450.2 and H.450.3 standards. Therefore, specify how
non-H.450 calls are to be handled by choosing one of the following options:
– Enable 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. 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. See the “Enabling H.450.12 Capabilities” section on
page 529.
Support for the H.450.12 standard is disabled by default and can be enabled globally or for
individual dial peers.
If the call does not have H.450 support, it can be handled by a VoIP-to-VoIP connection that you
configure using dial peers and the “Enabling H.323-to-H.323 Connection Capabilities” section
on page 531. The connection can be used for hairpin call routing or routing to an H.450 tandem
gateway.
– 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 a VoIP-to-VoIP connection that you configure using dial
peers and the“Enabling H.323-to-H.323 Connection Capabilities” section on page 531. This
connection can be used for hairpin call routing or routing to an H.450 tandem gateway.

Hairpin Call Routing

Cisco CME 3.1 and later supports hairpin call routing using a VoIP-to-VoIP connection 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.

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In 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, see “Configuring Transcoding Resources” on page 303.
Hairpin call routing provides the following benefits:
• Call transfer and forwarding is provided to non-H.450 endpoints, such as
Cisco Unified Communications Manager, 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 Unified CME automatically detects that the remote system is a
Cisco Unified Communications Manager.
Figure 28 on page 508 shows a call that is made from A to B. Figure 29 on page 509 shows that B has
forwarded all calls to C. Figure 30 on page 509 shows that A and C are connected by an H.323 hairpin.

Figure 28 Hairpin with H.323: A Calls B

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

Cisco Unified CME 2

146629

IP
Phone B

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Figure 29 Hairpin with H.323: Call is Forwarded to C

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

146630
IP Calls are forwarded
Phone B to phone C

Figure 30 Hairpin with H.323: A is Connected to C via B

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

Cisco Unified CME 2

146631
IP
Phone B

Tips for Using Hairpin Call Routing

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 version.
• 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 Unified Communications Manager, 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 H.450.12 Capabilities” section on page 529.
• No voice gateway is available to act as an H.450 tandem gateway.
For information about configuring Cisco Unified CME to forward calls using local hairpin routing, see
the “Forwarding Calls Using Local Hairpin Routing” section on page 533.

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Support for VoIP-to-VoIP connections is disabled by default and can be enabled globally. For
configuration information, see the “Enabling H.323-to-H.323 Connection Capabilities” section on
page 531.

H.450 Tandem Gateways

H.450 tandem gateways address the limitations of hairpin call routing using a manner similar to hairpin
call routing but without the double WAN link traversal created by hairpin connections. 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 Unified Communications Manager, 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.
An H.450 tandem gateway is configured with a dial peer that points to the
Cisco Unified Communications Manager 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 Unified CME systems in the private H.450 network. In this way, Cisco Unified CME and
the Cisco Unified Communications Manager 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.
An H.450 tandem gateway can also work as a PSTN gateway for remote Cisco Unified CME systems
and for Cisco Unified Communications Manager (or other non-H.450 system). Use different inbound
dial peers to separate Cisco Unified Communications Manager-to-PSTN G.711 calls from tandem
gateway-to-Cisco Unified CME G.729 calls.

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 Cisco IOS image on the selected router. Consult your Cisco Unified CME SE regarding the required
feature license. With Cisco IOS Release 12.3(7)T, you cannot use Cisco Unified 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 Unified Communications Manager.
For Cisco CME 3.1 and earlier, the only type of VoIP-to-VoIP connection supported by
Cisco Unified 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 Unified CME systems running Cisco Unity Express.
Figure 31 on page 511 shows a tandem voice gateway that is located between the central hub of the
network of a CPE-based Cisco CME 3.1 or later network and a Cisco Unified Communications Manager
network. This topology would work equally well with a Cisco BTS or Cisco PGW in place of the
Cisco Unified Communications Manager.

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In the network topology in Figure 31 on page 511, 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 Unified Communications Manager. 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 Unified 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 Unified CME 1.
Extension 1001 performs a consultation transfer to extension 2001 on phone 5, which is connected
to Cisco Unified 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 Unified CME 2.
5. Extension 4002 is connected with extension 2001.

Figure 31 H.450 Tandem Gateway

IP-to-IP
Gateway

Public VoIP Cisco Unified 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
Gate
3 PSTN
V

H.450.2 Message
Private VoIP Telephone
Cisco Unified CME 2
1 Cisco Unified CME 2

V V
2 5
4
IP IP IP IP
Phone 3 Phone 4 Phone 5 Phone 6
1001 1002 3001 3002
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Tips for Using H.450 Tandem Gateways

Use this procedure 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 Unified Communications Manager, Cisco BTS, and Cisco PGW.
Support for VoIP-to-VoIP connections is disabled by default and can be enabled globally. For more
information, see the “Enabling H.323-to-H.323 Connection Capabilities” section on page 531.
Use dial peers to set up an H.450 tandem gateway. See the “Dial Peers” section on page 512.

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. Dial peer settings override
the global settings for call forward and call transfer. For information about configuring dial peers, see
the Dial Peer Configuration on Voice Gateway Routers guide.

QSIG Supplementary Services

QSIG is an intelligent inter-PBX signaling system widely adopted by PBX vendors. It supports a range
of basic services, generic functional procedures, and supplementary services. Cisco Unified CME 4.0
introduces supplementary services features that allow Cisco Unified CME phones to seamlessly
interwork using QSIG with phones connected to a PBX. One benefit is that IP phones can use a PBX
message center with proper MWI notifications. Figure 32 illustrates a topology for a
Cisco Unified CME system with some phones under the control of a PBX.

Figure 32 Cisco Unified CME System with PBX

IP 1001 IP 2001

IP 1002 IP 2002

IP 1003 IP 2003

Remote Cisco CME IP network Local Cisco CME

QSIG 3001

3002
PBX
3003

Message
135562

center

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The following QSIG supplementary service features are supported in Cisco Unified CME systems. They
follow the standards from the European Computer Manufacturers Association (ECMA) and the
International Organization for Standardization (ISO) on PRI and BRI interfaces.
• Basic calls between IP phones and PBX phones.
• Calling Line/Name Identification (CLIP/CNIP) presented on an IP phone when called by a PBX
phone; in the reverse direction, such information is provided to the called endpoint.
• Connected Line/Name Identification (COLP/CONP) information provided when a PBX phone calls
an IP phone and is connected; in the reverse direction, such information presented on an IP phone.
• Call Forward using QSIG and H.450.3 to support any combination of IP phone and PBX phone,
including an IP phone in the Cisco Unified CME system that is connected to a PBX or an IP phone
in another Cisco Unified CME system across an H.323 network.
• Call forward to the PBX message center according to the configured policy. The other two endpoints
can be a mixture of IP phone and PBX phones.
• Hairpin call transfer, which interworks with a PBX in transfer-by-join mode. Note that
Cisco Unified CME does not support the actual signaling specified for this transfer mode (including
the involved FACILITY message service APDUs) which are intended for an informative purpose
only and not for the transfer functionality itself. As a transferrer (XOR) host, Cisco Unified CME
simply hairpins two call legs to create a connection; as a transferee (XEE) or transfer-to (XTO) host,
it will not be aware of a transfer that is taking place on an existing leg. As a result, the final endpoint
may not be updated with the accurate identity of its peer. Both blind transfer and consult transfer are
supported.
• Message-waiting indicator (MWI) activation or deactivation requests are processed from the PBX
message center.
• The PBX message center can be interrogated for the MWI status of a particular ephone-dn.
• A user can retrieve voice messages from a PBX message center by making a normal call to the
message center access number.
For information about enabling QSIG supplementary services, see the “Enabling H.450.7 and QSIG
Supplementary Services at a System-Level” section on page 535 and “Enabling H.450.7 and QSIG
Supplementary Services on a Dial Peer” section on page 536.
For more information about configuring Cisco Unified CME to integrate with voice-mail systems, see
“Integrating Voice Mail” on page 355.

Disabling SIP Supplementary Services for Call Forward and Call Transfer
If a destination gateway does not support supplementary services, you can disable REFER messages for
call transfers and redirect responses for call forwarding from being sent by Cisco Unified CME or
Cisco Unified SRST.
Disabling supplementary services is supported if all endpoints use SCCP or all endpoints use SIP. It is
not supported for a mix of SCCP and SIP endpoints.

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Typical Network Scenarios for Call Transfer and Call Forwarding

In a mixed network that involves two or more types of call agents or call-control systems, 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. This section provides descriptions of the specific mixed-network scenarios you might
encounter when configuring a router running Cisco CME 3.1 or a later version. Each of the following
sections point to the configuration instructions necessary to ensure call transfer and forwarding
capabilities throughout the network.
• Cisco CME 3.1 or Later and Cisco IOS Gateways, page 514
• Cisco CME 3.0 or an Earlier Version and Cisco IOS Gateways, page 515
• Cisco CME 3.1 or Later, Non-H.450 Gateways, and Cisco IOS Gateways, page 515
• Cisco Unified CME, Non-H.450 Gateways, and Cisco IOS Gateways, page 516
• Cisco CME 3.1 or Later, Cisco Unified Communications Manager, and Cisco IOS Gateways,
page 516
• Cisco CME 3.0 or an Earlier Version, Cisco Unified Communications Manager, and Cisco IOS
Gateways, page 517

Note Cisco Communications Manager Express 3.2 (Cisco CME 3.2) and later versions provide full
call-transfer and call-forwarding with call processing systems on the network that support H.450.2,
H.450.3, and H.450.12 standards. For interoperability with 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 special Tool Command Language (Tcl) script that was needed in earlier versions of
Cisco Unified CME.

Cisco CME 3.1 or Later and Cisco IOS Gateways

In a network with Cisco CME 3.1 or a later version and Cisco IOS gateways, 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 3.1 or
later features.
Configuration for this type of network consists of:
1. Setting 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 Call
Transfer and Forwarding at System-Level” section on page 519.
2. Enabling 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, we recommend it. See the “Enabling H.450.12
Capabilities” section on page 529.
3. Optionally setting 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 531.
4. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.

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Cisco CME 3.0 or an Earlier Version and Cisco IOS Gateways

Before Cisco CME 3.1, H.450.2 and H.450.3 standards are used for all calls by default and routers do
not support the H.450.12 standard.
Configuration for this type of network consists of:
1. Setting 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 Call
Transfer and Forwarding at System-Level” section on page 519.
2. Enabling 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 H.450.12 Capabilities” section on page 529.
3. Optionally setting 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 531.
4. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.

Cisco CME 3.1 or Later, Non-H.450 Gateways, and Cisco IOS Gateways
In a network with Cisco CME 3.1 or later, non-H.450 gateways, and Cisco IOS gateways, 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 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 using H.323-to-H.323 hairpin call routing.
Configuration for this type of network consists of:
1. Setting 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 H.450.3 capabilities on dial peers that point to non-H.450-capable systems such as
Cisco Unified Communications Manager, Cisco BTS, or Cisco PGW. See the “Enabling Call
Transfer and Forwarding at System-Level” section on page 519.
2. Enabling 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 H.450.12 Capabilities” section on
page 529.
3. Setting 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 531.
4. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.

Note If your network contains a Cisco Unified Communications Manager, also see the instructions in the
“Enabling Interworking with Cisco Unified Communications Manager” section on page 540.

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Cisco Unified CME, Non-H.450 Gateways, and Cisco IOS Gateways

Note Cisco CME 3.0 and Cisco ITS V2.1 systems do not have H.450.12 capabilities.

In a network that contains a mix of Cisco Unified CME versions and at least one non-H.450 gateway, the
simplest configuration approach 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:
1. Setting 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 Call
Transfer and Forwarding at System-Level” section on page 519.
2. Enabling 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 H.450.12 Capabilities” section on
page 529.
3. Setting 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 531.
4. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.

Note If your network contains a Cisco Unified Communications Manager, also see the instructions in the
“Enabling Interworking with Cisco Unified Communications Manager” section on page 540.

Cisco CME 3.1 or Later, Cisco Unified Communications Manager, and Cisco IOS Gateways
In a network with Cisco CME 3.1 or later, Cisco Unified Communications Manager, and Cisco IOS
gateways, Cisco CME 3.1 and later versions support automatic detection of calls to and from
Cisco Unified Communications Manager 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 Unified Communications Manager does not support. If a
call is detected to be coming from or going to a Cisco Unified Communications Manager 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 Unified Communications Manager call-processing systems.
Configuration for this type of network consists of:
1. Setting 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 Call
Transfer and Forwarding at System-Level” section on page 519.

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2. Enabling 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 H.450.12 Capabilities” section on
page 529.
3. Setting 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 Unified Communications Manager. See the “Enabling H.323-to-H.323 Connection
Capabilities” section on page 531.
4. Setting up specific parameters for Cisco Unified Communications Manager. See the instructions in
the “Enabling Interworking with Cisco Unified Communications Manager” section on page 540.
5. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.

Cisco CME 3.0 or an Earlier Version, Cisco Unified Communications Manager, and Cisco IOS
Gateways
Calls between the Cisco Unified Communications Manager 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 Unified Communications Manager 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.
See the configuration instructions in the “Configuring Call Transfer” chapter of the
Cisco CallManager Express 3.0 System Administrator Guide.
• Use a loopback-dn mechanism. See “Configuring Loopback Call Routing” on page 763.
• 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 regardless of whether the
call is from a Cisco Unified Communications Manager 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:
1. Setting 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 Call
Transfer and Forwarding at System-Level” section on page 519.
2. Leaving H.450.12 capability in its default disabled state. For more information, see the “Enabling
H.450.12 Capabilities” section on page 529.
3. Setting 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 Unified Communications Manager. See the “Enabling H.323-to-H.323 Connection
Capabilities” section on page 531.

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4. Setting up specific parameters for Cisco Unified Communications Manager. See the instructions in
the “Enabling Interworking with Cisco Unified Communications Manager” section on page 540.
5. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.

How to Configure Call Transfer and Forwarding

This section contains the following procedures:

SCCP
• Enabling Call Transfer and Forwarding at System-Level, page 519 (required)
• SCCP: Enabling Call Forwarding for a Directory Number, page 523 (required)
• SCCP: Enabling Call Transfer for a Directory Number, page 526 (required)
• SCCP: Configuring Call Transfer Options for Phones, page 527 (optional))
• SCCP: Verifying Call Transfer, page 528 (optional)
• Enabling H.450.12 Capabilities, page 529 (optional)
• Enabling H.323-to-H.323 Connection Capabilities, page 531 (optional)
• Forwarding Calls Using Local Hairpin Routing, page 533 (optional)
• Enabling H.450.7 and QSIG Supplementary Services at a System-Level, page 535 (optional)
• Enabling H.450.7 and QSIG Supplementary Services on a Dial Peer, page 536 (optional)
• Disabling SIP Supplementary Services for Call Forward and Call Transfer, page 538 (optional)
• Enabling Interworking with Cisco Unified Communications Manager, page 540 (optional)

SIP B2BUA
• SIP: Configuring SIP-to-SIP Phone Call Forwarding, page 546 (required)
• SIP: Configuring Call-Forwarding-All Soft Key URI, page 548 (optional)
• SIP: Specifying Number of 3XX Responses To be Handled, page 549 (optional)
• SIP: Configuring Call Transfer, page 550 (required)
• Disabling SIP Supplementary Services for Call Forward and Call Transfer, page 538 (optional)

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Enabling Call Transfer and Forwarding at System-Level

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 Unified CME system, perform the following steps.

Note H.450.2 and H.450.3 capabilities are enabled by default for transferred or forwarded parties and
transfer-destination or forward-destination parties. Dial peer settings override the global setting.

Prerequisites
Cisco CME 3.0 or a later version, or Cisco ITS V2.1.

Restrictions
• Call transfers are handled differently depending on the Cisco Unified CME version. See Table 26
on page 506 for recommendations on selecting a transfer method for your Cisco Unified CME
version.
• The transfer-system local-consult command is not supported if the transfer-to destination is on the
Cisco ATA, Cisco VG224, or a SCCP-controlled FXS port.
• The H.450.2 and H.450.3 standards are not supported by Cisco Unified Communications Manager,
Cisco BTS, or Cisco PGW.

SUMMARY STEPS

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

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 transfer-system {blind | full-blind | Specifies the call transfer method.
full-consult [dss] | local-consult}
• Cisco CME 3.0 and later versions—Use only the full-blind
or full-consult keyword.
Example:
Router(config-telephony)# transfer-system
• Before Cisco CME 3.0—Use the local-consult or blind
full-consult keyword. (Cisco ITS 2.1 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.)
• blind—Calls are transferred without consultation with a
single phone line using the Cisco proprietary method. This is
the default in Cisco CME versions earlier than 4.0.
• 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. Calls fall back to full-blind if the second line is
unavailable. This is the default in Cisco Unified CME 4.0 and
later versions.
• dss—(Optional) Calls are transferred with consultation to
idle monitored lines. All other call-transfer behavior is
identical to full-consult.
• 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.
Not supported if transfer-to destination is on the Cisco ATA,
Cisco VG224, or a SCCP-controlled FXS port.

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Command or Action Purpose

Step 5 transfer-pattern transfer-pattern [blind] Allows transfer of telephone calls by Cisco Unified 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
Example:
Router(config-telephony)#
phones.
transfer-pattern .T • transfer-pattern—String of digits for permitted call transfers.
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 For transfers to nonlocal numbers, 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 “Configuring Dialing Plans”
on page 267.
Step 6 call-forward pattern pattern Specifies the H.450.3 standard for call forwarding.
• Calling-party numbers that do not match the patterns defined
Example: with this command are forwarded using Cisco proprietary
Router(config-telephony)# call-forward call forwarding for backward compatibility, as described in
pattern .T the “Configuring Call Forwarding” chapter in the Cisco IOS
Telephony Services Version 2.1 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 For forwards to nonlocal numbers, pattern 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 “Configuring Dialing Plans” on
page 267.
Step 7 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 8 voice service voip (Optional) Enters voice-service configuration mode to establish
global call transfer and forwarding parameters.
Example:
Router(config)# voice service voip

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Command or Action Purpose

Step 9 supplementary-service h450.2 (Optional) Enables H.450.2 supplementary services capabilities
globally.
Example: • Default is enabled. Use the no form of this command to
Router(conf-voi-serv)# disable H.450.2 capabilities globally.
supplementary-service h450.2
• You can also use this command in dial-peer configuration
mode to enable H.450.2 services for a single dial peer.
Step 10 supplementary-service h450.3 (Optional) Enables H.450.3 supplementary services capabilities
globally.
Example: • Default is enabled. Use the no form of this command to
Router(conf-voi-serv)# disable H.450.3 capabilities globally.
supplementary-service h450.3
• You can also use this command in dial-peer configuration
mode to enable H.450.3 services for a single dial peer.
Step 11 exit (Optional) Exits voice-service configuration mode.

Example:
Router(conf-voi-serv)# exit
Step 12 dial-peer voice tag voip (Optional) Enters dial-peer configuration mode.

Example:
Router(config)# dial-peer voice 1 voip
Step 13 supplementary-service h450.2 (Optional) Enables H.450.2 supplementary services capabilities
for an individual dial peer.
Example: • Default is enabled. You can also use this command in
Router(config-dial-peer)# no voice-service configuration mode to enable H.450.2 services
supplementary-service h450.2 globally.
• 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 14 supplementary-service h450.3 (Optional) Enables H.450.3 supplementary services capabilities
exchange for an individual dial peer.
Example: • Default is enabled. You can also use this command in
Router(config-dial-peer)# no voice-service configuration mode to enable H.450.3 services
supplementary-service h450.3 globally.
• 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 15 end Returns to privileged EXEC mode.

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

SCCP: Enabling Call Forwarding for a Directory Number

To define the conditions and target numbers for call forwarding for individual ephone-dns, and set other
restrictions for call forwarding, perform the following steps.

Note When defining call forwarding 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 “Voice
Translation Rules and Profiles” section in “Configuring Dialing Plans” on page 267.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. call-forward pattern pattern
5. exit
6. ephone-dn dn-tag [dual-line]
7. number number [secondary number] [no-reg [both | primary]]
8. call-forward all target-number
9. call-forward busy target-number [primary | secondary] [dialplan-pattern]
10. call-forward noan target-number timeout seconds [primary | secondary] [dialplan-pattern]
11. call-forward night-service target-number
12. call-forward max-length length

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13. no forward local-calls

14. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)#
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.
pattern .T • pattern—Digits to match for call forwarding using the
H.450.3 standard. If an incoming calling-party number
matches the pattern, it is forwarded using the H.450.3
standard. A pattern of .T forwards all calling parties using the
H.450.3 standard.
Step 5 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 6 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an ephone-dn, and
optionally assigns it dual-line status.
Example: • dual-line—(Optional) Enables an ephone-dn with one voice
Router(config)# ephone-dn 20 port and two voice channels, which supports features such as
call waiting, call transfer, and conferencing with a single
ephone-dn.
Step 7 number number [secondary number] [no-reg Configures a valid extension number for this ephone-dn instance.
[both | primary]]

Example:
Router(config-ephone-dn)# number 2777
secondary 2778

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Command or Action Purpose

Step 8 call-forward all target-number Forwards all calls for this extension to the specified number.
• target-number—Phone number to which calls are forwarded.
Example: Note After you use this command to specify a target number,
Router(config-ephone-dn)# call-forward
all 2411
the phone user can activate and cancel the call-forward-all
state from the phone using the CFwdAll soft key or a
feature access code (FAC).
Step 9 call-forward busy target-number [primary Forwards calls for a busy extension to the specified number.
| secondary] [dialplan-pattern]

Example:
Router(config-ephone-dn)# call-forward
busy 2513
Step 10 call-forward noan target-number timeout Forwards calls for an extension that does not answer.
seconds [primary | secondary]
[dialplan-pattern]

Example:
Router(config-ephone-dn)# call-forward
noan 2513 timeout 45
Step 11 call-forward night-service target-number Automatically forwards incoming calls to the specified number
when night service is active.
Example: • target-number—Phone number to which calls are forwarded.
Router(config-ephone-dn)# call-forward
night-service 2879
Note Night service must also be configured. See “Configuring
Call-Coverage Features” on page 563.
Step 12 call-forward max-length length (Optional) Limits the number of digits that can be entered for a
target number when using the CfwdAll soft key on an IP phone.
Example: • length—Number of digits that can be entered using the
Router(config-ephone-dn)# call-forward CfwdAll soft key on an IP phone.
max-length 5
Step 13 no forward local-calls (Optional) Specifies that local calls (calls from ephone-dns on the
same Cisco Unified CME system) will not be forwarded from this
extension.
Example:
Router(config-ephone-dn)# no forward • If this extension is busy, an internal caller hears a busy signal.
local-calls
• If this extension does not answer, the internal caller hears
ringback.
Step 14 end Returns to privileged EXEC mode.

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

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SCCP: Enabling Call Transfer for a Directory Number

To enable call transfer for a specific directory number, perform the following steps. This procedure
overrides the global setting for blind or consultative transfer for individual directory numbers.

Prerequisites
Call transfer must be enabled globally. See the “Enabling Call Transfer and Forwarding at
System-Level” section on page 519.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. transfer-mode {blind | consult}
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an ephone-dn, and
optionally assigns it dual-line status.
Example: • dual-line—(Optional) Enables an ephone-dn with one voice
Router(config)# ephone-dn 20 port and two voice channels, which supports features such as
call waiting, call transfer, and conferencing with a single
ephone-dn.
Step 4 transfer-mode {blind | consult} Specifies the type of call transfer for an individual directory
number using the H.450.2 standard, allowing you to override the
global setting.
Example:
Router(config-ephone-dn)# transfer-mode • Default: system-level value set with the transfer-system
blind command.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

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SCCP: Configuring Call Transfer Options for Phones

To specify a maximum number of digits for transfer destinations or block transfers to external
destinations by individual phones, perform the following steps.

Restrictions
• Transfers made to speed-dial numbers are not blocked when the transfer-pattern blocked
command is used.
• Transfers made using speed-dial are not blocked by the after-hours block pattern command.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-template template-tag
4. transfer-pattern blocked
5. transfer max-length digit-length
6. exit
7. ephone phone-tag
8. ephone-template template-tag
9. restart
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-template template-tag Enters ephone-template configuration mode.
• template-tag—Unique sequence number that identifies this
Example: template during configuration tasks. Range is 1 to 20.
Router(config)# ephone-template 1
Step 4 transfer-pattern blocked (Optional) Prevents directory numbers on the phone to which this
template is applied from transferring calls to patterns specified in
the transfer-pattern (telephony-service) command.
Example:
Router(config-ephone-template)# Note This command is also available in ephone configuration
transfer-pattern blocked mode to block external transfers from individual phones
without using a template.

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Command or Action Purpose

Step 5 transfer max-length digit-length (Optional) Specifies the maximum number of digits the user can
dial when transferring a call.
Example: • digit-length—Number of digits allowed in a number to which
Router(config-ephone-template)# transfer a call is being transferred. Range: 3 to 16. Default: 16.
max-length 8
Step 6 exit Exits ephone-template configuration mode.

Example:
Router(config-ephone-template)# exit
Step 7 ephone phone-tag Enters ephone configuration mode.

Example:
Router(config)# ephone 25
Step 8 ephone-template template-tag Applies a template to a phone.
• template-tag—Template number that you want to apply to
Example: this phone.
Router(config-ephone)# ephone-template 1
Step 9 restart Performs a fast reboot of this phone without contacting the DHCP
server for updated information.
Example: • Repeat Step 6 to Step 9 for each phone on which you want to
Router(config-ephone)# restart limit transfer capabilities.
Step 10 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

SCCP: Verifying Call Transfer

Step 1 Use the show running-config command to verify your configuration. Transfer method and patterns are
listed in the telephony-service portion of the output. You can also use the show telephony-service
command to display this information.
Router# show running-config
!
telephony-service
fxo hook-flash
load 7910 P00403020214
load 7960-7940 P00305000600
load 7914 S00103020002
load 7905 CP7905040000SCCP040701A
max-ephones 100
max-dn 500
ip source-address 10.115.33.177 port 2000
max-redirect 20
no service directed-pickup
timeouts ringing 10
voicemail 7189
max-conferences 8 gain -6
moh music-on-hold.au

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web admin system name cisco password cisco

dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 92......
transfer-pattern 91..........
transfer-pattern 93......
transfer-pattern 94......
transfer-pattern 95......
transfer-pattern 96......
transfer-pattern 97......
transfer-pattern 98......
transfer-pattern 99......
transfer-pattern .T
secondary-dialtone 9
!
create cnf-files version-stamp 7960 Jul 13 2004 03:39:28

Step 2 If you have used the transfer-mode command to override the global transfer mode for an individual
ephone-dn, use the show running-config or show telephony-service ephone-dn command to verify that
setting.
Router# show running-config
!
ephone-dn 40 dual-line
number 451
description Main Number
huntstop channel
no huntstop
transfer-mode blind

Step 3 Use the show telephony-service ephone-template command to view ephone-template configurations.

Enabling H.450.12 Capabilities

To enable H.450.12 capabilities globally or by individual dial peer when not all gateway endpoints in
your network support H.450.2 and H.450.3 standards, perform the following steps. H.450.12 capabilities
are disabled by default to minimize the risk of compatibility issues with other types of H.323 systems.
Settings for individual dial peers override the global setting.

Restrictions
Cisco CME 3.0 and earlier versions do not support H.450.12.

SUMMARY STEPS

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

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7. supplementary-service h450.12
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip (Optional) Enters voice service configuration mode to establish
global call transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Step 4 supplementary-service h450.12 (Optional) Enables H.450.12 supplementary services capabilities
[advertise-only] globally for VoIP endpoints.
• This command enables call-by-call detection of H.450
Example: capabilities when some endpoints in your mixed network are
Router(conf-voi-serv)# H.450-capable and other endpoints are not. This command is
supplementary-service h450.12
disabled by default.
• advertise-only—(Optional) Advertises H.450 capabilities to
the remote end but does not require H.450.12 responses. Use
this keyword on Cisco CME 3.1 or later systems if you have
a mixed network containing Cisco CME 3.0 systems.
This command is also used in dial-peer configuration mode to
affect an individual dial peer.
Step 5 exit (Optional) Exits voice-service configuration mode.

Example:
Router(conf-voi-serv)# exit
Step 6 dial-peer voice tag voip (Optional) Enters dial-peer configuration mode.

Example:
Router(config)# dial-peer voice 1 voip

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Command or Action Purpose

Step 7 supplementary-service h450.12 (Optional) Enables H.450.12 supplementary services capabilities
for an individual dial peer. This command is disabled by default.
Example: This command is also used in voice-service configuration mode
Router(config-dial-peer)# to enable H.450.12 services globally.
supplementary-service h450.12
• 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 8 end Returns to privileged EXEC mode.

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

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 a later
version 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 H.450.12 Capabilities” section on page 529.
• Explicitly disable H.450.2 and H.450.3 capabilities globally or on the routes that your transfers and
forwards take. See the “Enabling Call Transfer and Forwarding at System-Level” section on
page 519.

Restrictions
• 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.
• 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 Unified 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.

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• H.323-to-SIP hairpin call routing is supported only with Cisco Unity Express. For more
information, see Integrating Cisco CallManager Express and Cisco Unity Express.
• Cisco Unified Communications Manager must use a media termination point (MTP), intercluster
trunk (ICT) mode, and slow start.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice service voip
4. allow-connections h323 to h323
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip Enters voice service configuration mode to establish global call
transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Step 4 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 5 end Returns to privileged EXEC mode.

Example:
Router(config-voi-serv)# end

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Forwarding Calls Using Local Hairpin Routing

When Cisco Unified CME is used to forward calls that originate on phones that do not support the
H.450.3 standard such as Cisco Unified Communications Manager phones, local hairpin routing must be
used to forward the calls. For calling parties whose numbers match the pattern specified, the system
automatically detects whether H.450.3 is supported and uses the appropriate method to forward calls.
To enable hairpin routing, you must denote the originating and terminating legs of the hairpin. To
forward calls to Cisco Unity Express, connections must be allowed to a SIP trunk.
Optionally, you can disable the use of H.450.3 but this is not required because the system automatically
detects calls on which H.450.3 is not supported and local hairpin routing is required when the
calling-party numbers match the pattern specified.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. call-forward pattern pattern
5. calling-number local
6. exit
7. voice service voip
8. allow connections from-type to to-type
9. supplementary-service h450.3
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 4 call-forward pattern pattern Specifies the calling-party numbers for which to allow call
forwarding with automatic detection of whether H.450.3 is
supported. If H.450.3 is supported, H.450.3 is used for the
Example:
Router(config-telephony)# call-forward
forward and, if not, local hairpin is used.
pattern 6000 • pattern—Digits to match for call forwarding. A pattern of .T
forwards all calling parties.
Step 5 calling-number local (Optional) Replaces a calling-party number and name with the
forwarding-party (local) number and name for hairpin-forwarded
calls only.
Example: • Before Cisco CME 3.3, this command must be used with Tool
Router(config-telephony)# calling-number Command Language (Tcl) script app-h450-transfer.2.0.0.7 or
local
a later version. The local-hairpin attribute-value (AV) pair
must be set to 1.
Step 6 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 7 voice service voip Enters voice-service configuration mode.

Example:
Router(config)# voice service voip
Step 8 allow connections from-type to to-type Allows connections between specific types of endpoints in a
network.
Example: • from-type—Originating endpoint type. Valid choices are
Router(conf-voi-serv)# allow connections h323 and sip.
h323 to sip
• to-type—Terminating endpoint type. Valid choices are h323
and sip.
Step 9 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)# no
can also be used in dial-peer configuration mode to disable
supplementary-service h450.3 H.450.3 functionality for a single dial peer.
Note If this command is disabled globally and either enabled or
disabled on a dial peer, the functionality is disabled for
the dial peer.
Step 10 end Returns to privileged EXEC mode.

Example:
Router(config-voi-serv)# end

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Enabling H.450.7 and QSIG Supplementary Services at a System-Level

To enable H.4350.7 capabilities and QSIG supplementary services on all dial peers, perform the
following steps.

Prerequisites
Cisco Unified CME 4.0 or a later version.

Restrictions
• QSIG integration supports SCCP phones only.
• QSIG integration is exclusive; once QSIG integration is configured, QSIG transit node capability is
disabled. There is no dial-peer control to enable either transit or originate/terminate capability on a
call by call basis.
• If you enable QSIG supplementary services at a system-level, you cannot disable the capability on
individual dial peers.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice service voip
4. supplementary-service h450.7
5. qsig decode
6. exit
7. voice service pots
8. supplementary-service qsig call-forward
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice service voip Enters VoIP voice-service configuration mode to define
global call transfer and forwarding parameters.
Example:
Router(config)# voice service voip

Step 4 supplementary-service h450.7 Enables H.450.7 supplementary services capabilities

exchange at a system-level.
Example:
Router(config-voi-serv)# supplementary-service
h450.7
Step 5 qsig decode Enables decoding for QSIG supplementary services.

Example:
Router(config-voi-serv)# qsig decode
Step 6 exit Exits VoIP voice-service configuration mode.

Example:
Router(config-voi-serv)# exit
Step 7 voice service pots Enters POTS voice-service configuration mode to define
global call transfer and forwarding parameters.
Example:
Router(config)# voice service pots
Step 8 supplementary-service qsig call-forward Enables QSIG call-forwarding supplementary services
(ISO 13873) to forward calls to another number.
Example:
Router(config-voi-serv)# supplementary-service
qsig call-forward
Step 9 end Returns to privileged EXEC mode.

Example:
Router(config-voi-serv)# end

Enabling H.450.7 and QSIG Supplementary Services on a Dial Peer

To enable H.4350.7 capabilities and QSIG supplementary services on an individual dial peer, perform
the following steps.

Prerequisites
Cisco Unified CME 4.0 or a later version.

Restrictions
• QSIG integration supports SCCP phones only.

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• QSIG integration is exclusive; once QSIG integration is configured, QSIG transit node capability is
disabled. There is no dial-peer control to enable either transit or originate/terminate capability on a
call by call basis.
• If you enable QSIG supplementary services at a system-level, you cannot enable or disable the
capability on individual dial peers.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice service voip
4. qsig decode
5. exit
6. dial-peer voice tag voip
7. supplementary-service h450.7
8. exit
9. dial-peer voice tag pots
10. supplementary-service qsig call-forward
11. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip Enters VoIP voice-service configuration mode to define
global call transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Step 4 qsig decode Enables decoding for QSIG supplementary services.

Example:
Router(config-voi-serv)# qsig decode
Step 5 exit Exits VoIP voice-service configuration mode.

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

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Command or Action Purpose

Step 6 dial-peer voice tag voip Enters dial-peer configuration mode to define parameters
for an individual dial peer.
Example:
Router(config)# dial-peer voice 1 voip
Step 7 supplementary-service h450.7 Enables H.450.7 supplementary services capabilities
exchange on a single dial peer.
Example:
Router(config-dial-peer)# supplementary-service
h450.7
Step 8 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit
Step 9 dial-peer voice tag pots Enters dial-peer configuration mode to define parameters
for an individual dial peer.
Example:
Router(config)# dial-peer voice 2 pots
Step 10 supplementary-service qsig call-forward Enables QSIG call-forwarding supplementary services
Router(config-dial-peer)# supplementary-service (ISO 13873) to forward calls to another number.
qsig call-forward
Step 11 end Returns to privileged EXEC mode.

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

Disabling SIP Supplementary Services for Call Forward and Call Transfer
To disable REFER messages for call transfers or redirect responses for call forwarding from being sent
to the destination by Cisco Unified CME, perform the following steps. You can disable these
supplementary features if the destination gateway does not support them.

Prerequisites
Cisco Unified CME 4.1 or a later version.

Restrictions
Disabling supplementary services is supported only when all endpoints are SCCP or all endpoints are
SIP. It does not support a mix of SCCP and SIP endpoints.

SUMMARY STEPS

1. enable
2. configure terminal

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3. voice service voip

or
dial-peer voice tag voip
4. no supplementary-service sip {moved-temporarily | refer}
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip Enters voice-service configuration mode to set global
or parameters for VoIP features.
dial-peer voice tag voip or
Enters dial peer configuration mode to set parameters for a
Example: specific dial peer.
Router(config)# voice service voip
or
Router(config)# dial-peer voice 99 voip
Step 4 no supplementary-service sip {moved-temporarily Disables SIP call forwarding or call transfer supplementary
| refer} services globally or for a dial peer.
• moved-temporarily—SIP redirect response for call
Example: forwarding.
Router(conf-voi-serv)# no supplementary-service
sip refer • refer—SIP REFER message for call transfers.
or • Sending REFER and redirect messages to the
Router(config-dial-peer)# no destination is the default behavior.
supplementary-service sip refer
Note This command is supported for calls between SIP
phones and calls between SCCP phones. It is not
supported for a mixture of SCCP and SIP endpoints.
Step 5 end Exits to privileged EXEC mode.

Example:
Router(config-voi-serv)# end
or
Router(config-dial-peer)# end

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Enabling Interworking with Cisco Unified Communications Manager

If Cisco CME 3.1 or later and Cisco Unified Communications Manager 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 Unified Communications Manager,
page 541
• Enabling Cisco Unified Communications Manager to Interwork with Cisco Unified CME, page 544
• Troubleshooting Transfer and Forwarding Configuration, page 544
Figure 33 shows a network containing Cisco Unified CME and Cisco Unified Communications
Manager systems.

Figure 33 Network with Cisco Unified CME and Cisco Unified Communications Manager

Cisco Unified CallManager 1 Cisco Unified CallManager 2

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

Media Termination Point (MTP)

V

Cisco Unified CallManager Network VoIP

Cisco Unified CME Network

PSTN

Cisco Unified CME 1 Cisco Unified CME 2 Cisco Unified CME 3

V V V
Telephone

IP IP IP IP IP IP
146621

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

1001 1002 2001 2002 3001 3002

Prerequisites
• Cisco Unified CME must be configured to forward calls using local hairpin routing. For
configuration information, see the “Forwarding Calls Using Local Hairpin Routing” section on
page 533.

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Configuring Cisco CME 3.1 or Later to Interwork with Cisco Unified Communications Manager
All of the Cisco IOS commands in this section are optional because they are set by default to work with
Cisco Unified Communications Manager. They are included here only to explain how to implement
optional capabilities or return nondefault settings to their defaults.

SUMMARY STEPS

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

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip Enters voice-service configuration mode to establish global
parameters.
Example:
Router(config)# voice service voip

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Command or Action Purpose

Step 4 h323 Enters H.323 voice-service configuration mode.

Example:
Router(conf-voi-serv)# h323
Step 5 telephony-service ccm-compatible (Optional) Globally enables a Cisco CME 3.1 or later system to
detect a Cisco Unified Communications Manager and exchange
calls with it. This is the default.
Example:
Router(conf-serv-h323)# telephony-service • Use the no form of this command to disable
ccm-compatible Cisco Unified Communications Manager detection and
exchange. We do not recommend using the no form of the
command.
• Using this command in an H.323 voice class definition allows
you to specify this behavior for an individual dial peer.
Step 6 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 Unified Communications Manager to
Example:
Router(conf-serv-h323)# h225 h245-address
generate local ringback for calls to Cisco Unified CME phones.
on-connect This is the default.
• The no form of this command disables the delay. We do not
recommend using the no form of the command.
• Using this command in an H.323 voice class definition allows
you to specify this behavior for an individual dial peer.
Step 7 exit Exits H.323 voice-service configuration mode.

Example:
Router(conf-serv-h323)# exit
Step 8 supplementary-service h225-notify (Optional) Globally enables H.225 messages with caller-ID
cid-update updates to be sent to Cisco Unified Communications Manager.
This is the default.
Example: • The no form of the command disables caller-ID update. We
Router(conf-voi-serv)# do not recommend using the no form of the command.
supplementary-service h225-notify
cid-update This command is also used in dial-peer configuration mode to
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 9 exit Exits voice-service configuration mode.

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

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Command or Action Purpose

Step 10 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 11 telephony-service ccm-compatible (Optional) Enables the dial peer to exchange calls with a
Cisco Unified Communications Manager system when this voice
class is applied to a dial peer. This is the default.
Example:
Router(config-voice-class)# • The no form of the command disables call exchange with
telephony-service ccm-compatible Cisco Unified Communications Manager. We do not
recommend using the no form of the command.
Step 12 h225 h245-address on-connect (Optional) 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 connected, when this voice class is applied
Example:
Router(config-voice-class)# h225
to a dial peer. The delay allows the playing of local ringback for
h245-address on-connect calls from Cisco Unified Communications Manager. This is the
default.
• The no form of this command disables the delay. We do not
recommend using the no form of the command.
Step 13 exit Exits voice-class configuration mode.

Example:
Router(config-voice-class)# exit
Step 14 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 15 supplementary-service h225-notify (Optional) Enables H.225 messages with caller-ID updates to
cid-update Cisco Unified Communications Manager for a specific dial peer.
This is the default.
Example: • The no form of the command disables caller-ID updates. We
Router(config-dial-peer)# no do not recommend using the no form of the command.
supplementary-service h225-notify
cid-update
Step 16 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 17 end Returns to privileged EXEC mode.

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

What to Do Next

Set up Cisco Unified Communications Manager using the configuration procedure in the “Enabling
Cisco Unified Communications Manager to Interwork with Cisco Unified CME” section on page 544.

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Enabling Cisco Unified Communications Manager to Interwork with Cisco Unified CME
To enable Cisco Unified Communications Manager to interwork with a Cisco CME 3.1 or later system,
perform the following steps in addition to the normal Cisco Unified Communications Manager
configuration.

SUMMARY STEPS

1. Set Cisco Unified Communications Manager service parameters.

2. Configure the Cisco CME 3.1 or later system as an ICT in the Cisco Unified Communications
Manager network.
3. Ensure that the Cisco Unified Communications Manager network uses an MTP.
4. Set up dial peers to establish routing.

DETAILED STEPS

Step 1 Set Cisco Unified Communications Manager service parameters. From Cisco Unified Communications
Manager Administration, choose Service Parameters. Choose the Cisco Unified Communications
Manager service, and make the following settings:
• Set the H323 FastStart Inbound service parameter to False.
• Set the Send H225 User Info Message service parameter to H225 Info for Ring Back.
Step 2 Configure the Cisco CME 3.1 or later system as an ICT in the Cisco Unified Communications Manager
network. For information about different intercluster trunk types and configuration instructions, see the
Cisco Unified Communications Manager documentation.
Step 3 Ensure that the Cisco Unified Communications Manager 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 Unified Communications Manager and
Cisco CME 3.1 or later must pass through the MTP because Cisco CM 3.1 does not support transcoding.
For more information, see the Cisco Unified Communications Manager documentation.
Step 4 Set up dial peers to establish routing using the instructions in the Dial Peer Configuration on Voice
Gateway Routers guide.

Troubleshooting Transfer and Forwarding Configuration

Step 1 If you encounter lack of ringback on direct calls from a Cisco Unified Communications Manager phone
to an IP phone on a Cisco Unified CME system, check the show running-config command output to
ensure that the following two commands do not appear: no h225 h245-address on-connect and no
telephony-service ccm-compatible. These commands should be enabled, which is their default state.
Step 2 Use the debug h225 asn1 command to display the H.323 messages that are sent from the
Cisco Unified CME system to the Cisco Unified Communications Manager system to see if the H.245
address is being sent too early.

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Step 3 For calls that are routed using VoIP-to-VoIP connections, use the show voip rtp connections detail
command to display 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:
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 172.27.82.2 209.165.202.129
Found 2 active RTP connections

Step 4 Use the show call prompt-mem-usage detail command to see 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 Unified 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 Unified CME system.
If you are not getting ringback tone when you should, use the show call prompt-mem-usage command
to ensure 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

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SIP: Configuring SIP-to-SIP Phone Call Forwarding

To configure SIP-to-SIP call forwarding using a back-to-back user agent (B2BUA) which allows call
forwarding on any dial peer, perform the following steps.

Prerequisites
• Connections between specific types of endpoints in a Cisco IP-to-IP gateway must be configured by
using the allow-connections command. For configuration information, see the “Enabling Calls in
Your VoIP Network” on page 96.
• Cisco CME 3.4 or a later version.

Restrictions
• SIP-to-SIP call forwarding is invoked only if that phone is dialed directly. Call forwarding is not
invoked when the phone number is called through a sequential, longest-idle, or peer hunt group.
• If call forwarding is configured for a hunt group member, call forward is ignored by the hunt group.
• In Cisco Unified CME 4.1 and later versions, Call Forward All requires SIP phones to be configured
with a directory number (using dn keyword in number command); direct line numbers are not
supported.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register dn dn-tag
4. call-forward b2bua all directory-number
5. call-forward b2bua busy directory-number
6. call-forward b2bua mailbox directory-number
7. call-forward b2bua noan directory-number timeout seconds
8. call-forward b2bua unreachable directory-number
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice register dn dn-tag Enters voice register dn mode to define a directory number
for a SIP phone, intercom line, voice port, or an MWI.
Example:
Router(config)# voice register dn 1
Step 4 call-forward b2bua all directory- number Enables call forwarding for a SIP back-to-back user agent so
that all incoming calls will be forwarded to the designated
directory-number.
Example:
Router(config-register-dn)# call-forward b2bua • If the call-forward b2bua all command is configured in
all 5005 voice register pool configuration mode, it applies to all
directory numbers on the phone.
• In Cisco CME 3.4 and later versions, this command is
also available in voice register pool configuration
mode. The configuration under voice register dn takes
precedence over the configuration under voice register
pool.
Step 5 call-forward b2bua busy directory- number Enables call forwarding for a SIP back-to-back user agent so
that incoming calls to an extension that is busy will be
forwarded to the designated directory number.
Example:
Router(config-register-dn)# call-forward b2bua • In Cisco CME 3.4 and later versions, this command is
busy 5006 also available in voice register pool configuration
mode. The configuration under voice register dn takes
precedence over the configuration under voice register
pool.
Step 6 call-forward b2bua mailbox directory- number Enables call forwarding for a SIP back-to-back user agent so
that incoming calls that have been forwarded to a busy or
no-answer extension will be forwarded to the recipient’s
Example:
Router(config-register-dn)# call-forward b2bua
voice mail.
mailbox 5007 • In Cisco CME 3.4 and later versions, this command is
also available in voice register pool configuration
mode. The configuration under voice register dn takes
precedence over the configuration under voice register
pool.
Step 7 call-forward b2bua noan directory- number Enables call forwarding for a SIP back-to-back user agent so
timeout seconds that incoming calls to an extension that does not answer will
be forwarded to the designated directory number.
Example: • In Cisco CME 3.4 and later versions, this command is
Router(config-register-dn)# call-forward b2bua also available in voice register pool configuration
noan 5010 timeout 10
mode. The configuration under voice register dn takes
or
precedence over the configuration under voice register
Router(config-register-pool)# call-forward pool.
b2bua noan 5010 timeout 10
• timeout seconds—Duration that a call can ring before it
is forwarded to the destination directory number.
Range: 3 to 60000. The default value is 20

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Command or Action Purpose

Step 8 call-forward b2bua unreachable directory- (Optional) Enables call forwarding for a SIP back-to-back
number user agent so that calls can be forwarded to a phone that has
not registered in Cisco Unified CME.
Example: • Target directory-number must be configured in
Router(config-register-dn)# call-forward b2bua Cisco Unified CME.
unreachable 5009
or • In Cisco CME 3.4 and Cisco Unified CME 4.0, this
command is also available in voice register pool
Router(config-register-pool)# call-forward
b2bua unreachable 5009 configuration mode. The configuration under voice
register dn takes precedence over the configuration
under voice register pool.
• This command was removed in Cisco Unified CME 4.1.
Step 9 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-dn)# end

SIP: Configuring Call-Forwarding-All Soft Key URI

To specify the uniform resource identifier (URI) for the call forward all (CfwdAll) soft key on supported
SIP phones, perform the following steps. This URI and the call forward number is sent to
Cisco Unified CME when a user enables Call Forward All on a SIP phone.

Prerequisites
• Cisco Unified CME 4.1 or a later version.
• The mode cme command must be enabled in Cisco Unified CME.
• Call Forward All must be enabled on the directory number. For information, see “SIP: Configuring
SIP-to-SIP Phone Call Forwarding” on page 546.

Restrictions
• This feature is supported only on Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE,
7970G, and 7971GE.
• If a user enables Call Forward All using the CfwdAll soft key, it is enabled on the primary line.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. call-feature-uri cfwdall service-uri
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
global parameters for all supported SIP phones in a
Cisco Unified CME environment.
Example:
Router(config)# voice register global
Step 4 call-feature-uri cfwdall service-uri Specifies the URI for soft keys on SIP phones connected to
a Cisco Unified CME router.
Example:
Router(config-register-global)#
call-feature-uri cfwdall
http://1.4.212.11/cfwdall
Step 5 end Exits to privileged EXEC mode.

Example:
Router(config-register-global)# end

SIP: Specifying Number of 3XX Responses To be Handled

To specify how many subsequent 3XX responses an originating SIP phone can handle for a single call
when the terminating side is a forwarding party which does not use B2BUA, perform the following steps.

Prerequisites
• Cisco CME 3.4 or a later version.
• The mode cme command must be enabled

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. phone-redirect-limit number
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 phone-redirect-limit number Changes the default number of 3XX responses a SIP phone
that originates a call can handle for a single call.
Example: • Default: 5.
Router(config-register-global)#
phone-redirect-limit 8
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end

SIP: Configuring Call Transfer

To create and apply a template to enable call transfer softkeys on an individual SIP phone in
Cisco Unified CME, perform the following steps.

Prerequisites
Cisco CME 3.4 or a later version.

Restrictions
• Blind transfer is not supported on Java-based phones, such as Cisco Unified IP Phone 7911G,
7941G, 7941GE, 7961G, 7961GE, 7970G, or 7971GE.
• In Cisco Unified CME 4.1, the soft key display can be customized for Java-based IP phones, such
as Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE. For
configuration information, see “SCCP: Modifying Soft-Key Display” on page 832.

SUMMARY STEPS

1. enable
2. configure terminal

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3. voice register template template-tag

4. transfer-attended
5. transfer-blind
6. voice register template template-tag
7. exit
8. voice register pool pool-tag
9. template template-tag
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register template template-tag Enters voice register template configuration mode to define
a template of common parameters for SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register template 1 • Range is 1 to 5.
Step 4 transfer-attended Enable a soft key for attended transfer on any supported SIP
phone that uses a template in which this command is
configure.
Example:
Router(config-register-template)#
transfer-attended

Step 5 transfer-blind Enable a soft key for blind transfer on any supported SIP
phone that uses a template in which this command is
configure.
Example:
Router(config-register-template)#
transfer-blind
Step 6 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-register-template)# exit
Step 7 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for SIP phones.
Example:
Router(config)# voice register pool 3

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Command or Action Purpose

Step 8 template template-tag Applies a template created with the voice register template
command.
Example: • template-tag—Range: 1 to 5.
Router(config-register-pool)# voice register
pool 1
Step 9 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-pool)# end

Configuration Examples for Call Transfer and Forwarding

The following configuration examples are included in this section:
• H.450.2 and H.450.3: Example, page 553
• Basic Call Forwarding: Example, page 553
• Call Forwarding Blocked for Local Calls: Example, page 553
• Selective Call Forwarding: Example, page 553
• Call Transfer: Example, page 554
• H.450.12: Example, page 554
• H.450.7 and QSIG Supplementary Services: Example, page 555
• Cisco Unified CME and Cisco Unified Communications Manager in Same Network: Example,
page 555
• H.450 Tandem Gateway Working with Cisco Unified CME and
Cisco Unified Communications Manager: Example, page 558
• Forwarding Calls to Cisco Unity Express: Example, page 559

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H.450.2 and H.450.3: Example

The following example sets all transfers and forwards that are initiated by a Cisco CME 3.0 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

Basic Call Forwarding: Example

The following example sets up forwarding for extension 2777 to extension 2513 on all calls, busy, and
no answer. During night service hours, calls are forwarded to a different number, extension 2879.
ephone-dn 20
number 2777
call-forward all 2513
call-forward busy 2513
call-forward noan 2513 timeout 45
call-forward night-service 2879

Call Forwarding Blocked for Local Calls: Example

In the following example, extension 2555 is configured to not forward local calls that are internal to the
Cisco Unified CME system. Extension 2222 dials extension 2555. If 2555 is busy, the caller hears a busy
tone. If 2555 does not answer, the caller hears ringback. The internal call is not forwarded.
ephone-dn 25
number 2555
no forward local-calls
call-forward busy 2244
call-forward noan 2244 timeout 45

Selective Call Forwarding: Example

The following example sets call forwarding on busy and no answer for ephone-dn 38 only for its primary
number, 2777. Callers who dial 2778 will hear a busy signal if the ephone-dn is busy or ringback if there
is no answer.
ephone-dn 38
number 2777 secondary 2778
call-forward busy 3000 primary
call-forward noan 3000 primary timeout 45

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Call Transfer: Example

The following example limits transfers from ephone 6, extension 2977, to numbers containing a
maximum of 8 digits.
telephony-service
load 7910 P00403020214
load 7960-7940 P00305000600
load 7914 S00103020002
load 7905 CP7905040000SCCP040701A
load 7912 CP7912040000SCCP040701A
max-ephones 100
max-dn 500
ip source-address 10.104.8.205 port 2000
max-redirect 20
system message XYZ Inc.
create cnf-files version-stamp 7960 Jul 13 2004 03:39:28
voicemail 7189
max-conferences 8 gain -6
moh music-on-hold.au
web admin system name admin1 password admin1
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 91..........
transfer-pattern 92......
transfer-pattern 93......
transfer-pattern 94......
transfer-pattern 95......
transfer-pattern 96......
transfer-pattern 97......
transfer-pattern 98......
transfer-pattern 99......
secondary-dialtone 9
fac standard

ephone-template 2
transfer max-length 8

ephone-dn 4
number 2977

ephone 6
button 1:4
ephone-template 2

H.450.12: 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

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H.450.7 and QSIG Supplementary Services: Example

The following example implements QSIG supplementary services on extension 74367 and globally
enables H.450.7 supplementary services and QSIG call-forwarding supplementary services.
telephony-service
voicemail 74398
transfer-system full-consult

ephone-dn 25
number 74367
mwi qsig
call-forward all 74000

voice service voip

supplementary-service h450.7

voice service pots

supplementary-service qsig call-forward

Cisco Unified CME and Cisco Unified Communications Manager in Same

Network: Example
The following example shows a running configuration for a Cisco CME 3.1 or later router that has a
Cisco Unified Communications Manager 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 pswd
!
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
!

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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
!
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 ipv4:10.22.22.38
!
!
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 ....
!

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

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H.450 Tandem Gateway Working with Cisco Unified CME and

Cisco Unified Communications Manager: 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 Cisco Unified Communications Manager.
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 pswd
!
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
!
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
!

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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 pswd
!
end

Forwarding Calls to Cisco Unity Express: Example

The following example enables the ability to forward calls that originate from
Cisco Unified Communications Manager phones and are routed through a Cisco Unified CME system to
a Cisco Unity Express extension. Call forwarding is enabled for all calling parties, H.450.3 is disabled,
and connections are allowed to SIP endpoints.
telephony-service
call-forward pattern .T

voice service voip

no supplementary-service h450.3
allow connections from h323 to sip

Where to Go Next
If you are finished modifying the configuration, generate a new configuration file and restart the phones.
See “Generating Configuration Files for Phones” on page 245.

Soft Keys
To block the function of the call-forward-all or transfer soft key without removing the key display or to
remove the soft key from one or more phones, see the “How to Customize Soft Keys” section on
page 832.

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Additional References

Feature Access Codes (FACs)

Phone users can activate and deactivate a phone’s call-forward-all setting by using a feature access code
(FAC) instead of a soft key on the phone if standard or custom FACs have been enabled for your system.
The following are the standard FACs for call forward all:
• callfwd all—Call forward all calls. Standard FAC is **1 plus an optional target extension.
• callfwd cancel—Cancel call forward all calls. Standard FAC is **2.
For more information about FACs, see “Configuring Feature Access Codes” on page 729.

Night Service
Calls can be automatically forwarded during night service hours, but you must define the night-service
periods, which are the dates or days and hours during which night service will be active. For instance,
you may want to designate night service periods that include every weeknight between 5 p.m. and 8 a.m.
and all day every Saturday and Sunday. For more information, see “Configuring Call-Coverage
Features” on page 563.

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Call Transfer and Forwarding

Feature Information for Call Transfer and Forwarding

Table 27 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 27 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 27 Feature Information for Call Transfer and Forwarding

Cisco Unified CME

Feature Name Version Feature Information
Call Forwarding 4.1 • Call Forward All synchronization between
Cisco Unified CME and SIP phones was added.
• Disabling SIP supplementary services for call forward
and call transfer was added.
4.0 • Automatic call forwarding during night service was
introduced.
• Selective call forwarding was introduced.
• Forwarding of local (internal) calls can be blocked.
• H.450.7 standards support and QSIG supplementary
services capability was introduced.
3.4 Calls into a SIP device can be forwarded to other SIP or
SCCP devices including Cisco Unity, third- party voice
mail systems, or an auto-attendant (AA) or other interactive
voice response (IVR) devices. SCCP devices may also be
forwarded to SIP devices.
3.1 • Number of digits that can be entered using the
CfwdALL (call-forward all) soft key can be limited.
• H.450.12 standards support, which provide dynamic
detection of H.450.2 and H.450.3 capabilities on a
call-by-call basis, was introduced.
3.0 • CFwdALL soft key was introduced.
• 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.
2.1 Call forwarding using the H.450.3 standard was introduced.

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Table 27 Feature Information for Call Transfer and Forwarding

Cisco Unified CME

Feature Name Version Feature Information
Call Forwarding 1.0 Call forwarding for all calls, busy conditions, and
no-answer conditions was introduced, using a
Cisco-proprietary method.
Call Transfer 4.1 • Disabling SIP supplementary services for call transfer
and call forward was added.
4.0 • Default for the transfer-system command was changed
from the blind keyword to the full-consult keyword.
• Transfers to phones outside the Cisco Unified CME
system can be blocked for individual ephones.
• Number of digits in transfer destination numbers can be
limited.
3.4 Support for attended and blind transfer s using SIP IP phone
directly connected to Cisco CME.
3.2 • Consultative transfer to monitored lines using direct
station select was introduced.
• 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.
3.1 Support was introduced for the following:
• Enhancements for VoIP networks which contain a mix
of platforms that support H.450.2 and H.450.3
standards, such as Cisco CME 3.1, Cisco CME 3.0,
Cisco ITS V2.1, and platforms that do not support
H.450.2 and H.450.3 standards, such as
Cisco Unified Communications Manager, Cisco BTS
Softswitch (BTS), and Cisco PSTN Gateway (PGW).
• H.450.12 standards, which provide dynamic detection
of H.450.2 and H.450.3 capabilities on a call-by-call
basis.
• Automatic detection of Cisco Unified Communications
Manager endpoints.
• Hairpin VoIP-to-VoIP call routing and routing to an
H.450 tandem gateway.
• Hairpin call routing does not require a Tcl script.
3.0 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.
2.1 Consultative transfer using the ITU-T H.450.2 standard was
introduced.
1.0 Call transfer was introduced, using a Cisco proprietary
method.

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Last Updated: June 18, 2007

This chapter describes features that can be used to provide appropriate, flexible coverage for incoming
calls in Cisco Unified Communications Manager Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Call Coverage”
section on page 636.

Contents
• Information About Call Coverage Features, page 563
• How to Configure Call Coverage Features, page 585
• Configuration Examples for Call Coverage Features, page 619
• Where to Go Next, page 633
• Additional References, page 634
• Feature Information for Call Coverage Features, page 636

Information About Call Coverage Features

To configure call coverage features, you should understand the following concepts:
• Call-Coverage Summary, page 564
• Call Hunt, page 565
• Call Pickup, page 566
• Call Waiting, page 568
• Callback Busy Subscriber, page 569
• Hunt Groups, page 569
• Night Service, page 579
• Overlaid Ephone-dns, page 581

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Call-Coverage Summary
Call coverage features are used to ensure that all incoming calls to Cisco Unified CME are answered by
someone, regardless of whether the called number is busy or does not answer.
Some single-dialed-number call-coverage features, such as hunt groups, can send incoming calls to a
single directory number to a pool of phone agents, while other features, such as call hunt, call waiting,
and call forwarding increase the chance of a call being answered by giving it another chance for a
connection if the dialed number is not available.
Multiple-dialed-number call-coverage features, such as call pickup, night service, and overlaid directory
numbers, provide different ways for one person to answer incoming calls to multiple numbers.
Any of the call-coverage features can be combined with other call-coverage features and with shared
lines and secondary numbers to design the call coverage plan that is best suited to your needs.
Table 28 summarizes call-coverage features.

Table 28 Call-Coverage Feature Summary

Feature Description Example How Configured

Call Forwarding Calls are automatically diverted Extension 3444 is configured to SCCP: Enabling Call
to a designated number on busy, send calls to extension 3555 Forwarding for a Directory
no answer, all calls, or only when it is busy or does not Number, page 523
during night-service hours. answer.
or
SIP: Configuring SIP-to-SIP
Phone Call Forwarding,
page 546
Call Hunt System automatically searches Three ephone-dns have the same SCCP: Configuring Call
for an available directory extension number, 755. One is Hunt, page 586
number from a matching group on the manager’s phone and the
or
of directory numbers until the others are on the assistants’
call is answered or the hunt is phones. Preference and huntstop SIP: Configuring Call Hunt,
stopped. are used to make sure that calls page 589
always come to the manager’s
phone first but if they can’t be
answered, they will ring on the
first assistant’s phone and if not
answered, on the second
assistant’s phone.
Call Pickup Calls to unstaffed phones can be Extension 201 and 202 are both SCCP: Creating Pickup
answered by other phone users in pickup group 22. A call is Groups, page 591
using a soft key or by dialing a received by 201, but no one is
short code. there to answer. The agent at 202
presses the GPickUp soft key to
answer the call.
Call Waiting Calls to busy numbers are Extension 564 is in conversation SCCP: Configuring
presented to phone users, giving when a call-waiting beep is Call-Waiting Indicator Tone,
them the option to answer them heard. The phone display shows page 593
or let them be forwarded. the call is from extension 568
or
and the phone user decides to let
the call go to voice mail. SIP: Enabling Call Waiting,
page 595

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Table 28 Call-Coverage Feature Summary (continued)

Feature Description Example How Configured

Cisco CME B-ACD Calls to a pilot number are The DID number 555-0125 is the See Cisco Unified CME
automatically answered by an pilot number for the XYZ B-ACD and Tcl
interactive application that Company. Incoming calls to this Call-Handling Applications.
presents callers with a menu of pilot number hear a menu of
choices before sending them to a choices; they can press 1 for
queue for a hunt group. sales, 2 for service, or 3 to leave
a message. The call is forwarded
appropriately when callers make
a choice.
Hunt Groups Calls are forwarded through a Extension 200 is a pilot number SCCP: Configuring Hunt
pool of agents until answered or for the sales department. Groups, page 596
sent to a final number. Extensions 213, 214, and 215
or
belong to sales agents in the hunt
group. When a call to extension SIP: Configuring Hunt
200 is received, it proceeds Groups, page 605.
through the list of agents until
one answers. If all the agents are
busy or do not answer, the call is
sent to voice mail.
Night Service Calls to ephone-dns that are not Extension 7544 is the cashier’s SCCP: Configuring Night
staffed during certain hours can desk but the cashier only works Service, page 609.
be answered by other phones until 3 p.m. A call is received at
using call pickup. 4:30 p.m. and the service
manager’s phone is notified. The
service manager uses call pickup
to answer the call.
Overlaid Ephone-dns Calls to several numbers can be Extensions 451, 452, and 453 all SCCP: Configuring Overlaid
answered by a single agent or appear on button 1 of a phone. A Ephone-dns, page 615.
multiple agents. call to any of these numbers can
be answered from button 1.

Call Hunt
Call hunt allows you to use multiple directory numbers to provide coverage for a single called number.
You do this by assigning the same number to several primary or secondary ephone-dns or by using
wildcards in the number associated with the directory numbers.
Calls are routed based on a match between the number dialed and the destination patterns that are
associated with dial peers. Through the use of wildcards in destination patterns, multiple dial peers can
match a particular called number. Call hunt is the ability to search through the dial peers that match the
called number until the call is answered. Call hunt uses a technique called preference to control the order
in which dial peers are matched to an incoming call and a technique called huntstop to determine when
the search for another matching peer ends.
In Cisco Unified CME, incoming calls search through the virtual dial peers that are automatically
created when you define directory numbers. These virtual dial peers are not directly configurable; you
must configure the directory number to control call hunt for virtual dial peers.

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Channel huntstop is used to stop the search for the two channels of a dual-line directory number. 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.
Huntstop prevents hunt-on-busy from redirecting a call from a busy phone into a dial peer that has been
setup with a catch-all default destination.
For configuration information, see the “SCCP: Configuring Call Hunt” section on page 586 and the
“SIP: Configuring Call Hunt” section on page 589.

Call Pickup
Call pickup, and pickup groups, enable phone users to answer a call that is ringing on a different
directory number other than their own. If both numbers to be answered are in the same pickup group, the
user presses fewer keys to pick up the call.
Call pickup has the following variations:
• Directed Call Pickup—Call pickup, explicit ringing extension. Any local phone user can pick up a
call that is on hold on another directory number in Cisco Unified CME. Phone user does not need to
belong to a pickup group to use this method. This is a default behavior.
• Group Pickup, Different Group—Call pickup, explicit group ringing extension. Phone user can
answer a ringing phone in any pickup group if the user knows the group number of the ringing
phone. If there is only one pickup group defined in Cisco Unified CME, the phone user can pick up
the call by pressing a soft key. Phone user does not need to belong to a pickup group to use this
method.
• Local Group Pickup—Call pickup, local group ringing extension. Phone users can pick up the called
number on another phone by pressing a soft key plus an asterisk (*) their own phone if both phones
are in the same pickup group.
Administrators can assign each ephone-dn independently to a maximum of one pickup group. 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 Unified CME system.
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 Unified CME system because
a pickup in group 17 will always be triggered before the user can enter the final 7 for 177.

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Figure 34 shows four call-pickup scenarios.

Figure 34 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.

For configuration information, see the “SCCP: Creating Pickup Groups” section on page 591.

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Call Waiting
Call waiting allows phone users to be alerted when they receive an incoming call while they are on
another call. Phone users hear a call-waiting tone when another party is trying to reach them and, on IP
phones, see the calling party information on the phone screen.
Call-waiting calls to IP phones with soft keys can be answered using the Answer soft key. Call-waiting
calls to analog phones controlled by Cisco Unified CME systems are answered using hookflash. When
phone users answer a call-waiting call, their original call is automatically put on hold. If a phone user
does not respond to a call-waiting notification, the call is forwarded as specified in the call-forward
noan command for that extension.
For an IP phone running SCCP, call waiting for single-line ephone-dns requires two ephone-dns to
handle the two calls. Call waiting on a dual-line ephone-dn requires only one ephone-dn because the two
channels of the ephone-dn handle the two calls. The audible call-waiting indicator can be either a
call-waiting beep or a call-waiting ring. For configuration information, see the “SCCP: Configuring
Call-Waiting Indicator Tone” section on page 593.
For a SIP phone, call waiting is automatically enabled when you configure a voice register pool. For SIP
phones directly connected to Cisco Unified CME, call waiting can be disabled at the phone-level. For
configuration information, see the “SIP: Enabling Call Waiting” section on page 595.
For information on call waiting using Overlaid ephone-dns, see the “Overlaid Ephone-dns” section on
page 581.

Call-Waiting Beep for SCCP Phones

Call-waiting beeps are enabled by default. You can disable the call-waiting beeps that are generated from
and accepted by directory numbers. If beep generation is disabled, incoming calls to the directory
number do not generate call-waiting beeps. If beep acceptance is disabled, the phone user does not hear
beeps when using the directory number for an active call.
Table 29 shows the possible beep behaviors of one ephone-dn calling another ephone-dn that is
connected to another caller.

Table 29 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

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Table 29 Call-Waiting Beep Behavior (continued)

Incoming
Active Call Call Expected
Ephone-dn 1 Configuration Ephone-dn 2 Configuration on DN on DN Behavior
no call-waiting beep accept — DN 1 DN 2 No beep
— no call-waiting beep DN 1 DN 1 Beep

Call-Waiting Ring for SCCP Phones

Instead of the standard call waiting beep sound through the handset, you can use a short ring for
call-waiting notification. The default is for directory numbers to accept call interruptions, such as call
waiting, and to issue a beeping sound for notification.
To use a ring sound, the directory number must accept call-waiting indicator tones. For configuration
information, see the “SCCP: Configuring Call-Waiting Indicator Tone” section on page 593 or the “SIP:
Enabling Call Waiting” section on page 595.

Callback Busy Subscriber

This feature allows callers who dial a busy extension number to request a callback from the system when
the called number is available. Callers can also request callbacks for extensions that do not answer, and
the system will notify them after the called phone is next used.
There can be only one callback request pending against a particular extension number, although a caller
can initiate more than one callback 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. To display a list of phones that have pending callback
requests, use the show ephone-dn callback command.

Hunt Groups
Hunt groups allow incoming calls to a specific number (pilot number) to be directed to a defined group
of directory numbers. Each hunt group can include up to 20 member directory numbers.
Incoming calls are redirected from a hunt group pilot number to the first directory number as defined by
the configuration. If the first directory number 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 directory number to
directory number in the list until it is answered or until the call reaches the number that was defined as
the final number.
The redirect from one directory number to the next in the list is also known as a hop. You can set the
maximum number of redirects for specific peer or longest-idle hunt groups, and for the maximum
number of redirects allowed in a Cisco Unified CME system, both inside and outside hunt groups. If a
call makes the maximum number of hops or redirects without being answered, the call is dropped.
For information on displaying hunt group statistics, see Cisco Unified CME B-ACD and Tcl
Call-Handling Applications.

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There are four different types of hunt groups. Each type uses a different strategy to determine the first
directory number that rings for successive calls to the hunt group pilot number. Hunt group types include
the following:
• Sequential Hunt Groups—Directory numbers 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 hunt groups.
• Peer Hunt Groups—The first directory number to ring is the number to the right of the directory number
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 in the hunt group configuration.
• Longest-Idle Hunt Groups—Calls go first to the directory number that has been idle the longest for the
number of hops specified when the hunt group was defined. The longest-idle time is determined from
the last time that a phone registered, reregistered, or went on-hook.
• Parallel Hunt Groups—Calls ring all directory numbers in the hunt group simultaneously.
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
a 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 35 on page 571 illustrates a sequential hunt group, Figure 36 on page 572 illustrates a peer hunt
group, and Figure 37 on page 573 illustrates a longest-idle hunt group.

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Sequential Hunt Groups

In a sequential hunt group, directory numbers 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
hunt groups.

Figure 35 Sequential 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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Peer Hunt Groups

In a peer hunt group, the first directory number to ring is the number to the right of the directory number 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 hunt group was defined.
Figure 36 illustrates a peer hunt group.

Figure 36 Peer 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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Longest-Idle Hunt Groups

In a longest-idle hunt group, the algorithm for choosing the next extension to receive a call is based on
a comparison of on-hook time stamps. The extension with the smallest on-hook time stamp value is
chosen when the next call comes to the hunt group.
The default behavior is that an on-hook time stamp value for an extension is updated only when the agent
answers a call. In Cisco Unified CME 4.0 and later versions, you can specify that an on-hook time stamp
is updated when a call rings an extension and also when a call is answered by an agent.
Figure 37 illustrates a longest-idle hunt group.

Figure 37 Longest-Idle 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, 504
final 6000
Phone 4 hops 3
Button 1 is extension 5044 IP preference 1
103299

timeout 30
no-reg

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Parallel Hunt Groups

In a parallel hunt group, calls simultaneously ring multiple phones. The use of parallel hunt groups is
also referred to as application-level forking because it enables the forking of a call to multiple
destinations. Parallel hunt group are supported by SIP phones only. (To enable similar functionality on
SCCP phones, use the ephone-dn overlay feature for shared lines. See the “Shared-Line Overlays”
section on page 582.)
In the following parallel hunt group example, when callers dial extension 1000, extension 1001, 1002,
and so on ring simultaneously. The first extension to answer is connected. If none of the extensions
answers, the call is forwarded to extension 2000, which is the number for the voice-mail service.
voice hunt-group 4 parallel
pilot 1000
list 1001, 1002, 1003, 1004
final 2000
timeout 20

The number of ringing calls that a parallel hunt group can support depends on whether call-waiting is
enabled on the SIP phones.
If call-waiting is enabled (the default), parallel hunt groups support multiple calls up to the limit of
call-waiting calls supported by a particular SIP phone model. You may not want to use unlimited
call-waiting however with parallel hunt-groups if agents do not want a large number of waiting calls
when they are already handling a call.
If call waiting is disabled, parallel hunt groups support only one call at a time in the ringing state. After
a call is answered (by one of the phones in the hunt group), a second call is allowed. The second and
subsequent calls ring only the idle phones in the hunt group, and bypass the busy phone that answered
the first call (because this phone is connected to the first call). After the second call is answered, a third
call is allowed, and so on until all the phones in the parallel hunt group are busy. The hunt group does
not accept further calls until at least one phone returns to the idle/on-hook state.
When two or more phones within the same parallel hunt group attempt to answer the same call, only one
phone can connect to the call. Phones that fail to connect must return to the on-hook state before they
can receive subsequent calls. Calls that arrive before a phone is placed on-hook are not presented to the
phone. For example, if a second call arrives after Phone 1 has answered the original call, but before
Phone 2 goes back on-hook, the second call bypasses Phone 2 (because it is offhook).
When a phone returns to the idle/on-hook state, it does not automatically re-synchronize to the next call
waiting to be answered. For example, in the previous scenario, if the second call is still ringing Phone 3
when Phone 2 goes on-hook, Phone 2 does not ring because it was offhook when the second call arrived.
For configuration information, see the “SIP: Configuring Hunt Groups” section on page 605.

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Hunt Group Agent Availability Options

Three options increase the flexibility of hunt group agents by allowing them to dynamically join and
leave hunt groups or to temporarily enter a not-ready state in which they do not receive calls.
Table 30 compares the following agent availability features:
• Dynamic Hunt Group Membership, page 577
• Agent Status Control, page 577
• Automatic Agent Status Not-Ready, page 578

Table 30 Comparison of Hunt Group Agent Availability Features

Comparison
Factor Dynamic Membership Agent Status Control Automatic Agent Status Not-Ready
Purpose Allows an authorized agent to join Allows an agent to manually Automatically puts an agent’s
and leave hunt groups. activate a toggle to temporarily phone in a not-ready state after a
enter a not-ready state, in which specified number of hunt-group
hunt-group calls bypass the calls are unanswered by the
agent’s phone. agent’s phone.
Example Agent A joins a hunt group at Agent A takes a coffee break at Agent B is suddenly called away
8 a.m. and takes calls until 1 p.m., 10 a.m. and puts his phone into a from her desk before she can
when he leaves the hunt group. not-ready status while he is on manually put her phone into the
While Agent A is a member of the break. When he returns he puts his not-ready status. After a
hunt group, he occupies one of the phone back into the ready status hunt-group call is unanswered at
wildcard slots in the list of and immediately starts receiving Agent B’s phone, the phone is
numbers configured for the hunt hunt-group calls again. He automatically placed in the
group. At 1 p.m., Agent B joins retained his wildcard slot while he not-ready status and it is not
the hunt group using the same was in the not-ready status. presented with further hunt-group
wildcard slot that Agent A calls. When Agent B returns, she
relinquished when he left. manually puts her phone back into
the ready status.
Hunt-group slot An agent joining a hunt group An agent who enters the not-ready An agent who enters the not-ready
availability occupies a wildcard slot in the state does not give up a slot in the does not give up a slot in the hunt
hunt group list. An agent leaving hunt group. The agent continues to group. The agent continues to
the group relinquishes the slot, occupy the slot regardless of occupy the slot regardless of
which becomes available for whether the agent is in the whether the agent is in the
another agent. not-ready status. not-ready status.

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Table 30 Comparison of Hunt Group Agent Availability Features (continued)

Comparison
Factor Dynamic Membership Agent Status Control Automatic Agent Status Not-Ready
Agent activation An authorized agent uses a feature An agent uses the HLog soft key An agent who is a member of a
method access code (FAC) to join a hunt to toggle agent status between hunt group configured with the
group and a different FAC to leave ready and not ready. Agents can auto logout command does not
the hunt group. also use the HLog ephone FAC or answer the specified number of
the HLog ephone-dn FAC to calls, and the agent’s phone is
toggle between ready and automatically changed to the
not-ready if FACs are enabled. not-ready status. The agent uses
the HLog soft key or a FAC to
If the HLog soft key is not
return to the ready status.
enabled, the DND soft key can be
used to put an agent in the If the HLog soft key or FAC has
not-ready status and the agent will not been enabled in the
not receive any calls. configuration, the agent uses the
DND soft key to return to the
ready status.
Configuration The system administrator uses the The system administrator uses the The system administrator uses the
list command to configure up to HLog keyword with the auto logout command to enable
20 wildcard slots in a hunt group hunt-group logout command to automatic agent status not-ready
and uses the ephone-hunt login provide an HLog soft key on for a hunt group.
command to authorize certain display phones and uses the fac
This functionality is disabled by
directory numbers to use these command to enable standard FACs
default.
wildcard slots. or create a custom FAC.
See SCCP: Configuring Hunt
See SCCP: Configuring Hunt See SCCP: Configuring Hunt
Groups, page 596.
Groups, page 596. Groups, page 596.
Optional The system administrator can The system administrator can use The system administrator can use
customizations establish custom FACs for agents the softkeys commands to change the auto logout command to
to use to enter or leave a hunt the position or prevent the display specify the number of unanswered
group. of the HLog soft key on individual calls that will trigger an agent
phones. status change to not-ready and
whether this feature applies to
dynamic hunt-group members,
static hunt-group members, or
both.
The system administrator can use
the hunt-group logout command
to specify whether an automatic
change to the not-ready status also
places a phone in DND mode.

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Dynamic Hunt Group Membership

Hunt groups allow you to set up pools of extension numbers to answer incoming calls. Up to 20 wildcard
slots can be entered in the list of hunt group extension numbers to allow dynamic group membership, in
which authorized phone users can join a hunt group whenever a vacant wildcard slot is available and
they can leave when they like. Each phone user who joins a group occupies one slot. If no slots are
available, a user who tries to join a group hears a busy signal.
Allowing dynamic membership in a hunt group is a three-step process:
1. Use the list command in ephone-hunt configuration mode to specify up to 20 wildcard slots in the
hunt group.
2. Use the ephone-hunt login command under each directory number that should be allowed to
dynamically join and leave hunt groups. Directory numbers are disallowed from joining hunt groups
by default, so you have to explicitly allow this behavior for each directory number that you want to
be able to log in to hunt groups.
3. Use the fac standard command to enable standard FACs or the fac custom command to define
custom FACs. FACs must be enabled so that agents can use them to join and leave hunt groups.
To dynamically join a hunt group, a phone user dials a standard or custom FAC for joining a hunt group.
The standard FAC to join a hunt group is *3. If multiple hunt groups have been created that allow
dynamic membership, the phone user must also dial the hunt group pilot number. For example, if the
following hunt groups are defined, a phone user dials *38000 to join the Sales hunt group:
ephone-hunt 24 sequential
pilot 8000
list 8001, 8002, *, *
description Sales Group
final 9000

ephone-hunt 25 sequential
pilot 7000
list 7001, 7002, *, *
description Service Group
final 9000

To leave a hunt group, a phone user dials the standard or custom FAC for leaving a hunt group. The
standard FAC to leave a hunt group is #3. See “Customizing Soft Keys” on page 829.

Note The Dynamic Membership feature is different from the Agent Status Control feature and the Automatic
Agent Status Not-Ready feature. Table 30 on page 575 compares the features.

Agent Status Control

The Agent Status Control feature allows ephone-hunt-group agents to control whether their phones are
in the ready or not-ready status. A phone in the ready status is available to receive calls from the hunt
group. A phone in the not-ready status blocks calls from the hunt group. Agents should use the not-ready
status for short breaks or other temporary interruptions during which they do not want to receive
hunt-group calls.
Agents who put their phones into the not-ready status do not relinquish their slots in the hunt group list.

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Agents use the HLog soft key or the DND soft key to put a phone into the not-ready status. When the
HLog soft key is used to put a phone in the not-ready status, it does not receive hunt group calls but can
receive other calls. If the DND soft key is used, the phone does not receive any calls until it is returned
to the ready status. The HLog and DND soft keys toggle the feature: if the phone is in the ready status,
pressing the key puts the phone in the not-ready status and vice-versa.
The DND soft key is visible on phones by default, but the HLog soft key must be enabled in the
configuration using the hunt-group logout command, which has the following options:
• HLog—Enables both an HLog soft key and a DND soft key on phones in the idle, seized, and
connected call states. When you press the HLog soft key, the phone is changed from the ready to
not-ready status or from the not-ready to ready status. When the phone is in the not-ready status, it
does not receive calls from the hunt group, but it is still able to receive calls that do not come through
the hunt group (calls that directly dial its extension). The DND soft key is also available to block all
calls to the phone if that is the preferred behavior.
• DND—Enables only a DND soft key on phones. The DND soft key also changes a phone from the
ready to not-ready status or from the not-ready to ready status, but the phone does not receive any
incoming calls, including those from outside hunt groups.
Phones without soft-key displays can use a FAC to toggle their status from ready to not-ready and back
to ready. The fac command must be used to enable the standard set of FACs or to create custom FACs.
The standard FAC to toggle the not-ready status at the directory number (extension) level is *4 and the
standard FAC to toggle the not-ready status at the ephone level (all directory numbers on the phone) is
*5. See Where to Go Next, page 633.

Note The Agent Status Control feature is different from the Dynamic Membership feature and the Automatic
Agent Status Not-Ready feature. Table 30 on page 575 compares the features.

Automatic Agent Status Not-Ready

Before Cisco Unified CME 4.0, this feature was known as Automatic Hunt Group Logout. If the auto
logout command was enabled for a hunt group, a phone was placed in DND mode when a line on the
phone did not answer a call for that hunt group within the time limit specified in the timeout command.
In Cisco Unified CME 4.0 and later versions, the name and behavior of this feature has changed,
although the Cisco IOS command remains the same. The auto logout command now specifies the
number of unanswered hunt group calls after which the agent status of an directory number is
automatically changed to not-ready. You can limit Automatic Agent Status Not-Ready to dynamic hunt
group members (those who log in using a wildcard slot in the list command) or to static hunt group
members (those who are explicitly named in the list command), or you can apply this behavior to all
hunt group members.
A related command, hunt-group logout, specifies whether the phones that are automatically changed to
the not-ready status should also be placed into DND mode. Phones in the not-ready status do not accept
calls from hunt groups, but they do accept calls that directly dial their extensions. Phones in DND mode
do not accept any calls. The default if the hunt-group logout command is not used is that the phones
that are automatically placed in the not-ready status are also placed in DND mode.
Agents whose phones are automatically placed into the not-ready status do not relinquish their slots in
the hunt group list.

Note The Automatic Agent Status Not-Ready feature is different from the Dynamic Membership feature and
the Agent Status Control feature. Table 30 on page 575 compares the features.

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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
(known as night-service directory numbers or night-service lines) send a special “burst” ring to phones
that have been specified to receive this special ring (the phones are known as night-service phones).
Phone users at the night-service phones can then use the call-pickup feature to answer the incoming calls
from the night-service directory numbers (Figure 38).
For example, the night-service feature can allow an employee working after hours to intercept and
answer calls that are presented to an unattended receptionist’s 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 Unified 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. A phone
user at any of the night-service 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. Step 4 to
Step 8 in the following procedure define the night-service period.
2. Night-service extensions (directory numbers)—When a night-service extension receives an
incoming call during the night-service period, night-service notification is triggered. Step 12 in the
following procedure specifies night service for an directory number.
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. Step 15 in the following procedure assigns 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. Before Cisco CME 3.3,
using the night-service code turned night service on or off only for directory numbers 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 directory number turns night service on or off for all phones with
night-service directory numbers. The following procedure defines a night-service toggle code.

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Figure 38 illustrates night service.

Figure 38 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

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Overlaid Ephone-dns
Overlaid ephone-dns are directory numbers that share the same button on a 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. They can have the same extension number or different numbers. The same
ephone-dns can appear on more than one phone and more than one phone can have the same set of
overlaid ephone-dns.
The order in which overlaid ephone-dns are used by incoming calls can be determined by the call hunt
commands, preference and huntstop. For example, ephone-dn 1 to ephone-dn 4 have the same
extension number, 1001. 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 second call
will roll over to the ephone-dn with the next-highest preference, and so forth. For more information, see
the “Call Hunt” section on page 565.
If the ephone-dns in an ephone-dn overlay use different numbers, incoming calls go to the ephone-dn
with the highest preference. If no preferences are configured, the dial-peer hunt command setting is
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.

Figure 39 on page 581 shows an overlay set with two directory numbers and one number that is shared
on two phones. Ephone-dn 17 has a 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 directory number 18.

Figure 39 Overlaid 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

When a call is answered on an ephone-dn, that ephone-dn is no longer available 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 displays on phone 1 and is removed from the screens of phone 2 and phone 3. All
actions pertaining to the call to extension 1001 (ephone-dn 17) are displayed on phone 1 only. If phone
1 puts extension 1001 on hold, the other phones will not be able to pick up the 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

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the ephone-dn while it is in use. When phone users press 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 must pick up 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).

Shared-Line Overlays
Dual-line ephone-dns can also 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.
The primary ephone-dn in a shared-line overlay set should be unique to the phone to guarantee that the
phone has a line available for outgoing calls, and to ensure that the phone user can obtain dial-tone even
when there are no idle lines available in the rest of the shared-line overlay set. Use a unique ephone-dn
to provide for a unique calling party identity on outbound calls made by the phone so that the called user
can see which specific phone is calling.
The following example shows the configuration for a simple shared-line overlay set. The primary
ephone-dn that is configured for each phone is unique while the remaining ephone-dns 10, 11, and 12
are shared in the overlay set on both phones:
!
ephone 1
mac-address 1111.1111.1111
button 1o1,10,11,12
!
ephone 2
mac-address 2222.2222.2222
button 1o2,10,11,12

For a more detailed example, see the “Shared-line Overlaid Ephone-dns: Example” section on page 628.
A more complex directory number configuration mixes overlaid directory numbers with shared directory
numbers and plain dual-line directory numbers on the same phones. Figure 40 on page 583 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
directory numbers, so the manager can have two active calls on this number simultaneously, one on each
button. The directory numbers 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 directory number and also shares the manager’s
directory numbers. Assistant 1 has all three directory numbers 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 (directory number 20).
2. A second call rings on 2001 and rolls over to the second button on the manager’s phone (directory
number 21). It also rings on both assistants’ phones, where it is also directory number 21, a shared
directory number.

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3. Assistant 2 answers the call. This is a shared overlay line (one directory number, 21, is shared among
three phones, and on two of them this directory number 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 directory number 22. The button is available because of the
additional directory numbers in the overlay set on the assistant 1 phone.
At this point, the manager is in conversation on directory number 20, assistant 1 is in conversation on
directory number 22, and assistant 2 is in conversation on directory number 21.

Figure 40 Overlaid 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

For configuration information, see the “SCCP: Configuring Overlaid Ephone-dns” section on page 615.

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 are
answered using hookflash. When phone users answer a call-waiting call, their original call is
automatically put on hold. If phone users ignore a call-waiting call, the caller is forwarded if
call-forward no-answer has been configured.
In Cisco CME 3.2.1 and later versions, call waiting is available for overlaid ephone-dns. 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-dns without call waiting use the o keyword. For configuration information, see the
“SCCP: Configuring Overlaid Ephone-dns” section on page 615.

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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
“SCCP: Configuring Call-Waiting Indicator Tone” section on page 593.) 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.
• In Cisco Unified CME 4.0 and later versions, up to six waiting calls can be displayed on
Cisco Unified IP Phone 7940G, 7941G, 7941G-GE, 7960G, 7961G, 7961G-GE, 7970G, and
7971G-GE. For all other phones and earlier Cisco Unified CME versions, two calls to numbers in
an overlaid ephone-dn set can be announced. Subsequent calls must wait in line 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 Cisco Unified IP Phone 7910 (maximum two call-waiting calls) 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 generates call-waiting notification. 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 can then talk to the person who called ephone-dn 2. The call
to ephone-dn 3 issues call-waiting notification while the call to ephone-dn 4 waits in line. (The
Cisco Unified IP Phone 7960 supports six calls waiting.) Phones configured for call waiting do not
generate call-waiting notification when they are transferring calls or hosting conference calls.
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 are forwarded to the configured destination. If
call-forward-no-answer is not configured, incoming calls receive ringback tones until the calls are
answered.
More than one phone can use the same set of overlaid ephone-dns. In this case, the call-waiting behavior
is slightly different. The following example demonstrates call waiting for overlaid ephone-dns that are
shared on two phones.
ephone 1
button 1c1,2,3,4
!
ephone 2
button 1c1,2,3,4

1. A call to ephone-dn 1 rings on ephone 1 and on ephone 2. Ephone 1 answers, and the call is no longer
visible to ephone 2.
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.
3. 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.
4. 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. (Note that the call maximum is six for those phones that are able to handle six
call-waiting calls, as previously described.)

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Note Ephone-dns accept call interruptions, such as call waiting, by default. For call waiting to work, the
default must be active. For more information, see the “SCCP: Configuring Call-Waiting Indicator Tone”
section on page 593.

Extending Calls for Overlaid Ephone-dns to Other Buttons on the Same Phone
Phones with overlaid ephone-dns can use the button command with the x keyword to dedicate one or
more additional buttons to receive overflow calls. If an overlay button is busy, an incoming call to any
of the other ephone-dns in the overlay set rings on the first available overflow button on each phone that
is configured to receive the overflow. This feature works only for overlaid ephone-dns that are
configured with the button command and the o keyword; it is not supported with overlaid ephone-dns
that are configured using the button command and the c keyword or other types of ephone-dns that are
not overlaid.
Using the button command with the c keyword results in multiple calls on one button (the button is
overlaid with multiple ephone-dns that have call waiting), whereas using the button command with the
o keyword and the x keyword results in one call per button and calls on multiple buttons.
For example, an ephone has an overlay button with ten numbers assigned to it using the button command
and the o keyword. The next two buttons on the phone are configured using the button command and
the x keyword. These buttons are reserved to receive additional calls to the overlaid extensions on the
first button when the first button is in use.
ephone 276
button 1o24,25,26,27,28,29,30,31,32,33 2x1 3x1

For configuration information, see the “SCCP: Configuring Overlaid Ephone-dns” section on page 615.

How to Configure Call Coverage Features

This section contains the following procedures:

Call Hunt
• SCCP: Configuring Call Hunt, page 586 (required)
• SCCP: Verifying Call Hunt, page 587 (optional)
• SIP: Configuring Call Hunt, page 589 (required)

Call Pickup
• SCCP: Creating Pickup Groups, page 591 (required)
• SCCP: Verifying Call Pickup, page 592 (optional)

Call Waiting
• SCCP: Configuring Call-Waiting Indicator Tone, page 593 (optional)
• SIP: Enabling Call Waiting, page 595 (required)

Hunt Groups
• SCCP: Configuring Hunt Groups, page 596 (required)
• SCCP: Verifying Hunt Groups, page 603 (optional)

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• SIP: Configuring Hunt Groups, page 605 (required)

Night Service
• SCCP: Configuring Night Service, page 609 (required)
• SCCP: Verifying Night Service, page 613 (optional)

Overlaid Ephone-dns
• SCCP: Configuring Overlaid Ephone-dns, page 615 (required)
• SCCP: Verifying Overlaid Ephone-dns, page 618 (optional)

SCCP: Configuring Call Hunt

To configure a group of directory numbers to provide call coverage for a single called number, perform
the following steps for each directory number in the group.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. preference preference-order [secondary secondary-order]
6. huntstop
7. huntstop channel
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode for the purpose of
configuring a directory number.
Example:
Router(config)# ephone-dn 20 dual-line

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Command or Action Purpose

Step 4 number number [secondary number] [no-reg Associates a telephone or extension number with the directory
[both | primary]] number.
• Assign the same number to several primary or secondary
Example: ephone-dns to create a group of virtual dial peers through
Router(config-ephone-dn)# number 101 which the incoming called number must search.
Step 5 preference preference-order [secondary Sets the preference value for the ephone-dn.
secondary-order]
• Default: 0.
• Increment the preference order for subsequent ephone-dns
Example:
Router(config-ephone-dn)# preference 2
with the same number. That is, the first directory number is
preference 0 by default and you must specify 1 for the second
ephone-dn with the same number, 2 for the next, and so on.
• secondary secondary-order—(Optional) Preference value
for the secondary number of an ephone-dn. Default is 0.
Step 6 no huntstop Explicitly enables call hunting behavior for a directory number.
or
• Configure no huntstop for all ephone-dns, except the final
huntstop ephone-dn, within a set of ephone-dns with the same number.
• Configure the huntstop command for the final ephone-dn
Example: within a set of ephone-dns with the same number.
Router(config-ephone-dn)# no huntstop

or
Router(config-ephone-dn)# huntstop
Step 7 huntstop channel (Optional) Enables channel huntstop, which keeps a call from
hunting to the next channel of an ephone-dn if the first channel is
busy or does not answer.
Example:
Router(config-ephone-dn)# huntstop • Required for dual-line ephone-dns.
channel
Step 8 end Returns to privileged EXEC mode.

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

What to Do Next
If you want to collect statistics for hunt groups, see Cisco Unified CME B-ACD and Tcl Call-Handling
Applications.

SCCP: Verifying Call Hunt

To verify the configuration for call hunt, perform the following steps.

SUMMARY STEPS

1. show running-config
2. show telephony-service ephone-dn

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3. show telephony-service all

or
show telephony-service dial-peer

DETAILED STEPS

Step 1 show running-config

This command displays your configuration. Preference and huntstop information is listed in the
ephone-dn portion of the output.
Router# show running-config

ephone-dn 2 dual-line
number 126
description FrontDesk
name Receptionist
preference 1
call-forward busy 500
huntstop channel
no huntstop

Step 2 show telephony-service ephone-dn

This command displays ephone-dn preference and huntstop configuration information.
Router# show telephony-service ephone-dn

ephone-dn 243
number 1233
preference 1
huntstop

Step 3 show telephony-service all

or
show telephony-service dial-peer
These commands display preference and huntstop configurations for ephone-dn dial peers.
Router# show telephony-service dial-peer

!
dial-peer voice 20026 pots
destination-pattern 5002
huntstop
call-forward noan 5001 timeout 45
port 50/0/2

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SIP: Configuring Call Hunt

To configure the call hunting feature and prevent hunt-on-busy from redirecting a call from a busy phone
into a dial peer that has been setup with a catch-all default destination, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register dn dn-tag
4. number number
5. preference preference-order
6. huntstop
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or an MWI.
Example:
Router(config)# voice register dn 1
Step 4 number number Associates a phone number with the directory number.
• Assign the same number to several directory numbers
Example: to create a group of virtual dial peers through which the
Router(config-register-dn)# number 5001 incoming called number must search.
Step 5 preference preference-order Creates the preference order for matching the VoIP dial
peers created for the number associated with this directory
number to establish the hunt strategy for incoming calls.
Example:
Router(config-register-dn)# preference 4 • Default is 0, which is the highest preference.

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Command or Action Purpose

Step 6 huntstop Disables call-hunting behavior for an extension on a SIP
phone.
Example:
Router(config-register-dn)# huntstop
Step 7 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-dn)# end

What to Do Next
If you want to collect statistics for hunt groups, see Cisco Unified CME B-ACD and Tcl Call-Handling
Applications.

SCCP: Enabling Local-Group Call Pickup at a System-Level

To enable local-group call pickup at a system-level, and disable directed call pickup, perform the
following steps.

Note To selectively disable directed call pickup for one or more SCCP phones, use the features blocked
command in ephone-template mode. For configuration information, see “SCCP: Enabling Ephone
Templates” on page 883.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. no service directed-pickup
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 no service directed-pickup Disables directed call pickup.
• Changes the action of the PickUp soft key to perform
Example: local group call pickup rather than directed call pickup.
Router(config-telephony)# no service
directed-pickup
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

SCCP: Creating Pickup Groups

To create a pickup group, perform the following steps.

Prerequisites
Directory numbers to be added to a pickup group must be configured in Cisco Unified CME. For
configuration information, see “SCCP: Creating Directory Numbers” on page 158.

Restrictions
• Each directory number can be independently assigned to a maximum of one pickup group.
• There is no limit to the number of directory numbers that can be assigned to a single pickup group.
• There is no limit to the number of pickup groups that can be defined in Cisco Unified CME.
• Pickup group numbers may be of varying length, but must have unique leading digits.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. pickup-group number
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode for the purpose of
configuring a directory number.
Example:
Router(config)# ephone-dn 20 dual-line
Step 4 pickup-group number Creates a pickup group and assigns the directory number
being configured to the group.
Example: • number—Digit string of up to 32 characters. Group
Router(config-ephone-dn)# pickup-group 2345 numbers may be of varying length, but they must have
unique leading digits. For example, if there is a group
number 17, there cannot also be a group number 177.
Step 5 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

SCCP: Verifying Call Pickup

Step 1 Use the show running-config command to verify your configuration. Call pickup groups are listed in
the ephone-dn portion of the output.
Router# show running-config
!
ephone-dn 34 dual-line
ring feature secondary
number 330 secondary 331
pickup-group 30
call-forward noan 500 timeout 10 secondary
huntstop channel
no huntstop

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Step 2 Use the show telephony-service ephone-dn command to display call pickup configuration information.
Router# show telephony-service ephone-dn

ephone-dn 2
number 5002
pickup group 30
call-forward noan 5001 timeout 8

SCCP: Configuring Call-Waiting Indicator Tone

To specify the type of audible call-waiting indicator on a SCCP phone, perform the following steps. The
default is for directory numbers to accept call interruptions, such as call waiting, and to issue a beep tone.
Instead of the standard call waiting beep, you can enable a ring tone for call-waiting.

Restrictions
• The call-waiting ring option is not supported if the ephone-dn is configured with the no call-waiting
beep accept command.
• If you configure a button to have a silent ring, you will not hear a call-waiting beep or call-waiting
ring regardless of whether the ephone-dn associated with the button is configured to generate a
call-waiting beep or call-waiting ring. To configure a button for silent ring, see the “SCCP:
Assigning Directory Numbers to Phones” on page 160.
• The call-waiting beep volume cannot be adjusted through Cisco Unified CME for the
Cisco Unified IP Phone 7902G, Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7912G,
Cisco ATA-186, and Cisco ATA-188.
• The call-waiting ring option is not supported on the Cisco Unified IP Phone 7902G,
Cisco Unified IP Phone 7905G, or Cisco Unified IP Phone 7912G.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. call-waiting beep [accept | generate]
5. call-waiting ring
6. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an
ephone-dn, and optionally assigns it dual-line status.
Example:
Router(config)# ephone-dn 20 dual-line
Step 4 call-waiting beep [accept | generate] Enables an ephone-dn to generate or accept call-waiting
beeps.
Example: • Default is directory number both accepts and generates
Router(config-ephone-dn)# no call-waiting beep call waiting beep.
accept
• The beep is heard only if the other ephone-dn is
configured to accept call-waiting beeps (default).
Step 5 call-waiting ring (Optional) Enables an ephone-dn to use a ring indicator for
call-waiting notification.
Example: • To use this command, do not disable call-waiting beep
Router(config-ephone-dn)# call-waiting ring by using the no call-waiting beep accept command.
Step 6 end Returns to privileged EXEC mode.

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

SCCP: Verifying Call-Waiting Indicator Tone

Step 1 Use the show running-config command to verify your configuration. Call-waiting settings are listed in
the ephone-dn portion of the output. 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.
Router# show running-config
!
ephone-dn 3 dual-line
number 126
name Accounting
preference 2 secondary 9
huntstop
huntstop channel
call-waiting beep
!

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Step 2 Use the show telephony-service ephone-dn command to display call-waiting configuration
information.
Router# show telephony-service ephone-dn

ephone-dn 1 dual-line
number 126 secondary 1261
preference 0 secondary 9
no huntstop
huntstop channel
call-forward busy 500 secondary
call-forward noan 500 timeout 10
call-waiting beep

SIP: Enabling Call Waiting

To enable call waiting on an individual SIP phone, perform the following steps.

Prerequisites
• Cisco Unified CME 3.4 or a later version.
• mode cme command must be configured in Cisco Unified CME.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register pool pool-tag
4. call-waiting
5. exit
6. voice register global
7. hold-alert timeout
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone in
Cisco Unified CME.
Example:
Router(config)# voice register pool 3
Step 4 call-waiting Configures call waiting on the SIP phone being configured.
Note This step is included to illustrate how to enable the
Example: command if it was previously disabled.
Router(config-register-pool)# call-waiting
• Default: Enabled.
Step 5 exit Exits voice register pool configuration mode.

Example:
Router(config-register-pool)# exit
Step 6 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 7 hold-alert timeout Sets an audible alert notification when a call is on hold on a
SIP phone. Default is disabled.
Example: • timeout—Interval after which an audible alert
Router(config-register-global)# hold-alert 30 notification is repeated, in seconds. Range: 15 to 300.
Step 8 end Exits to privileged EXEC mode.

Example:
Router(config-register-global)# end

SCCP: Configuring Hunt Groups

To define a hunt group and optional agent availability parameters, perform the following steps.

Prerequisites
Directory numbers to be included in a hunt group must be already configured in Cisco Unified CME.
For configuration information, see “SCCP: Creating Directory Numbers” on page 158.

Restrictions
• The HLog soft key is available only on display phones. It is not available on Cisco Unified IP
Phones 7902, 7905, and 7912; Cisco IP Communicator; and Cisco VG 224.
• Shared ephone-dns cannot use the Agent Status Control or Automatic Agent Not-Ready feature.
• The Agent Status Control feature and the HLog soft key require the user locale to be set to US. To
display the HLog soft key on a Cisco Unified IP Phone 7940 or Cisco Unified IP Phone 7960,
change the user locale to any locale other than US and reset the phone. Then change the user locale
to US and reset the phone again.

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• If directory numbers that are members of a hunt group are to be configured for called-name display,
the following restrictions apply:
– The primary or secondary pilot number must be defined using at least one wildcard character.
– The phone numbers in the list command cannot contain wildcard characters.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-hunt hunt-tag {longest-idle | peer | sequential}
4. pilot number [secondary number]
5. list number[, number...]
6. final final-number
7. hops number
8. timeout seconds[, seconds...]
9. max-timeout seconds
10. preference preference-order [secondary secondary-order]
11. no-reg [both | pilot]
12. fwd-final {orig-phone | final}
13. forward local-calls
14. secondary start [current | next | agent-position]
15. present-call {idle-phone | onhook-phone}
16. from-ring
17. description text-string
18. display-logout text-string
19. exit
20. telephony-service
21. max-redirect number
22. hunt-group logout {DND | HLog}
23. exit
24. ephone-dn dn-tag
25. ephone-hunt login
26. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-hunt hunt-tag {longest-idle | peer | Enters ephone-hunt configuration mode to define an ephone
sequential} hunt group.
• hunt-tag—Unique sequence number that identifies this
Example: hunt group during configuration tasks. Range: 1 to 100.
Router(config)# ephone-hunt 23 peer
• longest-idle—Calls go 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.
• peer—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.
• sequential—Ephone-dns ring in the left-to-right order
in which they are listed when the hunt group is defined.
Step 4 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 up to 27 characters. The
Router(config-ephone-hunt)# pilot 5601 dialplan pattern can be applied to the pilot number.
• secondary—(Optional) Defines an additional pilot
number for the ephone hunt group.
Step 5 list number[, number...] Defines the list of numbers (from 2 and 20) to which the
ephone hunt group redirects the incoming calls.
Example: • number—E.164 number up to 27 characters. Primary or
Router(config-ephone-hunt)# list 5001, 5002, secondary number assigned to an ephone-dn.
5017, 5028

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Command or Action Purpose

Step 6 final final-number Defines the last number in the ephone hunt group, after
which the call is no longer redirected. Can be an ephone-dn
primary or secondary number, a voice-mail pilot number, a
Example:
Router(config-ephone-hunt)# final 6000
pilot number of another hunt group, or an FXS number.
Note When 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.

Note This command is not used for ephone hunt groups

that are part of a Cisco Unified CME B-ACD
service. The final destination for those groups is
determined by the B-ACD service.
Step 7 hops number (Optional; peer and longest-idle hunt groups only) Sets the
number of hops before a call proceeds to the final number.
Example: • number—Number of hops before the call proceeds to
Router(config-ephone-hunt)# hops 7 the final ephone-dn. Range is 2 to 20, but the value must
be less than or equal to the number of extensions that
are specified in the list command. Default
automatically adjusts to the number of hunt group
members.
Step 8 timeout seconds[, seconds...] (Optional) Sets the number of seconds after which an
unanswered call is redirected to the next number in the
hunt-group list.
Example:
Router(config-ephone-hunt)# timeout 7, 10, 15 • seconds—Number of seconds. Range: 3 to 60000.
Multiple entries can be made, separated by commas,
that must correspond to the number of ephone-dns in
the list command. Each number in a multiple entry
specifies the time that the corresponding ephone-dn
will ring before a call is forwarded to the next number
in the list. If a single number is entered, it is used for
the no-answer period for each ephone-dn.
• If this command is not used, the default is the number
of seconds set by the timeouts ringing command,
which defaults to 180 seconds. Note that the default of
180 seconds may be greater than you desire.
Step 9 max-timeout seconds (Optional) Sets the maximum combined timeout for the
no-answer periods for all ephone-dns in the ephone-hunt
list. The call proceeds to the final destination when this
Example:
Router(config-ephone-hunt)# max-timeout 25
timeout expires, regardless of whether it has completed the
hunt cycle.
• seconds—Number of seconds. Range is 3 to 60000.
• If this command is not used, the default is that no
combined timeout limit is set.

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Command or Action Purpose

Step 10 preference preference-order [secondary (Optional) Sets a preference order for the ephone-dn
secondary-order] associated with the hunt-group pilot number.
• preference-order—See the CLI help for a range of
Example: numeric values, 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. See the CLI help
for a range of numeric values, where 0 is the highest
preference. Default is 9.
Step 11 no-reg [both | pilot] (Optional) Prevents the hunt-group pilot number from
registering with an H.323 gatekeeper. If this command is
not used, the default is that the pilot number registers with
Example:
Router(config-ephone-hunt)# no-reg
the H.323 gatekeeper.
• both—(Optional) Both the primary and secondary pilot
numbers are not registered.
• pilot—(Optional) Only the primary pilot number is not
registered.
• In Cisco CME 3.1 and later versions, if this command
is used without the either the both or pilot keywords,
only the secondary number is not registered.
Step 12 fwd-final {orig-phone | final} (Optional) For calls that have been transferred into an
ephone hunt group by a local extension, determines the final
destination of a call that is not answered in the hunt group.
Example:
Router(config-ephone-hunt)# fwd-final • final—Forwards the call to the ephone-dn number that
orig-phone is specified in the final command.
• orig-phone—Forwards the call to the primary
directory number of the phone that transferred the call
into the hunt group.
Step 13 forward local-calls (Optional; sequential hunt groups only) Specifies that local
calls (calls from ephone-dns on the same
Cisco Unified CME system) will not be forwarded past the
Example:
Router(config-ephone-hunt)# no forward
first list member in a hunt group. If the first member is busy,
local-calls the internal caller hears busy. If the first number does not
answer, the internal caller hears ringback.

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Command or Action Purpose

Step 14 secondary start [current | next | (Optional) For calls that are parked by hunt group member
list-position] phones, returns them to a different entry point in the hunt
group (as specified in this command) if the calls are recalled
Example: from park to the secondary pilot number or transferred from
Router(config-ephone-hunt)# secondary start park to an ephone-dn that forwards the call to the secondary
next pilot number.
• current—The ephone-dn that parked the call.
• next—The ephone-dn in the hunt group list that follows
the ephone-dn that parked the call.
• list-position—The ephone-dn at the specified position
in the list specified by the list command. Range is
1 to 10.
Step 15 present-call {idle-phone | onhook-phone} (Optional) Presents ephone-hunt-group calls only to
member phones that are idle or onhook, as specified.
Example: • idle-phone—A call from the ephone-hunt group is
Router(config-ephone-hunt)# present-call presented to an ephone only if all lines on the phone are
idle-phone idle. This option ignores monitored lines that have been
configured on the phone using the button m command.
• onhook-phone—A call from the ephone-hunt group is
presented to an ephone only if the phone is in the
on-hook state. When this keyword is configured, calls
in the ringing or hold state that are unrelated to the hunt
group do not prevent the presentation of calls from the
ephone-hunt group.
Step 16 from-ring (Optional) Specifies that on-hook time stamps should be
recorded when calls ring extensions and when calls are
answered. The default is that on-hook time stamps are
Example:
Router(config-ephone-hunt)# from-ring
recorded only when calls are answered.
Step 17 description text-string (Optional) Defines text that will appear in configuration
output.
Example:
Router(config-ephone-hunt)# description
Marketing Hunt Group
Step 18 display-logout text-string (Optional) Defines text that will appear on IP phones that
are members of a hunt group when all the hunt-group
members are in the not-ready status. This string can be used
Example:
Router(config-ephone-hunt)# display-logout
to inform hunt-group members where the calls are being
Night Service sent when all members are unavailable to take calls.
Step 19 exit Exits ephone-hunt configuration mode.

Example:
Router(config-ephone-hunt)# exit
Step 20 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 21 max-redirect number (Optional) Sets the number of times that a call can be
redirected within a Cisco Unified CME system.
Example: • number—Range is 5 to 20. Default is 5.
Router(config-telephony)# max-redirect 8
Note This command is required if the number of hops is
greater than 5.
Step 22 hunt-group logout {DND | HLog} (Optional) Specifies whether agent not-ready status applies
only to ephone hunt group extensions on a phone (HLog
mode) or to all extensions on a phone (DND mode). Agent
Example:
Router(config-telephony)# hunt-group logout
not-ready status can activated by an agent using the HLog
HLog soft key or a FAC, or it can be activated automatically after
the number of calls specified in the auto logout command
are not answered.
The default if this command is not used is DND.
• DND—When phones are placed in agent not-ready
status, all ephone-dns on the phone will not accept
calls.
• HLog—Enables the display of the HLog soft key.
When phones are placed in the agent not-ready status,
only the ephone-dns assigned to ephone hunt groups
will not accept calls.
Step 23 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 24 ephone-dn dn-tag (Optional) Enters ephone-dn configuration mode.
• dn-tag—Tag number for the ephone-dn to be
Example: authorized to join and leave ephone hunt groups.
Router(config)# ephone-dn 29
Step 25 ephone-hunt login (Optional) Enables this ephone-dn to join and leave ephone
hunt groups (dynamic membership).
Example:
Router(config-ephone-dn)# ephone-hunt login
Step 26 end Returns to privileged EXEC mode.

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

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SCCP: Verifying Hunt Groups

Step 1 Use the show running-config command to verify your configuration. Ephone hunt group parameters are
listed in the ephone-hunt portion of the output.
Router# show running-config

ephone-hunt 1 longest-idle
pilot 500
list 502, 503, *
max-timeout 30
timeout 10, 10, 10
hops 2
from-ring
fwd-final orig-phone
!
!
ephone-hunt 2 sequential
pilot 600
list 621, *, 623
final 5255348
max-timeout 10
timeout 20, 20, 20
fwd-final orig-phone
!
!
ephone-hunt 77 longest-idle
from-ring
pilot 100
list 101, *, 102
!
Step 2 To verify the configuration of ephone hunt group dynamic membership, use the show running-config
command. Look at the ephone-hunt portion of the output to ensure at least one wildcard slot is
configured. Look at the ephone-dn section to see whether particular ephone-dns are authorized to join
ephone hunt groups. Look at the telephony-service section to see whether FACs are enabled.
Router# show running-config

ephone-hunt 1 longest-idle
pilot 500
list 502, 503, *
max-timeout 30
timeout 10, 10, 10
hops 2
from-ring
fwd-final orig-phone
!
!
ephone-dn 2 dual-line
number 126
preference 1
call-forward busy 500
ephone-hunt login
!
telephony-service
fac custom alias 5 *5 to *35000
fac custom ephone-hunt cancel #5

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Step 3 Use the show ephone-hunt command for detailed information about hunt groups, including dial-peer
tag numbers, hunt-group agent status, and on-hook time stamps. This command also displays the
dial-peer tag numbers of all ephone-dns that have joined dynamically and are members of the group at
the time that the command is run.
Router# show ephone-hunt

Group 1
type: peer
pilot number: 450, peer-tag 20123
list of numbers:
451, aux-number A450A0900, # peers 5, logout 0, down 1
peer-tag dn-tag rna login/logout up/down
[20122 42 0 login up ]
[20121 41 0 login up ]
[20120 40 0 login up ]
[20119 30 0 login up ]
[20118 29 0 login down]
452, aux-number A450A0901, # peers 4, logout 0, down 0
peer-tag dn-tag rna login/logout up/down
[20127 45 0 login up ]
[20126 44 0 login up ]
[20125 43 0 login up ]
[20124 31 0 login up ]
453, aux-number A450A0902, # peers 4, logout 0, down 0
peer-tag dn-tag rna login/logout up/down
[20131 48 0 login up ]
[20130 47 0 login up ]
[20129 46 0 login up ]
[20128 32 0 login up ]
477, aux-number A450A0903, # peers 1, logout 0, down 0
peer-tag dn-tag rna login/logout up/down
[20132 499 0 login up ]
preference: 0
preference (sec): 7
timeout: 3, 3, 3, 3
max timeout : 10
hops: 4
next-to-pick: 1
E.164 register: yes
auto logout: no
stat collect: no
Group 2
type: sequential
pilot number: 601, peer-tag 20098
list of numbers:
123, aux-number A601A0200, # peers 1, logout 0, down 0
peer-tag dn-tag rna login/logout up/down
[20097 56 0 login up ]
622, aux-number A601A0201, # peers 3, logout 0, down 0
peer-tag dn-tag rna login/logout up/down
[20101 112 0 login up ]
[20100 111 0 login up ]
[20099 110 0 login up ]
623, aux-number A601A0202, # peers 3, logout 0, down 0
peer-tag dn-tag rna login/logout up/down
[20104 122 0 login up ]
[20103 121 0 login up ]
[20102 120 0 login up ]
*, aux-number A601A0203, # peers 1, logout 0, down 1
peer-tag dn-tag rna login/logout up/down
[20105 0 0 - down]
*, aux-number A601A0204, # peers 1, logout 0, down 1

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peer-tag dn-tag rna login/logout up/down

[20106 0 0 - down]
final number: 5255348
preference: 0
preference (sec): 9
timeout: 5, 5, 5, 5, 5
max timeout : 40
fwd-final: orig-phone
E.164 register: yes
auto logout: no
stat collect: no
Group 3
type: longest-idle
pilot number: 100, peer-tag 20142
list of numbers:
101, aux-number A100A9700, # peers 3, logout 0, down 3
on-hook time stamp 7616, off-hook agents=0
peer-tag dn-tag rna login/logout up/down
[20141 132 0 login down]
[20140 131 0 login down]
[20139 130 0 login down]
*, aux-number A100A9701, # peers 1, logout 0, down 1
on-hook time stamp 7616, off-hook agents=0
peer-tag dn-tag rna login/logout up/down
[20143 0 0 - down]
102, aux-number A100A9702, # peers 2, logout 0, down 2
on-hook time stamp 7616, off-hook agents=0
peer-tag dn-tag rna login/logout up/down
[20145 142 0 login down]
[20144 141 0 login down]
all agents down!
preference: 0
preference (sec): 7
timeout: 100, 100, 100
hops: 0
E.164 register: yes
auto logout: no
stat collect: no

SIP: Configuring Hunt Groups

To redirect calls for a specific number (hunt-group pilot number) to a defined group of directory
numbers on SIP phones, perform the following steps.

Prerequisites
Directory numbers to be added to a hunt group must be configured in Cisco Unified CME. For
configuration information, see “SIP: Creating Directory Numbers” on page 162.

Restrictions
• SIP-to-H.323 calls are not supported.
• If call forward is configured for a hunt group member, call forward is ignored by the hunt group.
• Forwarding or transferring to a voice hunt group is not supported.

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• Voice-class with codec list can be configured under voice register pool, and more than one list
member will not be supported for B2BUA call.
• Caller ID update is not supported for supplementary services.
• 100 voice hunt groups is the maximum number of hunt group supported.
• Voice hunt groups are subject to max-redirect restriction.
• A pilot dial peer cannot be used as a voice hunt group and a hunt group at the same time.
• If call-waiting is enabled (the default), parallel hunt groups support multiple calls up to the limit of
call-waiting calls supported by the particular SIP phone model. If call waiting is disabled, parallel
hunt groups support only one call at a time in the ringing state. Phones that fail to connect must
return to the on-hook state before they can receive other calls.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice hunt-group hunt-tag [longest-idle | parallel | peer | sequential]
4. pilot number [secondary number]
5. list dn-number, dn-number[, dn-number...]
6. final final-number
7. preference preference-order [secondary secondary-order]
8. hops number
9. timeout seconds
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice hunt-group hunt-tag [longest-idle | Enters voice hunt-group configuration mode to define a
parallel | peer | sequential] hunt group.
• hunt-tag—Unique sequence number of the hunt group
Example: to be configured. Range is 1 to 100.
Router(config)# voice hunt-group 1 longest-idle
• longest idle—Hunt group in which calls go to the
directory number that has been idle for the longest time.
• parallel—Hunt group in which calls simultaneously
ring multiple phones.
• peer—Hunt group in which the first directory number
is selected round-robin from the list.
• sequential—Hunt group in which directory numbers
ring in the order in which they are listed, left to right.
• To change the hunt-group type, remove the existing
hunt group first by using the no form of the command;
then, recreate the group.
Step 4 pilot number [secondary number] Defines the telephone number that callers dial to reach a
voice hunt group.
Example: • number—String of up to 16 characters that represents
Router(config-voice-hunt-group)# pilot number an E.164 telephone number.
8100
• Number string may contain alphabetic characters when
the number is to be dialed only by the
Cisco Unified CME router, as with an intercom
number, and not from telephone keypads.
• secondary number—(Optional) Keyword and
argument combination defines the number that follows
as an additional pilot number for the voice hunt group.
• Secondary numbers can contain wild cards. A wildcard
is a period (.), which matches any entered digit.
Step 5 list directory-number, directory-number Creates a list of extensions that are members of a voice hunt
[,directory-number...] group. To remove a list from a router configuration, use the
no form of this command.
Example: • directory-numbers—List of extensions to be added as
Router(config-voice-hunt-group)# list 8000, members to the voice hunt group. Separate the
8010, 8020, 8030
extensions with commas.
• Add or delete all extensions in a hunt-group list at one
time. You cannot add or delete a single number in an
existing list.
• There must be from 2 to 10 extensions in the
hunt-group list, and each number must be a primary or
secondary number.
• Any number in the list cannot be a pilot number of a
parallel hunt group.

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Command or Action Purpose

Step 6 final directory-number Defines the last extension in a voice hunt group.
• If a final number in one hunt group is configured as a
Example: pilot number of another hunt group, the pilot number of
Router(config-voice-hunt-group)# final 8888 the first hunt group cannot be configured as a final
number in any other hunt group.
Step 7 preference preference-order [secondary Sets the preference order for the directory number
secondary-order] associated with a voice hunt-group pilot number.
Note We recommend that the parallel hunt-group pilot
Example: number be unique in the system. Parallel hunt
Router(config-voice-hunt-group)# preference 6 groups may not work if there are more than one
partial or exact dial-peer match. For example, if the
pilot number is “8000” and there is another dial peer
that matches “8…”. If multiple matches cannot be
avoided, give parallel hunt groups the highest
priority to run by assigning a lower preference to the
other dial peers. Note that 10 is the lowest
preference value. By default, dial peers created by
parallel hunt groups have a preference of 0.

• preference-order—Range is 0 to 8, where 0 is the

highest preference, and 8 is the lowest preference.
Default is 0.
• secondary secondary-order—(Optional) Keyword and
argument combination is used to set the preference
order for the secondary pilot number. Range is 0 to 10,
where 0 is the highest preference and 10 is the lowest
preference. Default is 9.
Step 8 hops number For configuring a peer or longest-idle voice hunt group
only. Defines the number of times that a call can hop to the
next number in a peer or longest-idle voice hunt group
Example:
Router(config-voice-hunt-group)# hops 2
before the call proceeds to the final number.
• number—Number of hops. Range is 2 to 10, and the
value must be less than or equal to the number of
extensions specified by the list command.
• Default is the same number as there are destinations
defined under the list command.
Step 9 timeout seconds Defines the number of seconds after which a call that is not
answered is redirected to the next directory number in a
voice hunt-group list.
Example:
Router(config-voice-hunt-group)# timeout 100 • Default: 180 seconds.
Step 10 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-voice-hunt-group)# end

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SCCP: Configuring Night Service

This procedure defines night-service hours, an optional night-service code, the ephone-dns that trigger
the notification process, and the ephones that will receive notification.

Restrictions
In Cisco Unified CME 4.0 and later, silent ringing, configured on the phone by using the s keyword with
the button command, is suppressed when used with the night service feature. Silent ringing is
overridden and the phone audibly rings during designated night-service periods.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. night-service day day start-time stop-time
5. night-service date month date start-time stop-time
6. night-service everyday start-time stop-time
7. night-service weekday start-time stop-time
8. night-service weekend start-time stop-time
9. night-service code digit-string
10. exit
11. ephone-dn dn-tag
12. night-service bell
13. exit
14. ephone phone-tag
15. night-service bell
16. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 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 5 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 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 a
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 6 night-service everyday start-time stop-time Defines a recurring night-service time period to be effective
everyday.
Example: • start-time stop-time—Beginning and ending times for
Router(config-telephony)# night-service night service, in an HH:MM format using a 24-hour
everyday 1200 1300 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, “19:00 07:00” means “from 7 p.m.
to 7 a.m. the next morning.” The value 24:00 is not
valid. If 00:00 is entered as a stop time, it is changed to
23:59. If 00:00 is entered for both start time and stop
time, the night service feature will be activated for the
entire 24-hour period.

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Command or Action Purpose

Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 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 5 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 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 a
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 6 night-service everyday start-time stop-time Defines a recurring night-service time period to be effective
everyday.
Example: • start-time stop-time—Beginning and ending times for
Router(config-telephony)# night-service night service, in an HH:MM format using a 24-hour
everyday 1200 1300 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, “19:00 07:00” means “from 7 p.m.
to 7 a.m. the next morning.” The value 24:00 is not
valid. If 00:00 is entered as a stop time, it is changed to
23:59. If 00:00 is entered for both start time and stop
time, the night service feature will be activated for the
entire 24-hour period.

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Command or Action Purpose

Step 7 night-service weekday start-time stop-time Defines a recurring night-service time period to be effective
on all weekdays.
Example: • start-time stop-time—Beginning and ending times for
Router(config-telephony)# night-service weekday night service, in an HH:MM format using a 24-hour
1700 0700 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, “19:00 07:00” means “from 7 p.m.
to 7 a.m. the next morning.” The value 24:00 is not
valid. If 00:00 is entered as a stop time, it is changed to
23:59. If 00:00 is entered for both start time and stop
time, the night service feature will be activated for the
entire 24-hour period.
Step 8 night-service weekend start-time stop-time Defines a recurring night-service time period to be effective
on all weekend days. Weekend is defined as Saturday and
Sunday.
Example:
Router(config-telephony)# night-service weekend • start-time stop-time—Beginning and ending times for
00:00 00:00 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, “19:00 07:00” means “from 7 p.m.
to 7 a.m. the next morning.” The value 24:00 is not
valid. If 00:00 is entered as a stop time, it is changed to
23:59. If 00:00 is entered for both start time and stop
time, the night service feature will be activated for the
entire 24-hour period.
Step 9 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 assigned to night service in the system. The
Example:
Router(config-telephony)# night-service code
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 (*).
Step 10 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 11 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 12 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.

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Command or Action Purpose

Step 13 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 14 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 15 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 16 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

SCCP: Verifying Night Service

Step 1 Use the show running-config command to verify the night-service parameters, which are listed in the
telephony-service portion of the output, or use the show telephony-service command to display the
same parameters.
Router# show running-config

telephony-service
fxo hook-flash
load 7910 P00403020214
load 7960-7940 P00303020214
max-ephones 48
max-dn 288
ip source-address 10.50.50.1 port 2000
application segway0
caller-id block code *321
create cnf-files version-stamp 7960 Mar 07 2003 11:19:18
voicemail 79000
max-conferences 8
call-forward pattern .....
moh minuet.wav
date-format yy-mm-dd
transfer-system full-consult
transfer-pattern .....
secondary-dialtone 9
night-service code *1234
night-service day Tue 00:00 23:00
night-service day Wed 01:00 23:59
!
!

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Router# show telephony-service

CONFIG (Version=4.0(0))
=====================
Version 4.0(0)
Cisco Unified CallManager Express
For on-line documentation please see:
www.cisco.com/en/US/products/sw/voicesw/tsd_products_support_category_home.html

ip source-address 10.103.3.201 port 2000

load 7910 P00403020214
load 7961 TERM41.7-0-1-1
load 7961GE TERM41.7-0-1-1
load 7960-7940 P00307020300
max-ephones 100
max-dn 500
max-conferences 8 gain -6
dspfarm units 2
dspfarm transcode sessions 4
dspfarm 1 MTP00059a3d7441
dspfarm 2
hunt-group report delay 1 hours
Number of hunt-group configured: 14
hunt-group logout DND
max-redirect 20
voicemail 7189
cnf-file location: system:
cnf-file option: PER-PHONE-TYPE
network-locale[0] US (This is the default network locale for this box)
user-locale[0] US (This is the default user locale for this box)
moh flash:music-on-hold.au
time-format 12
date-format mm-dd-yy
timezone 0 Greenwich Standard Time
secondary-dialtone 9
call-forward pattern .T
transfer-pattern 92......
transfer-pattern 91..........
transfer-pattern .T
after-hours block pattern 1 91900 7-24
after-hours block pattern 2 9976 7-24
after-hours block pattern 4 91...976.... 7-24
night-service date Jan 1 00:00 23:59
night-service day Mon 17:00 07:00
night-service day Wed 17:00 07:00
keepalive 30
timeout interdigit 10
timeout busy 10
timeout ringing 100
caller-id name-only: enable
system message XYZ Company
web admin system name xyz password xxxx
web admin customer name Customer
edit DN through Web: enabled.
edit TIME through web: enabled.
Log (table parameters):
max-size: 150
retain-timer: 15
create cnf-files version-stamp Jan 01 2002 00:00:00
transfer-system full-consult
multicast moh 239.10.10.1 port 2000
fxo hook-flash
local directory service: enabled.

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Step 2 Use the show running-config command to verify that the correct ephone-dns and ephones are
configured with the night-service bell command. You can also use the show telephony-service
ephone-dn and show telephony-service ephone commands to display these parameters.
Router# show running-config

ephone-dn 24 dual-line
number 2548
description FrontDesk
night-service bell

ephone 1
mac-address 110F.80C0.FE0B
type 7960 addon 1 7914
no dnd feature-ring
keep-conference
button 1f40 2f41 3f42 4:30
button 7m20 8m21 9m22 10m23
button 11m24 12m25 13m26
night-service bell

SCCP: Configuring Overlaid Ephone-dns

To create ephone-dns, then assign multiple ephone-dns to a single phone button by using the o or c
keyword with the button command, perform the following steps.

Restrictions
• Call waiting is disabled when you configure ephone-dn overlays using the o keyword with the
button command. To enable call waiting, you must configure ephone-dn overlays using the
c keyword with the button command.
• Rollover of overlay calls to another phone button by using the x keyword with the button command
only works to expand coverage if the overlay button is configured with the o keyword in the button
command. Overlay buttons with call waiting that use the c keyword in the button command are not
eligible for overlay rollover.
• In Cisco Unified CME 4.0(3), the Cisco Unified IP Phone 7931G cannot support overlays that
contain ephone-dn configured for dual-line mode.
• The primary ephone-dn on each phone in a shared-line overlay set should be an ephone-dn that is
unique to the phone to guarantee that the phone will have a line available for outgoing calls, and to
ensure that the phone user can obtain dial-tone even when there are no idle lines available in the rest
of the shared-line overlay set. Use a unique ephone-dn in this manner to provide for a unique calling
party identity on outbound calls made by the phone so that the called user can see which specific
phone is calling.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]

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4. number number
5. preference preference-value
6. huntstop
or
no huntstop
7. call-forward noan
8. call-forward busy
9. huntstop channel
10. exit
11. ephone phone-tag
12. mac-address mac-address
13. button button-number{o | c}dn-tag,dn-tag[,dn-tag...] button-number{x}overlay-button-number
14. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn phone-tag [dual-line] Enters ephone-dn configuration mode to create an extension
(ephone-dn) for a Cisco Unified IP phone line.
Example: • For shared-line overlay set: Primary ephone-dn on a
Router(config)# ephone-dn 10 dual-line phone should be an ephone-dn that is unique to the
phone.
Step 4 number number Associates a telephone or extension number with the
ephone-dn.
Example:
Router(config-ephone-dn)# number 1001
Step 5 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). Type
Router(config-ephone-dn)# preference 1 ? for a range of numeric options, where 0 is the highest
preference. Default: 0.

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Command or Action Purpose

Step 6 no huntstop Explicitly enables call hunting behavior for a directory
or number.
huntstop • Set this command on all ephone-dns in the overlay set
except the final instance.
Example: • Required to allow call hunting allow call hunting across
Router(config-ephone-dn)# no huntstop multiple numbers on the same line button on an IP
or phone.
or
Example:
Router(config-ephone-dn)# huntstop
Disables call hunting behavior for a directory number.
• Set this command on the last ephone-dn within a
overlay set.
• Required to limit the call hunting to an overlay set.
Step 7 call-forward noan (Optional) Forwards incoming unanswered call to next line
in the overlay set.
• Set this command on all ephone-dns in the overlay set.
Step 8 call-forward busy (Optional) Forwards incoming call if line is busy.
• Set this command on the last ephone-dn in the overlay
set only.
Step 9 huntstop channel Only for dual-line ephone-dns in overlay set; 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 • Reserves the second channel for outgoing calls, such as
a consultation call to be placed during a call transfer
attempt, or for conferencing
Step 10 exit Exits ephone-dn configuration mode

Example:
Router(config-ephone-dn)# exit
Step 11 ephone phone-tag Enters ephone 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 12 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 13 button Creates a set of ephone-dns overlaid on a single button.
button-number{o | c}dn-tag,dn-tag[,dn-tag...]
button-number{x}overlay-button-number • o—Overlay button. Multiple ephone-dns share this
button. A maximum of 25 ephone-dns can be specified
for a single button, separated by commas.
Example:
Router(config-ephone)# button 1o15,16,17,18,19 • c—Overlay button with call-waiting. Multiple
2c20,21,22 3x1 4x1 ephone-dns share this button. A maximum of
25 ephone-dns can be specified for a single button,
separated by commas.
• x—Separator that creates a rollover button for an
overlay button that was defined using the o keyword.
When the overlay button specified in this command is
occupied by an active call, a second call to one of its
ephone-dns will be presented on this button.
• dn-tag—Unique identifier previously defined with the
ephone-dn command for the ephone-dn to be added to
this overlay set.
• overlay-button-number—Number of the overlay button
that should overflow to this button. Note that the button
must have been defined using the o keyword and not the
c keyword.
Note For other keywords, see the button command in the
Cisco Unified Communications Manager Express
Command Reference.
Step 14 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

SCCP: Verifying Overlaid Ephone-dns

Step 1 Use the show running-config command or the show telephony-service ephone command to view
button assignments.
Router# show running-config

ephone 5
description Cashier1
mac-address 0117.FBC6.1985
type 7960
button 1o4,5,6,200,201,202,203,204,205,206 2x1 3x1

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Step 2 Use the show ephone overlay command to display the configuration and current status of registered
overlay ephone-dns.
Router# show ephone overlay

ephone-1 Mac:0007.0EA6.353A TCP socket:[1] activeLine:0 REGISTERED

mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:10.2.225.205 52486 Telecaster 7960 keepalive 2771 max_line 6
button 1: dn 11 number 60011 CH1 IDLE overlay
button 2: dn 17 number 60017 CH1 IDLE overlay
button 3: dn 24 number 60024 CH1 IDLE overlay
button 4: dn 30 number 60030 CH1 IDLE overlay
button 5: dn 36 number 60036 CH1 IDLE CH2 IDLE overlay
button 6: dn 39 number 60039 CH1 IDLE CH2 IDLE overlay
overlay 1: 11(60011) 12(60012) 13(60013) 14(60014) 15(60015) 16(60016)
overlay 2: 17(60017) 18(60018) 19(60019) 20(60020) 21(60021) 22(60022)
overlay 3: 23(60023) 24(60024) 25(60025) 26(60026) 27(60027) 28(60028)
overlay 4: 29(60029) 30(60030) 31(60031) 32(60032) 33(60033) 34(60034)
overlay 5: 35(60035) 36(60036) 37(60037)
overlay 6: 38(60038) 39(60039) 40(60040)

Step 3 Use the show dialplan number command to display 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.

Configuration Examples for Call Coverage Features

This section contains the following configuration examples:
• Call Hunt: Examples, page 619
• Call Pickup: Examples, page 621
• Call-Waiting Beep: Example, page 622
• Call-Waiting Ring: Example, page 622
• Hunt Group: Examples, page 622
• Night Service: Examples, page 625
• Overlaid Ephone-dns Examples, page 626

Call Hunt: Examples

This section contains the following examples:
• Ephone-dn Dial-Peer Preference: Example, page 620
• Huntstop Disabled: Example, page 620
• Channel Huntstop: Example, page 621
• SIP Call Hunt: Example, page 621

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Ephone-dn Dial-Peer Preference: 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
number 3001
preference 2

Huntstop Disabled: Example

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.
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

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Channel Huntstop: Example

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

SIP Call Hunt: Example

The following example shows a typical configuration in which huntstop is required. The huntstop
command is enabled and prevents calls to extension 5001 from being rerouted to the on-net H.323 dial
peer for 5... when extension 5001 is busy (three periods are used as wild cards).
voice register dn 1
number 5001
huntstop

voice register pool 4

number 1 dn 1
id-mac 0030.94c3.8724

dial-peer voice 5000 voip

destination-pattern 5...
session target ipv4:192.168.17.225
session protocol sipv2

Call Pickup: Examples

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

The following example globally disables directed call pickup and changes the action of the PickUp soft
key to perform local group call pickup rather than directed call pickup.
telephony-service
no service directed-pickup

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Call-Waiting Beep: Example

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

Call-Waiting Ring: Example

The following example specifies that a short ring will indicate a call is waiting for extension 5533.
ephone-dn 20
number 5533
call-waiting ring

Hunt Group: Examples

This section contains the following examples:
• Sequential Ephone Hunt Group: Example, page 622
• Peer Ephone Hunt Group: Example, page 623
• Longest-Idle Ephone Hunt Group: Example, page 623
• Longest-Idle Ephone Hunt Group Using From-Ring Option: Example, page 623
• Logout Display: Example, page 624
• Dynamic Membership: Example, page 624
• Agent Status Control: Example, page 624
• Automatic Agent Not-Ready: Example, page 625

Sequential Ephone Hunt Group: 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 2 sequential
pilot 600
list 621, *, 623
final 5255348
max-timeout 10
timeout 20, 20, 20
fwd-final orig-phone

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Peer Ephone Hunt Group: Example

The following example defines peer ephone hunt group 10 with a pilot number 450, a final number 500,
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 10 peer
pilot 450
list 451, 452, 453, 477
final 500
max-timeout 10
timeout 3, 3, 3, 3

Longest-Idle Ephone Hunt Group: Example

The following example defines longest-idle ephone hunt group 1 with a pilot number 7501 and 11
numbers in the list. After a call is redirected five times, it is redirected to the final number.
ephone-hunt 1 longest-idle
pilot 7501
list 7001, 7002, 7023, 7028, 7045, 7062, 7067, 7072, 7079, 7085, 7099
final 8000
preference 1
hops 5
timeout 20
no-reg

Longest-Idle Ephone Hunt Group Using From-Ring Option: 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. Because the from-ring command is used, on-hook time stamps
will be recorded when calls ring extensions and when calls are answered. After a call is redirected six
times (makes six hops), it is redirected to the final number, 8000. The max-redirect command is used
to increase the number of redirects that are allowed because the number of hops (six) is larger than the
default number of redirects that are allowed in the system (five).
ephone-hunt 1 longest-idle
pilot 7501
list 7001, 7002, 7023, 7028, 7045, 7062, 7067, 7072, 7079, 7085, 7099
final 8000
from-ring
preference 1
hops 6
timeout 20

telephony-service
max-redirect 8

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Logout Display: Example

In the following example, the description is set to “Marketing Hunt Group.” This information will be
shown in the configuration output and also on the display of IP phones that are receiving calls from this
hunt group. The display-logout message is set to “Night Service,” which will be displayed on IP phones
that are members of the hunt group when all the members are logged out.
ephone-hunt 17 sequential
pilot 3000
list 3011, 3021, 3031
timeout 10
final 7600
description Marketing Hunt Group
display-logout Night Service

Dynamic Membership: Example

The following example creates four ephone-dns and a hunt group that includes the first ephone-dn and
two wildcard slots. The last three ephone-dns are enabled for group hunt dynamic membership. Each of
them can join and leave the hunt group whenever one of the wildcard slots is available. Standard FACs
have been enabled, and the agents use standard FACs to join (*3) and leave (#3) the hunt group. You
can also use the fac command to create custom FACs for these actions if you prefer.
ephone-dn 22
number 4566

ephone-dn 24
number 4568
ephone-hunt login

ephone-dn 25
number 4569
ephone-hunt login

ephone-dn 26
number 4570
ephone-hunt login

ephone-hunt 1 peer
list 4566,*,*
timeout 10
final 7777

telephony-service
fac standard

Agent Status Control: Example

The following example sets up a peer ephone hunt group. It also establishes the appearance and order of
soft keys for phones that are configured with ephone-template 7. These phones will have the HLog key
available when they are idle, when they have seized a line, or when they are connected to a call. Phones
without soft keys can use the standard HLog codes to toggle ready and not-ready status.
ephone-hunt 10 peer
pilot 450
list 451, 452, 453, 477
final 500
timeout 45

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telephony-service
hunt-group logout HLog
fac standard

ephone-template 7
softkeys connected Endcall Hold Transfer HLog
softkeys idle Newcall Redial Pickup Cfwdall HLog
softkeys seized Endcall Redial Pickup Cfwdall HLog

Automatic Agent Not-Ready: Example

The following example enables automatic status change to not-ready after one unanswered hunt group
call (the default) for both dynamic and static hunt group members (the default). It also specifies that the
phones which are automatically put into the not-ready status should only be blocked from further
hunt-group calls and that they should be able to receive calls that directly dial their extensions.
ephone-hunt 3 peer
pilot 4200
list 1001, 1002, 1003
timeout 10
auto logout
final 4500

telephony-service
hunt-group logout HLog

The following example enables automatic status change to not-ready after two unanswered hunt group
calls for any ephone-dn that dynamically logs in to the hunt group using the wildcard slot in the hunt
group list. Phones that are automatically placed in the not-ready status when they do not answer two
hunt-group calls are also placed into DND status (they will also not accept directly dialed calls).
ephone-hunt 3 peer
pilot 4200
list 1001, 1002, *
timeout 10
auto logout 2 dynamic
final 4500

telephony-service
hunt-group logout DND

Night Service: Examples

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. Incoming calls to extension 1000 during the night-service period 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.

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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
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

Overlaid Ephone-dns Examples

This section contains the following examples:
• Overlaid Ephone-dn: Example, page 627
• Overlaid Dual-Line Ephone-dn: Example, page 627
• Shared-line Overlaid Ephone-dns: Example, page 628
• Overlaid Ephone-dn with Call Waiting: Example, page 629
• Overlaid Ephone-dns with Rollover Buttons: Example, page 630
• Called Directory Name Display for Overlaid Ephone-dns: Example, page 630
• Called Ephone-dn Name Display for Overlaid Ephone-dns: Example, page 632

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Overlaid Ephone-dn: 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

Overlaid Dual-Line Ephone-dn: Example

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

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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

Shared-line Overlaid Ephone-dns: Example

The following is an example of a unique ephone-dn as the primary dn in a simple shared-line overlay
configuration. The no huntstop command is configured for all the ephone-dns except ephone-dn 12, the
last one in the overlay set. Because the ephone-dns are dual-line dns, the huntstop-channel command
is also configured to ensure that the second channel remains free for outgoing calls and for conferencing.
ephone-dn 1 dual-line
number 101
huntstop-channel
!
ephone-dn 2 dual-line
number 102
huntstop-channel
!
ephone-dn 10 dual-line
number 201
no huntstop
huntstop-channel
!
ephone-dn 11 dual-line
number 201
no huntstop
huntstop-channel
!
ephone-dn 12 dual-line
number 201

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huntstop-channel
!
!The following ephone configuration includes (unique) ephone-dn 1 as the primary line in a
shared-line overlay

ephone 1
mac-address 1111.1111.1111
button 1o1,10,11,12
!
!The next ephone configuration includes (unique) ephone-dn 2 as the primary line in
another shared-line overlay
!
ephone 2
mac-address 2222.2222.2222
button 1o2,10,11,12

Overlaid Ephone-dn with Call Waiting: 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 noan 7000 timeout 30

ephone-dn 11 dual-line
number 1111
preference 1
no huntstop
huntstop channel
call-forward noan 7000 timeout 30

ephone-dn 12 dual-line
number 1111
preference 2
huntstop channel
call-forward noan 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

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Overlaid Ephone-dns with Rollover Buttons: Example

The following example configures a “3x3” shared-line setup for three ephones and nine shared lines
(ephone-dns 20 to 28). Each ephone has a unique ephone-dn for each of its three buttons (ephone-dns
11 to 13 on ephone 1, ephone-dns 14 to 16 on ephone 2, and ephone-dns 17 to 19 on ephone 3). The rest
of the ephone-dns are shared among the three phones. Three phones with three buttons each can take
nine calls. The overflow buttons provide the ability for an incoming call to ring on the first available
button on each phone.
ephone-dn 11
number 2011

ephone-dn 12
number 2012

ephone-dn 13
number 2013

ephone-dn 14
number 2014
.
.
.
ephone-dn 28
number 2028

ephone 1
button 1o11,12,13,20,21,22,23,24,25,26,27,28 2x1 3x1

ephone 2
button 1o14,15,16,20,21,22,23,24,25,26,27,28 2x1 3x1

ephone 3
button 1o17,18,19,20,21,22,23,24,25,26,27,28 2x1 3x1

Called Directory Name Display for Overlaid Ephone-dns: Example

The following example demonstrates the display of a directory name for a called ephone-dn that is part
of an overlaid ephone-dn set. For configuration information, see “Configuring Directory Services” on
page 689.
This configuration of overlaid ephone-dns uses wildcards in the secondary numbers for the ephone-dns.
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 5550101 rings on button 1 on phone 1 to phone 3, “doctor1” is displayed on all three phones.

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telephony-service
service dnis dir-lookup

directory entry 1 5550101 name doctor1

directory entry 2 5550102 name doctor2
directory entry 3 5550103 name doctor3

directory entry 4 5550110 name doctor4

directory entry 5 5550111 name doctor5
directory entry 6 5550112 name doctor6

directory entry 7 5550120 name doctor7

directory entry 8 5550121 name doctor8
directory entry 9 5550122 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

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 Unified 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, see the “Hunt Groups” section on page 569. 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

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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

Called Ephone-dn Name Display for Overlaid Ephone-dns: Example

The following example demonstrates the display of the name assigned to the called ephone-dn using the
name command. For information about configuring this feature, see “Configuring Directory Services”
on page 689.
In this example, three phones have button 1 assigned to pick up three shared 800 numbers for three
different catalogs.
The default display for the phones is the number of the first ephone-dn listed in the overlay set
(18005550100). A call is made to the first ephone-dn (18005550100), and the caller ID (for example,
4085550123) is visible on all phones. The user for phone 1 answers the call. The caller ID (4085550123)
remains visible on phone 1, and the displays on phone 2 and phone 3 return to the default display
(18005550100). A call to the second ephone-dn (18005550101) is made. The default display on phone
2 and phone 3 is replaced with the called ephone-dn's name (catalog1) and number (18005550101).
telephony-service
service dnis overlay

ephone-dn 1
number 18005550100

ephone-dn 2
name catalog1
number 18005550101

ephone-dn 3
name catalog2
number 18005550102

ephone-dn 4
name catalog3
number 18005550103

ephone 1
button 1o1,2,3,4

ephone 2
button 1o1,2,3,4

ephone 3
button 1o1,2,3,4

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Where to Go Next

Where to Go Next
Dial-Peer Call Hunt and Hunt Groups
Dial peers other than ephone-dn dial peers can be directly configured as hunt groups or rotary groups,
in which multiple dial peers can match incoming calls. (These are not the same as Cisco Unified CME
ephone hunt groups.) For more information, see the “Hunt Groups” section of the “Dial Peers Features
and Configuration” chapter of Dial Peer Configuration on Voice Gateway Routers.

Called-Name Display
This feature allows you to specify that the name of the called party, rather than the number, should be
displayed for incoming calls. This feature is very helpful for agents answering calls for multiple
ephone-dns that appear on a single line button in an ephone-dn overlay set. For more information, see
“Configuring Directory Services” on page 689.

Soft Key Control

If the hunt-group logout command is used with the HLog keyword, the HLog soft key appears on
phones during the idle, connected, and seized call states. The HLog soft key is used to toggle an agent
from the ready to not-ready status or from the not-ready to ready status. To move or remove the HLog
soft key on one or more phones, create and apply an ephone template that contains the appropriate
softkeys commands.
For more information, see “Customizing Soft Keys” on page 829.

Feature Access Codes (FACs)

Dynamic membership allows agents at authorized ephones to join or leave a hunt group using a feature
access code (FAC) after standard or custom FACs are enabled.
In Cisco Unified CME 4.0 and later versions, you can activate call pickup using a feature access code
(FAC) instead of a soft key when standard or custom FACs have been enabled for your system. The
following are the standard FACs for call pickup:
• Pickup group—Dial the FAC and a pickup group number to pick up a ringing call in a different
pickup group than yours. Standard FAC is **4.
• Pickup local—Dial the FAC to pick up a ringing call in your pickup group. Standard FAC is **3.
• Pickup direct—Dial the FAC and the extension number to pick up a ringing call at any extension.
Standard FAC is **5.
For more information about FACs, see “Configuring Feature Access Codes” on page 729.

Controlling Use of the Pickup Soft Keys

To block the functioning of the group pickup (GPickUp) or local pickup (Pickup) soft key without
removing the key display, create and apply an ephone template that contains the features blocked
command. For more information, see Configuring Call Blocking, page 467.
To remove the group pickup (GPickUp) or local pickup (Pickup) soft key from one or more phones,
create and apply an ephone template that contains the appropriate softkeys command. For more
information, see “Customizing Soft Keys” on page 829.

Ephone-dn Templates
The ephone-hunt login command authorizes an ephone-dn to dynamically join and leave an ephone
hunt group. It can be included in an ephone-dn template that is applied to one or more individual
ephone-dns. For more information, see “Creating Templates” on page 881.

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Additional References

Ephone Hunt Group Statistics Reports

Several different types of statistics can help you track whether your current ephone hunt groups are
meeting your call-coverage needs. These statistics can be displayed on-screen or written to files.
For more information, see the “Cisco Unified CME Basic Automatic Call Distribution and
Auto-Attendant Service” chapter in Cisco Unified CME B-ACD and Tcl Call-Handling Applications.

Do Not Disturb
The Do Not Disturb (DND) feature can be used as an alternative to the HLog function for preventing
incoming calls from ringing on a phone. The difference is that HLog prevents only hunt group calls from
ringing, while DND prevents all calls from ringing. For more information, see “Configuring Do Not
Disturb” on page 709.

Automatic Call Forwarding During Night-Service

To have an ephone-dn forward all its calls automatically during night-service hours, use the
call-forward night-service command. For more information, see the “SCCP: Enabling Call Forwarding
for a Directory Number” section on page 523.

Ephone Templates
The night-service bell command specifies that a phone will receive night-service notification when calls
are received at ephone-dns configured as night-service ephone-dns. This command can be included in
an ephone template that is applied to one or more individual ephones.
For more information, see “Creating Templates” on page 881.

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

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Additional References

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Call Coverage Features

Feature Information for Call Coverage Features

Table 31 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified Communications Manager Express and Cisco IOS Software Version Compatibility
Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 31 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 31 Feature Information for Call Coverage

Cisco Unified CME

Feature Name Version Modification
Call Hunt 3.4 Added support for configuring call hunt features on SIP IP
phones connected directly to Cisco Unified CME.
3.0 • Preference for secondary numbers was introduced.
• Channel huntstop was introduced.
1.0 • Ephone-dn dial-peer preference was introduced.
• Huntstop was introduced.
Call Pickup 4.0 • The ability to globally disable directed call pickup was
introduced.
• Feature access codes for call pickup were introduced.
• The ability to block call pickup on individual phones
was introduced.
3.2 The ability to remove or rearrange soft keys on individual
phones was introduced.
3.0 Call pickup groups were introduced.
Call Waiting 3.4 Added support for configuring call waiting for SIP phones
directly connected to Cisco Unified CME.
Callback Busy Subscriber 3.0 Callback busy subscriber was introduced.

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Table 31 Feature Information for Call Coverage

Cisco Unified CME

Feature Name Version Modification
Hunt Groups 4.0 Added support for the following on IP phones running
SCCP:
• Maximum number of hunt groups in a system was
increased from 20 to 100 and maximum number of
agents in a hunt group was increased from 10 to 20.
• Maximum number of hops automatically adjusts to the
number of agents.
• A description can be added to phone displays and
configuration output to provide hunt group information
associated with ringing and answered calls.
• A configurable message can be displayed on agent
phones when all agents are in the not-ready status to
advise the destination to which calls are being
forwarded or other useful information.
• No-answer timeouts can be set individually for each
ephone-dn in the list and a cumulative no-answer
timeout can be set for all ephone-dns.
• Automatic logout trigger criterion was changed from
exceeding the specified timeout to exceeding the
specified number of calls. The name of this feature was
changed from automatic logout to automatic agent
status not-ready.
• Dynamic hunt group membership is introduced. Agents
can join and leave hunt groups whenever a wildcard slot
is available.
• Agent status control using an HLog soft key or feature
access code (FAC) is introduced. Agents can put their
lines into not-ready state to temporarily block hunt
group calls without relinquishing their slots in group.
• Calls can be blocked from agent phones that are not idle
or on hook.
• Calls that are not answered by the hunt group can be
returned to the party who transferred them into the hunt
group.
• Calls parked by hunt group agents can be returned to a
different entry point.
• (Sequential hunt groups only) Local calls to a hunt
group can be restricted so that they will not be
forwarded past the initial agent that is rung.
• (Longest-idle hunt groups only) A new command, the
from-ring command, specifies that on-hook time
stamps should be updated when a call rings an agent
and when a call is answered by an agent.

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Table 31 Feature Information for Call Coverage

Cisco Unified CME

Feature Name Version Modification
3.4 Added support for configuring hunt groups for SIP phones
directly connected to Cisco Unified CME.
3.2.1 • Maximum number of hunt groups in a system was
increased to 20.
• Automatic logout capability was introduced.
3.2 Longest-idle hunt groups were introduced.
3.1 Secondary pilot numbers were introduced.
3.0 Peer and sequential ephone hunt groups were introduced.
Night Service 4.0 The night-service everyday, night-service weekday, and
night-service weekend commands were introduced.
3.3 The behavior of the night-service code was changed.
Previously, using the night-service code at a phone either
enabled or disabled night service for the ephone-dns on that
phone. Now, using the night-service code at a phone enables
or disables night service for all night-service ephone-dns.
3.0 Night service was introduced.
Overlaid Ephone-dns 4.0 • The number of ephone-dns that can be overlaid on a
single button using the button command and the o or c
keyword was increased from 10 to 25.
• The ability to extend calls for overlaid ephone-dns to
other buttons (rollover buttons) on the same phone was
introduced. Rollover buttons are created by using the x
keyword with the button command.
• The number of waiting calls that can be displayed for
overlaid ephone-dns that have call waiting configured
has been increased to six for the following phone types:
Cisco Unified IP Phone 7940G, 7941G, 7941G-GE,
7960G, 7961G, 7961G-GE, 7970G, and 7971G-GE.
3.2.1 Call waiting for overlaid ephone-dns was introduced and the
c keyword was added to the button command.
3.0 Overlaid ephone-dns were introduced and the o keyword
was added to the button command.

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Last Updated: March 26, 2007

This chapter describes the caller-ID (CLID) blocking feature in Cisco Unified Communications
Manager Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Caller ID Blocking”
section on page 646.

Contents
• Restrictions for Caller ID Blocking, page 639
• Information about Caller ID Blocking, page 639
• How to Configure Caller ID Blocking, page 640
• Configuration Examples for Caller ID Blocking, page 644
• Additional References, page 644
• Feature Information for Caller ID Blocking, page 646

Restrictions for Caller ID Blocking

Caller ID blocking on outbound calls does not apply to PSTN calls 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 you to subscribe to their caller ID blocking service.

Information about Caller ID Blocking

To enable Caller ID Blocking, you should understand the following concept:
• Caller ID Blocking on Outbound Calls, page 640

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How to Configure Caller ID Blocking

Caller ID Blocking on Outbound Calls

Phone users can block caller-ID displays on calls from a particular ephone-dn, or you can selectively
choose to block the name or number on outbound calls from a particular dial peer.
The display of caller ID information for outgoing calls from a particular ephone-dn can be blocked on a
per-call basis, allowing users to maintain their privacy when necessary. The system administrator defines
a code for caller ID blocking in Cisco Unified CME. Users then dial the code before making any call on
which they do not want their number displayed on the called-party phone. The caller ID is sent, but its
presentation parameter is set to “restricted” so that the caller ID is not displayed.
Blocking CLID displays for local calls from a particular extension tells 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 and independently block the CLID
name or number on outbound VoIP calls. This configuration has the benefit of allowing caller-ID display
for local calls while preventing caller-ID display for external calls going over VoIP. This feature can be
used for PSTN calls that go out over ISDN.

How to Configure Caller ID Blocking

This section contains the following tasks:
• Blocking Caller ID For Local Calls From a Directory Number, page 640 (optional)
• Blocking Caller ID For All Outbound Calls, page 642 (optional)
• Verifying Caller ID Blocking, page 643 (optional)

Blocking Caller ID For Local Calls From a Directory Number

To define a code that phone users can dial to block caller ID display on all local calls from a particular
ephone-dn, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. caller-id block code code-string
5. exit
6. ephone-dn dn-tag
7. caller-id block
8. end

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How to Configure Caller ID Blocking

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 caller-id block code code-string (Optional) Defines a code that users can enter before making
calls on which the caller ID should not be displayed.
Example: • code-string—Digit string of up to 16 characters. The first
Router(config-telephony)# caller-id block character must be an asterisk (*).
code *1234
Step 5 exit Exits telephony-service configuration mode.

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

Example:
Router(config)# ephone-dn 3
Step 7 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 Unified IP phone.
• This command tells the far-end gateway device to block
display of calling-party information for the calls received
from this ephone-dn.
Step 8 end Returns to privileged EXEC mode.

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

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How to Configure Caller ID Blocking

Blocking Caller ID For All Outbound Calls

To block the CLID name or number on outbound VoIP calls from a particular dial peer, perform the
following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag [pots | voip]
4. clid strip
5. clid strip name
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 dial-peer voice tag [pots | voip] Enters dial-peer configuration mode.
Note You can configure caller-ID blocking on POTS dial
Example: peers if the POTS interface is ISDN. This feature is not
Router(config)# dial-peer voice 3 voip available on FXO/CAS lines.
Step 4 clid strip (Optional) Removes the calling-party number from the CLID
information being sent with VoIP calls.
Example:
Router(config-dial-peer)# clid strip
Step 5 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 6 end Returns to privileged EXEC mode.

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

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Verifying Caller ID Blocking

Step 1 Use the show running-config command to display caller ID blocking parameters, which may appear in
the telephony-service, ephone-dn, or dial-peer portions of the output.
Router# show running-config

dial-peer voice 450002 voip

translation-profile outgoing 457-456
destination-pattern 457
session target ipv4:10.43.31.81
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
clid strip
!
telephony-service
fxo hook-flash
load 7960-7940 P00305000600
load 7914 S00103020002
max-ephones 100
max-dn 500
ip source-address 10.115.34.131 port 2000
max-redirect 20
no service directed-pickup
timeouts ringing 10
system message XYZ Company
voicemail 7189
max-conferences 8 gain -6
moh music-on-hold.au
caller-id block code *1234
web admin system name cisco password cisco
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 92......
transfer-pattern 91..........
transfer-pattern 93......
transfer-pattern 94......
transfer-pattern 95......
transfer-pattern 96......
transfer-pattern 97......
transfer-pattern 98......
transfer-pattern .T
secondary-dialtone 9
after-hours block pattern 1 91900 7-24
after-hours block pattern 2 9976 7-24
!
create cnf-files version-stamp 7960 Jul 13 2004 03:39:28
!
ephone-dn 2 dual-line
number 126
preference 1
call-forward busy 500
caller-id block

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Configuration Examples for Caller ID Blocking

Configuration Examples for Caller ID Blocking

This section contains the following examples:
• Caller ID Blocking Code: Example, page 644
• Caller ID Blocking for Outbound Calls: Example, page 644

Caller ID Blocking Code: Example

The following example defines a code of *1234 for phone users to enter to block caller ID on their
outgoing calls:
telephony-service
caller-id block code *1234

Caller ID Blocking for Outbound Calls: Example

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

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

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Additional References

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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 Configuring Caller ID Blocking
Feature Information for Caller ID Blocking

Feature Information for Caller ID Blocking

Table 32 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 32 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 32 Feature Information for Caller ID Blocking

Cisco Unified CME

Feature Name Version Feature Information
Caller ID Blocking 3.0 Caller ID blocking per local call was introduced.
1.0 Caller ID blocking for outbound calls was introduced.

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 Configuring Conferencing

Last Updated: June 18, 2007

This chapter describes the conferencing support in Cisco Unified Communications Manager Express
(Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Conferencing”
section on page 687.

Contents
• Restrictions for Conferencing, page 647
• Information About Conferencing, page 648
• How to Configure Conferencing, page 651
• Configuration Examples for Conferencing, page 671
• Where to Go Next, page 686
• Additional References, page 686
• Feature Information for Conferencing, page 687

Restrictions for Conferencing

When you are configuring dial peers or ephone-dns, including park slots and conferencing extensions,
on Cisco Integrated Services Router Voice Bundles, the following message may appear to warn you that
free memory is not available:
%DIALPEER_DB-3-ADDPEER_MEM_THRESHOLD: Addition of dial-peers limited by available
memory
To configure more dial peers or ephone-dns, increase the DRAM in the system. A moderately complex
configuration may exceed the default 256 MB DRAM and require 512 MB DRAM. Note that many
factors contribute to memory usage, in addition to the number of dial peers and ephone-dns configured.

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Information About Conferencing

Information About Conferencing

To enable conferencing, you should understand the following concepts:
• Conferencing Overview, page 648
• Secure Conferencing Limitation, page 648
• Ad Hoc Conferencing, page 648
• Meet-Me Conferencing, page 650
• Soft Keys for Conference Functions, page 650

Conferencing Overview
Conferencing allows you to join three or more parties in a telephone conversation. Two types of
conferencing are available in Cisco Unified CME: ad hoc and meet-me.
Ad hoc conferences are created when one party calls another party, then either party adds one or more
parties to the conference call. Ad hoc conferences can be hardware-based or software-based.
Hardware-based ad hoc conferencing uses digital signal processors (DSPs) to enhance ad hoc
conferencing by allowing more parties than software-based ad hoc conferencing, which allows three
parties only.
Meet-me conferences are created by parties calling a designated conference number. Meet-me
conferencing is hardware-based only. If you configure software-based conferencing, you cannot have
meet-me conferences.

Secure Conferencing Limitation

Cisco Unified CME cannot use the secure conference DSP farm capability. If Cisco Unified CME needs
a conference DSP farm resource for multiparty ad hoc or meet-me conferencing, it will use a secure or
nonsecure DSP farm resource depending on what resources have been registered with Cisco Unified
CME. If Cisco Unified CME happens to pick a secure DSP farm resource, the conference itself will not
be secure, which is a waste of the more expensive, in terms of sessions capacity, secure DSP farm
resource.
To avoid using valuable secure DSP farm resources, we recommend that you do not register a secure
conference DSP Farm profile to a Cisco Unified CME because Cisco Unified CME cannot use the DSP
farm’s secure capabilities.

Ad Hoc Conferencing
Before Cisco Unified CME 4.1, support for conferencing is limited to three-party ad hoc conference
calls using a G.711 codec. To have an ad hoc conference with a party that is not using a G.711 codec,
transcoding is necessary. For more information, see the “Transcoding When a Remote Phone Uses
G.729r8” section on page 305.
The maximum number of simultaneous conferences is platform-specific to the type of
Cisco Unified CME router, and each individual Cisco Unified 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.

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Information About Conferencing

Conference Gain Levels

In Cisco Unified CME 3.3 and later versions, 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.

End-of-Conference Options
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.
Cisco Unified IP phones can be configured 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.
In Cisco Unified CME 4.0 and later versions, behavior for the end of three-way conferences can be
configured at a phone level. The options specify whether the last party that joined a conference can be
dropped from the conference and whether the remaining two parties should be allowed to continue their
connection after the conference initiator has left the conference.

Multi-Party Ad Hoc Conferencing for More Than Three Parties

In Cisco Unified CME 4.1 and later versions, multi-party ad hoc conferences allow more than three
parties. Ad hoc conferences are created when one party calls another, then either party decides to add
another party to the call. Ad hoc conferences can be created in several ways.
The conference shown in Figure 41 is created when extension 1215 dials extension 1225. The two parties
decide to add a third party, extension 1235. Extensions 1215, 1225, and 1235 are now parties in an ad
hoc conference. Extension 1215 is the creator.

Figure 41 Simple Ad Hoc Conference Using the Conf Soft Key

x1215 x1225
1
IP IP

x1235
170458

IP

You can configure ad hoc conferencing so that only the creator can add parties to the conference. The
default is that any party can add other parties to the conference.

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Information About Conferencing

You can configure conferencing so that the conference drops when the creator hangs up, and you can
configure it so that the conference drops when the last local party hangs up. The default is that the
conference is not dropped, regardless of whether the creator hangs up, provided three parties remain in
the conference.
For configuration information, see the “SCCP: Configuring Conferencing Options for a Phone” section
on page 667 for more information.

Meet-Me Conferencing
In Cisco Unified CME 4.1 and later versions, meet-me conferences consist of at least three parties
dialing a meet-me conference number predetermined by a system administrator. For example, the
conference shown in Figure 42 is created when the conference creator at extension 1215 presses the
MeetMe soft key and hears a confirmation tone, then dials the meet-me conference number 1500.
Extension 1225 and extension 1235 join the meet-me conference by dialing 1500. Extensions 1215,
1225, and 1235 are now parties in a meet-me conference on extension 1500.

Figure 42 Simple Meet-Me Conference Scenario

x1500

1 3 2

IP IP
x1215 x1225
170459

IP
x1235

Configuring Maximum Parties

You can configure the maximum number of conference parties to be lower than the actual maximum of
32 for meet-me conferences. See the “SCCP: Configuring the DSP Farm” section on page 661 for more
information.

Freeing Conference Resources

If only one party remains in the meet-me conference, for example, if one party has forgotten to hang up,
the conference call is disconnected after five minutes to free system resources.
If the creator is waiting for parties to join the conference and is the only party on the conference, the
conference is not disconnected because significant resources are not being used.

Soft Keys for Conference Functions

In Cisco Unified CME 4.1 and later versions, the following soft keys provide conferencing functions for
multi-party conferencing enhancements on your phone:

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• ConfList—Conference list. Lists all parties in a conference. For ad hoc conferences, this soft key is
available for all parties in a conference. For meet-me conferences, this soft key is available for the
creator only. Press Update to update the list of parties in the conference, for instance, to verify that
a party has been removed from the conference.
• Join—Joins an established call to conference. After you press Select to choose an established call
or conference, press Join to join that call or conference to the established call or conference.
• RmLstC—Remove last caller. Removes the last party added to the conference. This soft key works
for the creator only.
• Select—Selects a call or conference to join to conference and selects a call to remove from a
conference. The creator can remove other parties by pressing the ConfList soft key, then use the
Select and Remove soft keys to remove the appropriate parties.
• MeetMe—Initiates a meet-me conference. This soft key is pressed by the creator before dialing the
conference number. Other meet-me conference parties dial the conference number only to join the
conference. This soft key must be configured before you can initiate meet-me conferences.

How to Configure Conferencing

This section contains the following tasks:

Three-Party Ad Conferencing
• Modifying the Default Configuration for Three-Party Ad Hoc Conferencing, page 651 (optional)
• SCCP: Configuring Conferencing Options on a Phone, page 653 (optional)
• SIP: Configuring Conferencing Options on a Phone, page 655 (optional)

Multi-Party Ad Hoc and Meet-Me Conferencing

• SCCP: Configuring Multi-Party Ad Hoc and Meet-Me Conferencing, page 657 (required)

Modifying the Default Configuration for Three-Party Ad Hoc Conferencing

To globally modify the default configuration and change any of the following parameters for three-party
ad hoc conferencing, perform the following steps.
• Maximum number of three-party conferences that are supported simultaneously by the
Cisco Unified CME router. Maximum number of simultaneous three-party conferences supported
by a router is platform-dependent. The default value is half of the maximum number.
• Increase the sound volume of VoIP and public switched telephony network (PSTN) parties joining
a conference call

Restrictions
• When a three-way conference is established, a participant cannot use call transfer to join the
remaining conference participants to a different number.
• Three-party ad hoc conferencing does not support meet-me conferences.

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SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. max-conferences max-conference-number [gain -6 | 0 | 3 | 6]
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)#
Step 4 max-conferences max-conference-number Sets the maximum number of simultaneous three-party
[gain -6 | 0 | 3 | 6] conferences supported by the router.
• max-conference-number—Maximum number of
Example: simultaneous three-party conferences supported by a router,
Router(config-telephony)# max-conferences which is platform-dependent. The default is half of the
6
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 a higher release)
• 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 5 end Exits to privileged EXEC mode.

Example:
Router(config-telephony)# end

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What to Do Next
• To configure optional end-of-conference options for three-party ad hoc conferencing on SCCP
phones, see “SCCP: Configuring Conferencing Options on a Phone” section on page 653
• To configure optional end-of-conference options for three-party ad hoc conferencing on SCCP
phones, see “SIP: Configuring Conferencing Options on a Phone” section on page 655

SCCP: Configuring Conferencing Options on a Phone

To configure optional end-of-conference options for three-party ad hoc conferencing on a
Cisco Unified IP phone running SCCP, perform the following steps for each phone to be configured.

Prerequisites
• Conferencing uses call transfer to connect the two remaining parties of a conference when a
conference initiator leaves the conference. To use this feature, you must configure the
transfer-system command. For configuration information, see “Configuring Call Transfer and
Forwarding” on page 499.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. keep-conference [drop-last] [endcall] [local-only]
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies this
Example: ephone during configuration tasks.
Router(config)# ephone 1

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Command or Action Purpose

Step 4 keep-conference [drop-last] [endcall] Allows conference initiators to exit from conference calls and to
[local-only] either end or maintain the conference for the remaining parties.
• no keep-conference—(Default; the no form of the
Example: command) The conference initiator can hang up or press the
Router(config-ephone)# keep-conference EndCall soft key to end the conference and disconnect all
endcall
parties or press the Confrn soft key to drop only the last party
that was connected to the conference.
• keep-conference—(No keywords used) The conference
initiator can press the EndCall soft key to end the conference
and disconnect all parties or hang up to leave the conference
and keep the other two parties connected. The conference
initiator can also use the Confrn soft key (IP phone) or
hookflash (analog phone) to break up the conference but stay
connected to both parties.
• drop-last—The action of the Confrn soft key is changed; the
conference initiator can press the Confrn soft key (IP phone)
or hookflash (analog phone) to drop the last party.
Note Analog phones connected to the Cisco Unified CME
system through a Cisco VG 224 require Cisco IOS
Release 12.3(11)YL1 or a later release to use this feature.

• endcall—The action of the EndCall soft key is changed; the

conference initiator can hang up or press the EndCall soft key
to leave the conference and keep the other two parties
connected.
• local-only—The conference initiator can hang up to end the
conference and leave the other two parties connected only if
one of the remaining parties is local to the
Cisco Unified CME system (an internal extension).
Step 5 end Exits to privileged EXEC mode.

Example:
Router(config)# end

What to Do Next
If you are finished modifying the configuration, you are ready to generate configuration files for the
phones to be connected. See “SCCP: Generating Configuration Files for SCCP Phones” on page 247.

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SIP: Configuring Conferencing Options on a Phone

To configure optional end-of-conference options for three-party ad hoc conferencing on a
Cisco Unified IP phone running SIP, perform the following steps for each phone to be configured.

Prerequisites
• To facilitate call transfer by using the Confrn softkey, conference and transfer attended or transfer
blind must be enabled. For configuration information, see “Configuring Call Transfer and
Forwarding” on page 499.

Restrictions
Music on hold (MOH) is not supported for call hold invoked from a SIP phone. A caller hears only
silence when placed on hold by a SIP phone.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register pool pool-tag
4. keep-conference
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for SIP phones.
Example: • pool-tag unique sequence number of the SIP phone to
Router(config)# voice register pool 3 be configured. Range is 1 to 100 or the upper limit as
defined by max-pool command.

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Command or Action Purpose

Step 4 keep-conference Allows a Cisco Unified IP phone conference initiator to exit
from conference calls and keeps the remaining parties
connected.
Example:
Router(config-register-pool)# keep-conference Note This step is included to illustrate how to enable the
command if it was previously disabled.

• Default is enabled.
• Remaining calls are transferred without consultation as
enabled by the transfer-attended (voice register
template) or transfer-blind (voice register template)
commands.
Step 5 end Exits to privileged EXEC mode.

Example:
Router(config-register-pool)# end

What to Do Next
• If you are finished modifying the configuration, you are ready to generate configuration files for the
phones to be connected. See “SIP: Generating Configuration Profiles for SIP Phones” on page 250.

Verifying Three-Party Ad Hoc Conferencing

Step 1 Use the show running-config command to verify your configuration. Any non-default conferencing
parameters are listed in the telephony-service portion of the output, and end-of-conference options are
listed in the ephone portion.
Router# show running-config
!
ephone-dn 1 dual-line
ring feature secondary
number 126 secondary 1261
description Sales
name Smith
call-forward busy 500 secondary
call-forward noan 500 timeout 10
huntstop channel
no huntstop
no forward local-calls
!
ephone 1
mac-address 011F.92A0.C10B
type 7960 addon 1 7914
no dnd feature-ring
keep-conference

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Troubleshooting Three-Party Ad Hoc Conferencing

Step 1 Use the debug ephone commands to observe messages and states associated with an ephone. For more
information, see the Cisco Unified CME Command Reference.

SCCP: Configuring Multi-Party Ad Hoc and Meet-Me Conferencing

To configure multi-party ad hoc conference support for 3-8 parties plus Meet-Me conferencing for up to
32 parties, perform the following tasks:
• SCCP: Configuring Join and Leave Tones, page 658 (optional)
• SCCP: Configuring SCCP for Cisco Unified CME, page 660 (required)
• SCCP: Configuring the DSP Farm, page 661 (required)
• SCCP: Associating Cisco Unified CME with a DSP Farm Profile, page 663 (required)
• SCCP: Enabling Multi-Party Ad Hoc and Meet-Me Conferencing, page 664 (required)
• SCCP: Configuring Multi-Party Ad Hoc Conferencing and Meet-Me Numbers, page 665 (required)
• SCCP: Configuring Conferencing Options for a Phone, page 667 (required)
• SCCP: Verifying Multi-Party Ad Hoc and Meet-Me Conferencing, page 670 (optional)

Prerequisites
• Cisco Unified CME 4.1 or a later version
• You must have a PVDM2-8, PVDM2-16, PVDM2-32, or PVDM2-64 high-density packet voice
digital signal processor module hosted on the motherboard or on a module such as the NM-HDV2
or NM-HD-2VE.
• For Cisco Unified IP Phone 7985, firmware version 4-1-2-0 or a later version

Restrictions
• The maximum number of meet-me conference parties is 32 for one DSP using the G.711 codec and
16 for the G.729 codec.
• A participant cannot join more than one conference at the same time.
• Ad hoc conferencing for more than three parties is not supported on the Cisco Unified IP Phone
7906 and 7910 and Cisco Unified IP Phone 7914 Expansion Module.
• Ad hoc conferencing for more than three parties is not supported on Cisco Unified IP phones
running SIP.

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SCCP: Enabling DSP Farm Services for a Voice Card

To enable DSP farm services for a voice card to support multi-party ad hoc and meet-me conferences,
perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice-card slot
4. dsp services dspfarm
5. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice-card slot Enters voice-card configuration mode and configure a voice
card.
Example:
Router(config)# voice-card 2
Step 4 dsp services dspfarm Enables digital-signal-processor (DSP) farm services for a
particular voice network module.
Example:
Router(config-voicecard)# dsp services dspfarm
Step 5 exit Exits voice-card configuration mode.

Example:
Router(config-voicecard)# exit

SCCP: Configuring Join and Leave Tones

To configure tones to be played when parties join and leave ad hoc and meet-me conferences, perform
the following steps for each tone to be configured.

SUMMARY STEPS

1. enable
2. configure terminal

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3. voice class custom-cptone cptone-name

4. dualtone conference
5. frequency frequency-1 [frequency-2]
6. cadence {cycle-1-on-time cycle-1-off-time [cycle-2-on-time cycle-2-off-time] [cycle-3-on-time
cycle-3-off-time] [cycle-4-on-time cycle-4-off-time]} | continuous
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice class custom-cptone cptone-name Creates a voice class for defining custom call-progress
tones to be detected.
Example:
Router(config)# voice class custom-cptone
jointone
Step 4 dualtone conference Configures conference join and leave tones.

Example:
Router(cfg-cptone)# dualtone conference
Step 5 frequency frequency-1 [frequency-2] Defines the frequency components for a call-progress tone.

Example:
Router(cfg-cp-dualtone)# frequency 600 900
Step 6 cadence {cycle-1-on-time cycle-1-off-time Defines the tone-on and tone-off durations for a
[cycle-2-on-time cycle-2-off-time] call-progress tone.
[cycle-3-on-time cycle-3-off-time]
[cycle-4-on-time cycle-4-off-time]} |
continuous

Example:
Router(cfg-cp-dualtone)# cadence 300 150 300
100 300 50
Step 7 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(cfg-cp-dualtone)# exit

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SCCP: Configuring SCCP for Cisco Unified CME

To enable Skinny Client Control Protocol (SCCP) on Cisco Unified CME, perform the following steps:

SUMMARY STEPS

1. enable
2. configure terminal
3. sccp local interface-type interface-number [port port-number]
4. sccp ccm {ip-address | dns} identifier identifier-number [priority priority] [port port-number]
[version version-number]
5. sccp ccm group group-number
6. bind interface interface-type interface-number
7. exit
8. sccp
9. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 sccp local interface-type interface-number Selects the local interface that SCCP applications
[port port-number] (transcoding and conferencing) use to register with
Cisco Unified CME.
Example:
Router(config)# sccp local FastEthernet0/0
Step 4 sccp ccm {ip-address | dns} identifier Adds a Cisco Unified CME router to the list of available
identifier-number [priority priority] [port servers and set various parameters—including IP address or
port-number] [version version-number]
Domain Name System (DNS) name, port number, and
version number.
Example: • version-number—Must be 4.0 or later.
Router(config)# sccp ccm 1.4.158.3 identifier
100 version 4.0
Step 5 sccp ccm group group-number Creates a Cisco Unified CME group.

Example:
Router(config)# sccp ccm group 123

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Command or Action Purpose

Step 6 bind interface interface-type interface-number Binds an interface to a Cisco Unified CME group.

Example:
Router(config-sccp-cm)# bind interface
fastethernet 0/0
Step 7 exit Exits SCCP Cisco Unified CME configuration mode.

Example:
Router(config-sccp-cm)# exit
Step 8 sccp Enables SCCP and its related applications (transcoding and
conferencing).
Example:
Router(config)# sccp
Step 9 exit Exits global configuration mode.

Example:
Router(config)# exit

SCCP: Configuring the DSP Farm

To configure the DSP farm profile for multi-party ad hoc and meet-me conferencing, perform the
following steps.

Note The DSP farm can be on the same router as the Cisco Unified CME or on a different router.

SUMMARY STEPS

1. enable
2. configure terminal
3. dspfarm profile profile-identifier conference
4. codec {codec-type | pass-through}
5. conference-join custom-cptone cptone-name
6. conference-leave custom-cptone cptone-name
7. maximum conference-party max-parties
8. maximum sessions number
9. associate application sccp
10. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 dspfadrm profile profile-identifier conference Enters DSP farm profile configuration mode and defines a
profile for DSP farm services.
Example:
Router(config)# dspfarm profile 1 conference
Step 4 codec {codec-type | pass-through} Specifies the codecs supported by a DSP farm profile.
Note Repeat this step as necessary to specify all the
Example: supported codecs.
Router(config-dspfarm-profile)# codec g711ulaw
Step 5 conference-join custom-cptone cptone-name Associates a custom call-progress tone to indicate joining a
conference with a DSP farm profile.
Example: Note The cptone-name argument in this step must be the
Router(config-dspfarm-profile)# conference-join same as the cptone-argument in the voice class
custom-cptone jointone custom-cptone command configured in the “SCCP:
Enabling DSP Farm Services for a Voice Card”
section on page 658.
Step 6 conference-leave custom-cptone cptone-name Associates a custom call-progress tone to indicate leaving a
conference with a DSP farm profile.
Example: Note The cptone-name argument in this step must be the
Router(config-dspfarm-profile)# same as the cptone-argument in the voice class
conference-leave custom-cptone leavetone custom-cptone command configured in the “SCCP:
Enabling DSP Farm Services for a Voice Card”
section on page 658.
Step 7 maximum conference-party max-parties (Optional) Configures the maximum number of conference
parties allowed in each meet-me conference. The maximum
is codec-dependent.
Example:
Router(config-dspfarm-profile)# maximum
conference-party 32
Step 8 maximum sessions number Specifies the maximum number of sessions that are
supported by the profile.
Example:
Router(config-dspfarm-profile)# maximum
sessions 8

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Command or Action Purpose

Step 9 associate application sccp Associates SCCP with the DSP farm profile.

Example:
Router(config-dspfarm-profile)# associate
application sccp
Step 10 end Exits to privileged EXEC mode.

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

SCCP: Associating Cisco Unified CME with a DSP Farm Profile

To associate a DSP farm profile with a group of Cisco Unified CME routers that control DSP services,
perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. sccp ccm group group-number
4. associate ccm identifier-number priority priority-number
5. associate profile profile-identifier register device-name
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 sccp ccm group group-number Creates a Cisco Unified CME group.

Example:
Router(config)# sccp ccm group 1
Step 4 associate ccm identifier-number priority Associates a Cisco Unified CME router with the group and
priority-number establishes its priority within the group.

Example:
Router(config-sccp-ccm)# associate ccm 100
priority 1

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Command or Action Purpose

Step 5 associate profile profile-identifier register Associates a DSP farm profile with the Cisco Unified CME
device-name group.
• device-name is a maximum of 16 characters.
Example:
Router(config-sccp-ccm)# associate profile 2
Note Repeat this step for every conferencing DSP farm
register confdsp1 and transcoding DSP farm.
Step 6 end Exits to privileged EXEC mode.

Example:
Router(config-sccp-ccm)# end

SCCP: Enabling Multi-Party Ad Hoc and Meet-Me Conferencing

To allow multi-party ad hoc conferences with more than three parties and meet-me conferences, perform
the following steps.

Note Configuring multi-party ad hoc conferencing in Cisco Unified CME disables three-party ad hoc
conferencing.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. conference hardware
5. sdspfarm units number
6. sdspfarm tag number device-name
7. sdspfarm conference mute-on mute-on-digits mute-off mute-off-digits
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 conference hardware Configures a Cisco Unified CME system for multi-party
conferencing only.
Example:
Router(config-telephony)# conference hardware
Step 5 sdspfarm units number Specifies the maximum number of DSP farms that are
allowed to be registered to the SCCP server.
Example:
Router(config-telephony)# sdspfarm units 3
Step 6 sdspfarm tag number device-name Permits a DSP farm to register to Cisco Unified CME and
associates it with a SCCP client interface's MAC address.
Example: Note The device-name in this step must be the same as the
Router(config-telephony)# sdspfarm tag 2 device-name in the associate profile command in
confdsp1 Step 5 of the “SCCP: Associating Cisco Unified
CME with a DSP Farm Profile” section on
page 663.
Step 7 sdspfarm conference mute-on mute-on-digits Defines mute-on and mute-off digits for conferencing.
mute-off mute-off-digits
• Maximum: 3 digits. Valid values are the numbers and
symbols that appear on your telephone keypad: 1, 2, 3,
Example: 4, 5, 6, 7, 8, 9, 0, *, and #.
Router(config-telephony)# sdspfarm conference
mute-on 111 mute-off 222 • Mute-on and mute-off digits can be the same.
Step 8 end Exits to privileged EXEC mode.

Example:
Router(config-telephony)# end

SCCP: Configuring Multi-Party Ad Hoc Conferencing and Meet-Me Numbers

To configure numbers for multi-party ad hoc and meet-me ad hoc conferencing, based on the maximum
number of conference participants you configure, perform the following steps. Ad hoc conferences
require four extensions per conference, regardless of how many extensions are actually used by the
conference parties.

Note Ensure that you configure enough directory numbers to accommodate the anticipated number of
conferences. The maximum number of parties in a multi-party ad hoc conference on an IP phone is eight;
the maximum on an analog phone is three.

SUMMARY STEPS

1. enable
2. configure terminal

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

4. number number [secondary number] [no-reg [both | primary]]
5. conference {ad-hoc | meetme}
6. preference preference-order [secondary secondary-order]
7. no huntstop [channel]
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag dual-line Enters ephone-dn configuration mode for the purposes of
creating and configuring an extension (ephone-dn) for a
Cisco Unified IP phone line.
Example:
Router(config)# ephone-dn 18 dual-line • Each ephone-dn can carry two parties if it is configured
as a dual line.
• Configure enough ephone-dns to accommodate the
maximum number of conference participants to be
supported.
• For multi-party ad hoc conferencing, maximum number
of directory numbers is 8, but you can configure a lower
maximum.
• For meet-me conferencing, maximum number of
directory numbers is 32, but you can configure a lower
maximum.
• Minimum number of directory numbers required: 2.
Step 4 number number [secondary number] [no-reg [both Associates a telephone or extension number with an
| primary]] ephone-dn in a Cisco Unified CME system.
• Each DN for a conference must have the same primary
Example: and secondary number.
Router(config-ephone-dn)# number 6789

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Command or Action Purpose

Step 5 conference ad-hoc Configures a number as a placeholder for ad hoc
or conferencing to associate the call with the DSP farm.
conference meetme or
(Optional) Associates meet-me conferencing with a
Example: directory number.
Router(config-ephone-dn)# conference ad-hoc
or
Router(config-ephone-dn)# conference meetme
Step 6 preference preference-order [secondary Sets dial-peer preference order for an extension
secondary-order] (ephone-dn) associated with a Cisco Unified IP phone.
• Remember to configure “preference x” with low value
Example: to last DN.
Router(config-ephone-dn)# preference 1
• The lower the value of the preference-order argument,
the higher the preference of the extension.
Step 7 no huntstop [channel] Continues call hunting behavior for an extension
(ephone-dn) or an extension channel.
Example: • Remember to configure no huntstop for all DNs except
Router(config-ephone-dn)# no huntstop the last one.
Step 8 end Exits to privileged EXEC mode.

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

SCCP: Configuring Conferencing Options for a Phone

To configure a template of conferencing features such as the add party mode, drop party mode, and soft
keys, for multi-party ad hoc, and meet-me conferences and apply the template to a phone, perform the
following steps.

Note The following commands can also be configured in ephone configuration mode. Commands configured
in ephone configuration mode have priority over commands in ephone-template configuration mode.

Restrictions

• The ConfList (including the Remove, Update, and Exit soft keys within the ConfList function) and
RmLstC soft keys do not work on a Cisco Unified IP Phone 7902, 7935, and 7936.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-template template-tag
4. conference add-mode [creator]
5. conference drop-mode [creator | local]

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6. conference admin
7. softkeys connected [Acct] [ConfList] [Confrn] [Endcall] [Flash] [HLog] [Hold] [Join] [Park]
[RmLstC] [Select] [Trnsfer]
8. softkeys hold [Join] [Newcall] [Resume] [Select]
9. softkeys idle [Cfwdall] [ConfList] [Dnd] [Gpickup] [HLog] [Join] [Login] [Newcall] [Pickup]
[Redial] [RmLstC]
10. softkeys seized [CallBack] [Cfwdall] [Endcall] [Gpickup] [HLog] [MeetMe] [Pickup] [Redial]
11. exit
12. ephone phone-tag
13. ephone-template template-tag
14. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-template template-tag Enter ephone-template configuration mode to create an
ephone template to configure a set of phone features.
Example:
Router(config)# ephone-template 1
Step 4 conference add-mode [creator] (Optional) Configures the mode for adding parties to
conferences.
Example: • creator—Only the creator can add parties to the
Router(config-ephone-template)# conference conference.
add-mode creator
Step 5 conference drop-mode [creator | local] (Optional) Configures the mode for dropping parties from
multi-party ad hoc and meet-me conferences.
Example: • creator—The active conference terminates when the
Router(config-ephone-template)# conference creator hangs up.
drop-mode creator
• local—The active conference terminates when the last
local party in the conference hangs up or drops out of
the conference.

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Command or Action Purpose

Step 6 conference admin (Optional) Configures the ephone as the conference
administrator. The administrator can:
Example: • Dial in to any conference directly through the
Router(config-ephone-template)# conference conference number
admin
• Use the ConfList soft key to list conference parties
• Remove any party from any conference
Step 7 softkeys connected [Acct] [ConfList] [Confrn] Configures an ephone template for soft-key display during
[Endcall] [Flash] [HLog] [Hold] [Join] [Park] the connected call stage.
[RmLstC] [Select] [Trnsfer]
• The soft keys added are RmLstC, ConfList, Join, and
Select.
Example:
Router(config-ephone-template)# softkeys • The number and order of soft key keywords you enter
connected Hold Trnsfer Park Endcall Confrn in this command correspond to the number and order of
ConfList Join Select RmLstC soft keys on your phone.
Step 8 softkeys hold [Join] [Newcall] [Resume] Configures an ephone template to modify soft-key display
[Select] during the call-hold call stage.
• The soft keys added are Join and Select.
Example:
Router(config-ephone-template)# softkeys hold
• The number and order of soft key keywords you enter
Join Newcall Resume Select in this command correspond to the number and order of
soft keys on your phone.
Step 9 softkeys idle [Cfwdall] [ConfList] [Dnd] Configures an ephone template for soft-key display during
[Gpickup] [HLog] [Join] [Login] [Newcall] the idle call stage.
[Pickup] [Redial] [RmLstC]
• The soft keys added for multi-party conferencing are
RmLstC, ConfList, and Join.
Example:
Router(config-ephone-template)# softkeys idle • The number and order of soft key keywords you enter
ConfList Gpickup Join Login Newcall Pickup in this command correspond to the number and order of
Redial RmLstC soft keys on your phone.
Step 10 softkeys seized [CallBack] [Cfwdall] [Endcall] (Optional) Configures an ephone template for soft-key
[Gpickup] [HLog] [MeetMe] [Pickup] [Redial] display during the seized call stage.
• You must configure the MeetMe soft key in the seized
Example: state for the ephone to initiate a meet-me conference.
Router(config-ephone-template)# softkeys seized
Redial Endcall Cfwdall Pickup Gpickup Callback • The number and order of soft key keywords you enter
Meetme in this command correspond to the number and order of
soft keys on your phone.
Step 11 exit Exits ephone-template configuration mode.

Example:
Router(config-ephone-template)# exit
Step 12 ephone phone-tag Enters Ethernet phone (ephone) configuration mode for an
IP phone for the purposes of creating and configuring an
ephone.
Example:
Router(config)# ephone 1

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Command or Action Purpose

Step 13 ephone-template template-tag Applies an ephone-dn template to an ephone-dn.
Note The template-tag must be the same as the
Example: template-tag in Step 3.
Router(config-ephone)# ephone-dn-template 1
Step 14 end Exits to privileged EXEC mode.

Example:
Router(config-ephone)# exit

What to Do Next
If you are finished modifying the configuration, you are ready to generate configuration files for the
phones to be connected. See “SCCP: Generating Configuration Files for SCCP Phones” on page 247.

SCCP: Verifying Multi-Party Ad Hoc and Meet-Me Conferencing

Use the following show commands to verify multi-party ad hoc and meet-me conferencing:
• show ephone-dn conference—Displays information about ad hoc and meet-me conferences.
• show telephony-service conference hardware—Displays information about hardware-based
conferences.

show ephone-dn conference: Example

type active inactive numbers
=======================================
Meetme 0 8 2345
DN tags: 9, 10, 11, 12

Ad-hoc 0 8 A001
DN tags: 13, 14, 15, 16

Meetme 0 8 1234
DN tags: 20, 21, 22, 23

show telephony-service conference hardware detail: Example

Conference Type Active Max Peak Master MasterPhone Last
cur(initial)
==================================================================
8889 Ad-hoc 3 8 3 8044 29 ( 29) 8012
Conference parties:
8012
8006
8044

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Configuration Examples for Conferencing

This section provides the following configuration examples:
• Basic Conferencing: Example, page 671
• End of Conference Options: Example, page 671
• DSP Farm and Cisco Unified CME on the Same Router: Example, page 672
• DSP Farm and Cisco Unified CME on Different Routers: Example, page 677

Basic Conferencing: Example

The following example sets the maximum number of conferences for a Cisco Unified 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

End of Conference Options: Example

In the following example, extension 3555 initiates a three-way conference. After the conference is
established, extension 3555 can press the Confrn soft key to disconnect the last party that was connected
and remain connected to the first party that was connected. If extension 3555 hangs up from the
conference, the other two parties remain connected if one of them is local to the Cisco Unified CME
system.
ephone-dn 35
number 3555

ephone 24
button 1:35
keep-conference drop-last local-only

In the following example, extension 3666 initiates a three-way conference. After the conference is
established, extension 3666 can press the Confrn soft key to disconnect the last party that was connected
and remain connected to the first party that was connected. Also, extension 3666 can hang up or press
the EndCall soft key to leave the conference and keep the other two parties connected.
ephone-dn 36
number 3666

ephone 25
button 1:36
keep-conference drop-last endcall

In the following example, extension 3777 initiates a three-way conference. After the conference is
established, extension 3777 can press the Confrn soft key to disconnect the last party that was connected
and remain connected to the first party that was connected. Also, extension 3777 can hang up or press
the EndCall soft key to leave the conference and keep the other two parties connected only if one of the
two parties is local to the Cisco Unified CME system.
ephone-dn 38
number 3777

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ephone 27
button 1:38
keep-conference drop-last endcall local-only

In the following example, extension 3999 initiates a three-way conference. After the conference is
established, extension 3999 can hang up or press the EndCall soft key to leave the conference and keep
the other two parties connected only if one of the two parties is local to the Cisco Unified CME system.
Extension 3999 can also use the Confrn soft key to break up the conference but stay connected to both
parties.
ephone-dn 39
number 3999

ephone 29
button 1:39
keep-conference endcall local-only

DSP Farm and Cisco Unified CME on the Same Router: Example
In this example, the DSP farm and Cisco Unified CME are on the same router as shown in Figure 43.

Figure 43 CME and the DSP Farm on the Same Router

IP
Cisco Unified CME
DSP farm SIP

WAN
LAN
SCCP FXS VG224 H323 FXS
SCCP

IP
PSTN
IPC

IP link 170540

Current configuration : 16345 bytes

!
version 12.4
service timestamps debug datetime msec
service timestamps log uptime
no service password-encryption
service internal
!
hostname cmedsprtr
!
boot-start-marker
boot-end-marker
!
logging buffered 90000 debugging
!
no aaa new-model
!

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resource policy
!
no network-clock-participate slot 1
no network-clock-participate wic 0
ip cef
!
!
ip dhcp pool phone1
host 10.4.188.66 255.255.0.0
client-identifier 0100.0ab7.b144.4a
default-router 10.4.188.65
option 150 ip 10.4.188.65
!
ip dhcp pool phone2
host 1.4.188.67 255.255.0.0
client-identifier 0100.3094.c269.35
default-router 10.4.188.65
option 150 ip 10.4.188.65
!
!
voice-card 1
dsp services dspfarm
!
!
voice call send-alert
voice call carrier capacity active
!
voice service voip
allow-connections h323 to h323
supplementary-service h450.12
h323
!
!
!
!
controller E1 1/0
framing NO-CRC4
!
controller E1 1/1
!
!
interface FastEthernet0/0
ip address 10.4.188.65 255.255.0.0
duplex auto
speed auto
no keepalive
no cdp enable
no clns route-cache
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
no clns route-cache
!
ip route 10.4.0.0 255.255.0.0 FastEthernet0/0
ip route 192.168.254.254 255.255.255.255 10.4.0.1
!
ip http server
!
!
control-plane
!

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!
sccp local FastEthernet0/0
sccp ccm 10.4.188.65 identifier 1 version 4.0
sccp
!
sccp ccm group 123
associate ccm 1 priority 1
associate profile 1 register mtp00097c5e9ce0
keepalive retries 5
!
!
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 6
associate application SCCP
!
dial-peer cor custom
!
!
!
dial-peer voice 6 voip
destination-pattern 6...
session target ipv4:10.4.188.90
!
telephony-service
conference hardware
load 7960-7940 P00307020400
load 7905 CP7905060100SCCP050309A.sbin
max-ephones 48
max-dn 180
ip source-address 10.4.188.65 port 2000
timeouts ringing 500
system message MY MELODY (2611)
sdspfarm units 4
sdspfarm tag 1 mtp00097c5e9ce0
max-conferences 4 gain -6
call-forward pattern ....
transfer-system full-consult
transfer-pattern 7...
transfer-pattern ....
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-template 1
softkeys hold Newcall Resume Select Join
softkeys idle Cfwdall ConfList Dnd Gpickup HLog Join Login Newcall Pickup Redial RmLstC
softkeys seized Redial Pickup Gpickup HLog Meetme Endcall
softkeys connected Acct ConfList Confrn Endcall Flash HLog Hold Join Park RmLstC Select
Trnsfer
!
!
ephone-dn 1 dual-line
number 8001
name melody-8001
!
!
ephone-dn 2 dual-line
number 8002
!

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!
ephone-dn 3 dual-line
number 8003
!
!
ephone-dn 4 dual-line
number 8004
!
!
ephone-dn 5 dual-line
number 8005
!
!
ephone-dn 6 dual-line
number 8006
!
!
ephone-dn 7 dual-line
number 8007
!
!
ephone-dn 8 dual-line
number 8008
!
!
ephone-dn 60 dual-line
number 8887
conference meetme
no huntstop
!
!
ephone-dn 61 dual-line
number 8887
conference meetme
preference 1
no huntstop
!
!
ephone-dn 62 dual-line
number 8887
conference meetme
preference 2
no huntstop
!
!
ephone-dn 63 dual-line
number 8887
conference meetme
preference 3
!
!
ephone-dn 64 dual-line
number 8889
name Conference
conference ad-hoc
no huntstop
!
!
ephone-dn 65 dual-line
number 8889
name Conference
conference ad-hoc
preference 1
no huntstop

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!
!
ephone-dn 66 dual-line
number 8889
name Conference
conference ad-hoc
preference 2
no huntstop
!
!
ephone-dn 67 dual-line
number 8889
name Conference
conference ad-hoc
preference 3
!
!
ephone 1
ephone-template 1
mac-address 0030.94C2.6935
type 7960
button 1:1 2:2
!
!
ephone 2
ephone-template 1
mac-address 000A.B7B1.444A
type 7940
button 1:4 2:8
!
line con 0
exec-timeout 0 0
line aux 0
exec-timeout 0 0
line vty 0 4
exec-timeout 0 0
login
line vty 5 15
login
!
!
end

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DSP Farm and Cisco Unified CME on Different Routers: Example

In this example, the DSP farm and Cisco Unified CME are on different routers as shown in Figure 44.

Figure 44 Cisco Unified CME and the DSP Farm on Different Routers

IP
Cisco Unified CME SIP

WAN
LAN
SCCP FXS VG224 H323 FXS
SCCP

IP
PSTN
IPC
PSTN call

IP link

170541
DSP farm

This section contains configuration examples for the following routers:

• Cisco Unified CME Router Configuration: Example, page 677
• DSP Farm Router Configuration: Example, page 684

Cisco Unified CME Router Configuration: Example

Current configuration : 5659 bytes
!
version 12.4
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
!
boot-start-marker
boot-end-marker
!
!
card type command needed for slot 1
logging buffered 3000000 debugging
!
no aaa new-model
!
resource policy
!
no network-clock-participate slot 1
no network-clock-participate aim 0
!

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voice-card 1
no dspfarm
!
voice-card 3
dspfarm
!
ip cef
!
!
no ip dhcp use vrf connected
!
ip dhcp pool IPPhones
network 10.15.15.0 255.255.255.0
option 150 ip 10.15.15.1
default-router 10.15.15.1
!
!
interface FastEthernet0/0
ip address 10.3.111.102 255.255.0.0
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
duplex auto
speed auto
!
interface FastEthernet0/1.1
encapsulation dot1Q 10
ip address 10.15.14.1 255.255.255.0
!
interface FastEthernet0/1.2
encapsulation dot1Q 20
ip address 10.15.15.1 255.255.255.0
!
ip route 0.0.0.0 0.0.0.0 10.5.51.1
ip route 0.0.0.0 0.0.0.0 10.3.0.1
!
ip http server
!
!
!
!
control-plane!
!
!
!
dial-peer voice 1 voip
destination-pattern 3...
session target ipv4:10.3.111.101
!
!
telephony-service
conference hardware
load 7910 P00403020214
load 7960-7940 P003-07-5-00
max-ephones 50
max-dn 200
ip source-address 10.15.15.1 port 2000
sdspfarm units 4
sdspfarm transcode sessions 12
sdspfarm tag 1 confer1
sdspfarm tag 4 xcode1
max-conferences 8 gain -6

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moh flash:music-on-hold.au
multicast moh 239.0.0.0 port 2000
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-template 1
softkeys hold Resume Newcall Select Join
softkeys idle Redial Newcall ConfList RmLstC Cfwdall Join Pickup Login HLog Dnd Gpickup
softkeys seized Endcall Redial Cfwdall Meetme Pickup Callback
softkeys alerting Endcall Callback
softkeys connected Hold Endcall Confrn Trnsfer Select Join ConfList RmLstC Park Flash
!
ephone-dn 1 dual-line
number 6000
!
!
ephone-dn 2 dual-line
number 6001
!
!
ephone-dn 3 dual-line
number 6002
!
!
ephone-dn 4 dual-line
number 6003
!
!
ephone-dn 5 dual-line
number 6004
!
!
ephone-dn 6 dual-line
number 6005
!
!
ephone-dn 7 dual-line
number 6006
!
!
ephone-dn 8 dual-line
number 6007
!
!
ephone-dn 9 dual-line
number 6008
!
!
ephone-dn 10 dual-line
number 6009
!
!
ephone-dn 11
number 6011
!
!
ephone-dn 12
number 6012
!
!
ephone-dn 13
number 6013
!

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!
ephone-dn 14
number 6014
!
!
ephone-dn 15
number 6015
!
!
ephone-dn 16
number 6016
!
!
ephone-dn 17
number 6017
!
!
ephone-dn 18
number 6018
!
!
ephone-dn 19
number 6019
!
!
ephone-dn 20
number 6020
!
!
ephone-dn 21
number 6021
!
!
ephone-dn 22
number 6022
!
!
ephone-dn 23
number 6023
!
!
ephone-dn 24
number 6024
!
!
ephone-dn 25 dual-line
number 6666
conference meetme
preference 1
no huntstop
!
!
ephone-dn 26 dual-line
number 6666
conference meetme
preference 2
no huntstop
!
!
ephone-dn 27 dual-line
number 6666
conference meetme
preference 3
no huntstop

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!
!
ephone-dn 28 dual-line
number 6666
conference meetme
preference 4
no huntstop
!
!
ephone-dn 29 dual-line
number 8888
conference meetme
preference 1
no huntstop
!
!
ephone-dn 30 dual-line
number 8888
conference meetme
preference 2
no huntstop
!
!
ephone-dn 31 dual-line
number 8888
conference meetme
preference 3
no huntstop
!
!
ephone-dn 32 dual-line
number 8888
conference meetme
preference 4
!
!
ephone-dn 33
number 6033
!
!
ephone-dn 34
number 6034
!
!
ephone-dn 35
number 6035
!
!
ephone-dn 36
number 6036
!
!
ephone-dn 37
number 6037
!
!
ephone-dn 38
number 6038
!
!
ephone-dn 39
number 6039
!
!

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ephone-dn 40
number 6040
!
!
ephone-dn 41 dual-line
number 6666
conference meetme
preference 5
no huntstop
!
!
ephone-dn 42 dual-line
number 6666
conference meetme
preference 6
no huntstop
!
!
ephone-dn 43 dual-line
number 6666
conference meetme
preference 7
no huntstop
!
!
ephone-dn 44 dual-line
number 6666
conference meetme
preference 8
no huntstop
!
!
ephone-dn 45 dual-line
number 6666
conference meetme
preference 9
no huntstop
!
!
ephone-dn 46 dual-line
number 6666
conference meetme
preference 10
no huntstop
!
!
ephone-dn 47 dual-line
number 6666
conference meetme
preference 10
no huntstop
!
!
ephone-dn 48 dual-line
number 6666
conference meetme
preference 10
!
!
ephone-dn 51 dual-line
number A0001
name conference
conference ad-hoc
preference 1

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no huntstop
!
!
ephone-dn 52 dual-line
number A0001
name conference
conference ad-hoc
preference 2
no huntstop
!
!
ephone-dn 53 dual-line
number A0001
name conference
conference ad-hoc
preference 3
no huntstop
!
!
ephone-dn 54 dual-line
number A0001
name conference
conference ad-hoc
preference 4
!
!
ephone 1
ephone-template 1
mac-address C863.B965.2401
type anl
button 1:1
!
!
!
ephone 2
ephone-template 1
mac-address 0016.C8BE.A04A
type 7920
!
!
!
ephone 3
ephone-template 1
mac-address C863.B965.2400
type anl
button 1:2
!
!
!
ephone 4
no multicast-moh
ephone-template 1
mac-address 0017.952B.7F5C
type 7912
button 1:4
!
!
!
ephone 5
ephone-template 1
ephone 6
no multicast-moh
ephone-template 1
mac-address 0017.594F.1468

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type 7961GE
button 1:6
!
!
!
ephone 11
ephone-template 1
mac-address 0016.C8AA.C48C
button 1:10 2:15 3:16 4:17
button 5:18 6:19 7:20 8:21
button 9:22 10:23 11:24 12:33
button 13:34 14:35 15:36 16:37
button 17:38 18:39 19:40
!
!
line con 0
line aux 0
line vty 0 4
login
!
!
end

DSP Farm Router Configuration: Example

Current configuration : 2179 bytes
!
! Last configuration change at 05:47:23 UTC Wed Jul 12 2006
!
version 12.4
service timestamps debug datetime msec localtime
no service timestamps log uptime
no service password-encryption
hostname dspfarmrouter
!
boot-start-marker
boot-end-marker
!
!
card type command needed for slot 1
logging buffered 4096 debugging
enable password lab
!
no aaa new-model
!
resource policy
!
no network-clock-participate slot 1
!
!
ip cef
!
!
no ip domain lookup
!
!
voice-card 0
no dspfarm
!
voice-card 1
no dspfarm
dsp services dspfarm

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interface GigabitEthernet0/0
ip address 10.3.111.100 255.255.0.0
duplex auto
speed auto
!
interface GigabitEthernet0/1.1
encapsulation dot1Q 100
ip address 192.168.1.10 255.255.255.0
!
interface GigabitEthernet0/1.2
encapsulation dot1Q 200
ip address 192.168.2.10 255.255.255.0
!
interface GigabitEthernet0/1.3
encapsulation dot1Q 10
ip address 10.15.14.10 255.255.255.0
!
interface GigabitEthernet0/1.4
encapsulation dot1Q 20
ip address 10.15.15.10 255.255.255.0
!
ip route 10.0.0.0 255.0.0.0 10.3.0.1
ip route 192.168.0.0 255.0.0.0 10.3.0.1
!
!
ip http server
!
!
!
!
control-plane
!
sccp local GigabitEthernet0/0
sccp ccm 10.15.15.1 identifier 1 version 4.1
!
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 101 register confer1
associate profile 103 register xcode1
!
!
dspfarm profile 103 transcode
codec g711ulaw
codec g711alaw
codec g729r8
maximum sessions 6
associate application SCCP
!
dspfarm profile 101 conference
codec g711ulaw
codec g711alaw
codec g729r8
maximum sessions 5
associate application SCCP
!
!
!
!
line con 0
exec-timeout 0 0
line aux 0

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Where to Go Next

line vty 0 4
session-timeout 300
exec-timeout 0 0
password
no login
!
scheduler allocate 20000 1000
!
end

Where to Go Next
Controlling Use of the Conference Soft Key
To block the functioning of the conference (Confrn) soft key without removing the key display, create
and apply an ephone template that contains the features blocked command. For more information, see
“Creating Templates” on page 881.
To remove the conference (Confrn) soft key from one or more phones, create and apply an ephone
template that contains the appropriate softkeys command. For more information, see “Customizing Soft
Keys” on page 829.

Additional References
The following sections provide references related to conferencing.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS voice configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Voice Command Reference
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Technical Support & Documentation http://www.cisco.com/techsupport
website contains thousands of pages of searchable
technical content, including links to products,
technologies, solutions, technical tips, and tools.
Registered Cisco.com users can log in from this page to
access even more content.

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Feature Information for Conferencing

Table 33 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 33 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 33 Feature Information for Conferencing

Cisco Unified CME

Feature Name Version Feature Information
Meet-me Conferences 4.1 Added support for hardware-based meet-me conferences
created by parties calling a designated conference number.
Multi-party Ad Hoc Conferencing 4.1 Added support for hardware-based Multi-party
Conferencing Enhancements which uses digital signal
processors (DSPs) to enhance ad hoc conferencing by
allowing more parties than software-based ad hoc
conferencing. Configuring multi-party ad hoc conferencing
disables three-party ad hoc conferencing.
Three-Party Ad Hoc Conferencing 4.0 • End-of-conference options were introduced.
• Phones connected in a three-way conference display
“Conference.”
3.2.2 Conference gain control for external calls was introduced.
3.2 Conference initiator drop-off control was introduced.
2.0 Support for software-based conferencing was introduced.

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Last Updated: March 26, 2007

This chapter describes the directory services support available in Cisco Unified Communications
Manager Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Directory Services”
section on page 707.

Contents
• Information About Directory Services, page 689
• How to Configure Directory Services, page 691
• Configuration Examples for Directory Services, page 701
• Additional References, page 706
• Feature Information for Directory Services, page 707

Information About Directory Services

To enable directory services, you should understand the following concepts:
• Local Directory, page 690
• External Directory, page 690
• Called-Name Display, page 690

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Local Directory
Cisco Unified CME automatically creates a local phone directory containing the telephone numbers that
are assigned in the directory number configuration of the phone. You can make additional entries to the
local directory in telephony services configuration mode. Additional entries can be nonlocal numbers
such as telephone numbers on other Cisco Unified CME systems used by your company.
When a phone user selects the Directories > Local Directory menu, the phone displays a search page
from Cisco Unified CME. After a user enters the search information, the phone sends the information to
Cisco Unified CME, which searches for the requested number or name pattern in the directory number
configuration and sends the response back to the phone, which displays the matched results. The phone
can display up to 32 directory entries. If a search results in more than 32 entries, the phone displays an
error message and the user must refine the search criteria to narrow the results.
The order of the names in the directory entries can display with first names first or last names first.
The local directory that is displayed on an IP phone is an XML page that is accessed through HTTP
without password protection. The directory HTTP service can be disabled to suppress the availability of
the local directory.
For configuration information, see the “Configuring Local Directory Service” section on page 691.

External Directory
Cisco Unified IP Phones can support URLs in association with the four programmable feature buttons
on IP phones, including the Directories button. Operation of these services is determined by the Cisco
Unified IP phone capabilities and the content of the referenced URL. Provisioning the directory URL to
select an external directory resource disables the Cisco Unified CME local directory service.

Called-Name Display
When phone agents answer calls for several different departments or people, it is often helpful for them
to see a display of the name, rather than the number, of the called party. For example, if order-entry
agents are servicing three catalogs with individual 800 numbers configured in one overlay ephone-dn
set, 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 called-name display feature can display either of the
following types of name:
• Name for a directory number in a local directory
• Name associated with an overlay directory number. Calls to the first directory number in a set of
overlay numbers will display a caller ID. Calls to the remaining directory numbers in the overlay set
will display the name associated with the directory number.

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How to Configure Directory Services

This section contains the following tasks:
• Configuring Local Directory Service, page 691
• SCCP: Defining a Name for a Directory Number, page 692
• SCCP: Adding an Entry to a Local Directory, page 693
• SCCP: Configuring External Directory Service, page 694
• SCCP: Enabling Called-Name Display, page 696
• Verifying Called-Name Display, page 697
• SIP: Defining a Name for a Directory Number, page 698
• SIP: Configuring External Directory Service, page 699
• Verifying Directory Services, page 700

Configuring Local Directory Service

To define the format for local directory names or block the local directory display on all phones, perform
the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. directory {first-name-first | last-name-first}
5. no service local-directory
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 4 directory {first-name-first | Defines the format for entries in the local directory.
last-name-first}
• Default is first-name-first.

Example:
Router(config-telephony)# directory
last-name-first
Step 5 no service local-directory Disables local directory service on IP phones.

Example:
Router(config-telephony)# no service
local-directory
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

SCCP: Defining a Name for a Directory Number

To define a name for a directory number in a local directory, perform the following steps.

Prerequisites
• Cisco CME 3.0 or a later version.
• Directory number for which you are defining a directory entry must already have a number assigned
by using the number (ephone- dn) command. For configuration information, see “SCCP: Creating
Directory Numbers” on page 158.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag
4. name name
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode.

Example:
Router(config)# ephone-dn 55
Step 4 name name Associates a name with this directory number. This name is
used for caller-ID displays and in the local directory
listings.
Example:
Router(config-ephone-dn)# name Smith, John • Must follow the name order that is specified with the
directory command.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

SCCP: Adding an Entry to a Local Directory

To add an entry to the local directory, perform the following steps.

Restrictions
If the directory entry being configured is to be used for called-name display, the number being
configured must contain at least one wildcard character.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. directory entry {entry-tag number name name | clear}
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 directory entry {entry-tag number name Creates a telephone directory entry that is displayed on an IP
name | clear} phone. Entries appear in the order in which they are entered.
• entry-tag—Unique sequence number that identifies this
Example: directory entry during all configuration tasks. Range is
Router(config-telephony)# directory entry 1 to 100.
1 5550111 name Sales
• If this name is to be used for called-name display, the number
associated with the names must contain at least one wildcard
character.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

SCCP: Configuring External Directory Service

To enable an external directory resource on supported Cisco Unified IP phones and disable local
directory services on those same phones, perform the following steps.

Prerequisites
To use a Cisco Unified Communications Manager directory as an external directory source for Cisco
Unified CME phones, the Cisco Unified Communications Manager must be made aware of the phones.
You must list the MAC addresses of the Cisco Unified CME phones in the Cisco Unified
Communications Manager and reset the phones from the Cisco Unified Communications Manager. It is
not necessary for you to assign ephone-dns to the phones or for the phones to register with Cisco Unified
Communications Manager.

Restrictions
Provisioning of the directory URL to select an external directory resource disables the
Cisco Unified CME local directory service.

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SUMMARY STEPS

1. enable
2. configure terminal
3. telephony -service
4. url {directory | service} url
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 url {directory | service} url Associates a URL with the programmable Directories and
Services feature buttons on supported Cisco Unified IP
phones in Cisco Unified CME.
Example:
Router(config-telephony)# url directory • Provisioning the directory URL to select an external
http://10.0.0.11/localdirectory directory resource disables the Cisco Unified CME
local directory service.
• Operation of these services is determined by the
Cisco Unified IP phone capabilities and the content of
the specified URL.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-telephony)# end

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SCCP: Enabling Called-Name Display

To enable called-name display, perform the following steps.

Prerequisites
• For directory numbers other than overlaid directory numbers—To display a name in the called-name
display, the name to be displayed must be defined in the local directory. See the “SCCP: Adding an
Entry to a Local Directory” section on page 693.
• For overlaid directory numbers—To display a name in the called-name display for a directory
number that is in a set of overlaid directory numbers, the name to be displayed must be defined. See
the “SCCP: Defining a Name for a Directory Number” section on page 692

Restrictions
• The service dnis overlay command can only be used to configure overlaid ephone-dns.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. service dnis dir-lookup
5. service dnis overlay
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)#

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Command or Action Purpose

Step 4 service dnis dir-lookup Specifies that incoming calls to a called number should display
the name that was defined for this directory number with the
directory entry command.
Example:
Router(config-telephony)# service dnis • If the service dnis dir-lookup and service dnis overlay
dir-lookup commands are both used in one configuration, the service
dnis dir-lookup command takes precedence.
Step 5 service dnis overlay (For overlaid directory numbers only.) Specifies that incoming
calls to a called number should display the name that was defined
for this directory number with the name command.
Example:
Router(config-telephony)# service dnis Note If the service dnis dir-lookup and service dnis overlay
overlay commands are both used in one configuration, the service
dnis dir-lookup command takes precedence.
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Verifying Called-Name Display

Step 1 Use the show running-config command to verify your configuration. Called-name display is shown in
the telephony-service part of the output.
Router# show running-config

telephony-service
service dnis overlay

Step 2 Use the show telephony-service directory-entry command to display current directory entries.
Router# show telephony-service directory-entry

directory entry 1 5550341 name doctor1

directory entry 2 5550772 name doctor1
directory entry 3 5550263 name doctor3

Step 3 Use the show telephony-service ephone-dn command to verify that you have used at least one wildcard
(period or .) in the ephone-dn primary or secondary number or to verify that you have entered a name
for the number.
Router# show telephony-service ephone-dn

ephone-dn 2
number 5002 secondary 200.
name catalogN
huntstop
call-forward noan 5001 timeout 8

Step 4 Use the show ephone overlay command to verify the contents of overlaid ephone-dn sets.
Router# show ephone overlay

ephone-1 Mac:0007.0EA6.353A TCP socket:[1] activeLine:0 REGISTERED

mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0

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IP:10.2.225.205 52486 Telecaster 7960 keepalive 2771 max_line 6

button 1: dn 11 number 60011 CH1 IDLE overlay
button 2: dn 17 number 60017 CH1 IDLE overlay
button 3: dn 24 number 60024 CH1 IDLE overlay
button 4: dn 30 number 60030 CH1 IDLE overlay
button 5: dn 36 number 60036 CH1 IDLE CH2 IDLE overlay
button 6: dn 39 number 60039 CH1 IDLE CH2 IDLE overlay
overlay 1: 11(60011) 12(60012) 13(60013) 14(60014) 15(60015) 16(60016)
overlay 2: 17(60017) 18(60018) 19(60019) 20(60020) 21(60021) 22(60022)
overlay 3: 23(60023) 24(60024) 25(60025) 26(60026) 27(60027) 28(60028)
overlay 4: 29(60029) 30(60030) 31(60031) 32(60032) 33(60033) 34(60034)
overlay 5: 35(60035) 36(60036) 37(60037)
overlay 6: 38(60038) 39(60039) 40(60040

SIP: Defining a Name for a Directory Number

To define name for a directory number on a SIP phone, perform the following steps.

Prerequisites
• Cisco CME 3.4 or a later version.
• Directory number for which you are defining a name must already have a number assigned by using
the number (voice register dn) command. For configuration information, see “SIP: Creating
Directory Numbers” on page 162.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register dn dn-tag
4. name name
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or a message-waiting indicator (MWI).
Example:
Router(config-register-global)# voice register
dn 17
Step 4 name name Associates a name with a directory number in
Cisco Unified CME and provides caller ID for calls
originating from a SIP phone.
Example:
Router(config-register-dn)# name Smith, John • Name must follow the order specified by using the
or directory (telephony-service) command.

Example:
Router(config-register-dn)# name John Smith
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-dn)# end

SIP: Configuring External Directory Service

To enable an external directory resource on supported Cisco Unified IP phones and disable local
directory services on those same phones, perform the following steps.

Prerequisites
Cisco CME 3.4 or a later version.

Restrictions
• Provisioning of the directory URL to select an external directory resource disables the
Cisco Unified CME local directory service.
• Supported only on Cisco Unified IP Phone 7960s and 7960Gs and Cisco Unified IP Phone 7940s
and 7940Gs.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. url {directory | service} url
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 url {directory | service} url Associates a URL with the programmable Directories and
Services feature buttons on supported Cisco Unified Ip
phones in Cisco Unified CME.
Example:
Router(config-register-global)# url directory • Provisioning the directory URL to select an external
http://10.0.0.11/localdirectory directory resource disables the Cisco Unified CME
Router(config-register-global)# url service
local directory service.
http://10.0.0.4/CCMUser/123456/urltest.html
• Operation of these services is determined by the
Cisco Unified IP phone capabilities and the content of
the specified URL.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end

Verifying Directory Services

To verify the configuration for local directory services, perform the following steps.

SUMMARY STEPS

1. show running-config
2. show telephony-service
3. show telephony-service directory-entry

DETAILED STEPS

Step 1 show running-config

This command displays the running configuration. Directory configuration commands are listed in the
telephony-service portion of the output.
Router# show running-config
.

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.
.
timeout busy 10
timeout ringing 100
caller-id name-only: enable
system message XYZ Company
web admin system name admin1 password admin1
web admin customer name Customer
edit DN through Web: enabled.
edit TIME through web: enabled.
Log (table parameters):
max-size: 150
retain-timer: 15
create cnf-files version-stamp Jan 01 2002 00:00:00
transfer-system full-consult
multicast moh 239.12.20.123 port 2000
fxo hook-flash
local directory service: enabled.

Step 2 show telephony-service

This command displays only the telephony-service configuration information.
Step 3 Use the show telephony-service directory-entry command to display the entries made using the
directory entry command.

Configuration Examples for Directory Services

This section contains the following examples:
• Local Directory, page 701
• Called-Name Display, page 702

Local Directory
The following example defines the naming order for the local directory on IP phones served by the
Cisco Unified CME router:
telephony-service
directory last-name-first

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

The following example disables the local directory on IP phones served by the Cisco Unified CME
router:
telephony-service
no service local-directory

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Called-Name Display
This section contains the following examples:
• First Ephone-dn in the Overlay Set: Example, page 702
• Directory Name for an Overlaid Ephone-dn Set: Example, page 702
• Directory Name for a Hunt Group with Overlaid Ephone-dns: Example, page 703
• Directory Name for Non-Overlaid Ephone-dns: Example, page 704
• Ephone-dn Name for Overlaid Ephone-dns: Example, page 705

First Ephone-dn in the Overlay Set: Example

The following example shows 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 18005550100
ephone-dn 2
name department1
number 18005550101
ephone-dn 3
name department2
number 18005550102

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
(18005550100). A call is made to the first ephone-dn (18005550100), 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 (18005550100). 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 (18005550101).

Directory Name for an Overlaid Ephone-dn Set: 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

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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

ephone 2
button 1o1,2,3
mac-address 2222.2222.2222

ephone 3
button 1o1,2,3
mac-address 3333.3333.3333

For more information about making directory entries, see the “Local Directory” section on page 690. For
more information about overlaid ephone-dns, see “Configuring Call-Coverage Features” on page 563.

Directory Name for a Hunt Group with Overlaid Ephone-dns: Example

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 Unified CME matches the hunt-group pilot secondary number
(555....), rings button 1 on one of the two phones, and displays “doctor1.”
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

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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

For more information about hunt-group behavior, see “Configuring Call-Coverage Features” on
page 563. Note that wildcards are used only in secondary numbers and cannot be used with primary
numbers. For more information about making directory entries, see the “Local Directory” section on
page 690. For more information about overlaid ephone-dns, see “Configuring Call-Coverage Features”
on page 563.

Directory Name for Non-Overlaid Ephone-dns: 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 doctor6.
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

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

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Configuration Examples for Directory Services

For more information about making directory entries, see the “Local Directory” section on page 690.

Ephone-dn Name for Overlaid Ephone-dns: 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

For more information about overlaid ephone-dns, see “Configuring Call-Coverage Features” on
page 563.

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Directory Services

Feature Information for Directory Services

Table 34 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 34 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 34 Feature Information for Directory Services

Cisco Unified CME

Feature Name Version Feature Information
Called-Name Display 3.2 Called-name display was introduced.
Local Directory Service 4.0(2) Added support for transferring a call directly to a selected
External Directory Service number listed in the directory. If directory transfer is not
supported, the user must press Transfer and then use the
keypad to manually enter the number of the monitored line
to transfer the incoming call.
3.4 Added support of directory services for SIP phones directly
connected in Cisco Unified CME.
3.0 The ability to add local directory entries in addition to
those that are automatically added from phone
configurations was introduced. Authentication for local
directory display was introduced.
2.1 The ability to block the display of the local directory on
phones was introduced.
2.0 The specification of name format in the local directory was
introduced.

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Last Updated: March 26, 2007

This chapter describes the do-not-disturb feature in Cisco Unified Communications Manager Express
(Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Do Not Disturb”
section on page 716.

Contents
• Information About Do Not Disturb, page 709
• How to Configure Do Not Disturb, page 710
• Configuration Examples for Do Not Disturb, page 714
• Where to Go Next, page 714
• Additional References, page 715
• Feature Information for Do Not Disturb, page 716

Information About Do Not Disturb

To configure do not disturb, you should understand the following concept:
• Do Not Disturb, page 709

Do Not Disturb
The Do Not Disturb (DND) feature allows you to set your phone to forward calls without ringing the
phone. Enable DND service using the DND soft key on Cisco Unified IP phones that support soft keys.
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. 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.

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Pressing the DND soft key during an incoming call forwards the call to the call-forward no-answer
destination if call-forward no-answer is enabled. If call-forward no-answer is not enabled, pressing the
DND soft key disables the ringer and visual alerting.
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.
In Cisco CME 3.2.1 and later versions, DND can be blocked from phones with the feature-ring function.
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-ring function.
You can disable the DND function on feature-ring lines. In the preceding example, salespeople could
activate DND on their phones and still hear calls to their private lines.

How to Configure Do Not Disturb

This section contains the following tasks:
• SCCP: Blocking Do Not Disturb, page 710
• SCCP: Verifying Do Not Disturb, page 711
• SIP: Configuring Do Not Disturb, page 712

SCCP: Blocking Do Not Disturb

To block DND on phones that have buttons configured for feature ringing, perform the following steps.
DND is enabled by using the DND soft key on Cisco Unified IP phones that support soft keys.

Prerequisites
• Cisco Unified 3.2.1 or a later version.
• Call-forwarding no-answer must be set for a phone to use DND to forward calls. No other
configuration is necessary for basic DND.

Restrictions
• Phone users cannot enable DND for a shared line in a hunt group. The soft key displays in the idle
and ringing states but does not enable DND for shared lines in hunt groups.

SUMMARY STEPS

1. enable
2. configure terminal

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3. ephone phone-tag
4. no dnd feature-ring
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: the ephone to be configured.
Router(config)# ephone 10
Step 4 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 5 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

SCCP: Verifying Do Not Disturb

show ephone dnd
Use this command to display a list of SCCP phones that have DND enabled.
Router# show ephone dnd

ephone-1 Mac:0007.0EA6.353A TCP socket:[1] activeLine:0 REGISTERED

mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:1.2.205.205 52486 Telecaster 7960 keepalive 2729 max_line 6 DnD
button 1: dn 11 number 60011 CH1 IDLE

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SIP: Configuring Do Not Disturb

To configure the Do-not-Disturb (DND) feature on a SIP phone with softkeys and use DND to forward
calls when a a directory number does not answer, perform the following steps.

Note You can enable the Do-Not-Disturb (DND) soft key on one or more SIP phones by using the
dnd-control command in voice register template configuration mode. For information about
configuring templates, see “Creating Templates” on page 881.

Prerequisites
Cisco CME 3.4 or a later version.

Restrictions
If the DND soft key is disabled by a user on the Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G,
7961GE, 7970G, or 7971GE, it does not display after the phone is reset or restarted. DND must be
enabled both in Cisco Unified CME and by using the DND soft key on the phone.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register pool pool-tag
4. dnd
5. exit
6. voice register dn dn-tag
7. call-forward b2bua noan directory- number timeout seconds
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
parameters for SIP phone to be configured.
Example:
Router(config)# voice register pool 1

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Command or Action Purpose

Step 4 dnd Enables DND.
Note If call-forward-no-answer is not configured for the
Example: extension, pressing the DND key mutes the ringer
Router(config-register-pool)# dnd-control until the caller terminates the call.
Step 5 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-register-pool)# exit
Step 6 voice register dn dn-tag Enters voice register dn mode to define a directory number
for a SIP phone, intercom line, voice port, or an MWI.
Example:
Router(config)# voice register dn 1
Step 7 call-forward b2bua noan directory- number Enables call forwarding for a SIP back-to-back user agent
timeout seconds so that incoming calls to an extension that does not answer
will be forwarded to the designated directory number.
Example: • In Cisco CME 3.4 and Cisco Unified CME 4.0, this
Router(config-register-dn)# call-forward b2bua command is also available in voice register pool
noan 5010 timeout 10
configuration mode. The information under voice
register dn takes precedence over the information
configured under voice register pool.
• timeout seconds—Duration that a call can ring before
it is forwarded to the destination directory number.
Range: 3 to 60000. Default: 20
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-pool)# end

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Configuration Examples for Do Not Disturb

Configuration Examples for Do Not Disturb

In 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

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

ephone 2
button 1f2
button 2o10,11
no dnd feature-ring

Where to Go Next
Agent Status Control for Ephone Hunt Groups and Cisco Unified CME B-ACD
Ephone hunt group agents can control their ready/not-ready status (their ability to receive calls) using
the DND function or the HLog function of their phones. When they use the DND soft key, they do not
receive calls on any extension on their phones. When they use the HLog soft key, they do not receive
calls on hunt group extensions, but they do receive calls on other extensions. For more information on
agent status control and the HLog function, see “Configuring Call-Coverage Features” on page 563.

Call Forwarding
To use the DND soft key to forward calls, enable call-forwarding no-answer. See “Configuring Call
Transfer and Forwarding” on page 499.

Feature Access Codes (FACs)

DND can be activated and deactivated using a feature access code (FAC) instead of the DND soft key
when standard or custom FACs are enabled. The following is the standard FAC for DND:
• DND—**7
See “Configuring Feature Access Codes” on page 729.

Soft-Key Display
You can remove or change the position of the DND soft key. See “Customizing Soft Keys” on page 829.

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Do Not Disturb

Feature Information for Do Not Disturb

Table 35 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 35 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 35 Feature Information for Do Not Disturb

Cisco Unified CME

Feature Name Version Feature Information
Do Not Disturb 3.4 Added support for Do-not-disturb (DND) soft key on
supported Cisco Unified IP phones that are connected to a
Cisco Unified CME router and running SIP.
3.2.1 DND bypass for feature-ring phones was introduced.
3.2 DND was introduced.

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 Configuring Extension Mobility

Last Updated: July 5, 2007

First Published: June 18, 2007

This module describes features in Cisco Unified Communications Manager Express

(Cisco Unified CME) that provide support for phone mobility for end users.

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Extension Mobility”
section on page 728.

Contents
• Information About Extension Mobility, page 717
• How to Enable Extension Mobility, page 718
• Configuration Examples for Extension Mobility, page 725
• Where to Go Next, page 726
• Additional References, page 727
• Feature Information for Extension Mobility, page 728

Information About Extension Mobility

To configure interoperability, you should understand the following concepts:
• Extension Mobility, page 717

Extension Mobility
Extension mobility in Cisco Unified CME 4.2 and later versions provides the benefit of phone mobility
for end users.

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A user login service allows phone users to temporarily access a physical phone other than their own
phone and utilize their personal settings, such as directory number, speed-dial lists, and services, as if
the phone is their own desk phone. The phone user can make and receive calls on that phone using the
same personal directory number as is on their own desk phone.
Each Cisco Unified IP phone that is enabled for extension mobility is configured with a logout profile.
This profile determines the default appearance of a phone that is enabled for extension mobility when
there is no phone user logged into that phone. Minimally, the logout profile allows calls to emergency
services such as 911. A single logout profile can be applied to multiple phones.
After a Cisco Unified IP phone that is enabled for extension mobility boots up, the Services button on
the phone is configured with a login service URL hosted by Cisco Unified CME that points to the
extension mobility Login page.
A phone user logs in to a Cisco Unified IP phone that is enabled for extension mobility by pressing the
Services button or a Unified CCX agent can log in using a Unified CCX Cisco Agent Desktop. User
authentication and authorization is performed by Cisco Unified CME. If the login is successful,
Cisco Unified CME retrieves the appropriate user profile, based on user name and password match, and
replaces the phone’s logout profile with the user profile.
After the phone user is logged in, the service URL points to a logout URL hosted by Cisco Unified CME
to provide a logout prompt on the phone. Logging into a different device automatically closes the first
session and start a new session on the new device. When a phone user is not logged in to any phone,
incoming calls to the phone user’s directory number are sent to the phone user’s voice mailbox.
For button appearance, extension mobility associates directory numbers, then speed-dial numbers in the
logout profile or user profile to phone buttons in a sequence. If the profile contains more numbers than
there are buttons on the physical phone to which the profile is downloaded, the remaining numbers in
the profile are ignored.
For configuration information, see the “How to Enable Extension Mobility” section on page 718.

How to Enable Extension Mobility

Perform the following tasks to enable extension mobility in Cisco Unified CME:
• Configuring a Logout Profile for an IP Phone, page 719 (required)
• Enabling an IP Phone for Extension Mobility, page 721 (required)
• Configuring a User Profile, page 723 (required)

Prerequisites
• Cisco Unified CME 4.2 or a later version.

Restrictions
• Extension mobility on remote Cisco Unified CME routers is not supported; a phone user can log into
any local Cisco Unified IP phone only.

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Configuring a Logout Profile for an IP Phone

To create a logout profile to define the default appearance for a Cisco Unified IP phone that is enabled
for extension mobility, perform the following steps.

Prerequisites
• All directory numbers to be included in a logout profile or a voice-user profile must be already
configured in Cisco Unified CME. For configuration information, see “Configuring Phones to Make
Basic Calls” on page 147.

Restrictions
• For button appearance, extension mobility associates directory numbers, then speed-dial definitions
in the logout profile or voice-user profile to phone buttons in a sequence beginning with numbers,
followed by speed dials. If the profile contains more directory numbers and speed-dial numbers than
there are buttons on the physical phone to which the profile is downloaded, not all numbers will be
downloaded to buttons.
• The first number to be configured for line appearance cannot be a monitored directory number.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice logout-profile tag
4. username username password password
5. number number type type
6. speed-dial speed-tag number [label label] [blf]
7. pin pin
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice logout-profile profile-tag Enters voice logout-profile configuration mode for the
purpose of creating a logout profile to define the default
appearance for a Cisco Unified IP phone enabled for
Example:
Router(config)# voice logout-profile 1
extension mobility.
• profile-tag—Unique number that identifies this profile
during configuration tasks. Range: 1 to maximum
number of phones supported by the Cisco Unified CME
router. Type ? to display the maximum number.
Step 4 username name password password Creates credential to be used by a TAPI phone device to log
into a Cisco Unified CME.
Example: • name—Alphanumeric string.
Router(config-logout-profile)# user 23C2-8
password 43214
• password—Alphanumeric string.
Step 5 number number[,...number] type type Creates line definition.
• number—Directory number to be associated with and
Example: displayed next to a button on a Cisco Unified IP phone
Router(config-logout-profile)# number 3001 type that is configured with this profile.
silent-ring
Router(config-logout-profile)# number 3002 type • [, ...number]—(Optional) For overlay lines only, with
beep-ring or without callwaiting. The directory number that is the
Router(config-logout-profile)# number 3003 type far left in command list is the highest priority. Can
feature-ring contain up to 25 numbers. Individual numbers must be
Router(config-logout-profile)# number 3004 type
monitor-ring
separated by commas (,).
Router(config-logout-profile)# number 3005,3006 • type type—Denotes characteristics to be associated
type overlay
with this line. Type ? for list of options.
Router(config-logout-profile)# number 3007,3008
type cw-overly
Step 6 speed-dial speed-tag number [label label] [blf] Creates speed-dial definition.
• speed-tag—Unique sequence number that identifies a
Example: speed-dial definition during configuration tasks.
Router(config-logout-profile)# speed-dial 1 Range: 1 to 36.
2001
Router(config-logout-profile)# speed-dial 2 • number—Digits to be dialed when the speed-dial
2002 blf button is pressed.
• label label—(Optional) String that contains identifying
text to be displayed next to the speed-dial button.
Enclose the string in quotation marks if the string
contains a space.
• blf—(Optional) Enables Busy Lamp Field (BLF)
monitoring for a speed-dial number.

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Command or Action Purpose

Step 7 pin pin Sets a personal identification number (PIN) to be used by a
phone user to disable the call blocking configuration for a
Cisco Unified IP phone on which this profile is
Example:
Router(config-logout-profile)# pin 1234
downloaded.
• pin—Numeric string containing four to eight digits.
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-logout-profile)# end

Enabling an IP Phone for Extension Mobility

To enable the Extension Mobility feature on an individual Cisco Unified IP phone in
Cisco Unified CME, perform the following steps.

Prerequisites
• Logout profile to be assigned to a phone must be configured in Cisco Unified CME.

Restrictions
• Extension mobility is not supported on Cisco Unified IP phones without phone screens.
• Extension mobility is not supported for SIP IP phones.
• Extension mobility is not supported for analog devices.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. mac-address mac-address
5. type phone-type
6. logout-profile profile-tag
7. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone phone-tag Enables phone configuration mode.
• phone-tag—Unique number that identifies this phone
Example: during configuration tasks. Range is 1 to maximum
Router(config)# ephone 1 number supported phones, where maximum is platform
and version dependent and defined by using the
max-ephone command.
Step 4 mac-address mac-address Associates a physical phone with this ephone configuration.
• mac-address—MAC address of phone, which is found
Example: on a sticker located on the bottom of the phone.
Router(config-ephone)# mac-address
000D.EDAB.3566
Step 5 logout-profile profile-tag Enables Cisco Unified IP phone for extension mobility and
assigns a logout profile to this phone.
Example: • tag—Unique identifier of logout profile to be used
Router(config-ephone)# logout-profile 1 when no phone user is logged in to this phone. This tag
number corresponds to a tag number created when this
logout profile was configured by using the voice
logout-profile command.
Step 6 type phone-type Defines a phone type for the phone being configured.

Example:
Router(config-ephone)# type 7960
Step 7 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone)# end

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Configuring a User Profile

To configure a user profile for a phone user who logs into a Cisco Unified IP phone that is enabled for
extension mobility, perform the following steps.

Note Templates created using the ephone-template and ephone-dn-template commands can be applied to a
user profile for extension mobility.

Prerequisites
• All directory numbers to be included in a logout profile or voice-user profile must be already
configured in Cisco Unified CME. For configuration information, see “Configuring Phones to Make
Basic Calls” on page 147.

Restrictions
• For button appearance, extension mobility associates directory numbers, then speed-dial definitions
in the logout profile or voice-user profile to phone buttons in a sequence beginning with numbers,
followed by speed dials. If the profile contains more directory numbers and speed-dial numbers than
there are buttons on the physical phone to which the profile is downloaded, not all numbers will be
downloaded to buttons.
• The first number to be configured for line appearance cannot be a monitored directory number.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice user-profile profile-tag
4. name username password password
5. number number type type
6. speed-dial speed-tag number [label label] [blf]
7. pin pin
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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How to Enable Extension Mobility

Command or Action Purpose

Step 3 voice user-profile profile-tag Enters voice user-profile configuration mode for the
purpose of configuring a user profile for extension mobility.
Example: • profile-tag—Unique number that identifies this profile
Router(config)# voice user-profile 1 during configuration tasks. Range: 1 to three times the
maximum number supported phones, where maximum
is platform dependent. Type ? to display value.
Step 4 name username password password Creates credential to be authenticated by
Cisco Unified CME before allowing the phone user to log
into a Cisco Unified IP phone phone enabled for extension
Example:
Router(config-user-profile)# name me password
mobility.
pass123 • username—Name of authorized user.
• password—Password for authorized user.
Step 5 number number[,...number] type type Creates line definition.
• number—Directory number to be associated with and
Example: displayed next to a button on a phone that is configured
Router(config-user-profile)# number 2001 type with this profile.
silent-ring
Router(config-user-profile)# number 2002 type • [, ...number]—(Optional) For overlay lines only, with
beep-ring or without call waiting. The directory number that is far
Router(config-user-profile)# number 2003 type left in the command list is given the highest priority.
feature-ring Can contain up to 25 numbers. Individual numbers
Router(config-user-profile)# number 2004 type
monitor-ring
must be separated by commas (,)
Router(config-user-profile)# number 2005,2006 • type type—Denotes characteristics to be associated
type overlay
with this line. Type ? for list of options.
Router(config-user-profile)# number 2007,2008
type cw-overly
Step 6 speed-dial speed-tag number [label label] [blf] Creates speed-dial definition.
• speed-tag—Unique sequence number that identifies a
Example: speed-dial definition during configuration tasks.
Router(config-user-profile)# speed-dial 1 3001 Range: 1 to 36.
Router(config-user-profile)# speed-dial 2 3002
blf • number—Digits to be dialed when the speed-dial
button is pressed.
• label label—(Optional) String that contains identifying
text to be displayed next to the speed-dial button.
Enclose the string in quotation marks if the string
contains a space.
• blf—(Optional) Enables Busy Lamp Field (BLF)
monitoring for a speed-dial number.

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Command or Action Purpose

Step 7 pin pin Sets a personal identification number (PIN) to be used by a
phone user to disable the call blocking configuration for a
Cisco Unified IP phone on which this profile is
Example:
Router(config-user-profile)# pin 12341
downloaded.
• pin—Numeric string containing four to eight digits.
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-user-profile)# end

Configuration Examples for Extension Mobility

This section contains the following configuration examples:
• Logout Profile: Example, page 725
• Enabling an IP Phone for Extension Mobility: Example, page 725
• Voice-User Profile: Example, page 726

Logout Profile: Example

The following example shows the configuration for a logout profile that defines the default appearance
for a Cisco Unified IP phone that is enabled for extension mobility. Which lines and speed-dial buttons
in this profile are configured on a phone depends on phone type. For example, for a Cisco Unified
IP Phone 7970, all buttons are configured according to logout profile1. However, if the phone is a
Cisco Unified IP Phone 7960, all six lines are mapped to phone buttons and the speed dial is ignored
because there is no button available for speed dial.
pin 9999
user 23C2-8 password 43214
number 3001 type silent-ring
number 3002 type beep-ring
number 3003 type feature-ring
number 3004 type monitor-ring
number 3005,3006 type overlay
number 3007,3008 type cw-overly
speed-dial 1 2000
speed-dial 2 2001 blf

Enabling an IP Phone for Extension Mobility: Example

The following example shows the ephone configurations for three IP phones. All three phones are
enabled for extension mobility and share the same logout profile number 1, to be downloaded when these
phones boot and when no phone user is logged into the phone:
ephone 1
mac-address 000D.EDAB.3566
type 7960
logout-profile 1

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Where to Go Next

ephone 2
mac-address 0012.DA8A.C43D
type 7970
logout-profile 1

ephone 3
mac-address 1200.80FC.9B01
type 7911
logout-profile 1

Voice-User Profile: Example

The following example shows the configuration for a voice-user profile to be downloaded when a phone
user logs into a Cisco Unified IP phone that is enabled for extension mobility. Which lines and
speed-dial buttons in this profile are configured on a phone after the user logs in depends on phone type.
For example, if the user logs into a Cisco Unified IP Phone 7970, all buttons are configured according
to voice-user profile1. However, if the phone user logs into a Cisco Unified IP Phone 7960, all six lines
are mapped to phone buttons and the speed dial is ignored because there is no button available for speed
dial.
pin 12345
user me password pass123
number 2001 type silent-ring
number 2002 type beep-ring
number 2003 type feature-ring
number 2004 type monitor-ring
number 2005,2006 type overlay
number 2007,2008 type cw-overly
speed-dial 1 3001
speed-dial 2 3002 blf

Where to Go Next
• If you created a new or modified an existing logout or voice-user profile, you must restart the phones
to propagate the changes. See “Resetting and Restarting Phones” on page 257.
• If you enabled one or more Cisco Unified IP phones for extension mobility, generate a new
configuration file and restart the phones. See “Generating Configuration Files for Phones” on
page 245.

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Extension Mobility

Feature Information for Extension Mobility

Table 36 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified Communications Manager Express and Cisco IOS Software Version Compatibility
Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 36 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 36 Feature Information for Extension Mobility

Cisco Unified CME

Feature Name Version Modification
Extension Mobility 4.2 Provides the benefit of phone mobility for end users by
enabling the user to log into any local Cisco Unified IP
phone that is enabled for extension mobility.

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 Configuring Feature Access Codes

Last Updated: March 26, 2007

This chapter describes the feature access codes support in Cisco Unified Communications Manager
Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Feature Access
Codes” section on page 735.

Contents
• Information About Feature Access Codes, page 729
• How to Configure Feature Access Codes, page 731
• Configuration Examples for Feature Access Codes, page 733
• Additional References, page 734
• Feature Information for Feature Access Codes, page 735

Information About Feature Access Codes

To enable Feature Access Codes, you should understand the following concept:
• Feature Access Codes, page 730

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Information About Feature Access Codes

Feature Access Codes

Feature Access Codes (FACs) are special patterns of characters that are dialed from a telephone keypad
to invoke particular features. For example, a phone user might press **1, then press 2345 to forward all
incoming calls to extension 2345.
Typically, FACs are invoked using a short sequences of digits that are dialed using the keypad on an
analog phone, while IP phones users select soft keys to invoke the same features. In Cisco Unified CME
4.0 and later, the same FACs that are available for analog phones can be enabled on IP phones. This
allows phone users to select a particular feature or activate/deactivate a function in the same manner
regardless of phone type.
FACs are disabled on IP phones until they are explicitly enabled. You can enable all standard FACs for
all SCCP phones registered in Cisco Unified CME or you can define a custom FAC or alias to enable
one or more individual FACs.
All FACs except the call-park FAC are valid only immediately after a phone is taken off hook. The
call-park FAC is considered a transfer to a call-park slot and therefore is only valid after the Trnsfer soft
key (IP phones) or hookflash (analog phones) is used to initiate a transfer.
Table 37 contains a list of the standard predefined FACs.

Table 37 Standard FACs

Standard FAC Description

**1 plus optional extension number Call forward all.
**2 Call forward all cancel.
**3 Pick up local group.
**4 plus group number Pick up a ringing call in the specified pickup group. Specified
pickup group must already configured in Cisco Unified
CME.
**5 plus extension number Pick up direct extension.
**6 plus optional park-slot number Call park, if the phone user has an active call and if the phone
user presses the Transfer soft key (IP phone) or hookflash
(analog phone) before dialing this FAC. Target park slot must
be already configured in Cisco Unified CME.
**7 Do not disturb.
**8 Redial.
**9 Dial voice-mail number.
*3 plus hunt group pilot number Join ephone-hunt group. If multiple hunt groups have been
created that allow dynamic membership, the hunt group to be
joined is identified by its pilot number.
*4 Activate or deactivate hunt group logout functionality to
toggle between ready/not-ready status of an extension when
an hunt group agent is off-hook.

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Table 37 Standard FACs (continued)

Standard FAC Description

*5 Activate or deactivate phone-level hunt group logout to
toggle between ready/not-ready status of all extensions on a
individual phone that is a member of an ephone hunt group
when the phone is idle.
#3 Leave ephone-hunt group. Telephone or extension number
must already be configured as a dynamic member of a hunt
group.

How to Configure Feature Access Codes

This section contains the following tasks:
• Enabling Feature Access Codes, page 731
• Verifying Feature Access Codes, page 732

Enabling Feature Access Codes

To enable standard FACs or create custom FACs, perform the following steps:

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. fac {standard | custom {alias alias-tag custom-fac to existing-fac [extra-digits]} | feature
custom-fac}}
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 4 fac {standard | custom {alias alias-tag Enables standard FACs or creates a custom FAC or alias.
custom-fac to existing-fac [extra-digits]} |
feature custom-fac}} • standard—Enables standard FACs for all phones.
• custom—Creates a custom FAC for a FAC type.
Example: • alias—Creates a custom FAC for an existing FAC or a
Router(config-telephony)# fac custom callfwd existing FAC plus extra digits.
*#5
• alias-tag—Unique identifying number for this alias.
Range: 0 to 9.
• custom-fac—User-defined code to be dialed using the
keypad on an IP or analog phone. Custom FAC can be
up to 256 characters long and contain numbers 0 to 9
and * and #.
• to—Maps custom FAC to specified target.
• existing-fac—Already configured custom FAC that is
automatically dialed when the phone user dials the
custom FAC being configured.
• extra-digits—(Optional) Additional digits that are
automatically dialed when the phone user dials the
custom FAC being configured.
• feature—Predefined alphabetic string that identifies a
particular feature or function. Type ? for a list.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Verifying Feature Access Codes

To verify the FAC configuration, perform the following step.

Step 1 show telephony-service fac

This command displays a list of FACs that are configured on the Cisco Unified CME router. The
following example shows the output when standard FACs are enabled:
Router# show telephony-service fac

telephony-service fac standard

callfwd all **1
callfwd cancel **2
pickup local **3
pickup group **4
pickup direct **5
park **6
dnd **7
redial **8
voicemail **9
ephone-hunt join *3
ephone-hunt cancel #3
ephone-hunt hlog *4

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ephone-hunt hlog-phone *5

The following example shows the output when custom FACs are configured:
Router# show telephony-service fac

telephony-service fac custom

callfwd all #45
alias 0 #1 to **4121
alias 1 #2 to **4122
alias 4 #4 to **4124

Configuration Examples for Feature Access Codes

This section contains the following configuration example:
• FAC: Example, page 733

FAC: Example
The following example shows how to enable standard FACs for all phones:
Router# telephony-service
Router(config-telephony)# fac standard
fac standard is set!
Router(config-telephony)#

The following example shows how the standard FAC for the Call Forward All feature is changed to a
custom FAC (#45). Then an alias is created to map a second custom fac to #45 plus an extension (1111).
The custom FAC (#44) allows the phone user to press #44 to forward all calls all calls to extension 1111,
without requiring the phone user to dial the extra digits that are the extension number.
Router# telephony-service
Router(config-telephony)# fac custom callfwd all #45
fac callfwd all code has been configured to #45
Router(config-telephony)# fac custom alias 0 #44 to #451111
fac alias0 code has been configurated to #44!
alias0 map code has been configurated to #451111!

The following example shows how to define an alias for the group pickup of group 123. The alias
substitutes the digits #4 for the standard FAC for group pickup (**4) and add s the the group number
(123) to the dial pattern. Using this custom FAC, a phone user can dial #4 to pick up a ringing call in
group 123, instead of dialing the standard FAC **4 plus the group number 123.
Router# telephony-service
Router(config-telephony)# fac custom alias 5 #4 to **4123

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Feature Access Codes

Feature Information for Feature Access Codes

Table 38 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 38 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 38 Feature Information for Feature Access Codes

Cisco Unified CME

Feature Name Version Feature Information
Feature Access Codes 4.0 Feature access codes (FACs) were introduced.

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 Configuring Fax Relay

First Published: June 18, 2007

This module describes how to enable Skinny Client Control Protocol (SCCP) Fax Relay for analog
foreign exchange service (FXS) ports under the control of Cisco Unified CME.

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Fax Relay” section
on page 744.

Contents
• Prerequisites for Fax Relay, page 737
• Restrictions for Fax Relay, page 738
• Information About Fax Relay, page 738
• How to Configure Fax Relay, page 740
• Configuration Examples for Fax Relay, page 742
• Additional References, page 743
• Feature Information for Fax Relay, page 744

Prerequisites for Fax Relay

• Cisco Unified CME 4.0(3) or a later version.
• Cisco IOS Release 12.4(11)T or a later release.
• SCCP Telephony Control (STC) application is enabled.

Note If your voice gateway is a separate router than the Cisco Unified CME router, it must use an IP
voice image of Cisco IOS Release 12.4(11)T or later.

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Restrictions for Fax Relay

Note For Cisco Unified CME versions before Cisco Unified CME, 4.0(3), there are two manually-controlled
options for setting up facsimiles:
• Fax Gateway Protocol
– Configure the Cisco VG 224, FXS port, or analog telephone adaptor (ATA) to use H.323 or
Session Initiation Protocol (SIP) with a specific fax relay protocol. See the Cisco IOS Fax and
Modem Services over IP Application Guide.
• G.711 Fax Pass-Through with SCCP
– This is the default setup for facsimile on Cisco VG 224 and FXS ports before
Cisco Unified CME 4.0(3). See the Cisco IOS Fax and Modem Services over IP Application
Guide.

Restrictions for Fax Relay

• RFC2833 dual tone multifrequency (DTMF) digit relay under Cisco Unified CME for SCCP FXS
ports is not supported.
• SCCP FXS ports under Cisco Unified CME control do not natively support RFC2833 DTMF-relay.
However, Cisco Unified CME can support conversion of DTMF digits to and from RFC2833
DTMF-relay on its H323 and SIP interfaces when used with SCCP-controlled FXS ports.
• Cisco Fax Relay is only supported on those Cisco IOS gateways and network modules listed in
Table 39, Supported Gateways, Modules, and VICs for Fax Relay.

Information About Fax Relay

To configure the fax relay feature, you should understand the following:
• Fax Relay and Equipment, page 738
• Feature Design of Cisco Fax Relay, page 739

Fax Relay and Equipment

• The fax relay feature supports the use of existing customer premises equipment (CPE) in voice
networks by allowing legacy analog phones attached to a Cisco IOS gateway to be controlled by
Cisco Unified CME, and by providing feature interoperability between analog and IP endpoints.
• The voice gateway can be the same router that is being used for Cisco Unified CME or it may be a
separate router (for example, the Cisco VG 224).
• The fax relay feature facilitates replacement of the PSTN time-division multiplexing (TDM)
infrastructure with VoIP.

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Information About Fax Relay

Feature Design of Cisco Fax Relay

Cisco Fax Relay is a proprietary fax relay implementation that uses Real-time Transport Protocol (RTP)
to transport fax data. It is the default fax relay type on Cisco voice gateways. The fax relay feature
provides enhanced supplementary feature capability on analog ports connected to a Cisco integrated
services router (ISR) or Cisco VG 224 analog gateway. Calls through the analog FXS ports are
controlled by a Cisco Unified CME system.
Before the introduction of SCCP-enhanced features, SCCP gateways supported fax pass-through only.
SCCP-enhanced features add support for Cisco Fax Relay and Super Group 3 (SG3) to G3 fax relay. This
feature allows the fax stream between two SG3 fax machines to negotiate down to G3 speeds (less than
14.4 kbps) allowing SG3 fax machines to interoperate over fax relay with G3 fax machines.
The SCCP telephony control (STC) application on the Cisco voice gateway presents the locally attached
analog telephones as individual endpoints to the call-control system, which allows the analog phones to
be controlled in the same way as IP phones. With this capability, gateway-attached endpoints share the
same telephony features that are available on IP phones directly connected to Cisco Unified CME.
SCCP-enhanced features provide analog endpoint to analog endpoint interoperability within the IP
telephony network.
Figure 45 shows a multisite deployment of the fax relay feature in a Cisco Unified CME topology.

Figure 45 Cisco Unified CME Fax Relay Deployment

VoIP WAN

Cisco gateway FXS Cisco Unified CME Cisco Unified CME FXS Cisco gateway
FXS
FXS V V
PSTN
LAN 1 LAN 2

IP IP

230565
SCCP

For information on configuring gateway-controlled fax relay features, see the “How to Configure Fax
Relay” section on page 740.

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How to Configure Fax Relay

Supported Gateways, Modules, and Voice Interface Cards for Fax Relay
Table 39 lists supported gateways, modules, and voice interface cards (VICs).

Table 39 Supported Gateways, Modules, and VICs for Fax Relay

Network Modules and

Gateways Extension Modules Expansion Modules VICs
• Cisco 2801 — • NM-HD-1V • VIC2-2FXS
• Cisco 2811 • NM-HD-2V • VIC-4FXS/DID
• Cisco 2821 • NM-HD-2VE • VIC2-2BRI-NT/TE
• Cisco 2851
• Cisco 3825
• Cisco 3845
• Cisco 2801 • EVM-HD • EVM-HD-8FXS/DID —
• Cisco 2821 • EM-3FXS/4FXO
• Cisco 2851 • EM-HDA-8FXS
• Cisco 3825 • EM-4BRI-NT/TE
• Cisco 3845
• Cisco 2801 — • NM-HDV2 • VIC2-2FXS
• Cisco 2811 • NM-HDV2-1T1/E1 • VIC-4FXS/DID
• Cisco 2821 • NM-HDV2-2T1/E1 • VIC2-2BRI-NT/TE
• Cisco 2851
• Cisco 3825
• Cisco 3845
• Cisco VG 224 — — —

How to Configure Fax Relay

This section contains the following tasks:
• Configuring Fax Relay, page 740 (required)
• Verifying and Troubleshooting Fax Relay Configuration, page 742 (optional)

Configuring Fax Relay

To configure the fax relay features on Cisco Unified CME, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal

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3. voice service voip

4. fax protocol cisco
5. fax-relay sg3-to-g3
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip Enters voice service configuration mode and specifies VoIP
encapsulation.
Example:
Router(config)# voice service voip
Step 4 fax protocol cisco Specifies the Cisco-proprietary Fax Protocol as the fax
protocol for SCCP analog endpoints.
Example: • This command is enabled by default.
Router(config-voi-serv)# fax protocol cisco
Step 5 fax-relay sg3-to-g3 (Optional) Enables the fax stream between two SG3 fax
machines to negotiate down to G3 speeds.
Example:
Router(config-voi-serv)# fax relay sg3-to-g3
Step 6 exit Exits the current configuration mode.

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

Troubleshooting Tips
The following commands can help troubleshoot SCCP fax relay features:
• debug voip application stcapp all
• debug voip vtsp all

Note For more information on these and other commands, see the Cisco IOS Voice Command Reference,
Cisco IOS Debug Command Reference, Release 12.4T, Cisco Unified Communications Manager
Express Command Reference, and Cisco IOS Configuration Fundamentals Command Reference,
Release 12.4.

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Configuration Examples for Fax Relay

Verifying and Troubleshooting Fax Relay Configuration

To verify the configuration of Cisco Fax Relay, use the show-running config command. Sample output
is located in the “Configuration Examples for Fax Relay” section on page 742.
Use the following commands to verify and troubleshoot SCCP gateway-controlled Fax Relay:
• show voice call summary—Displays fax relay voice port settings.
• show voice dsp—Displays fax relay digital signal processor (DSP) channel status.
• debug voip application stcapp all— Displays SCCP telephony control (STC) application fax relay
information.
• debug voip dsm all—Displays fax relay DSP stream manager (DSM) messages.
• debug voip dsmp all—Displays fax relay distributed stream media processor (DSMP) messages.
• debug voip hpi all—Displays gateway DSP fax relay information on RTP packet events.
• debug voip vtsp all—Displays gateway voice telephony service provider (VTSP) debugging
information for fax calls.

Note For more information on these and other commands, see the Cisco IOS Voice Command Reference,
Cisco IOS Debug Command Reference, Release 12.4T, Cisco Unified Communications Manager
Express Command Reference, and Cisco IOS Configuration Fundamentals Command Reference,
Release 12.4.

Configuration Examples for Fax Relay

This section contains the following example:
• Fax Relay: Example, page 742

Fax Relay: Example

voice service voip
fax-relay sg3-to-g3

ephone-dn 44
number 1234
name fax machine

ephone 33
mac-address 1111.2222.3333
button 1:44
type anl

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Additional References

Additional References
The following sections provide references related to Cisco Fax Relay.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration Cisco Unified Communications Manager Express System
Administrator Guide
Cisco Unified CME command reference Cisco Unified Communications Manager Express Command
Reference
Cisco IOS debugging Cisco IOS Debug Command Reference, Release 12.4T
Cisco IOS voice commands Cisco IOS Voice Command Reference
Cisco IOS voice configuration Cisco IOS Voice Configuration Library
Fax and modem transmission on Cisco Voice over IP Cisco IOS Fax and Modem Service over IP Application Guide
(VoIP) networks
SCCP gateway controlled feature mode call control Feature Mode for SCCP FXS Ports in Cisco IOS
SCCP gateway controlled VMWI VMWI for SCCP FXS Ports in Cisco IOS
SCCP gateway controlled supplementary features SCCP Controlled Analog (FXS) Ports with Supplementary Features
in Cisco IOS Gateways
Platform-specific documentation for the following: http://www.cisco.com/en/US/products/hw/routers/index.html
• Cisco 2800 Series Integrated Services Routers http://www.cisco.com/en/US/products/sw/voicesw/index.html
• Cisco 3800 Series Integrated Services Routers
• Cisco VG 224 Voice Gateway Router

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies.
To receive security and technical information about
your products, you can subscribe to various services,
such as the Product Alert Tool (accessed from Field
Notices), the Cisco Technical Services Newsletter, and
Really Simple Syndication (RSS) Feeds.
Access to most tools on the Cisco Support website
requires a Cisco.com user ID and password.

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Feature Information for Fax Relay

Feature Information for Fax Relay

Table 40 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 40 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 40 Feature Information for Cisco Fax Relay

Feature Name Cisco Unified CME Feature Information

Version
Fax Relay 4.0(3) Enables Fax Relay on analog FXS ports on Cisco IOS
voice gateways under the control of a Cisco Unified CME
system.

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 Configuring Headset Auto-Answer

Last Updated: March 26, 2007

This chapter describes the headset auto-answer feature in Cisco Unified Communications Manager
Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Headset
Auto-Answer” section on page 751.

Contents
• Information About Headset Auto-Answer, page 745
• How to Configure Headset Auto-Answer, page 748
• Configuration Examples for Headset Auto-answer, page 749
• Additional References, page 750
• Feature Information for Headset Auto-Answer, page 751

Information About Headset Auto-Answer

To enable the Headset Auto-Answer feature, you should understand the following concepts:
• Auto-Answering Calls Using a Headset, page 746
• Difference Between a Line and a Button, page 746

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Information About Headset Auto-Answer

Auto-Answering Calls Using a Headset

In Cisco Unified CME 4.0 and later versions you can configure lines on specific phones to automatically
connect to incoming calls when the headset key is activated. The phone cannot be busy with an active
call and the headset key must be engaged to automatically answer calls. Incoming calls are automatically
answered one by one on the phone as long as the headset light remains lit. For each ephone, you can
specify one or more lines for headset auto-answer.
After a phone is configured for headset auto-answer, the phone user must press the headset key to start
auto-answer. The headset light is lit to indicate that auto-answer is active for the lines that are designated
in the configuration. When the phone auto-answers a call, a zip tone is played to alert the phone user that
a call is present. To stop auto-answer, the phone user presses the headset key again and the headset light
goes out. At this time, the phone user can answer calls in a normal manner using the handset.

Difference Between a Line and a Button

Note that a line is similar to, but not exactly the same as, a button on the phone. A line represents a
phone’s capability to make a call connection, so each button that can make a call connection becomes a
line. (For example, unoccupied buttons or speed-dial buttons are not lines.) Note also that a line is not
the same as an ephone-dn. A button with overlaid ephone-dns is only one line, regardless of whether it
has several ephone-dns (extension numbers) associated with it. In most cases an ephone’s line numbers
do match its button numbers, but in a few cases they do not.
Figure 46 illustrates a comparison of line numbers and button numbers for different types of ephone
configurations.

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Information About Headset Auto-Answer

Figure 46 When is a Line the Same as a Button?

Most of the time, a line number is the same as the button number on which
it appears.
In this example, line 1 is button 1, line 2 is button 2, and line 3 is button 3.

ephone-dn 21
number 2001
ephone-dn 22
number 2002
ephone-dn 23
number 2003
2001 Li ne 1
ephone 2
2002 Li ne 2
button 1:21 2:22 3:23
2003 Li ne 3 headset auto-answer line 1
headset auto-answer line 2
But not always. In the following case, line 2 is button 3, because
button3 is the second button that has an ephone-dn to be connected
to a phone call. Button 2 is unoccupied and cannot take calls.

ephone-dn 33
number 2889
ephone-dn 34
number 2887
ephone 2
2889 Li ne 1 button 1:33 3:34
headset auto-answer line 1
2887 Li ne 2 headset auto-answer line 2

ephone-dn 21
In the following example, button 2 has three overlay ephone- number 2001
dns (22, 23, and 24). Button 2 is defined as one line because
only one of those ephone-dns can be connected to a call ephone-dn 22
using this button at any one time. number 2002
ephone-dn 23
number 2003
ephone-dn 24
number 2004
ephone-dn 25
number 2005
2001 Li ne 1
2002, 2003, 2004 Li ne 2 ephone 2
2005 Li ne 3 button 1:21 2o22,23,24 3:25
headset auto-answer line 2
headset auto-answer line 3

An expansion, or rollover, line for overlaid ephone-dns also ephone-dn 21

counts as one line. Button 2 in this example is also line 2. number 2001
ephone-dn 22
number 2002
ephone-dn 23
number 2003
ephone-dn 24
number 2004
2001, 2002, 2003 Li ne 1
(rollover) ephone 2
Li ne 2
button 1o21,2 2,23 2x1 3:24
135076

2004 Li ne 3
headset auto-a nswer line 1
headset auto-a nswer line 2

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How to Configure Headset Auto-Answer

How to Configure Headset Auto-Answer

This section contains the following tasks:
• Enabling Headset Auto-Answer, page 748 (required)
• Verifying Headset Auto-Answer, page 749 (optional)

Enabling Headset Auto-Answer

To enable headset auto-answer, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. headset auto-answer line line-number
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies this
Example: ephone during configuration tasks. The maximum number of
Router(config)# ephone 25 ephones for a particular Cisco Unified CME system is
version- and platform-specific. For the range of values, see
the CLI help.
Step 4 headset auto-answer line line-number Specifies a line on an ephone that will be answered automatically
when the headset button is depressed.
Example: • line-number—Number of the phone line that should be
Router(config-ephone)# headset automatically answered.
auto-answer line 1
Note Repeat this command to add additional lines.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

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Configuration Examples for Headset Auto-answer

Verifying Headset Auto-Answer

Step 1 Use the show running-config command to verify your configuration. Headset auto-answer is listed in
the ephone portion of the output.
Router# show running-config

ephone 1
headset auto-answer line 1
headset auto-answer line 2
headset auto-answer line 3
headset auto-answer line 4
username "Front Desk"
mac-address 011F.92B0.BE03
speed-dial 1 330 label “Billing”
type 7960 addon 1 7914
no dnd feature-ring
keep-conference
button 1f40 2f41 3f42 4:30
button 5:405 7m20 8m21 9m22
button 10m23 11m24 12m25 13m26
button 14m499 15:1 16m31 17f498
button 18s500
night-service bell

Step 2 Use the show telephony-service ephone command to display only the ephone configuration portion of
the running configuration.

Configuration Examples for Headset Auto-answer

The following example enables headset auto-answer on ephone 3 for line 1 (button 1) and
line 4 (button 4).
ephone 3
button 1:2 2:4 3:6 4o21,22,23,24,25
headset auto-answer line 1
headset auto-answer line 4

The following example enables headset auto-answer on ephone 17 for line 2 (button 2), which has
overlaid ephone-dns, and line 3 (button 3), which is an overlay rollover line.
ephone 17
button 1:2 2o21,22,23,24,25 3x2
headset auto-answer line 2
headset auto-answer line 3

The following example enables headset auto-answer on ephone 25 for line 2 (button 3) and
line 3 (button 5). In this case, the button numbers do not match the line numbers because buttons 2 and 4
are not used.
ephone 25
button 1:2 3:4 5:6
headset auto-answer line 2
headset auto-answer line 3

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Headset Auto-Answer

Feature Information for Headset Auto-Answer

Table 41 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 41 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 41 Feature Information for Headset Auto-Answer

Cisco Unified CME

Feature Name Version Feature Information
Headset Auto-Answer 4.0 Headset auto-answer was introduced.

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Feature Information for Headset Auto-Answer

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 Configuring Intercom Lines

Last Updated: March 26, 2007

This chapter describes the intercom features in Cisco Unified Communications Manager Express
(Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Intercom Lines”
section on page 761.

Contents
• Information About Intercom Lines, page 753
• How to Configure Intercom Lines, page 755
• Configuration Examples for Intercom Lines, page 759
• Where to Go Next, page 759
• Additional References, page 760
• Feature Information for Intercom Lines, page 761

Information About Intercom Lines

To enable intercom lines, you should understand the following concept:
• Intercom Auto-Answer Lines, page 753

Intercom Auto-Answer Lines

An intercom line is a dedicated two-way audio path between two phones. Cisco Unified CME supports
intercom functionality for one-way and press-to-answer voice connections using a dedicated pair of
intercom directory numbers on two phones that speed-dial each other.
When an intercom speed-dial button is pressed, a call is speed-dialed to the directory that is the other
half of the dedicated pair. The called phone 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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Information About Intercom Lines

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 pressing the
Mute button or, on phones such as the Cisco Unified IP Phone 7910, lifting the handset.
In Cisco CME 3.2.1 and later versions, you can deactivate the speaker-mute function on intercom calls.
For example, if phone user 1 makes an intercom call to phone user 2, both users hear each other on
connection when no-mute is configured. The benefit is that people who receive intercom calls can be
heard without them having to disable the mute function. The disadvantage is that nearby background
sounds and conversations can be heard the moment a person receives an intercom call, regardless of
whether they are ready to take a call or not.
Intercom lines cannot be used in shared-line configurations. If a directory number is configured for
intercom operation, it must be associated with one IP phone only. The intercom attribute causes an IP
phone line to operate as an autodial line for outbound calls and as an autoanswer-with-mute line for
inbound calls. Figure 47 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 nonintercom call), you can assign the intercom a directory number that
includes 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 Unified CME router. For example, the intercom ephone-dns in Figure 47 are
assigned numbers with alphabetic characters so that only the receptionist can call the manager on his or
her intercom line, and no one except the manager can call the receptionist on his or her intercom line.

Note An intercom requires the configuration of two ephone-dns, one each on a separate phone.

Figure 47 Intercom Lines

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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How to Configure Intercom Lines

How to Configure Intercom Lines

This section contains the following tasks:
• SCCP: Configuring an Intercom Auto-Answer Line, page 755 (required)
• SIP: Configuring an Intercom Auto-Answer Line, page 757 (required)

SCCP: Configuring an Intercom Auto-Answer Line

To enable a two-way audio path between two phones, perform the following steps for each SCCP phone
at both ends of the two-way voice path.

Restrictions
• Intercom lines cannot be dual-line.
• If a directory number is configured for intercom operation, it can be associated with only one
Cisco Unified IP phone.
• A separate configuration is required for each phone at both ends of the two-way voice path.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag
4. number number
5. name name
6. intercom directory-number [barge-in | no-auto-answer] [label label] [no-mute]
7. exit
8. ephone phone-tag
9. button button-number:dn-tag [[button-number:dn-tag] ...]
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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How to Configure Intercom Lines

Command or Action Purpose

Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode.
• Do not use the dual-line keyword with this
Example: command. Intercom ephone-dns cannot be
Router(config)# ephone-dn 11 dual-line.
Step 4 number number Assigns a valid intercom number.
• Using one or more alphabetic characters in an
Example: intercom number ensures that the number can
Router(config-ephone-dn)# number A2345 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 5 name name Sets a name to be associated with the ephone-dn.
• This name is used for caller-ID displays and also
Example: shows up in the local directory associated with
Router(config-ephone-dn)# name intercom the ephone-dn.
Step 6 intercom directory-number [barge-in | Defines the directory number that is speed-dialed for
no-auto-answer] [label label] [no-mute] the intercom feature when this line is used.

Example:
Router(config-ephone-dn)# intercom A2346 label
Security
Step 7 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 8 ephone phone-tag Enters ephone configuration mode.

Example:
Router(config)# ephone 24
Step 9 button button-number:dn-tag [[button-number:dn-tag] Assigns a button number to the intercom ephone-dn
...] being configured.
• Use the colon separator (:) between the button
Example: number and the intercom ephone-dn tag to
Router(config-ephone)# button 1:1 2:4 3:14 indicate a normal ring for the intercom line.
Step 10 end Exits configuration mode and enters privileged
EXEC mode.
Example:
Router(config)# exit

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How to Configure Intercom Lines

SIP: Configuring an Intercom Auto-Answer Line

To enable the Intercom Auto-Answer feature for SIP phones, perform the following steps for each SIP
phone at both ends of the two-way voice path.

Prerequisites
Cisco CME 3.4 or a later version.

Restrictions
• If a directory number is configured for intercom operation, it can be associated with only one
Cisco Unified IP phone.
• A separate configuration is required for each phone at each end of the two-way voice path.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register dn dn-tag
4. number number
5. auto-answer
6. exit
7. voice register pool pool-tag
8. id mac address
9. type phone-type
10. number tag dn dn-tag
11. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or an MWI.
Example:
Router(config-register-global)# voice register
dn 1

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How to Configure Intercom Lines

Command or Action Purpose

Step 4 number number Defines a valid number for the directory number being
configured.
Example: • To prevent non intercom originators from manually
Router(config-register-dn)# number A5001 dialing an intercom destination, the number string can
contain alphabetic characters enabling the number to be
dialed only by the Cisco Unified CME router and not
from telephone keypads.
Step 5 auto-answer Enables the Intercom Auto Answer feature on the directory
number being configured.
Example:
Router(config-register-dn)# auto-answer
Step 6 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-register-dn)# exit
Step 7 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone in
Cisco Unified CME.
Example:
Router(config)# voice register pool 3
Step 8 id {network address mask mask | ip address mask Explicitly identifies a locally available individual SIP phone to
mask | mac address} support a degree of authentication.

Example:
Router(config-register-pool)# id mac
0009.A3D4.1234
Step 9 type phone-type Defines a phone type for the SIP phone being configured.

Example:
Router(config-register-pool)# type 7960-7940
Step 10 number tag dn dn-tag Associates a directory number with the SIP phone being
configured.
Example:
Router(config-register-pool)# number 1 dn 17
Step 11 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-pool)# end

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Configuration Examples for Intercom Lines

Configuration Examples for Intercom Lines

This section contains the following example:
• Intercom Lines: Example, page 759

Intercom Lines: Example

The following example shows an intercom between two Cisco Unified 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 Unified IP phone 4.
Ephone-dn 19 is associated with line button 2 on Cisco Unified IP phone 5. The two ephone-dns provide
a two-way intercom between the two Cisco Unified 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

Where to Go Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.

Paging
The paging feature sets up a one-way audio path to deliver information to a group of phones at one time.
For more information, see “Configuring Paging” on page 785.

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Intercom Lines

Feature Information for Intercom Lines

Table 42 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 42 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 42 Feature Information for Intercom Lines

Cisco Unified CME

Feature Name Version Feature Information
Intercom Lines 3.4 Adds intercom feature, with no-mute function, for
supported Cisco Unified IP phones that are connected to a
Cisco Unified CME router and running SIP.
3.2.1 The no-mute function was introduced.
2.0 Intercom feature was introduced.

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 Configuring Loopback Call Routing

Last Updated: March 26, 2007

This chapter describes the loopback call-routing feature in Cisco Unified Communications Manager
Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Loopback Call
Routing” section on page 770.

Contents
• Information About Loopback Call Routing, page 763
• How to Configure Loopback Call Routing, page 764
• Configuration Examples for Loopback Call Routing, page 768
• Additional References, page 769
• Feature Information for Loopback Call Routing, page 770

Information About Loopback Call Routing

To enable loopback call routing, you should understand the following concept:
• Loopback Call Routing, page 763

Loopback Call Routing

Loopback call routing in a Cisco Unified 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.
Loopback call routing and loopback-dn restricts 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

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(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 on
configuring Cisco Unity Express, see the Cisco Unity Express documentation.

Note A preferred alternative to loopback call routing was introduced in Cisco CME 3.1. This alternative
blocks H.450-based supplementary service requests by using the following Cisco IOS commands:
no supplementary-service h450.2, no supplementary-service h450.3, and supplementary-service
h450.12. For more information, see “Configuring Call Transfer and Forwarding” on page 499.

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 for use in VoIP network interworking where the 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.
Loopback call routing requires two extensions (ephone-dns) to be separately configured, each as half of
a loopback-dn pair. Ephone-dns that are defined as a loopback-dn pair can only be used for loopback call
routing. In addition to defining the loopback-dn pair, you must specify preference, huntstop, class of
restriction (COR), and translation rules.

How to Configure Loopback Call Routing

This section contains the following tasks:
• Enabling Loopback Call Routing, page 764
• Verifying Loopback Call Routing, page 768

Enabling Loopback Call Routing

To enable loopback call-routing, perform the following steps for each ephone-dn that is part of the
loopback-dn pair.

Restrictions
Loopback-dns do not support T.38 fax relay.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag

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4. number number [secondary number] [no-reg [both | primary]]

5. caller-id {local | passthrough}
6. no huntstop
7. preference preference-order [secondary secondary-order]
8. cor {incoming | outgoing} cor-list-name
9. translate {called | calling} translation-rule-tag
10. 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}]
11. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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.
Note Ephone-dns used for loopback cannot be dual-line
ephone-dns.
Step 4 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.

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Command or Action Purpose

Step 5 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 6 no huntstop Disables huntstop and allows call hunting behavior for an
extension (ephone-dn).
Example:
Router(config-ephone-dn)# no huntstop
Step 7 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: 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 0 to 10, where 0 is the highest preference and 10
is the lowest preference. Default is 9.
Step 8 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 “Dial Peer Configuration on
Voice Gateway Routers”.
Step 9 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
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 10 loopback-dn dn-tag [forward Enables H.323 call transfer and call forwarding by using
number-of-digits | strip number-of-digits] hairpin call routing for VoIP endpoints that do not support
[prefix prefix-digit-string] [suffix
suffix-digit-string] [retry seconds]
Cisco-proprietary or H.450-based call-transfer and
[auto-con] [codec {g711alaw | g711ulaw}] call-forwarding.
• 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 being configured. The paired ephone-dn
forward 15 prefix 415353.... 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 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 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 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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Command or Action Purpose

Step 11 end Exits to privileged exec mode.

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

Verifying Loopback Call Routing

Step 1 Use the show running-config or show telephony-service ephone-dn command to display ephone-dn
configurations.

Configuration Examples for Loopback Call Routing

This section contains the following example:
• Enabling Loopback Call Routing: Example, page 768

Enabling Loopback Call Routing: 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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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Loopback Call Routing

Feature Information for Loopback Call Routing

Table 43 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 43 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 43 Feature Information for Loopback Call Routing

Cisco Unified CME

Feature Name Version Feature Information
Loopback Call Routing 2.0 Loopback call routing was introduced.

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 Configuring Music on Hold

Last Updated: May 24, 2007

This chapter describes the music on hold features in Cisco Unified Communications Manager Express
(Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Music on Hold”
section on page 783.

Contents
• Information About Music on Hold, page 772
• Prerequisites for Music on Hold, page 771
• Restrictions for Music on Hold, page 771
• How to Configure Music on Hold, page 774
• Configuration Examples for Music on Hold, page 781
• Additional References, page 782
• Feature Information for Music on Hold, page 783

Prerequisites for Music on Hold

• Phones receiving MOH in a system using G.729 require transcoding between G.711 and G.729. For
information about transcoding, see “Configuring Transcoding Resources” on page 303.

Restrictions for Music on Hold

• IP phones do not support multicast at 224.x.x.x addresses.
• Cisco Unified CME 3.3 and earlier versions do not support MOH for local Cisco Unified CME
phones that are on hold with other Cisco Unified CME phones; these parties hear a periodic
repeating tone instead.

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• Cisco Unified CME 4.0 and later versions support MOH for internal calls only if the multicast moh
command is used to enable the flow of packets to the subnet on which the phones are located.
• Internal extensions that are connected through an analog voice gateway (Cisco VG 224) or through
a WAN (remote extensions) do not hear MOH on internal calls.
• Multicast MOH is not supported on a phone if the phone is configured with the mtp command or
the paging-dn command with the unicast keyword.

Information About Music on Hold

To enable music on hold, you should understand the following concept:
• Music on Hold, page 772

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 Unified CME system. This audio stream is intended to reassure
callers that they are still connected to their calls.
When the phone receiving MOH is part of a system that uses a G.729 codec, 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, see “Configuring Transcoding Resources” on page 303.
If the MOH audio stream is also identified as a multicast source, the Cisco Unified CME router
additionally transmits the stream on the physical IP interfaces of the Cisco Unified CME router that you
specify during configuration, which permits external devices to have access to it.
Certain IP phones do not support IP multicast and, therefore, do not support multicast MOH. You can
disable multicast MOH to individual phones that do not support multicast. Callers hear a repeating tone
when they are placed on hold.
The audio stream that is used for MOH can derive from one of two sources:
• Audio file—A MOH audio stream from an audio file is supplied from an .au or .wav file held in
router flash memory.
• Live feed—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.
If both are configured concurrently on the Cisco Unified 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.

Music on Hold for SIP Phones

In Cisco Unified CME 4.1 and later versions, the MOH feature is supported when a call is put on hold
from a SIP phone and when the user of a SIP phone is put on hold by a SIP, SCCP, or POTS endpoint.
The holder (party that pressed the hold key) or holdee (party who is put on hold) can be on the same
Cisco Unified CME or a different Cisco Unified CME connected through a SIP trunk. MOH is also
supported for call transfers and conferencing, with or without a transcoding device.

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Information About Music on Hold

Configuring MOH for SIP phones is the same as configuring MOH for SCCP phones. For configuration
information, see the “How to Configure Music on Hold” section on page 774.

Music on Hold from a Live Feed

The live-feed feature is typically used to connect to a CD jukebox player. 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. Only one live MOH feed is supported per system.
Using an analog E&M port as the live-feed MOH interface 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 use an FXO port as the live-feed MOH interface, connect the MOH source to the FXO port using
a MOD-SC cable if the MOH source has a different connector than the FXO RJ-11 connector. MOH
from a live feed is supported on the VIC2-2FXO, VIC2-4FXO, EM-HDA-3FXS/4FXO,
EM-HDA-6FXO, and EM2-HDA-4FXO.
You can directly connect a live-feed source to an FXO port if the signal loop-start live-feed command
is configured on the voice port; otherwise, the port must connect through an external third-party adapter
to provide a battery feed. An external adapter must supply normal telephone company (telco) battery
voltage with the correct polarity to the tip and ring leads of the FXO port and it must 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 Unified CME MOH system 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 Unified 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 Unified CME router.

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How to Configure Music on Hold

How to Configure Music on Hold

This section contains the following tasks:
• Configuring Music on Hold from an Audio File, page 774 (optional)
• Configuring Music on Hold from a Live Feed, page 776 (optional)
• Verifying Music on Hold, page 780 (optional)

Configuring Music on Hold from an Audio File

To configure MOH when you are using a file to supply the audio stream, perform the following steps.

Prerequisites
• SIP phones require Cisco Unified CME 4.1 or a later version.
• 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
• If MOH from an audio file and MOH from a live feed are both configured on the Cisco Unified 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. 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.
• The volume level of a MOH file cannot be adjusted through Cisco IOS software, so it cannot be
changed when 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 before downloading the file to router flash memory.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. moh filename
5. multicast moh ip-address port port-number [route ip-address-list]
6. exit
7. ephone phone-tag
8. multicast-moh
9. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 moh filename Enables music on hold using the specified file.
• filename—Source of the audio stream for MOH.
Example: Note If you specify a file with this command and
Router(config-telephony)# moh minuet.au
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.
Step 5 multicast moh ip-address port port-number [route Specifies that the MOH audio stream should also be
ip-address-list] multicast as specified.
Note This command is required to use MOH for
Example: internal calls and it must be configured after
Router(config-telephony)# multicast moh 239.10.16.4 MOH is enabled with the moh command.
port 2123 route 10.10.29.17 10.10.29.33
• ip-address—Specifies that this audio stream is to
be used for multicast and also for MOH, and
specifies the destination IP address for multicast.
• port port-number—Media port for multicast.
Range is 2000 to 65535. We recommend
port 2000 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 for 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.
Note For MOH on internal calls, packet flow must
be enabled to the subnet on which the phones
are located.

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Command or Action Purpose

Step 6 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 7 ephone phone-tag Enters ephone configuration mode.

Example:
Router(config)# ephone 28
Step 8 multicast-moh (Optional) Enables multicast MOH on a phone. This
is the default.
Example: The no form of this command disables MOH for
Router(config-ephone)# no multicast-moh phones that do not support multicast. Callers hear a
repeating tone when they are placed on hold.
Note This command can also be used in an ephone
template that is applied to one or more
phones.
Step 9 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

Configuring Music on Hold from a Live Feed

To configure music on hold from a live feed, perform the following steps.

Prerequisites
• SIP phones require Cisco Unified CME 4.1 or a later version.
• VIC2-2FXO, VIC2-4FXO, EM-HDA-3FXS/4FXO, EM-HDA-6FXO, or EM2-HDA-4FXO
• For a live feed from VoIP, VAD must be disabled.

Restrictions
• A foreign exchange station (FXS) port cannot be used for a live feed.
• The signal loop-start live-feed command for FXO ports is supported in Cisco IOS
Release 12.4(11)XJ, 12.4(15)T, and later releases.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice-port port
4. input gain decibels

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5. auto-cut-through (E&M only)

6. operation 4-wire (E&M only)
7. signal immediate (E&M only)
8. signal loop-start live-feed (FXO only)
9. no shutdown
10. exit
11. dial peer voice tag pots
12. destination-pattern string
13. port port
14. exit
15. ephone-dn dn-tag
16. number number
17. moh [out-call outcall-number] [ip ip-address port port-number [route ip-address-list]]
18. exit
19. ephone phone-tag
20. multicast-moh
21. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice-port port Enters voice-port configuration mode.
Note Port argument is platform-dependent; type ? to
Example: display syntax. For more information, see the
Router(config)# voice-port 1/1/0 Cisco IOS Voice Command Reference.
Step 4 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 5 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

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Command or Action Purpose

Step 6 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 7 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.
Step 8 signal loop-start live-feed (FXO ports only) Enables an MOH audio stream from a live
feed to be directly connected to the router through an FXO
port.
Example:
Router(config-voice-port)# signal loop-start
live-feed
Step 9 no shutdown Activates the voice port.
• To shut the voice port down and disable MOH from a
Example: live feed, use the shutdown command.
Router(config-voice-port)# no shutdown
Step 10 exit Exits voice-port configuration mode.

Example:
Router(config-voice-port)# exit
Step 11 dial peer voice tag pots Enters dial-peer configuration mode.

Example:
Router(config)# dial peer voice 7777 pots
Step 12 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 13 port port Associates the dial peer with the voice port that was
specified in Step 3.
Example:
Router(config-dial-peer)# port 1/1/0
Step 14 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit
Step 15 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
Router(config)# ephone-dn 55 1 to 288.

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Command or Action Purpose

Step 16 number number Configures a valid extension number for this ephone-dn.
• This number is not assigned to any phone; it is only
Example: used to make and receive calls that contain an audio
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.
Step 17 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 the source for an MOH stream. If this
command is used without the out-call keyword, the MOH
Example: 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 for MOH and
10.10.29.45
specifies the number to be called. Forces a connection
to the local voice port that was specified in Step 3.
• ip ip-address—(Optional) Indicates that this audio
stream is to be used as a multicast source and also 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 2000 to 65535. We recommend
port 2000 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 18 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 19 ephone phone-tag Enters ephone configuration mode.

Example:
Router(config)# ephone 28

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Command or Action Purpose

Step 20 multicast-moh (Optional) Enables multicast MOH on a phone.
• This command is enabled by default.
Example: The no form of this command disables MOH for phones that
Router(config-ephone)# no multicast-moh
do not support multicast. Callers hear a repeating tone when
they are placed on hold.
Note This command can also be made part of an ephone
template that is applied to one or more phones.
Step 21 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

Verifying Music on Hold

Step 1 Use the show running-config command to display the running configuration. MOH commands are
listed in the telephony-service part of the output.
Router# show running-config

telephony-service
fxo hook-flash
load 7960-7940 P00307020300
load 7914 S00104000100
max-ephones 100
max-dn 500
ip source-address 10.123.23.231 port 2000
max-redirect 20
timeouts ringing 100
system message XYZ Company
voicemail 7189
max-conferences 8 gain -6
call-forward pattern .T
moh flash:music-on-hold.au
multicast moh 239.15.10.1 port 2000
web admin system name admin1 password admin1
dn-webedit
time-webedit
transfer-system full-consult
secondary-dialtone 9
fac custom callfwd all **1
fac custom callfwd cancel **2
fac custom pickup local **3
fac custom pickup group *7
fac custom pickup direct **5
fac custom park *8
fac custom dnd **7
fac custom redial #8
fac custom voicemail **9
fac custom ephone-hunt join *3
fac custom ephone-hunt cancel #3
create cnf-files version-stamp Jan 01 2002 00:00:00

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Configuration Examples for Music on Hold

Step 2 Use the show telephony-service command to display only the telephony-service configuration
information.

Configuration Examples for Music on Hold

This section contains the following examples:
• MOH from an Audio File: Example, page 781
• MOH from a Live Feed: Example, page 781

MOH from an Audio File: Example

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

MOH from a Live Feed: 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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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Music on Hold

Feature Information for Music on Hold

Table 44 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 44 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 44 Feature Information for Music on Hold

Cisco Unified CME

Feature Name Version Feature Information
Music on Hold 4.1 Music on hold for SIP phones was supported.
4.0 • Music on hold was introduced for internal calls.
• The ability to disable multicast MOH per phone was
introduced.
3.0 The ability to use a live audio feed as a multicast source
was introduced.
2.1 Music on hold from a live audio feed was introduced for
external calls.
2.0 Music on hold from an audio file was introduced for
external calls.

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Last Updated: March 26, 2007

This chapter describes the paging feature in Cisco Unified Communications Manager Express
(Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Paging” section on
page 795.

Contents
• Information About Paging, page 785
• How to Configure Paging, page 787
• Configuration Examples for Paging, page 791
• Where to Go Next, page 794
• Additional References, page 794
• Feature Information for Paging, page 795

Information About Paging

To enable paging, you should understand the following concept:
• Audio Paging, page 785

Audio Paging
A paging number can be defined to relay audio pages to a group of designated 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.

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Information About 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.
After you have created two or more simple paging groups, you can unite them into combined paging
groups. By creating combined paging groups, you provide phone users with the flexibility to page a
small local paging group (for example, paging four phones in a store’s jewelry department) or to page a
combined set of several paging groups (for example, by paging a group that consists of both the jewelry
department and the accessories department).
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).
Figure 48 shows a paging group with two phones.

Figure 48 Paging Group

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

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How to Configure Paging

How to Configure Paging

This section contains the following tasks:
• Configuring a Simple Paging Group, page 787 (required)
• Configuring a Combined Paging Group, page 788 (optional)
• Verifying Paging, page 791 (optional)

Configuring a Simple Paging Group

To set up a paging number that relays incoming pages to a group of phones, perform the following steps.

Restrictions
IP phones do not support multicast at 224.x.x.x addresses.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn paging-dn-tag
4. number number
5. name name
6. paging [ip multicast-address port udp-port-number]
7. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 1 to 288.
Note Do not use the dual-line keyword with this command.
Paging ephone-dns cannot be dual-line.

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Command or Action Purpose

Step 4 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 5 name name Assigns to the paging number a name to appear in caller-ID displays
and directories.
Example:
Router(config-ephone-dn)# name paging4
Step 6 paging [ip multicast-address port Specifies that this ephone-dn is to be used to broadcast 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. We
recommend port 2000 because it is already used for normal
non-multicast RTP media streams between phones and the
Cisco Unified CME router.
Note IP phones do not support multicast at 224.x.x.x addresses.
Step 7 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Configuring a Combined Paging Group

To set up a combined paging group consisting of two or more simple paging groups, perform the
following steps.

Prerequisites
Simple paging groups must be configured. See the “Configuring a Simple Paging Group” section on
page 787.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn paging-dn-tag
4. number number
5. name name
6. paging group paging-dn-tag,paging-dn-tag[[,paging-dn-tag]...]

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7. exit
8. ephone phone-tag
9. paging-dn paging-dn-tag {multicast | unicast}
10. exit
11. Repeat Step 8 to Step 10 to add additional IP phones to the paging group.
12. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn paging-dn-tag Enters ephone-dn configuration mode to create a paging number for
a combined paging group.
Example: • paging-dn-tag—A unique sequence number that identifies this
Router(config)# ephone-dn 42 paging ephone-dn during all configuration tasks. This is the
ephone-dn that is dialed to initiate a page. This ephone-dn is not
associated with a physical phone. Range is 1 to 288.
Note Do not use the dual-line keyword with this command.
Paging ephone-dns cannot be dual-line.
Step 4 number number Defines an extension number associated with the combined group
paging ephone-dn. This is the number that people call to initiate a
page to the combined group.
Example:
Router(config-ephone-dn)# number 3556
Step 5 name name (Optional) Assigns to the combined group 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 6 paging group paging-dn-tag,paging-dn-tag Sets the paging directory number for a combined group. This
[[,paging-dn-tag]...] command combines the individual paging group ephone-dns that
you specify into a combined group so that a page can be sent to more
Example: than one paging group at a time.
Router(config-ephone-dn)# paging group • paging-dn-tag—Unique sequence number associated with the
20,21
paging number for an individual paging group. 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.
Note Configure the paging command for all ephone-dns in a
paging group before configuring the paging group
command for that group.
Step 7 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 8 ephone phone-tag Enters ephone configuration mode to add IP phones to the paging
group.
Example: • phone-tag—Unique sequence number of a phone to receive
Router(config)# ephone 2 audio pages when the paging ephone-dn is called.
Step 9 paging-dn paging-dn-tag {multicast | Associates this ephone with an ephone-dn tag that is used for a
unicast} paging ephone-dn (the number that people call to deliver a page).
Note that the paging ephone-dn tag is not associated with a line
Example: button on this ephone.
Router(config-ephone)# paging-dn 42 The paging mechanism supports audio distribution using IP
multicast
multicast, replicated unicast, and a mixture of both (so that
multicast is used where possible and unicast is allowed to specific
phones that cannot be reached through multicast).
• paging-dn-tag—Unique sequence number for a paging
ephone-dn.
• multicast—(Optional) Multicast paging for groups. By default,
paging is transmitted to the Cisco Unified IP phone using
multicast.
• unicast—(Optional) Unicast paging for a single
Cisco Unified IP phone. This keyword indicates that the
Cisco Unified IP phone is not capable of receiving paging
through multicast and requests that the phone receive 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.
Step 10 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

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Configuration Examples for Paging

Command or Action Purpose

Step 11 Repeat Step 8 to Step 10 to add additional IP —
phones to a paging group.
Step 12 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Verifying Paging
Step 1 Use the show running-config command to display the running configuration. Paging ephone-dns are
listed in the ephone-dn portion of the output. Phones that belong to paging groups are listed in the ephone
part of the output.
Router# show running-config

ephone-dn 48
number 136
name PagingCashiers
paging ip 239.1.1.10 port 2000

ephone 2
headset auto-answer line 1
headset auto-answer line 4
ephone-template 1
username "FrontCashier"
mac-address 011F.2A0.A490
paging-dn 48
type 7960
no dnd feature-ring
no auto-line
button 1f43 2f44 3f45 4:31

Step 2 Use the show telephony-service ephone-dn and show telephony-service ephone commands to display
only the configuration information for ephone-dns and ephones.

Configuration Examples for Paging

This section contains the following examples:
• Simple Paging Group: Example, page 792
• Combined Paging Groups: Example, page 792

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Configuration Examples for Paging

Simple Paging Group: Example

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 Unified IP phones 1 and 2, and paging calls
to 2001 go to Cisco Unified 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

Combined Paging Groups: 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.

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Configuration Examples for Paging

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-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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Where to Go Next

Where to Go Next
Intercom
The intercom feature is similar to paging because it allows a phone user to deliver an audio message to
a phone without the called party having to answer. The intercom feature is different than paging because
the audio path between the caller and the called party is a dedicated audio path and because the called
party can respond to the caller. See “Configuring Intercom Lines” on page 753.

Speed Dial
Phone users who make frequent pages may want to include the paging ephone-dn numbers in their list
of speed-dial numbers. See “Configuring Speed Dial” on page 847.

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Paging

Feature Information for Paging

Table 45 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 45 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 45 Feature Information for Paging

Cisco Unified CME

Feature Name Version Feature Information
Paging 2.0 Paging was introduced.

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Last Updated: March 26, 2007

This module describes presence support in a Cisco Unified Communications Manager Express
(Cisco Unified CME) system.

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Presence Service”
section on page 818.

Contents
• Prerequisites for Presence Service, page 797
• Restrictions for Presence Service, page 798
• Information About Presence Service, page 798
• How to Configure Presence Service, page 799
• Configuration Examples for Presence, page 813
• Additional References, page 817
• Feature Information for Presence Service, page 818

Prerequisites for Presence Service

• Cisco Unified CME 4.1 or a later version.
• Cisco IOS Release 12.4(11)XJ, 12.4(15)T, or a later release.
• Cisco Unified IP Phones 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE require
firmware load 8.2(1) or a later version.

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Restrictions for Presence Service

• Presence features such as Busy Lamp Field (BLF) notification are supported for SIP trunks only;
these features are not supported on H.323 trunks.
• Presence requires that SIP phones are configured with a directory number (using dn keyword in
number command); direct line numbers are not supported.

Information About Presence Service

To configure presence service in a Cisco Unified CME system, you should understand the following
concept:
• Presence Service, page 798

Presence Service
A presence service, as defined in RFC 2778 and RFC 2779, is a system for finding, retrieving, and
distributing presence information from a source, called a presence entity (presentity), to an interested
party called a watcher. When you configure presence in a Cisco Unified CME system with a SIP WAN
connection, a phone user, or watcher, can monitor the real-time status of another user at a directory
number, the presentity. Presence enables the calling party to know before dialing whether the called
party is available. For example, a directory application may show that a user is busy, saving the caller
the time and inconvenience of not being able to reach someone.
Presence uses SIP SUBSCRIBE and NOTIFY methods to allow users and applications to subscribe to
changes in the line status of phones in a Cisco Unified CME system. Phones act as watchers and a
presentity is identified by a directory number on a phone. Watchers initiate presence requests
(SUBSCRIBE messages) to obtain the line status of a presentity. Cisco Unified CME responds with the
presentity’s status. Each time a status changes for a presentity, all watchers of this presentity are sent a
notification message. SIP phones and trunks use SIP messages; SCCP phones use presence primitives in
SCCP messages.
Presence supports Busy Lamp Field (BLF) notification features for speed-dial buttons and directory call
lists for missed calls, placed calls, and received calls. SIP and SCCP phones that support the BLF
speed-dial and BLF call-list features can subscribe to status change notification for internal and external
directory numbers.
Figure 49 shows a Cisco Unified CME system supporting BLF notification for internal and external
directory numbers. If the watcher and the presentity are not both internal to the Cisco Unified CME
router, the subscribe message is handled by a presence proxy server.

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Figure 49 BLF Notification Using Presence

SIP

Subscribe
Notify
V

Cisco Unified CME

PSTN

Subscribe
Notify

155790
IP IP IP IP
IP IP IP IP

The following line states display through BLF indicators on the phone:
• Line is idle—Displays when this line is not being used.
• Line is in-use—Displays when the line is in the ringing state and when a user is on the line, whether
or not this line can accept a new call.
• BLF indicator unknown—Phone is unregistered or this line is not allowed to be watched.
Cisco Unified CME acts as a presence agent for internal lines (both SIP and SCCP) and as a presence
server for external watchers connected through a SIP trunk, providing the following functionality:
• Processes SUBSCRIBE requests from internal lines to internal lines. Notifies internal subscribers
of any status change.
• Processes incoming SUBSCRIBE requests from a SIP trunk for internal SCCP and SIP lines.
Notifies external subscribers of any status change.
• Sends SUBSCRIBE requests to external presentities on behalf of internal lines. Relays status
responses to internal lines.
Presence subscription requests from SIP trunks can be authenticated and authorized. Local subscription
requests cannot be authenticated.
For configuration information, see the “How to Configure Presence Service” section on page 799.

How to Configure Presence Service

This section contains the following tasks:
• Enabling Presence for Internal Lines, page 800
• Enabling a Directory Number to be Watched, page 801
• Enabling a SCCP Phone to Monitor BLF Status for Speed-Dials and Call Lists, page 803
• Enabling a SIP Phone to Monitor BLF Status for Speed-Dials and Call Lists, page 806
• Configuring Presence to Watch External Lines, page 808

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• Verifying Presence Configuration, page 810

• Troubleshooting Presence, page 811

Enabling Presence for Internal Lines

Perform the following steps to enable the router to accept incoming presence requests from internal
watchers and SIP trunks.

Restrictions
• A presentity can be identified by a directory number only.
• BLF monitoring indicates the line status only.
• Instant Messaging is not supported.

SUMMARY STEPS

1. enable
2. configure terminal
3. sip-ua
4. presence enable
5. exit
6. presence
7. max-subscription number
8. presence call-list
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 sip-ua Enters SIP user-agent configuration mode to configure the
user agent.
Example:
Router(config)# sip-ua

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Command or Action Purpose

Step 4 presence enable Allows the router to accept incoming presence requests.

Example:
Router(config-sip-ua)# presence enable
Step 5 exit Exits SIP user-agent configuration mode.

Example:
Router(config-sip-ua)# exit
Step 6 presence Enables presence service and enters presence configuration
mode.
Example:
Router(config)# presence
Step 7 presence call-list Globally enables BLF monitoring for directory numbers in
call lists and directories on all locally registered phones.
Example: • Only directory numbers that you enable for watching
Router(config-presence)# presence call-list with the allow watch command display BLF status
indicators.
• This command enables the BLF call-list feature
globally. To enable the feature for a specific phone, see
the “Enabling a SCCP Phone to Monitor BLF Status for
Speed-Dials and Call Lists” section on page 803.
Step 8 max-subscription number (Optional) Sets the maximum number of concurrent watch
sessions that are allowed.
Example: • number—Maximum watch sessions. Range: 100 to the
Router(config-presence)# max-subscription 128 maximum number of directory numbers supported on
the router platform. Type ? to display range.
Default: 100.
Step 9 end Exits to privileged EXEC mode.

Example:
Router(config-presence)# end

Enabling a Directory Number to be Watched

To enable a line associated with a directory number to be monitored by a phone registered to a
Cisco Unified CME router, perform the following steps. The line is enabled as a presentity and phones
can subscribe to its line status through the BLF call-list and BLF speed-dial features. There is no
restriction on the type of phone that can have its lines monitored; any line on any IP phone or on an
analog phone on supported voice gateways can be a presentity.

Restrictions
• A presentity is identified by a directory number only.
• BLF monitoring indicates the line status only.

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SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag
or
voice register dn dn-tag
4. number number
5. allow watch
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters the configuration mode to define a directory number
or for an IP phone, intercom line, voice port, or a
voice register dn dn-tag message-waiting indicator (MWI).
• dn-tag—Identifies a particular directory number during
Example: configuration tasks. Range is 1 to the maximum number
Router(config)# ephone-dn 1 of directory numbers allowed on the router platform, or
or the maximum defined by the max-dn command. Type
? to display range.
Router(config)# voice register dn 1
Step 4 number number Associates a phone number with a directory number to be
assigned to an IP phone in Cisco Unified CME.
Example: • number—String of up to 16 characters that represents
Router(config-ephone-dn)# number 3001 an E.164 telephone number.
or
Router(config-register-dn)# number 3001

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Command or Action Purpose

Step 5 allow watch Allows the phone line associated with this directory number
to be monitored by a watcher in a presence service.
Example: • This command can also be configured in ephone-dn
Router(config-ephone-dn)# allow watch template configuration mode and applied to one or
or more phones. The ephone-dn configuration has priority
Router(config-register-dn)# allow watch over the ephone-dn template configuration.
Step 6 end Exits to privileged EXEC mode.

Example:
Router(config-ephone-dn)# end
or
Router(config-register-dn)# end

Enabling a SCCP Phone to Monitor BLF Status for Speed-Dials and Call Lists
A watcher can monitor the status of lines associated with internal and external directory numbers
(presentities) through the BLF speed-dial and BLF call-list presence features. To enable the BLF
notification features on an IP phone using SCCP, perform the following steps.

Prerequisites
• Presence must be enabled on the Cisco Unified CME router. See the “Enabling Presence for Internal
Lines” section on page 800.
• A directory number must be enabled as a presentity with the allow watch command to provide BLF
status notification. See the “Enabling a Directory Number to be Watched” section on page 801.

Restrictions
BLF Call-List
• Supported only on Cisco Unified IP Phone 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE.

BLF Speed-Dial
• Supported only on Cisco Unified IP Phone 7914, 7931, 7940, 7941G, 7941GE, 7960, 7961G,
7961GE, 7970G, and 7971GE.

Cisco Unified IP Phone 7931

• BLF status is displayed through monitor lamp only; BLF status icons are not displayed.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. button button-number{separator}dn-tag [,dn-tag...] [button-number{x}overlay-button-number]
[button-number...]

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5. blf-speed-dial tag number label string

6. presence call-list
7. exit
8. telephony-service
9. create cnf-files
10. restart
11. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode to set phone-specific
parameters for a SIP phone.
Example: • phone-tag—Unique sequence number of the phone to
Router(config)# ephone 1 be configured. Range is version and
platform-dependent; type ? to display range. You can
modify the upper limit for this argument with the
max-ephones command.
Step 4 button button-number{separator}dn-tag Associates a button number and line characteristics with a
[,dn-tag...] directory number on the phone.
[button-number{x}overlay-button-number]
[button-number...] • button-number—Number of a line button on an IP
phone.
Example: • separator—Single character that denotes the type of
Router(config-ephone)# button 1:10 2:11 3b12 characteristics to be associated with the button.
4o13,14,15
• dn-tag—Unique sequence number of the ephone-dn
that you want to appear on this button. For overlay lines
(separator is o or c), this argument can contain up to
25 ephone-dn tags, separated by commas.
• x—Separator that creates an overlay rollover button.
• overlay-button-number—Number of the overlay button
that should overflow to this button.

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Command or Action Purpose

Step 5 blf-speed-dial tag number label string Enables BLF monitoring of a directory number associated
with a speed-dial number on the phone.
Example: • tag—Number that identifies the speed-dial index.
Router(config-ephone)# blf-speed-dial 3 3001 Range: 1 to 33.
label sales
• number—Telephone number to speed dial.
• string—Alphanumeric label that identifies the
speed-dial button. String can contain a maximum of 30
characters.
Step 6 presence call-list Enables BLF monitoring of directory numbers that appear
in call lists and directories on this phone.
Example: • For a directory number to be monitored, it must have
Router(config-ephone)# presence call-list the allow watch command enabled.
• To enable BLF monitoring for call lists on all phones in
this Cisco Unified CME system, use this command in
presence mode. See the “Enabling Presence for Internal
Lines” section on page 800.
Step 7 exit Exits ephone configuration mode and enters privileged
EXEC mode.
Example:
Router(config-ephone)# exit
Step 8 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 9 create cnf-files Builds the XML configuration files that are required for
Cisco Unified CME phones.
Example:
Router(config-telephony)# create cnf-files
Step 10 restart {all [time-interval] | mac-address} Performs a fast reset of the specified phone or all phones
associated with this Cisco Unified CME router. Does not
contact the DHCP server.
Example:
Router(config-telephony)# restart all • all—Restarts all phones associated with a
Cisco Unified CME router.
• time-interval—(Optional) Time interval, in seconds,
between the beginning of each phone restart.
Range: 0 to 60. Default is 15.
• mac-address—Restarts the phone that has the specified
MAC address.
Step 11 end Exits to privileged EXEC mode.

Example:
Router(config-telephony)# end

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Enabling a SIP Phone to Monitor BLF Status for Speed-Dials and Call Lists
A watcher can monitor the status of lines associated with internal and external directory numbers
(presentities) through the BLF speed-dial and BLF call-list presence features. To enable the BLF
notification features on a SIP phone, perform the following steps.

Prerequisites
• Presence must be enabled on the Cisco Unified CME router. See the “Enabling Presence for Internal
Lines” section on page 800.
• A directory number must be enabled as a presentity with the allow watch command to provide BLF
status notification. See the “Enabling a Directory Number to be Watched” section on page 801.
• SIP phones must be configured with a directory number under voice register pool configuration
mode (use dn keyword in number command); direct line numbers are not supported.

Restrictions
BLF Call-List
• Supported only on Cisco Unified IP Phone 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE.

BLF Speed-Dial
• Supported only on Cisco Unified IP Phone 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register pool pool-tag
4. number tag dn dn-tag
5. blf-speed-dial tag number label string
6. presence call-list
7. exit
8. voice register global
9. mode cme
10. create profile
11. restart
12. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone.
Example: • pool-tag—Unique sequence number of the SIP phone
Router(config)# voice register pool 1 to be configured. Range is version and
platform-dependent; type ? to display range. You can
modify the upper limit for this argument with the
max-pool command.
Step 4 number tag dn dn-tag Assigns a directory number to the SIP phone.
• tag—Identifier when there are multiple number
Example: commands. Range: 1 to 10.
Router(config-register-pool)# number 1 dn 2
• dn-tag—Directory number tag that was defined using
the voice register dn command.
Step 5 blf-speed-dial tag number label string Enables BLF monitoring of a directory number associated
with a speed-dial number on the phone.
Example: • tag—Number that identifies the speed-dial index.
Router(config-register-pool)# blf-speed-dial 3 Range: 1 to 7.
3001 label sales
• number—Telephone number to speed dial.
• string—Alphanumeric label that identifies the
speed-dial button. The string can contain a maximum of
30 characters.
Step 6 presence call-list Enables BLF monitoring of directory numbers that appear
in call lists and directories on this phone.
Example: • For a directory number to be monitored, it must have
Router(config-register-pool)# presence the allow watch command enabled.
call-list
• To enable BLF monitoring for call lists on all phones in
this Cisco Unified CME system, use this command in
presence mode. See the “Enabling Presence for Internal
Lines” section on page 800.
Step 7 exit Exits voice register pool configuration mode and enters
privileged EXEC mode.
Example:
Router(config-register-pool)# exit

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Command or Action Purpose

Step 8 voice register global Enters voice register global configuration mode to set
global parameters for all supported SIP phones in a
Cisco Unified CME environment.
Example:
Router(config)# voice register global
Step 9 mode cme Enables mode for provisioning SIP phones in
Cisco Unified CME.
Example:
Router(config-register-global)# mode cme
Step 10 create profile Generates configuration profile files required for SIP
phones and writes the files to the location specified with the
tftp-path command.
Example:
Router(config-register-global)# create profile
Step 11 restart Performs a fast reset of all SIP phones associated with this
Cisco Unified CME router. Phones contact the TFTP server
for updated configuration information and reregister
Example:
Router(config-register-global)# restart
without contacting the DHCP server.
Step 12 end Exits to privileged EXEC mode.

Example:
Router(config-register-global)# end

Configuring Presence to Watch External Lines

To enable internal watchers to monitor external directory numbers on a remote Cisco Unified CME
router, perform the following steps.

Prerequisites
Presence service must be enabled for internal lines. See the “Enabling Presence for Internal Lines”
section on page 800.

SUMMARY STEPS

1. enable
2. configure terminal
3. presence
4. server ip-address
5. allow subscribe
6. watcher all
7. sccp blf-speed-dial retry-interval seconds limit number
8. exit
9. voice register global
10. authenticate presence

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11. authenticate credential tag location

12. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 presence Enables presence service and enters presence configuration
mode.
Example:
Router(config)# presence
Step 4 server ip-address Specifies the IP address of a presence server for sending
presence requests from internal watchers to external
presentities.
Example:
Router(config-presence)# server 10.10.10.1
Step 5 allow subscribe Allows internal watchers to monitor external directory
numbers.
Example:
Router(config-presence)# allow subscribe
Step 6 watcher all Allows external watchers to monitor internal directory
numbers.
Example:
Router(config-presence)# watcher all
Step 7 sccp blf-speed-dial retry-interval seconds (Optional) Sets the retry timeout for BLF monitoring of
limit number speed-dial numbers on phones running SCCP.
• seconds—Retry timeout in seconds. Range: 60 to 3600.
Example: Default: 60.
Router(config-presence)# sccp blf-speed-dial
retry-interval 90 limit number 15 • number—Maximum number of retries.
Range: 10 to 100. Default: 10.
Step 8 exit Exits presence configuration mode.

Example:
Router(config-presence)# exit
Step 9 voice register global Enters voice register global configuration mode to set
global parameters for all supported SIP phones in a
Cisco Unified CME environment.
Example:
Router(config)# voice register global

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Command or Action Purpose

Step 10 authenticate presence (Optional) Enables authentication of incoming presence
requests from a remote presence server.
Example:
Router(config-register-global)# authenticate
presence
Step 11 authenticate credential tag location (Optional) Specifies the credential file to use for
authenticating presence subscription requests.
Example: • tag—Number that identifies the credential file to use
Router(config-register-global)# authenticate for presence authentication. Range: 1 to 5.
credential 1 flash:cred1.csv
• location—Name and location of the credential file in
URL format. Valid storage locations are TFTP, HTTP,
and flash memory.
Step 12 end Exits to privileged EXEC mode.

Example:
Router(config-register-global)# end

Verifying Presence Configuration

Step 1 show running-config
Use this command to verify your configuration.
Router# show running-config
!
voice register global
mode cme
source-address 10.1.1.2 port 5060
load 7971 SIP70.8-0-1-11S
load 7970 SIP70.8-0-1-11S
load 7961GE SIP41.8-0-1-0DEV
load 7961 SIP41.8-0-1-0DEV
authenticate presence
authenticate credential 1 tftp://172.18.207.15/labtest/cred1.csv
create profile sync 0004550081249644
.
.
.
presence
server 10.1.1.4
sccp blf-speed-dial retry-interval 70 limit 20
presence call-list
max-subscription 128
watcher all
allow subscribe
!
sip-ua
presence enable

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Step 2 show presence global

Use this command to display presence configuration settings.
Router# show presence global

Presence Global Configuration Information:

=============================================
Presence feature enable : TRUE
Presence allow external watchers : FALSE
Presence max subscription allowed : 100
Presence number of subscriptions : 0
Presence allow external subscribe : FALSE
Presence call list enable : TRUE
Presence server IP address : 0.0.0.0
Presence sccp blfsd retry interval : 60
Presence sccp blfsd retry limit : 10
Presence router mode : CME mode

Step 3 show presence subscription [details | presentity telephone-number | subid subscription-id summary]
Use this command to display information about active presence subscriptions.
Router# show presence subscription summary

Presence Active Subscription Records Summary: 15 subscription

Watcher Presentity SubID Expires SibID Status
======================== ======================== ====== ======= ====== ======
[email protected] [email protected] 1 3600 0 idle
[email protected] [email protected] 6 3600 0 idle
[email protected] [email protected] 8 3600 0 idle
[email protected] [email protected] 9 3600 0 idle
[email protected] [email protected] 10 3600 0 idle
[email protected] [email protected] 12 3600 0 idle
[email protected] [email protected] 15 3600 0 idle
[email protected] [email protected] 17 3600 0 idle
[email protected] [email protected] 19 3600 0 idle
[email protected] [email protected] 21 3600 0 idle
[email protected] [email protected] 23 3600 24 idle
[email protected] [email protected] 121 3600 0 idle
[email protected] [email protected] 128 3600 129 idle
[email protected] [email protected] 130 3600 131 busy
[email protected] [email protected] 132 3600 133 idle

Troubleshooting Presence
Step 1 debug presence {all | asnl | errors | event | info | timer | trace | xml}
This command displays debugging information about the presence service.
Router# debug presence errors

*Sep 4 07:16:02.715: //PRESENCE:[0]:/presence_sip_line_update: SIP nothing to update

*Sep 4 07:16:02.723: //PRESENCE:[17]:/presence_handle_notify_done: sip stack response
code [29]
*Sep 4 07:16:02.723: //PRESENCE:[24]:/presence_handle_notify_done: sip stack response
code [29]
*Sep 4 07:16:02.791: //PRESENCE:[240]:/presence_handle_notify_done: sip stack response
code [17]

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*Sep 4 07:16:02.791: //PRESENCE:[766]:/presence_handle_notify_done: sip stack response

code [17]
*Sep 4 07:16:04.935: //PRESENCE:[0]:/presence_sip_line_update: SIP nothing to update
*Sep 4 07:16:04.943: //PRESENCE:[17]:/presence_handle_notify_done: sip stack response
code [29]
*Sep 4 07:16:04.943: //PRESENCE:[24]:/presence_handle_notify_done: sip stack response
code [29]
*Sep 4 07:16:04.995: //PRESENCE:[240]:/presence_handle_notify_done: sip stack response
code [17]
*Sep 4 07:16:04.999: //PRESENCE:[766]:/presence_handle_notify_done: sip stack response
code [17]

Step 2 debug ephone blf [mac-address mac-address]

This command displays debugging information for BLF presence features.
Router# debug ephone blf

*Sep 4 07:18:26.307: skinny_asnl_callback: subID 16 type 4

*Sep 4 07:18:26.307: ASNL_RESP_NOTIFY_INDICATION
*Sep 4 07:18:26.307: ephone-1[1]:ASNL notify indication message, feature index 4, subID
[16]
*Sep 4 07:18:26.307: ephone-1[1]:line status 6, subID [16]
*Sep 4 07:18:26.307: ephone-1[1]:StationFeatureStatV2Message sent, status 2
*Sep 4 07:18:26.307: skinny_asnl_callback: subID 23 type 4
*Sep 4 07:18:26.307: ASNL_RESP_NOTIFY_INDICATION
*Sep 4 07:18:26.307: ephone-2[2]:ASNL notify indication message, feature index 2, subID
[23]
*Sep 4 07:18:26.311: ephone-2[2]:line status 6, subID [23]
*Sep 4 07:18:26.311: ephone-2[2]:StationFeatureStatV2Message sent, status 2
*Sep 4 07:18:28.951: skinny_asnl_callback: subID 16 type 4
*Sep 4 07:18:28.951: ASNL_RESP_NOTIFY_INDICATION
*Sep 4 07:18:28.951: ephone-1[1]:ASNL notify indication message, feature index 4, subID
[16]
*Sep 4 07:18:28.951: ephone-1[1]:line status 1, subID [16]
*Sep 4 07:18:28.951: ephone-1[1]:StationFeatureStatV2Message sent, status 1
*Sep 4 07:18:28.951: skinny_asnl_callback: subID 23 type 4
*Sep 4 07:18:28.951: ASNL_RESP_NOTIFY_INDICATION
*Sep 4 07:18:28.951: ephone-2[2]:ASNL notify indication message, feature index 2, subID
[23]
*Sep 4 07:18:28.951: ephone-2[2]:line status 1, subID [23]
*Sep 4 07:18:28.951: ephone-2[2]:StationFeatureStatV2Message sent, status 1

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Configuration Examples for Presence

Configuration Examples for Presence

This section contains the following example:
• Presence in Cisco Unified CME: Example, page 813

Presence in Cisco Unified CME: Example

Router# show running-config

Building configuration...

Current configuration : 5465 bytes

!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname CME-3825
!
boot-start-marker
boot-end-marker
!
logging buffered 2000000 debugging
enable password lab
!
no aaa new-model
!
resource policy
!
no network-clock-participate slot 1
no network-clock-participate slot 2
ip cef
!
!
no ip domain lookup
!
voice-card 1
no dspfarm
!
voice-card 2
no dspfarm
!
!
voice service voip
allow-connections sip to sip
h323
sip
registrar server expires max 240 min 60
!
voice register global
mode cme
source-address 11.1.1.2 port 5060
load 7971 SIP70.8-0-1-11S
load 7970 SIP70.8-0-1-11S
load 7961GE SIP41.8-0-1-0DEV
load 7961 SIP41.8-0-1-0DEV
authenticate presence
authenticate credential 1 tftp://172.18.207.15/labtest/cred1.csv
create profile sync 0004550081249644

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Configuration Examples for Presence

!
voice register dn 1
number 2101
allow watch
!
voice register dn 2
number 2102
allow watch
!
voice register pool 1
id mac 0015.6247.EF90
type 7971
number 1 dn 1
blf-speed-dial 1 1001 label "1001"
!
voice register pool 2
id mac 0012.0007.8D82
type 7912
number 1 dn 2
!
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 11.1.1.2 255.255.255.0
duplex full
speed 100
media-type rj45
no negotiation auto
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
media-type rj45
negotiation auto
!
ip route 0.0.0.0 0.0.0.0 11.1.1.1
!
ip http server
!
!
!
tftp-server flash:Jar41sccp.8-0-0-103dev.sbn
tftp-server flash:cvm41sccp.8-0-0-102dev.sbn
tftp-server flash:SCCP41.8-0-1-0DEV.loads
tftp-server flash:P00303010102.bin
tftp-server flash:P00308000100.bin
tftp-server flash:P00308000100.loads
tftp-server flash:P00308000100.sb2
tftp-server flash:P00308000100.sbn
tftp-server flash:SIP41.8-0-1-0DEV.loads
tftp-server flash:apps41.1-1-0-82dev.sbn
tftp-server flash:cnu41.3-0-1-82dev.sbn
tftp-server flash:cvm41sip.8-0-0-103dev.sbn
tftp-server flash:dsp41.1-1-0-82dev.sbn
tftp-server flash:jar41sip.8-0-0-103dev.sbn
tftp-server flash:P003-08-1-00.bin
tftp-server flash:P003-08-1-00.sbn
tftp-server flash:P0S3-08-1-00.loads
tftp-server flash:P0S3-08-1-00.sb2
tftp-server flash:CP7912080000SIP060111A.sbin
tftp-server flash:CP7912080001SCCP051117A.sbin
tftp-server flash:SCCP70.8-0-1-11S.loads
tftp-server flash:cvm70sccp.8-0-1-13.sbn

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Configuration Examples for Presence

tftp-server flash:jar70sccp.8-0-1-13.sbn
tftp-server flash:SIP70.8-0-1-11S.loads
tftp-server flash:apps70.1-1-1-11.sbn
tftp-server flash:cnu70.3-1-1-11.sbn
tftp-server flash:cvm70sip.8-0-1-13.sbn
tftp-server flash:dsp70.1-1-1-11.sbn
tftp-server flash:jar70sip.8-0-1-13.sbn
!
control-plane
!
dial-peer voice 2001 voip
preference 2
destination-pattern 1...
session protocol sipv2
session target ipv4:11.1.1.4
dtmf-relay sip-notify
!
presence
server 11.1.1.4
sccp blf-speed-dial retry-interval 70 limit 20
presence call-list
max-subscription 128
watcher all
allow subscribe
!
sip-ua
authentication username jack password 021201481F
presence enable
!
!
telephony-service
load 7960-7940 P00308000100
load 7941GE SCCP41.8-0-1-0DEV
load 7941 SCCP41.8-0-1-0DEV
load 7961GE SCCP41.8-0-1-0DEV
load 7961 SCCP41.8-0-1-0DEV
load 7971 SCCP70.8-0-1-11S
load 7970 SCCP70.8-0-1-11S
load 7912 CP7912080000SIP060111A.sbin
max-ephones 100
max-dn 300
ip source-address 11.1.1.2 port 2000
url directories http://11.1.1.2/localdirectory
max-conferences 6 gain -6
call-forward pattern .T
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn 1 dual-line
number 2001
allow watch
!
!
ephone-dn 2 dual-line
number 2009
allow watch
application default
!
!
ephone-dn 3
number 2005
allow watch

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Configuration Examples for Presence

!
!
ephone-dn 4 dual-line
number 2002
!
!
ephone 1
mac-address 0012.7F57.62A5
fastdial 1 1002
blf-speed-dial 1 2101 label "2101"
blf-speed-dial 2 1003 label "1003"
blf-speed-dial 3 2002 label "2002"
type 7960
button 1:1 2:2
!
!
!
ephone 3
mac-address 0015.6247.EF91
blf-speed-dial 2 1003 label "1003"
type 7971
button 1:3 2:4
!
!
!
line con 0
exec-timeout 0 0
password lab
stopbits 1
line aux 0
stopbits 1
line vty 0 4
password lab
login
!
scheduler allocate 20000 1000
!
end

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Presence Service

Feature Information for Presence Service

Table 46 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 46 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 46 Feature Information for Presence Service

Cisco Unified CME

Feature Name Version Modification
Presence Service 4.1 Presence with BLF was introduced.

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 Configuring Ring Tones

Last Updated: March 26, 2007

This chapter describes ring tones features in Cisco Unified Communications Manager Express
(Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Ring Tones” section
on page 828.

Contents
• Information About Ring Tones, page 819
• How to Configure Ring Tones, page 821
• Configuration Examples for Ring Tones, page 826
• Additional References, page 827
• Feature Information for Ring Tones, page 828

Information About Ring Tones

To enable distinctive ringing or customized ring tones, you should understand the following concepts:
• Distinctive Ringing, page 820
• Customized Ring Tones, page 820
• On-Hold Indicator, page 820

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Information About Ring Tones

Distinctive Ringing
Distinctive ring is used to identify internal and external incoming calls. An internal calls is defined as a
call originating from any Cisco Unified IP phone that is registered in Cisco Unified CME or is routed
through the local FXS port.
In Cisco CME 3.4 and earlier versions, the standard ring pattern is generated for all calls to local SCCP
endpoints. In Cisco Unified CME 4.0, the following distinctive ring features are supported for SCCP
endpoints:
• Specify one of three ring patterns to be used for all types of incoming calls to a particular directory
number, on all phones on which the directory number appears. If a phone is already in use, an
incoming call is presented as a call-waiting call and uses a distinctive call-waiting beep.
• Specify whether the distinctive ring is used only if the incoming called number matches the primary
or secondary number defined for the ephone-dn. If no secondary number is defined for the
ephone-dn, the secondary ring option has no effect.
• Associate a feature ring pattern with a specific button on a phone so that different phones that share
the same directory number can use a different ring style.
For local SIP endpoints, the type of ring sound requested is signaled to the phone using an alert-info
signal. If distinctive ringing is enabled, Cisco Unified CME generates the alert-info for incoming calls
from any phone that is not registered in Cisco Unified CME, to the local endpoint. Alert-info from an
incoming leg can be relayed to an outgoing leg with the internally generated alert-info taking
precedence.
Cisco Unified IP phones use the standard Telcordia Technologies distinctive ring types.

Customized Ring Tones

Cisco Unified IP Phones have two default ring types: Chirp1 and Chirp2. Cisco Unified CME also
supports customized ring tones using pulse code modulation (PCM) files.
An XML file called RingList.xml specifies the ring tone options available for the default ring on an IP
phone registered to Cisco Unified CME. An XML file called DistinctiveRingList.xml specifies the ring
tones available on each individual line appearance on an IP phone registered to Cisco Unified CME.

On-Hold Indicator
On-hold indicator is an optional feature that generates a ring burst on idle IP phones that have placed a
call on hold. An option is available to generate call-waiting beeps for occupied phones that have placed
calls on hold. This feature is disabled by default. For configuration information, see the “SCCP:
Enabling On-Hold Indicator” section on page 824.
LED color display for hold state, also known as I-Hold, is supported in Cisco Unified CME 4.0(2) and
later versions. The I-Hold feature provides a visual indicator for distinguishing a local hold from a
remote hold on shared lines on supported phones, such as the Cisco Unified IP Phone 7931G. This
feature requires no additional configuration.

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How to Configure Ring Tones

How to Configure Ring Tones

This section contains the following tasks:
• SCCP: Enabling Distinctive Ringing, page 821
• SCCP: Enabling Customized Ring Tones, page 822
• SCCP: Enabling On-Hold Indicator, page 824
• SIP: Enabling Distinctive Ringing, page 825

SCCP: Enabling Distinctive Ringing

To set the ring pattern for all incoming calls to a directory number, perform the following steps.

Prerequisites
Cisco Unified CME 4.0 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. number number [secondary number] [no-reg [both | primary]]
5. ring {external | internal | feature} [primary | secondary]
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an
ephone-dn, and optionally assigns it dual-line status.
Example:
Router(config)# ephone-dn 29

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How to Configure Ring Tones

Command or Action Purpose

Step 4 number number [secondary number] [no-reg [both Configures a valid extension number for this ephone-dn.
| primary]]

Example:
Router(config-ephone-dn)# number 2333
Step 5 ring {external | internal | feature} [primary | Designates which ring pattern to be used for all types of
secondary] incoming calls to this directory number, on all phones on
which the directory number appears.
Example:
Router(config-ephone-dn)# ring internal
Step 6 end Returns to privileged EXEC mode.

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

SCCP: Enabling Customized Ring Tones

To create a customized ring tone, perform the following steps.

Prerequisites
Cisco Unified CME 4.0 or a later version.

SUMMARY STEPS

1. Create PCM file.

2. Edit RingList.xml and DistinctiveRingList.xml.
3. Copy PCM and XML files to system Flash.
4. tftp-server
5. Reboot phones.

DETAILED STEPS

Step 1 Create a PCM file for each customized ring tone (one ring per file). The PCM files must comply with
the following format guidelines.
• Raw PCM (no header)
• 8000 samples per second
• 8 bits per sample
• µLaw compression
• Maximum ring size—16080 samples
• Minimum ring size—240 samples

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How to Configure Ring Tones

• Number of samples in the ring must be evenly divisible by 240

• Ring should start and end at the zero crossing
Use an audio editing package that supports these file format requirements to create PCM files for
customized phone rings.
Sample ring files are in the ringtone.tar file at http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp
Step 2 Edit the RingList.xml and DistinctiveRingList.xml files using a text editor.
The RingList.xml and DistinctiveRingList.xml files contain a list of phone ring types. Each file shows
the PCM file used for each ring type and the text that is displayed on the Ring Type menu on a
Cisco Unified IP Phone for each ring.
Sample XML files are in the ringtone.tar file at http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp
The RingList.xml and DistinctiveRingList.xml files use the following format to specify customized
rings:
<CiscoIPPhoneRingList>
<Ring>
<DisplayName/>
<FileName/>
</Ring>
</CiscoIPPhoneRingList>

The XML ring files use the following tag definitions:

• Ring files contain two fields, DisplayName and FileName, which are required for each phone ring
type. Up to 50 rings can be listed.
• DisplayName defines the name of the customized ring for the associated PCM file that will be
displayed on the Ring Type menu of the Cisco Unified IP Phone.
• FileName specifies the name of the PCM file for the customized ring to associate with
DisplayName.
• The DisplayName and FileName fields can not exceed 25 characters.
The following sample RingList.xml file defines two phone ring types:
<CiscoIPPhoneRingList>
<Ring>
<DisplayName>Piano1</DisplayName>
<FileName>Piano1.raw</FileName>
</Ring>
<Ring>
<DisplayName>Chime</DisplayName>
<FileName>Chime.raw</FileName>
</Ring>
</CiscoIPPhoneRingList>

Step 3 Copy the PCM and XML files to system Flash on the Cisco Unified CME router. For example:
copy tftp://192.168.1.1/RingList.xml flash:
copy tftp://192.168.1.1/DistinctiveRingList.xml flash:
copy tftp://192.168.1.1/Piano1.raw flash:
copy tftp://192.168.1.1/Chime.raw flash:

Step 4 Use the tftp-server command to enable access to the files. For example:
tftp-server flash:RingList.xml
tftp-server flash:DistinctiveRingList.xml
tftp-server flash:Piano1.raw
tftp-server flash:Chime.raw

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 Configuring Ring Tones
How to Configure Ring Tones

Step 5 Reboot the IP phones. After reboot, the IP phones download the XML and ring tone files. Select the
customized ring by pressing the Settings button followed by the Ring Type menu option on a phone.

SCCP: Enabling On-Hold Indicator

The Call Hold feature is available by default. To define an audible indicator as a reminder that a call is
waiting on hold, perform the following steps.

SUMMARY STEPS
1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. hold-alert timeout {idle | originator | shared}
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an ephone-dn, and
optionally assigns it dual-line status.
Example:
Router(config)# ephone-dn 20
Step 4 hold-alert timeout {idle | originator | Sets audible alert notification on the Cisco Unified IP phone for
shared} alerting the user about on-hold calls.
Note From the perspective of the originator of the call on hold,
Example: the originator and shared keywords provide the same
Router(config-ephone-dn)# hold-alert 15 functionality.
idle
Step 5 end Returns to privileged EXEC mode.

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

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How to Configure Ring Tones

SIP: Enabling Distinctive Ringing

To set the ring pattern for distinguishing between external and internal incoming calls, perform the
following steps.

Prerequisites
Cisco Unified CME 3.4 or a later version.

Restrictions
bellcore-dr1 to bellcore-dr5 are the only Telcordia options that are supported for SIP phones.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. external-ring {bellcore-dr1 | bellcore-dr2 | bellcore-dr3 | bellcore-dr4 | bellcore-dr5}
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global

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 Configuring Ring Tones
Configuration Examples for Ring Tones

Command or Action Purpose

Step 4 external-ring {bellcore-dr1 | bellcore-dr2 | Specifies the type of audible ring sound to be used for
bellcore-dr3 | bellcore-dr4 | bellcore-dr5} external calls
• Default—Internal ring sound is used for all incoming
Example: calls.
Router(config-register-global)# external-ring
bellcore-dr3
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-global)# end

Configuration Examples for Ring Tones

This section contains the following examples:
• Distinctive Ringing for Internal Calls: Example, page 826
• On-Hold Indicator: Example, page 826

Distinctive Ringing for Internal Calls: Example

The following example sets distinctive ringing for internal calls on extension 2333.
ephone-dn 34
number 2333
ring internal

On-Hold Indicator: Example

In the following example, extension 2555 is configured to not forward local calls that are internal to the
Cisco Unified CME system. Extension 2222 dials extension 2555. If 2555 is busy, the caller hears a busy
tone. If 2555 does not answer, the caller hears ringback. The internal call is not forwarded.
ephone-dn 25
number 2555
no forward local-calls
call-forward busy 2244
call-forward noan 2244 timeout 45

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 Configuring Ring Tones
Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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 Configuring Ring Tones
Feature Information for Ring Tones

Feature Information for Ring Tones

Table 47 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 47 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 47 Feature Information for Ring Tones

Cisco Unified CME

Feature Name Version Feature Information
Distinctive Ringing 4.0 Supports ring tones choices for all incoming calls to an
individual directory number, for all SCCP phones on which
the directory number appears.
3.4 Generate the alert-info for incoming calls from any phone
that is not registered in Cisco Unified CME, to local SIP
endpoints.
Customized Ring Tones 4.0 Customized Ring Tones feature was introduced.
On-Hold Indictor 4.0(2) Controls LED color display for hold state to provide visual
indicator for distinguishing a local hold from a remote hold
on shared lines on supported phones, such as the
Cisco Unified IP Phone 7931G.
2.0 Audible on-hold indicator was introduced.
1.0 Call Hold was introduced.

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 Customizing Soft Keys

Last Updated: June 20, 2007

This chapter describes the soft-key features in Cisco Unified Communications Manager Express
(Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Soft Keys” section
on page 845.

Contents
• Information About Soft Keys, page 829
• How to Customize Soft Keys, page 832
• Configuration Examples for Soft-Keys, page 842
• Where to Go Next, page 844
• Additional References, page 844
• Feature Information for Soft Keys, page 845

Information About Soft Keys

To customize soft keys on IP phones, you should understand the following concepts:
• Soft Keys on IP Phones, page 830
• Account Code Entry, page 831
• Hookflash Soft Key, page 831
• Feature Blocking, page 832

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Soft Keys on IP Phones

You can customize the display and order of soft keys that appear during various call states on individual
IP phones. Soft keys that are appropriate in each call state are displayed by default. Using phone
templates, you can delete soft keys that would normally appear or change the order in which the soft
keys appear. For example, you might want to display the CFwdAll and Confrn soft keys on a manager's
phone and remove these soft keys from a receptionist's phone.
You can modify soft keys for the following call states:
• 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.
• Connected—When the connection to a remote point is established.
• Hold—When a connected party is still connected but there is temporarily no voice connection.
• Idle—Before a call is made and after a call is complete.
• Seized—When a caller is attempting a call but has not yet been connected.
• Ringing—For Cisco Unified CME 4.2 or a later version. After a call is received and before the call
is connected.
Not all soft keys are available in all call states. Use the CLI help to see the available soft keys for each
call state. The soft keys are as follows:
• Acct—Short for “account code.” Provides access to configured accounts.
• Answer—Picks up incoming call.
• Callback—Requests callback notification when a busy called line becomes free.
• CFwdALL—Short for “call forward all.” Forwards all calls.
• Confrn—Short for “conference.” Connects callers to a conference call.
• DND—Short for “do not disturb.” Enables the do-not-disturb features.
• 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.
• 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.
• HLog—Places the phone of an ephone-hunt group agent into the not-ready status or, if the phone is
in the not-ready status, it places the phone into the ready status.
• Hold—Places an active call on hold and resumes the call.
• Login—Provides personal identification number (PIN) access to restricted phone features.
• NewCall—Opens a line on a speakerphone to place a new call.
• Park—Places an active call on hold so it can be retrieved from another phone in the system.
• PickUp—Selectively picks up calls coming into another extension.
• Redial—Redials the last number dialed.
• Trnsfer—Short for “call transfer.” Transfers an active call to another extension.
You change the soft-key order by defining a phone template and applying the template to one or more
phones. You can create up to 20 phone templates for SCCP phones and ten templates for SIP phones.
Only one template can be applied to a phone. If you apply a second phone template to a phone that

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already has a template applied to it, the second template overwrites the first phone template information.
The new information takes effect only after you generate a new configuration file and restart the phone,
otherwise the previously configured template remains in effect.
In Cisco Unified CME 4.1, customizing the soft key display for IP phones running SIP is supported only
for the Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE.
For configuration information, see the “How to Customize Soft Keys” section on page 832.

Account Code Entry

The Cisco Unified IP Phones 7940 and 7940G and the Cisco Unified IP Phones 7960 and 7960G allow
phone users to enter account codes during call setup or when connected to an active call using the Acct
soft key. Account codes are inserted into call detail records (CDRs) on the Cisco Unified 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 in the “account-code” RADIUS vendor-specific attribute (VSA) for voice
authentication, authorization, and accounting (AAA).
To enter an account code during call setup or when in a connected state, press the Acct soft key, enter
the account code using the phone keypad, then press the # key to notify Cisco Unified CME that the last
digit of the code has been entered. The account code digits are processed upon receipt of the # and appear
in the show output after processing.
No configuration is required for this feature.

Note If Cisco Unified CME does not receive a #, each account code digit is processed only after a timer
expires. The timer is 30 seconds for the first digit entered, then x seconds for each subsequent digit,
where x equals the number of seconds configured with the timeouts interdigit (telephony-service)
command. The default value for the interdigit timeout is 10 seconds. The account code digits do not
appear in show output until after being processed.

Hookflash Soft Key

The Flash soft key provides hookflash functionality for calls made on FXO trunks. 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 hookflash
functionality for IP phones that use foreign exchange office (FXO) lines attached to the
Cisco Unified CME system.
When a Flash soft key is enabled on an IP phone, it can provide hookflash functionality during all calls
except for local IP-phone-to-IP-phone calls. Hookflash-controlled services can be activated only if they
are supported by the PSTN connection that is involved in the call. The availability of the Flash soft key
does not guarantee that hookflash-based services are accessible to the phone user.
For configuration information, see the “Enabling Flash Soft Key” section on page 838.

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Feature Blocking
In Cisco Unified CME 4.0 and later versions, individual soft-key features can be blocked on one or more
phones. You specify the features that you want blocked by adding the features blocked command to an
ephone template. The template is then applied under ephone configuration mode to one or more ephones.
If a feature is blocked using the features blocked command, the soft key is not removed, but it does not
function. For configuration information, see the “Configuring Feature Blocking” section on page 840.
To remove a soft-key display, use the appropriate no softkeys command. See the “SCCP: Modifying
Soft-Key Display” section on page 832.

How to Customize Soft Keys

This section contains the following tasks:
• SCCP: Modifying Soft-Key Display, page 832
• SIP: Modifying Soft-Key Display, page 835
• Verifying Soft-Key Configuration, page 837
• Enabling Flash Soft Key, page 838
• Verifying Flash Soft-Key Configuration, page 839
• Configuring Feature Blocking, page 840
• Verifying Feature Blocking, page 842

SCCP: Modifying Soft-Key Display

To modify the display of soft-keys, perform the following steps.

Prerequisites
• Cisco CME 3.2 or a later version.
• Cisco Unified 4.2 or a later version to enable soft keys during the ringing call state.
• The HLog soft key must be enabled with the hunt-group logout HLog command before it will be
displayed. For more information, see the “SCCP: Configuring Hunt Groups” section on page 596.
• The Flash soft key must be enabled with the fxo hook-flash command before it will be displayed.
For configuration information, see the “Enabling Flash Soft Key” section on page 838.

Restrictions
• The third soft-key button on the Cisco Unified IP Phone 7905G and Cisco Unified 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 Redial, DND, Message, PickUp, Login, and GPickUp soft keys.
• The NewCall soft key cannot be disabled on the Cisco Unified IP Phone 7905G or Cisco Unified IP
Phone 7912G.

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SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-template template-tag
4. softkeys alerting {[Acct] [Callback] [Endcall]}
5. softkeys connected {[Acct] [Confrn] [Endcall] [Flash] [Hlog] [Hold] [Park] [Trnsfer]}
6. softkeys hold {[Newcall] [Resume]}
7. softkeys idle {[Cfwdall] [Dnd] [Gpickup] [Hlog] [Login] [Newcall] [Pickup] [Redial]}
8. softkeys seized {[Cfwdall] [Endcall] [Gpickup] [Hlog] [Pickup] [Redial]}
9. softkeys ringing {[Answer] [Dnd] [HLog]}
10. exit
11. ephone phone-tag
12. ephone-template template-tag
13. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-template template-tag Enters ephone-template configuration mode to create an ephone
template.
Example: • template-tag—Unique identifier for the ephone template that
Router(config)# ephone-template 15 is being created. Range is 1 to 20.
Step 4 softkeys alerting {[Acct] [Callback] (Optional) Configures an ephone template for soft-key display
[Endcall]} during the alerting call state.
• You can enter any of the keywords in any order.
Example:
Router(config-ephone-template)# softkeys
• Default is all soft keys are displayed in alphabetical order.
alerting Callback Endcall • Any soft key that is not explicitly defined is disabled.
Step 5 softkeys connected {[Acct] [Confrn] (Optional) Configures an ephone template for soft-key display
[Endcall] [Flash] [Hlog] [Hold] [Park] during the call-connected state.
[Trnsfer]}
• You can enter any of the keywords in any order.

Example: • Default is all soft keys are displayed in alphabetical order.

Router(config-ephone-template)# softkeys • Any soft key that is not explicitly defined is disabled.
connected Endcall Hold Transfer Hlog

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Command or Action Purpose

Step 6 softkeys hold {[Newcall] {Resume]} (Optional) Configures an ephone template for soft-key display
during the call-hold state.
Example: • You can enter any of the keywords in any order.
Router(config-ephone-template)# softkeys
hold Resume
• Default is all soft keys are displayed in alphabetical order.
• Any soft key that is not explicitly defined is disabled.
Step 7 softkeys idle {[Cfwdall] [Dnd] [Gpickup] (Optional) Configures an ephone template for soft-key display
[Hlog] [Login] [Newcall] [Pickup] during the idle state.
[Redial]}
• You can enter any of the keywords in any order.

Example: • Default is all soft keys are displayed in alphabetical order.

Router(config-ephone-template)# softkeys • Any soft key that is not explicitly defined is disabled.
idle Newcall Redial Pickup Cfwdall Hlog
Step 8 softkeys seized {[Cfwdall] [Endcall] (Optional) Configures an ephone template for soft-key display
[Gpickup] [Hlog] [Pickup] [Redial]} during the seized state.
• You can enter any of the keywords in any order.
Example:
Router(config-ephone-template)# softkeys
• Default is all soft keys are displayed in alphabetical order.
seized Endcall Redial Pickup Cfwdall Hlog • Any soft key that is not explicitly defined is disabled.
Step 9 softkeys ringing {[Answer] [Dnd] [HLog]} (Optional) Configures an ephone template for soft-key display
during the ringing state.
Example: • You can enter any of the keywords in any order.
Router(config-ephone-template)# softkeys
ringing Answer Dnd Hlog
• Default is all soft keys are displayed in alphabetical order.
• Any soft key that is not explicitly defined is disabled.
Step 10 exit Exits ephone-template configuration mode.

Example:
Router(config-ephone-template)# exit
Step 11 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies this
Example: ephone during configuration tasks.
Router(config)# ephone 36
Step 12 ephone-template template-tag Applies an ephone template to the ephone that is being
configured.
Example:
Router(config-ephone)# ephone-template 15
Step 13 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

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What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “SCCP: Generating Configuration Files for SCCP Phones” section
on page 247.

SIP: Modifying Soft-Key Display

To modify the display of soft keys on SIP phones for different call states, perform the following steps.

Prerequisites
Cisco Unified CME 4.1 or a later version.

Restrictions
• This feature is supported only for Cisco Unified IP Phones 7911G, 7941G, 7941GE, 7961G,
7961GE, 7970G, and 7971GE.
• You can download a custom soft key XML file from a TFTP server, however if the soft key XML
file contains an error, the soft keys might not work properly on the phone. We recommend the
following procedure for creating a soft key template in Cisco Unified CME.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register template template-tag
4. softkeys connected {[Confrn] [Endcall] [Hold] [Trnsfer]}
5. softkeys hold {[Newcall] [Resume]}
6. softkeys idle {[Cfwdall] [Newcall] [Redial]}
7. softkeys seized {[Cfwdall] [Endcall] [Redial]}
8. exit
9. voice register pool pool-tag
10. template template-tag
11. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register template template-tag Enters voice register template configuration mode to create
a SIP phone template.
Example: • template-tag—Range: 1 to 10.
Router(config)# voice register template 9
Step 4 softkeys connected {[Confrn] [Endcall] [Hold] (Optional) Configures an SIP phone template for soft-key
[Trnsfer]} display during the call-connected state.
• You can enter the keywords in any order.
Example:
Router(config-register-template)# softkeys
• Default is all soft keys are displayed in alphabetical
connected Endcall Hold Transfer order.
• Any soft key that is not explicitly defined is disabled.
Step 5 softkeys hold {[Newcall] {Resume]} (Optional) Configures a phone template for soft-key display
during the call-hold state.
Example: • Default is that the NewCall and Resume soft keys are
Router(config-register-template)# softkeys hold displayed in alphabetical order.
Resume
• Any soft key that is not explicitly defined is disabled.
Step 6 softkeys idle {[Cfwdall] [Newcall] [Redial]} (Optional) Configures a phone template for soft-key display
during the idle state.
Example: • You can enter the keywords in any order.
Router(config-register-template)# softkeys idle
Newcall Redial Cfwdall
• Default is all soft keys are displayed in alphabetical
order.
• Any soft key that is not explicitly defined is disabled.
Step 7 softkeys seized {[Cfwdall] [Endcall] [Redial]} (Optional) Configures a phone template for soft-key display
during the seized state.
Example: • You can enter the keywords in any order.
Router(config-register-template)# softkeys
seized Endcall Redial Cfwdall
• Default is all soft keys are displayed in alphabetical
order.
• Any soft key that is not explicitly defined is disabled.
Step 8 exit Exits voice register template configuration mode.

Example:
Router(config-register-template)# exit

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Command or Action Purpose

Step 9 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone.
Example:
Router(config)# voice register pool 36
Step 10 template template-tag Applies a SIP phone template to the phone you are
configuring.
Example: • template-tag— Template tag that was created with the
Router(config-register-pool)# template 9 voice register template command in Step 3
Step 11 end Exits to privileged EXEC mode.

Example:
Router(config-register-pool)# end

What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “SIP: Generating Configuration Profiles for SIP Phones” section on
page 250.

Verifying Soft-Key Configuration

Step 1 show running-config
Use this command to verify your configuration. In the following example, the soft-key display is
modified in phone template 7 and the template is applied to SIP phone 2. All other phones use the default
arrangement of soft keys.
Router# show running-config
!
ephone-dn 1 dual-line
ring feature secondary
number 126 secondary 1261
description Sales
name Smith
call-forward busy 500 secondary
call-forward noan 500 timeout 10
huntstop channel
no huntstop
no forward local-calls
!
!
voice register template 7
session-transport tcp
softkeys hold Resume Newcall
softkeys idle Newcall Redial Cfwdall
softkeys connected Endcall Trnsfer Confrn Hold
voicemail 52001 timeout 30
.
.
.
voice register pool 2
id mac 0030.94C2.A22A

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number 1 dn 4
template 7
dialplan 3
!
Step 2 show telephony-service ephone-template
or
show voice register template template-tag
This command displays the contents of individual templates.
Router# show telephony-service ephone-template

ephone-template 1
softkey ringing Answer Dnd
conference drop-mode never
conference add-mode all
conference admin: No
Always send media packets to this router: No
Preferred codec: g711ulaw
User Locale: US
Network Locale: US

or
Router# show voice register template 7

Temp Tag 7
Config:
Attended Transfer is enabled
Blind Transfer is enabled
Semi-attended Transfer is enabled
Conference is enabled
Caller-ID block is disabled
DnD control is enabled
Anonymous call block is disabled
Voicemail is 52001, timeout 30
KPML is disabled
Transport type is tcp
softkey connected Endcall Trnsfer Confrn Hold
softkey hold Resume Newcall
softkey idle Newcall Redial Cfwdall

Enabling Flash Soft Key

To enable the flash soft key, perform the following steps.

Restrictions
The IP phone must support soft-key display.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service

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4. fxo hook-flash
5. restart all
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 fxo hook-flash Enables the Flash soft key on phones that support soft-key
display, on PSTN calls using an FXO port.
Example: Note The Flash soft key display is automatically disabled
Router(config-telephony)# fxo hook-flash for local IP-phone-to-IP-phone calls.
Step 5 restart all Performs a fast reboot of all phones associated with this
Cisco Unified CME router. Does not contact the DHCP or
TFTP server for updated information.
Example:
Router(config-telephony)# restart all
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Verifying Flash Soft-Key Configuration

Step 1 Use the show running-config command to display an entire configuration, including Flash soft key,
which is listed in the telephony-service portion of the output.
Router# show running-config

telephony-service
fxo hook-flash
load 7960-7940 P00305000600
load 7914 S00103020002
max-ephones 100
max-dn 500
.
.
.

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Step 2 Use the show telephony-service command to show only the telephony-service portion of the
configuration.

Configuring Feature Blocking

To configure feature blocking for SCCP phones, perform the following steps.

Prerequisites
Cisco Unified CME 4.0 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-template template-tag
4. features blocked [CFwdAll] [Confrn] [GpickUp] [Park] [PickUp] [Trnsfer]
5. exit
6. ephone phone-tag
7. ephone-template template-tag
8. restart
9. Repeat Step 5 to Step 8 for each phone to which the template should be applied.
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-template template-tag Enters ephone-template configuration mode.
• template-tag—Unique sequence number that identifies
Example: this template during configuration tasks. Range is
Router(config)# ephone-template 1 1 to 20.

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Command or Action Purpose

Step 4 features blocked [CFwdAll] [Confrn] [GpickUp] Prevents the specified soft key from invoking its feature.
[Park] [PickUp] [Trnsfer]
• CFwdAll—Call forward all calls.
• Confrn—Conference.
Example:
Router(config-ephone-template)# features • GpickUp—Group call pickup.
blocked Park Trnsfer
• Park—Call park.
• PickUp—Directed or local call pickup. This includes
pickup last-parked call and pickup from another
extension or park slot.
• Trnsfer—Call transfer.
Step 5 exit Exits ephone-template configuration mode.

Example:
Router(config-ephone-template)# exit
Step 6 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 25 number of ephones for a particular Cisco Unified CME
system is version- and platform-specific. For the range
of values, see the CLI help.
Step 7 ephone-template template-tag Applies an ephone template to an ephone.
• template-tag—Template number that you want to apply
Example: to this ephone.
Router(config-ephone)# ephone-template 1
Note To view your ephone-template configurations, use
the show telephony-service ephone-template
command.
Step 8 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 9 Repeat Step 5 to Step 8 for each phone to which the —
template should be applied.
Step 10 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

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Verifying Feature Blocking

Step 1 Use the show running-config command to display the running configuration, including ephone
templates and ephone configurations.
Step 2 Use the show telephony-service ephone-template command and the show telephony-service ephone
command to display only the contents of ephone templates and the ephone configurations.

Configuration Examples for Soft-Keys

This section contains the following examples:
• Modifying Soft-Key Display: Example, page 842
• Modifying the HLog Soft Key for Ephone Hunt Groups: Example, page 843
• Enabling Flash Soft Key for PSTN Calls: Example, page 843
• Park and Transfer Blocking: Example, page 843
• Conference Blocking: Example, page 843

Modifying Soft-Key Display: Example

The following example modifies soft-key display on four phones by creating two ephone templates.
Ephone template 1 is applied to ephone 11, 13, and 15. Template 2 is applied to ephone 34. The soft-key
displays on all other phones use the default arrangement of keys.
ephone-template 1
softkeys idle Redial Newcall
softkeys connected Endcall Hold Trnsfer

ephone-template 2
softkeys idle Redial Newcall
softkeys seized Redial Endcall Pickup
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

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Modifying the HLog Soft Key for Ephone Hunt Groups: Example
The following example establishes the appearance and order of soft keys for phones that are configured
with ephone-template 7. The Hlog key is available when a phone is idle, when it has seized a line, or
when it is connected to a call. Phones without soft keys can use the standard HLog codes to toggle ready
and not-ready status.
telephony-service
hunt-group logout HLog
fac standard
.
.
ephone-template 7
softkeys connected Endcall Hold Transfer Hlog
softkeys idle Newcall Redial Pickup Cfwdall Hlog
softkeys seized Endcall Redial Pickup Cfwdall Hlog

Enabling Flash Soft Key for PSTN Calls: Example

The following example enables the Flash soft key for PSTN calls through an FXO voice port.
telephony-service
fxo hook-flash

Park and Transfer Blocking: Example

The following example blocks the use of Park and Transfer soft keys on extension 2333.
ephone-template 1
features blocked Park Trnsfer

ephone-dn 2
number 2333

ephone 3
button 1:2
ephone-template 1

Conference Blocking: Example

The following example blocks the conference feature on extension 2579, which is on an analog phone.
ephone-template 1
features blocked Confrn

ephone-dn 78
number 2579

ephone 3
ephone-template 1
mac-address C910.8E47.1282
type anl
button 1:78

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Where to Go Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. For more information, see “Generating Configuration Files for Phones” on
page 245.

Ephone Templates
The softkeys commands are included in ephone templates that are applied to one or more individual
ephones. For more information about templates, see “Creating Templates” on page 881.

HLog Soft Key

The HLog soft key must be enabled with the hunt-group logout HLog command before it will be
displayed. For more information, see “Configuring Call-Coverage Features” on page 563.

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
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resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Soft Keys

Feature Information for Soft Keys

Table 48 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 48 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 48 Feature Information for Soft Keys

Cisco Unified CME

Feature Name Version Feature Information
Account Code Entry 3.0 Account code entry was introduced.
Feature Blocking 4.0 Feature blocking was introduced.
Flash Soft Key 3.0 Flash soft key was introduced.
Soft-Key Display 4.1 Configurable soft key display for IP phones running SIP is
supported for the Cisco Unified IP Phone 7911G, 7941G,
7941GE, 7961G, 7961GE, 7970G, and 7971GE
4.0 • An optional HLog soft key was added to the
connected, idle, and seized call states.
• The ability to customize soft-key display in the hold
call state was added.
3.2 Configurable soft-key display (the ability to customize
soft-key display in the alerting, connected, idle, and seized
call states) was introduced.

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 Configuring Speed Dial

Last Updated: May 14, 2007

This chapter describes the speed dial support available in Cisco Unified Communications Manager
Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Speed Dial” section
on page 866.

Contents
• Information About Speed Dial, page 847
• How to Configure Speed Dial, page 851
• Configuration Examples for Speed Dial, page 863
• Where to Go Next, page 864
• Additional References, page 865
• Feature Information for Speed Dial, page 866

Information About Speed Dial

To enable speed dial, you should understand the following concepts:
• Speed Dial Summary, page 848
• Speed Dial Buttons and Abbreviated Dialing, page 849
• Bulk-Loading Speed Dial Numbers, page 849
• Monitor-Line Button for Speed Dial, page 850
• DSS (Direct Station Select) Service, page 851

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Information About Speed Dial

Speed Dial Summary

Speed dial allows a phone user to quickly dial a number from a list. The different types of speed dial are
summarized in Table 49.

Table 49 Speed Dial Types

Speed Dial Type Availability of Numbers Description How Configured

Local Speed Dial System-level list of frequently Users invoke entries from the Enabling a Local Speed Dial
Menu called numbers that can be Directories > Local Speed Dial Menu, page 852.
programmed on all phones. menu on IP phones.
A maximum of 32 numbers can
be defined.
Numbers are set up by an
administrator using an XML File
speeddial.xml, which is placed in
the Cisco Unified CME router’s
flash memory.
Personal Speed Speed dial entries are local to a Users invoke entries from the • SCCP: Enabling a Personal
Dial Menu specific IP phone. Directories > Local Services > Speed Dial Menu, page 855
Personal Speed Dials menu on IP
A maximum of 24 numbers per • SIP: Configuring a Personal
phones.
phone can be defined. Speed-Dial Menu, page 861.
Speed Dial Up to 99 speed-dial codes per For IP phones, the first entries • SCCP: Defining Speed-Dial
Buttons and phone. that are set up occupy any unused Buttons and Abbreviated
Abbreviated line buttons and are invoked Dialing, page 856
Dialing when a user presses one of these
• SIP: Defining Speed-Dial
line buttons. Subsequent entries
Buttons, page 860.
are invoked when a phone user
dials the speed-dial code (tag)
and the Abbr soft key.
Analog phone users invoke speed
dial by entering an asterisk and
the speed-dial code (tag) number
of the desired entry.
Bulk-Loading There can be up to ten text files Phone users dial the following SCCP: Enabling Bulk-Loading
Speed Dial containing lists of many sequence: Speed-Dial, page 858.
Numbers speed-dial numbers that are prefix-code list-id index
loaded into flash, slot, or TFTP [extension-digits]
locations to be accessed by phone
users. The ten files can hold
10,000 numbers.

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Information About Speed Dial

Table 49 Speed Dial Types

Speed Dial Type Availability of Numbers Description How Configured

Monitor-Line IP phone buttons that are
Speed dial entries are local to a No additional configuration
Button for Speed specific IP phone. configured as monitor lines can required.
Dial There can be as many numbers as be used to speed-dial the line that
there are monitor lines on a is being monitored.
phone.
Direct Station All phones on which speed-dial Allows phone user to fast transfer SCCP: Enabling DSS Service,
Select (DSS) line or monitor line button is a call by pressing a single page 854.
Service configured. speed-dial line or monitor line
button.

Speed Dial Buttons and Abbreviated Dialing

In a Cisco Unified CME system, each phone can have up to 33 local speed-dial numbers (codes 1 to 33),
up to 99 system-level speed-dial numbers (codes 1 to 99), or a combination of the two. If you program
both a local and a system-level speed-dial number with the same speed-dial code (tag), the local number
takes precedence. Typically you will want to reserve codes 1 to 33 for local, per-phone speed-dial
numbers and use codes 34 to 99 for system-level speed-dial numbers so that there is no conflict.
On an IP phone, speed-dial entries are assigned to unused line buttons. Then, after all line buttons are
used, subsequent entries are added but do not have an assigned line button. The speed-dial entry is not
related to the physical button layout of the phone. Entries are assigned in order of speed-dial tag.
You can create local speed-dial codes with locked numbers that cannot be changed from the phone. You
can also create empty local speed-dial codes on an IP phone without a telephone number. These empty
speed-dial codes can be changed by the phone user to add a telephone number.
Changes to speed-dial entries are saved into the router’s nonvolatile random-access memory (NVRAM)
configuration after a timer-based delay.
For configuration information, see the “SCCP: Defining Speed-Dial Buttons and Abbreviated Dialing”
section on page 856.

Bulk-Loading Speed Dial Numbers

In Cisco Unified CME 4.0 and later versions, up to ten text files containing lists of many speed-dial
numbers can be loaded into flash, slot, or TFTP locations to be accessed by phone users. The ten files
can hold a total of up to 10,000 numbers. Each list holds numbers that are in an appropriate format for
dialing from IP phones and SCCP-enabled analog phones.
Up to ten bulk speed-dial lists can be created. These lists might be corporate directory lists, regional lists,
or local lists, for example. The speed-dial numbers in these lists can be system-level (available to all
ephones) or personal (available to one or more specified ephones). Each list receives a unique speed-dial
list ID number (sd-id) between 0 and 9.
Speed-dial list ID numbers that are not used for global speed-dial lists are available to identify personal,
custom lists that are associated with individual phones.

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Information About Speed Dial

Bulk speed-dial lists contain entries of speed-dial codes and the associated phone numbers to dial. Each
entry in a speed-dial list must appear on a separate line. The fields in each entry are separated by
commas (,). A line that begins with a semicolon (;) is handled as a comment. The format of each entry
is shown in the following line.
index,digits,[name],[hide],[append]
Table 50 explains the fields in a bulk speed-dial list entry.

Table 50 Bulk Speed-Dial List Entry

Field Description
index Zero-filled number that uniquely identifies this index entry.
Maximum length: 4 digits. All index entries must be the same
length.
digits Telephone number to dialed. Represents a fully qualified
E.164 number. Use a comma (,) to represent a one-second
pause.
name (Optional) Alphanumeric string to identify a name, up to 30
characters.
hide (Optional) Enter hide to block the display of the dialed
number.
append (Optional) Enter append to allow additional digits to be
appended to this number when dialed.

The following is a sample bulk speed-dial list:

01,5550140,voicemail,hide,append
90,914085550153,Cisco extension,hide,append
11,9911,emergency,hide,
91,9911,emergency,hide,
08,110,Paging,,append

To place a call to a speed-dial entry in a list, the phone user must first dial a prefix, followed by the list
ID number, then the index for the bulk speed-dial list entry to be called.
For configuration information, see the “SCCP: Enabling Bulk-Loading Speed-Dial” section on
page 858.

Monitor-Line Button for 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 is 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 is completed. For all other call states, the monitor-line lamp is on or lit.

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How to Configure Speed Dial

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

No additional configuration is required to enable a phone user to speed dial the number of a monitored
shared line, when the monitored line is in an idle call state.

DSS (Direct Station Select) Service

In Cisco Unified CME 4.0(2) and later versions, the DSS (Direct Station Select) Service feature allows
the phone user to press a single speed-dial line button to transfer an incoming call when the call is in the
connected state. This feature is supported on all phones on which monitor line buttons for speed dial or
speed-dial line buttons are configured.
When the DSS service is enabled, the system automatically generates a simulated transfer key event
when needed, eliminating the requirement for the phone user to press the Transfer button.
Disabling the service changes the behavior of the speed-dial line button on all IP phones so that a user
pressing a speed-dial button in the middle of a connected call will play out the speed-dial digits into the
call without transferring the call. When DSS service is disabled, the phone user must first press Transfer
and then press the monitor or speed-dial line button to transfer the incoming call.
For configuration information, see the “Enabling a Local Speed Dial Menu” section on page 852.

How to Configure Speed Dial

This section contains the following tasks:
• Enabling a Local Speed Dial Menu, page 852
• SCCP: Enabling DSS Service, page 854
• SCCP: Enabling a Personal Speed Dial Menu, page 855
• SCCP: Defining Speed-Dial Buttons and Abbreviated Dialing, page 856
• SCCP: Enabling Bulk-Loading Speed-Dial, page 858
• SIP: Defining Speed-Dial Buttons, page 860
• SIP: Configuring a Personal Speed-Dial Menu, page 861

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How to Configure Speed Dial

Enabling a Local Speed Dial Menu

To enable a local speed-dial menu for all phones, SCCP and SIP, in Cisco Unified CME, perform the
following steps:

Prerequisites
An XML file called speeddial.xml must be created and copied to the TFTP server application on the
Cisco Unified CME router. The contents of speeddial.xml must be valid as defined in the Cisco specified
directory DTD. See the Cisco IP Phone Services Application Development Notes.

Restrictions
• If a speed dial XML file contains incomplete information, for example the name or telephone
number is missing for an entry, any information in the file that is listed after the incomplete entry is
not displayed when the local speed dial directory option is used on a phone.
• Before Cisco CME 4.1, local speed-dial menu is not supported on SIP phones.
• Before Cisco CME 3.3, analog phones are limited to nine speed-dial numbers.

SUMMARY STEPS

1. enable
2. copy tftp flash
3. configure terminal
4. ip http server
5. ip http path flash:
6. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 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 3 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 4 ip http server Enables the Cisco web-browser user interface on the router.

Example:
Router(config)# ip http server
Step 5 ip http path flash: Sets the base HTTP path to flash memory.

Example:
Router(config)# ip http path flash:
Step 6 exit Returns to privileged EXEC mode.

Example:
Router(config)# exit

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How to Configure Speed Dial

SCCP: Enabling DSS Service

To enable DSS Service for all on all SCCP phones on which monitor line buttons for speed dial or
speed-dial line buttons are configured, perform the following steps.

Prerequisites
Cisco Unified CME 4.0(2) or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. service dss
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 service dss Configures DSS (Direct Station Select) service globally
for all phone users in Cisco Unified CME.
Example:
Router(config-telephony)# service dss
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-telephony)# end

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SCCP: Enabling a Personal Speed Dial Menu

To enable a personal speed-dial menu, perform the following steps.

Restrictions
• A personal speed-dial menu is available only on Cisco Unified IP Phones 7940, 7960, 7960G,
7970G and 7971G-GE.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. fastdial dial-tag number name name-string
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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.

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Command or Action Purpose

Step 4 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 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 a maximum of
24 alphanumeric characters. Personal speed dial is
handled through an XML request, so characters that
have special meaning to HTTP, such as ampersand (&),
percent sign (%), semicolon (;), angle brackets (< >),
and vertical bars (||), are not allowed.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

SCCP: Defining Speed-Dial Buttons and Abbreviated Dialing

To define speed-dial buttons and abbreviated dialing codes, perform the following steps for each
speed-dial definition to be configured.

Restrictions
• On-hook abbreviated dialing using the Abbr soft key is supported only on the following phone
types:
– Cisco Unified IP Phone 7905G
– Cisco Unified IP Phone 7912G
– Cisco Unified IP Phone 7920G
– Cisco Unified IP Phone 7970G
– Cisco Unified IP Phone 7971G-GE
• System-level speed-dial codes cannot be changed by the phone user, at the phone.
• Before Cisco CME 3.3, analog phones were limited to nine speed-dial numbers.
• Before to Cisco CME 3.3, speed-dial entries that were in excess of the number of physical phone
buttons available were ignored by IP phones.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. speed-dial speed-tag digit-string [label label-text]
5. exit

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6. telephony-service
7. directory entry {directory-tag number name name | clear}
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 4 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.
Example: Range is 1 to 33.
Router(config-ephone)# speed-dial 1 +5001 label
“Head Office”
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 configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-ephone)# exit
Step 7 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 8 directory entry {{directory-tag number name Adds a system-level directory and speed-dial definition.
name} | clear}
• directory-tag—Digit string that provides a unique
identifier for this entry. Range is 1 to 99.
Example:
Router(config-telephony)# directory entry 45
• If the same tags 1 through 33 are configured at a
8185550143 name Corp Acctg phone-level by using speed-dial command, and at a
system-level by using this command, the local
definition takes precedence. To prevent this conflict,
we recommend that you use only codes 34 to 99 for
system-level speed-dial numbers.

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Command or Action Purpose

Step 9 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

SCCP: Enabling Bulk-Loading Speed-Dial

To enable bulk-loading speed-dial numbers, perform the following steps:

Prerequisites
• Cisco Unified CME 4.0 or a letter version.
• The bulk speed-dial text files containing the lists must be available in a location that is available to
the Cisco Unified CME router: flash, slot, or TFTP location.

Restrictions
• Bulk speed dial is not supported on FXO trunk lines.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. bulk-speed-dial list list-id location
5. bulk-speed-dial prefix prefix-code
6. exit
7. ephone phone-tag
8. bulk-speed-dial list list-id location
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 bulk-speed-dial list list-id location Identifies the location of a bulk speed-dial list.
• list-id—Digit that identifies the list to be used. Range is
Example: 0 to 9.
Router(config-telephony)# bulk-speed-dial list 6
flash:sd_dept_0_1_8.txt
• location—Location of the bulk speed-dial text file in
URL format. Valid storage locations are TFTP, Slot 0/1,
and flash memory.
Step 5 bulk-speed-dial prefix prefix-code Sets the prefix code that phone users dial to access speed-dial
numbers from a bulk speed-dial list.
Example: • prefix-code—One- or two-character access code for
Router(config-telephony)# bulk-speed-dial prefix speed dial. Valid characters are digits from 0 to 9,
#7 asterisk (*), and pound sign (#). Default is #.
Step 6 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-telephony)# exit
Step 7 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies this
Example: ephone during configuration tasks.
Router(config)# ephone 25
Step 8 bulk-speed-dial list list-id location Identifies the location of a bulk speed-dial list.
• list-id—Digit that identifies the list to be used. Range is
Example: 0 to 9.
Router(config-ephone)# bulk-speed-dial list 7
flash:lmi_sd_list_08_24_95.txt
• location—Location of the bulk speed-dial text file in
URL format. Valid storage locations are TFTP, Slot 0/1,
and flash memory.
Step 9 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

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SCCP: Verifying Bulk Speed-Dial Parameters

show telephony-service bulk-speed-dial

Use this command to display information on speed-dial lists.
Router# show telephony-service bulk-speed-dial summary

List-id Entries Size Reference url

0 40 3840 Global tftp://192.168.254.254/phonedirs/uut.csv
1 20 1920 Global phoneBook.csv
8 15 1440 Global tftp://192.168.254.254/phonedirs/big.txt
9 20 1920 Global tftp://192.168.254.254/phonedirs/phoneBook.csv
6 24879 2388384 ephone-2 tftp://192.168.254.254/phonedirs/big.txt1
7 20 1920 ephone-2 phoneBook.csv
6 24879 2388384 ephone-3 big.txt1
7 20 1920 ephone-3 phoneBook.csv

4 Global List(s) 4 Local List(s)

SIP: Defining Speed-Dial Buttons

To define speed-dial buttons for Cisco SIP IP phones, perform the following steps.

Prerequisites
Cisco CME 3.4 or a later version.

Restrictions
• Certain SIP IP phones, such as the Cisco Unified IP Phone 7960 and 7940, cannot be configured to
enable speed dialing. Phone users with these phones must manually configure speed-dial numbers
by using the user interface at their Cisco Unified IP phone.
• On Cisco Unified IP phones, speed-dial definitions are assigned to available buttons that have not
been assigned to actual extensions. Speed-dial definitions are assigned in the order of their identifier
numbers.
• Phones with Cisco ATA devices are limited to a maximum of nine speed-dial numbers. Speed-dial
numbers cannot be programmed by using the user interface at the phone.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register pool pool-tag
4. speed-dial speed-tag digit-string [label label-text]
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
parameters for specified SIP phone.
Example:
Router(config)# voice register pool 23
Step 4 speed-dial speed-tag digit-string [label Creates a speed-dial definition in Cisco Unified CME for a
label-text] SIP phone or analog phone that uses an analog adapter
(ATA).
Example: • speed-tag—Unique sequence number that identifies the
router(config-register-pool)# speed-dial 2 speed-dial definition during configuration. Range is 1
+5001 label “Head Office”
to 5.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-pool)# end

Examples
The following example shows how to set speed-dial button 2 to dial the head office at extension 5001
and locks the setting so that the phone user cannot change the setting at the phone:
Router(config)# voice register pool 23
Router(config-register-pool)# speed-dial 2 +5001 label “Head Office”

SIP: Configuring a Personal Speed-Dial Menu

To define up to 24 personal speed-dial numbers for a SIP phone, perform the following steps.

Prerequisites
• Cisco Unified CME 4.1 or a later version.

Restrictions
• Cisco Unified IP Phone 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE—Personal
speed-dial numbers can only be created in Cisco Unified CME, using this procedure.

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• Cisco Unified IP Phone 7905, 7912, 7940, and 7960—Speed dial numbers can only be created by
the user directly on the phone and not in Cisco Unified CME.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register pool pool-tag
4. fastdial dial-tag number [name name-string]
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone.
Example: • pool-tag—Unique sequence number of the SIP phone
Router(config-register-pool)# voice register to be configured. Range is version and
pool 1 platform-dependent; type ? to display range. You can
modify the upper limit for this argument with the
max-pool command.
Step 4 fastdial dial-tag number [name name-string] Creates a personal speed-dial number on this SIP phone.
• dial-tag—Unique number to identify this entry during
Example: configuration. Range: 1 to 24.
Router(config-register-pool)# fastdial 1 5552
name Sales
• number—Telephone number or extension to be dialed.
• name name-string—(Optional) Label to appear in the
Personal Speed Dial menu, containing a string of a
maximum of 24 alphanumeric characters. Personal
speed dial is handled through an XML request, so
characters that have special meaning to HTTP, such as
ampersand (&), percent sign (%), semicolon (;), angle
brackets (< >), and vertical bars (||), are not allowed.
• Repeat this command for each personal speed-dial
number that you want to create on this phone.
Step 5 end Exits to privileged EXEC mode.

Example:
Router(config-register-pool)# end

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Configuration Examples for Speed Dial

Configuration Examples for Speed Dial

This section contains the following examples:
• Enabling a Local Speed Dial Menu: Example, page 863
• Personal Speed Dial Menu: Example, page 863
• Speed-Dial Buttons and Abbreviated Dialing: Example, page 864
• Bulk-Loading Speed Dial: Example, page 864

Enabling a Local Speed Dial Menu: Example

The following commands enable the Cisco web browser and set 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:

The following XML file—speeddial.xml, defines three speed-dial numbers that will appear to the user
after they press the Directories button on an IP phone.
<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>

Personal Speed Dial Menu: 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 NY Sales
fastdial 3 12135550112 name LA Sales

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Where to Go Next

Speed-Dial Buttons and Abbreviated Dialing: 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 are assigned. For instance, if two extensions are assigned on the
Cisco Unified IP Phones 7960 and 7960G, these speed-dial definitions appear on the third and fourth
buttons.
This example also defines two system-level 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

Bulk-Loading Speed Dial: Example

The following example changes the default bulk speed-dial prefix to #7 and enables global bulk
speed-dial list number 6 for all phones. It also enables a personal bulk speed-dial list for ephone 25.
telephony-service
bulk-speed-dial list 6 flash:sd_dept_01_1_87.txt
bulk-speed-dial prefix #7

ephone-dn 3
number 2555

ephone-dn 4
number 2557

ephone 25
button 1:3 2:4
bulk-speed-dial list 7 flash:lmi_sd_list_08_24_95.txt

Where to Go Next
If you are finished creating or modifying speed-dial configurations for individual phones, you must
reboot phones to download the modified configuration. See “Resetting and Restarting Phones” on
page 257.

DSS Call Transfer

Monitor-line button speed dial, also known as direct station select (DSS) call transfer, allows you to use
a monitored line button to speed-dial a call to that extension. If you want to allow consultation during
DSS transfers, see “Configuring Call Transfer and Forwarding” on page 499.

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Speed Dial

Feature Information for Speed Dial

Table 51 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CallManager Express and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 51 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 51 Feature Information for Speed Dial

Cisco Unified CME

Feature Name Version Feature Information
Speed Dial 4.1 Added support for local and personal speed-dial menus for
SIP phones in Cisco Unified CME.
4.0(2) Added support for DSS Service which allows phone user to
fast transfer a call by pressing a single speed-dial line or
monitor line button.
4.0 Added support for bulk speed-dial list for SCCP phones in
Cisco Unified CME.
3.4 Added support for speed dial buttons on SIP phones in
Cisco Unified CME.
3.0 • Added support for personal speed-dial from SCCP
phones in Cisco Unified CME.
• Number of speed-dial definitions that can be created
was increased from 4 to 33.
• The ability to program speed-dial numbers at the
phone was introduced.
• The ability to lock speed-dial numbers was introduced.
1.0 Speed dial using the speed-dial command was introduced.

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Endpoints

Last Updated: March 26, 2007

This chapter describes the video support for SCCP-based endpoints in Cisco Unified Communications
Manager Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the Feature Information for Video Support for
SCCP-Based Endpoints, page 880.

Contents
• Prerequisites for Video Support for SCCP-Based Endpoints, page 867
• Information About Video Support for SCCP-Based Endpoints, page 869
• How to Configure Video for SCCP-Based Endpoints, page 872
• Additional References, page 878
• Feature Information for Video Support for SCCP-Based Endpoints, page 880

Prerequisites for Video Support for SCCP-Based Endpoints

• Establish a working H.323 or SIP network for voice calls.
• Ensure that you have a Cisco IOS image that supports this feature. Access Cisco Feature Navigator
at http://www.cisco.com/go/cfn.
• Ensure that the Cisco Unified CME version is 4.0 or later.
• Ensure that the Cisco Unified Communications Manager version is 4.0 or later.
• Ensure that the Cisco Unified IP phones are registered with the Cisco Unified CME router. Use the
show ephone registered command to verify ephone registration.
• Ensure that the connection between the Cisco Unified Video Advantage application and the
Cisco Unified IP phone is up.

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Restrictions for Video Support for SCCP-Based Endpoints

From a PC with Cisco Unified Video Advantage version 1.02 or later installed, ensure that the line
between the Cisco Unified Video Advantage and the Cisco Unified IP phone is green. For more
information, see the Cisco Unified Video Advantage User Guide.
• Ensure that the correct video firmware is installed on the Cisco Unified IP phone. Use the show
ephone phone-load command to view current ephone firmware. The following lists the minimum
firmware version for video-enabled Cisco Unified IP phones:
– Cisco Unified IP Phone 7940G 6.0(4)
– Cisco Unified IP Phone 7960G 6.0(4)
– Cisco Unified IP Phone 7970G 7.0(3)
– Cisco Unified IP Phone 7941G 7.0(3)
– Cisco Unified IP Phone 7961G 7.0(3)

Note Other video-enabled endpoints, if registered with Cisco Unified Communications Manager,
can place a video call to one of the Cisco Unified IP phones listed above if it is registered
with Cisco Unified CME.

Restrictions for Video Support for SCCP-Based Endpoints

• This feature supports only the following video codecs:
– H.261
– H.263
• This feature supports only the following video formats:
– 4CIF—Resolution 704x576
– 16CIF—Resolution 1408x1152
– Common Intermediate Format (CIF)—Resolution 352x288
– One-Quarter Common Intermediate Format (QCIF)—Resolution 176x144
– Sub QIF (SQCIF)—Resolution 128x96
• The call start fast feature is not supported with an H.323 video connection. You must configure
call start slow for H.323 video.
• Video capabilities are configured per ephone, not per line.
• All call feature controls (for example, mute and hold) apply to both audio and video calls, if
applicable.
• This feature does not support the following:
– Dynamic addition of video capability—The video capability must be present before the call
setup starts to allow the video connection.
– T-120 data connection between two SCCP endpoints
– Video security
– Far-end camera control (FECC) for SCCP endpoints
– Video codec renegotiation—The negotiated video codec must match or the call falls back to
audio-only. The negotiated codec for the existing call can be used for a new call.

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– Video codec transcoding

– SIP endpoints— When a video-capable SCCP endpoint connects to a SIP endpoint, the call falls
back to audio-only.
– Video conferencing—The call falls back to audio-only.
– Features, such as conferencing, that mix the audio streams in Cisco Unified CME—In those
cases, the call falls back to audio-only.
– Video supplementary services between Cisco Unified CME and Cisco Unified
Communications Manager.
• If the Cisco Unified Communications Manager is configured for Media Termination Point (MTP)
transcoding, a video call between Cisco Unified CME and Cisco Unified Communications Manager
is not supported.
• Video telephony is not supported with CME MTP and codec g729/dspfarm-assist configuration
under ephone.
• If an SCCP endpoint calls an SCCP endpoint on the local Cisco Unified CME and one of the
endpoints transferred across an H323 network, a video-consult transfer between the
Cisco Unified CME systems is not supported.
• When a video-capable endpoint connects to an audio-only endpoint, the call falls back to audio-only.
During audio-only calls, video messages are skipped.
• For Cisco Unified CME, the video capabilities in the vendor configuration firmware is a global
configuration. This means that, although video can be enabled per ephone, the video icon shows on
all Cisco Unified IP phones supported by Cisco Unified CME.
• Because of the extra CPU consumption on RTP-stream mixing, the number of video calls supported
on Cisco Unified CME crossing an H.323 network is less than the maximum number of ephones
supported.
• Cisco Unified CME cannot differentiate audio-only streams and audio-in-video streams. You must
configure the DSCP values of audio and video streams in the H.323 dial-peers.
• If RSVP is enabled on the Cisco Unified CME, a video call is not supported.
• A separate VoIP dial peer, configured for fast-connect procedures, is required to complete a video
call from a remote H.323 network to a Cisco Unity Express system.

Information About Video Support for SCCP-Based Endpoints

To configure video support for SCCP endpoints, you should understand the following concepts:
• Video Support Overview, page 870
• Matching Endpoint Capabilities, page 870
• Retrieving Video Codec Information, page 870
• Call Fallback to Audio-Only, page 870
• Call Setup for Video Endpoints, page 871
• Flow of the RTP Video Stream, page 871

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Information About Video Support for SCCP-Based Endpoints

Video Support Overview

Video support allows you to pass a video stream, with a voice call, between two video-capable SCCP
endpoints and between SCCP and H.323 endpoints. Through the Cisco Unified CME router, the
video-capable endpoints can communicate with each other locally, to a remote H.323 endpoint through
a gateway, or through an H.323 network.
Video capabilities are not enabled by default, and enabling video capabilities on Cisco Unified CME
does not automatically enable video on all ephones. You must first enable video globally for all
video-capable ephones associated with a Cisco Unified CME router and then enable video on each
ephone individually.
Video parameters, like maximum bit rate, are set in video configuration mode.

Note After video is enabled globally, all video-capable ephones display the video icon.

Matching Endpoint Capabilities

During phone registration, information about endpoint capabilities is stored in the Cisco Unified CME.
These capabilities are used to match with other endpoints during call setup. Endpoints can update at any
time; however, the router recognizes endpoint-capability changes only during call setup. If a video
feature is added to a phone, the information about it is updated in the router’s internal data structure, but
that information does not become effective until the next call. If a video feature is removed, the router
continues to see the video capability until the call is terminated but no video stream is exchanged
between the two endpoints.

Note The endpoint-capability match is executed each time a new call is set up or an existing call is resumed.

Retrieving Video Codec Information

Voice gateways use dial-peer configurations to retrieve codec information for audio codecs. Video codec
selection is done by the endpoints and is not controlled by the H.323 service-provider interface (SPI)
through dial-peer or other configuration. The video-codec information is retrieved from the SCCP
endpoint using a capabilities request during call setup.

Call Fallback to Audio-Only

When a video-capable endpoint connects to an audio-only endpoint, the call falls back to an audio-only
connection. Also, for certain features, such as conferencing, where video support is not available, the
call falls back to audio-only.
Cisco Unified CME routers use a call-type flag to indicate whether the call is video-capable or
audio-only. The call-type flag is set to video when the video capability is matched or set to audio-only
when connecting to an audio-only TDM or an audio-only SIP endpoint.

Note During an audio-only connection, all video-related media messages are skipped.

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Call Setup for Video Endpoints

The process for handling SCCP video endpoints is the same as that for handling SCCP audio endpoints.
The video call must be part of the audio call. If the audio call setup fails, the video call fails.
During the call setup for video, media setup handling determines if a video-media-path is required. If
so, the corresponding video-media-path setup actions are taken.
• For an SCCP endpoint, video-media-path setup includes sending messages to the endpoints to open
a multimedia path and start the multimedia transmission.
• For an H.323 endpoint, video-media-path setup includes an exchange between the endpoints to open
a logical channel for the video stream.
A call-type flag is set during call setup on the basis of the endpoint-capability match. After call setup,
the call-type flag is used to determine whether an additional video media path is required. Call signaling
is managed by the Cisco Unified CME router, and the media stream is directly connected between the
two video-enabled SCCP endpoints on the same router. Video-related commands and flow-control
messages are forwarded to the other endpoint. Routers do not interpret these messages.

Call Setup Between Two Local SCCP Endpoints

For interoperation between two local SCCP endpoints (that exist on the same router), video call setup
uses all existing audio-call-setup handling, except during media setup. During media setup, a message
is sent to establish the video-media-path. If the endpoint responds, the video-media-path is established
and a start-multimedia-transmission function is called.

Call Setup Between SCCP and H.323 Endpoints

Call setup between SCCP and H.323 endpoints is the same as it is between SCCP endpoints except that
if video capability is selected, the event is posted to the H.323 call leg to send out a video open logical
channel (OLC) and the gateway generates an OLC for the video channel. Because the router needs to
both terminate and originate the media stream, video must be enabled on the router before call setup
begins.

Call Setup Between Two SCCP Endpoints Across an H.323 Network

If call setup between SCCP endpoints occurs across an H.323 network, the setup is a combination of the
processes listed in the previous two sections. The router controls the video media setup between the two
endpoints, and the event is posted to the H.323 call leg so that the gateway can generate an OLC.

Flow of the RTP Video Stream

For video streams between two local SCCP endpoints, the Real-Time Transport Protocol (RTP) stream
is in flow-around mode. For video streams between SCCP and H.323 endpoints or two SCCP endpoints
on different Cisco Unified CME routers, the RTP stream is in flow-through mode.
• Media flow-around mode enables RTP packets to stream directly between the endpoints of a VoIP
call without the involvement of the gateway. By default, the gateway receives the incoming media,
terminates the call, and then reoriginates it on the outbound call leg. In flow-around mode, only
signaling data is passed to the gateway, improving scalability and performance.

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• With flow-through mode, the video media path is the same as for an audio call. Media packets flow
through the gateway, thus hiding the networks from each other.
Use the show voip rtp connection command to display information about RTP named-event packets,
such as caller-ID number, IP address, and port for both the local and remote endpoints, as show in the
following sample output.
Router# show voip rtp connections

VoIP RTP active connections :

No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 102 103 18714 18158 10.1.1.1 192.168.1.1
2 105 104 17252 19088 10.1.1.1 192.168.1.1
Found 2 active RTP connections
============================

How to Configure Video for SCCP-Based Endpoints

This section contains the following tasks:
• Enabling Slow Connect Procedures, page 872
• Enabling Video Capabilities Globally, page 873
• Enabling Video Capabilities on a Specific Phone, page 874
• Setting Video Parameters, page 876

Enabling Slow Connect Procedures

Video streams require slow-connect procedures for Cisco Unified CME. H.323 endpoints also require
slow-connect because the endpoint capability match occurs after the connect message. To enable slow
connect procedures in Cisco Unified CME for H.323 endpoints, perform the following steps.

Note For more information on slow-connect procedures, see Configuring Quality of Service for Voice.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice service voip
4. h323
5. call start slow
6. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip Enters voice-service configuration mode.

Example:
Router(config)# voice service voip
Step 4 h323 Enters H.323 voice-service configuration mode.

Example:
Router(config-voi-serv)# h323
Step 5 call start slow Forces an H.323 gateway to use slow-connect procedures
for all VoIP calls.
Example:
Router(config-serv-h323)# call start slow
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-serv-h323)# end

Enabling Video Capabilities Globally

To enable video for all video-capable phones in a Cisco Unified CME system, perform the following
steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. service phone videoCapability [0 | 1]
5. end

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DETAILED STEP

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 service phone videoCapability [0 | 1] Enables or disables video capabilities for
Cisco Unified CME.
Example: • 0—(Optional) Disables capabilities.
Router(config-telephony)# service phone
videoCapability 1
• 1—(Optional) Enables capabilities.
Note This command is case sensitive. If it is not entered
exactly as shown, Cisco Unified CME accepts the
command, but video capabilities are not enabled.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.

Enabling Video Capabilities on a Specific Phone

To enable video for an individual ephone associated with a Cisco Unified CME router, perform the
following steps.

Prerequisites
Use the show ephone registered command to display which phones are registered to
Cisco Unified CME and have video capability. The following example shows that ephone 1 has video
capabilities and ephone 2 is an audio-only phone.
Router# show ephone registered

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ephone-1 Mac:0011.5C40.75E8 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 6 + Video

and Server in ver 5
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:7
IP:10.1.1.6 51833 7970 keepalive 35 max_line 8
button 1: dn 1 number 8003 CH1 IDLE CH2 IDLE

ephone-2 Mac:0006.D74B.113D TCP socket:[2] activeLine:0 REGISTERED in SCCP ver 6 and

Server in ver 5
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:7
IP:10.1.1.4 51123 Telecaster 7960 keepalive 36 max_line 6
button 1: dn 2 number 8004 CH1 IDLE CH2 IDLE
button 2: dn 4 number 8008 CH1 IDLE CH2 IDLE
===========================================

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone phone-tag
4. video
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies an
Example: ephone during configuration tasks. The maximum
Router(config)# ephone 6 number is platform-dependent.
Step 4 video Enables video capabilities on the specified ephone. Repeat
as necessary for all ephones that require video enabled.
Example:
Router(config-ephone)# video
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

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What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.

Setting Video Parameters

To set the maximum bit rate for all video-capable phones associated with a Cisco Unified CME router,
perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. video
5. maximum bit-rate value
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 video Enters video configuration mode.

Example:
Router(config-telephony)# video

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Command or Action Purpose

Step 5 maximum bit-rate value Sets the maximum IP phone video bandwidth, in kbps. The
range is 0 to 10000000. The default is 10000000.
Example:
Router(conf-tele-video)# maximum bit-rate 256
Step 6 end Returns to privileged EXEC mode.

Example:
Router(conf-tele-video)# end

Verifying Video Support for SCCP-Based Endpoints

Step 1 Use the show running-config command to verify the video settings in the configuration.
• See the telephony-service portion of the output for commands that configure video support on the
Cisco Unified CME.
• See the ephone portion of the output for commands that configure video support for a specific
ephone.

The following example shows the telephony-service portion of the output:

telephony-service
video
maximum bit-rate 256
load 7960-7940 P00306000404
max-ephones 24
max-dn 24
ip source-address 10.0.180.130 port 2000
service phone videoCapability 1
timeouts interdigit 4
timeouts ringing 100
create cnf-files version-stamp Jan 01 2002 00:00:00
keepalive 60
max-conferences 4 gain -6
call-park system redirect
call-forward pattern .T
web admin system name cisco password cisco
web customize load xml.jeff
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T

The following example shows the ephone portion of the output:

ephone 6
video
maximum bit-rate 256
mac-address 000F.F7DE.CAA5
type 7960
button 1:6

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Additional References

Troubleshooting Video Support for SCCP-Based Endpoints

Step 1 For SCCP endpoint troubleshooting, use the following debug commands:
• debug cch323 video—Enables video debugging trace on the H.323 service-provider interface (SPI).
• debug ephone detail—Debugs all Cisco Unified IP phones that are registered to the router, and
displays error and state levels.
• debug h225 asn1—Displays Abstract Syntax Notation One (ASN.1) contents of H.225 messages
that have been sent or received.
• debug h245 asn1—Displays ASN.1 contents of H.245 messages that have been sent or received.
• debug voip ccapi inout—Displays the execution path through the call-control application
programming interface (CCAPI).
Step 2 For ephone troubleshooting, use the following debug commands:
• debug ephone message—Enables message tracing between Cisco Unified IP phones.
• debug ephone register—Sets registration debugging for Cisco Unified IP phones.
• debug ephone video—Sets ephone video traces, which provide information about different video
states for the call, including video capabilities selection, start, and stop.
Step 3 For basic video-to-video call checking, use the following show commands:
• show call active video—Displays call information for SCCP video calls in progress.
• show ephone offhook—Displays information and packet counts for ephones that are off hook.
• show ephone registered—Displays the status of registered ephones.
• show voip rtp connections—Displays information about RTP named-event packets, such as caller
ID number, IP address, and port for both the local and remote endpoints.

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

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Additional References

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Video Support for SCCP-Based Endpoints

Feature Information for Video Support for SCCP-Based

Endpoints
Table 52 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note The following table lists the Cisco Unified CME version that introduced support for a given feature.
Unless noted otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 52 Feature Information for Video Support for SCCP-Based Endpoints

Cisco Unified CME

Feature Name Version Feature Information
Video Support for SCCP-Based Endpoints 4.0 Video support for SCCP-based endpoints was introduced.

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Last Updated: March 26, 2007

This chapter describes templates support available in Cisco Unified Communications Manager Express
(Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Creating
Templates” section on page 891.

Contents
• Information About Templates, page 881
• How to Configure Templates, page 882
• Configuration Examples for Creating Templates, page 888
• Where to Go Next, page 889
• Additional References, page 889
• Feature Information for Creating Templates, page 891

Information About Templates

To enable templates you should understand the following concepts:
• Phone Templates, page 882
• Ephone-dn Templates, page 882

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Phone Templates
An ephone or voice-register template is a set of features that can be applied to one or more individual
phones using a single command.
Ephone templates were introduced in Cisco CME 3.2 to manipulate soft-key display and order on IP
phones. In Cisco Unified CME 4.0, ephone templates were significantly enhanced to include a number
of additional phone features. Templates allow you to uniformly and easily implement the features you
select for a set of phones. A maximum of 20 ephone templates can be created in a Cisco Unified CME
system, although an ephone can have only one template applied to it at a time.
If you use an ephone template to apply a command to a phone and you also use the same command in
ephone configuration mode for the same phone, the value set in ephone configuration mode has priority.
Voice-register templates were introduced in Cisco CME 3.4 to enable sets of features to be applied to
individual SIP IP phones that are connected directly in Cisco Unified CME. Typically, features to be
enabled by using a voice-register template are not configurable in other configuration modes. A
maximum 10 voice-register templates can be defined in Cisco Unified CME, although a phone can have
only one template applied to it at a time.
Type ? in ephone-template or voice-register-template configuration mode to display a list of features that
can be implemented by using templates.

Ephone-dn Templates
Ephone-dn templates allow you to apply a standard set of features to ephone-dns. A maximum of 15
ephone-dn templates can be created in a Cisco Unified CME system, although an ephone-dn can have
only one template applied to it at a time.
If you use an ephone-dn template to apply a command to an ephone-dn and you also use the same
command in ephone-dn configuration mode for the same ephone-dn, the value that you set in ephone-dn
configuration mode has priority.

Note The features that can be implemented using ephone-dn templates are available in the CLI help by
entering a question mark:
Router(config)# ephone-dn-template 1
Router(config-ephone-dn-template)# ?

How to Configure Templates

This section contains the following tasks:
• SCCP: Enabling Ephone Templates, page 883
• SCCP: Enabling Ephone-dn Templates, page 884
• SCCP: Verifying Templates, page 885
• SIP: Creating and Applying Templates to SIP Phones, page 886

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SCCP: Enabling Ephone Templates

To create an ephone template and apply it to a phone, perform the following steps:

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-template template-tag
4. command
5. exit
6. ephone phone-tag
7. ephone-template template-tag
8. restart
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-template template-tag Enters ephone-template configuration mode to create an
ephone template.
Example: • template-tag—Unique identifier for the ephone
Router(config)# ephone-template 15 template that is being created. Range is 1 to 20.
Step 4 command Applies the specified command to the ephone template that
is being created. See the CLI help for a list of commands
that can be used in this step. Repeat this step for each
Example:
Router(config-ephone-template)# features
command that you want to add to the ephone template.
blocked Park Trnsfer
Step 5 exit Exits ephone-template configuration mode.

Example:
Router(config-ephone-template)# exit
Step 6 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: this ephone during configuration tasks.
Router(config)# ephone 36

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Command or Action Purpose

Step 7 ephone-template template-tag Applies an ephone template to the ephone that is being
configured.
Example:
Router(config-ephone)# ephone-template 15
Step 8 restart Performs a fast reboot of this ephone. Does not contact the
DHCP or TFTP server for updated information.
Example: Note Restart all ephones using the restart all command
Router(config-ephone)# restart in telephony-service configuration mode.
Step 9 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

SCCP: Enabling Ephone-dn Templates

To create an ephone-dn template and apply it to an ephone-dn, perform the following steps:

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn-template template-tag
4. command
5. exit
6. ephone-dn dn-tag
7. ephone-dn-template template-tag
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn-template template-tag Enters ephone-dn-template configuration mode to create an
ephone-dn template.
Example: • template-tag—Unique identifier for the ephone-dn
Router(config)# ephone-dn-template 3 template that is being created. Range is 1 to 20.

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Command or Action Purpose

Step 4 command Applies the specified command to the ephone-dn template
that is being created. See the CLI help for a list of
commands that can be used in this step. Repeat this step to
Example:
Router(config-ephone-dn-template)#
add more commands to the template.
call-forwarding busy 4000
Step 5 exit Exits ephone-dn-template configuration mode.

Example:
Router(config-ephone-dn-template)# exit
Step 6 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—Unique sequence number that identifies this
Example: ephone-dn during configuration tasks.
Router(config)# ephone-dn 23
Step 7 ephone-dn-template template-tag Applies an ephone-dn template to the ephone-dn that is
being configured.
Example:
Router(config-ephone-dn)# ephone-dn-template 3
Step 8 end Returns to privileged EXEC mode.

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

SCCP: Verifying Templates

To view the configuration of a template, and verify to which phone or directory number a template is
applied, perform the following steps.

SUMMARY STEPS

1. show telephony-service ephone

2. show telephony-service ephone-template
3. show telephony-service ephone-dn
4. show telephony-service ephone-dn-template

DETAILED STEPS

Step 1 show telephony-service ephone

Use is command to display information about SCCP phones in Cisco Unified CME, including which
template-tags are enabled in the configuration for a phone.
Router# show telephony-service ephone 1
ephone-dn-template 1
description Call Center Line 1
call-forward busy 500
call-forward noan 500 timeout 10
pickup-group 33!
!

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Step 2 show telephony-service ephone-template

Use is command to display information about an ephone template in Cisco Unified CME, including a
list of features enabled in the configuration.
Step 3 show telephony-service ephone-dn
Use is command to display information about directory numbers, including which template-tags are
enabled in the configuration for a directory number.
Router# show telephony-service ephone-dn 4
!
ephone-dn 4 dual-line
number 136
description Desk4
ephone-dn template 1
ephone-hunt login

Step 4 show telephony-service ephone-dn-template

Use is command to display information about an ephone-dn template in Cisco Unified CME, including
a list of features enabled in the configuration.

SIP: Creating and Applying Templates to SIP Phones

To create templates of common features and softkeys that can be applied to individual Cisco SIP IP
phones, follow the steps in this section.

Prerequisites
• Cisco CME 3.4 or a later version.
• The mode cme command must be enabled in Cisco Unified CME.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register template template-tag
4. command
5. exit
6. voice register pool pool-tag
7. template template-tag
8. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register template template-tag Enters voice register template configuration mode to define
a template of common parameters for SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register template 1 • Range is 1 to 5.
Step 4 command Applies the specified command to this template and enables
the corresponding feature on any supported SIP phone that
uses a template in which this command is configure.
Example:
Router(config-register-template)# anonymous • Type ? to display list of commands that can be used in
block a voice register template.
• Repeat this step for each feature to be added to this
voice register template.
Step 5 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-register-template)# exit
Step 6 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for SIP phones.
Example: • pool-tag—Unique sequence number of the Cisco SIP
Router(config)# voice register pool 3 phone to be configured. Range is 1 to 100 or the upper
limit as defined by max-pool command.
Step 7 template template-tag Applies a template created with the voice register template
command.
Example: • template-tag—Unique sequence number of the
Router(config-register-pool)# voice register template to be applied to the SIP phone specified by the
pool 1 voice register pool command. Range is 1 to 5.
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-pool)# end

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Examples
The following example shows templates 1 and 2 and how to do the following:
• Apply template 1 to SIP phones 1 to 3
• Apply template 2 to SIP phone 4
• Remove a previously created template 5 from SIP phone 5.
Router(config)# voice register template 1
Router(config-register-temp)# anonymous block
Router(config-register-temp)# caller-id block
Router(config-register-temp)# voicemail 5001 timeout 15

Router(config)# voice register template 2

Router(config-register-temp)# anonymous block
Router(config-register-temp)# caller-id block
Router(config-register-temp)# no conference
Router(config-register-temp)# no transfer-attended
Router(config-register-temp)# voicemail 5005 timeout 15

Router(config)# voice register pool 1

Router(config-register-pool)# template 1

Router(config)# voice register pool 2

Router(config-register-pool)# template 1

Router(config)# voice register pool 3

Router(config-register-pool)# template 1

Router(config)# voice register pool 4

Router(config-register-pool)# template 2

Router(config)# voice register pool 5

Router(config-register-pool)# no template 5

Configuration Examples for Creating Templates

This section contains the following examples:
• Using Ephone Template to Block The Use of Park and Transfer Soft Keys, page 888
• Using Ephone-dn Template to Set Call Forwarding, page 889

Using Ephone Template to Block The Use of Park and Transfer Soft Keys
The following example creates an ephone template to block the use of Park and Transfer soft keys. It is
applied to ephone 36 and extension 2333.
ephone-template 15
features blocked Park Trnsfer

ephone-dn 2
number 2333

ephone 36
button 1:2
ephone-template 15

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Using Ephone-dn Template to Set Call Forwarding

The following example creates ephone-dn template 3, which sets call forwarding on busy and no answer
to forward calls to extension 4000 and sets the pickup group to 4. Ephone-dn template 3 is then applied
to ephone-dn 23 and ephone-dn 33, which appear on ephones 13 and 14, respectively.
ephone-dn-template 3
call-forwarding busy 4000
call-forwarding noan 4000 timeout 30
pickup group 4

ephone-dn 23
number 2323
ephone-dn-template 3

ephone-dn 33
number 3333
ephone-dn-template 3

ephone 13
button 1:23

ephone 14
button 1:33

Where to Go Next
Soft-Key Display
The display of soft keys during different call states is managed using ephone templates. For more
information, see “Customizing Soft Keys” on page 829.

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

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Additional References

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Creating Templates

Feature Information for Creating Templates

Table 53 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 53 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 53 Feature Information for Templates

Cisco Unified CME

Feature Name Version Feature Information
Ephone Templates 4.0 • The number of ephone templates that can be created
was increased from 5 to 20.
• More commands can be included in ephone templates.
3.2 Ephone templates were introduced to manage soft keys.
The only commands that can be used in ephone templates
are the softkeys commands.
Ephone-dn Templates 4.0 Ephone-dn templates were introduced.
Phone Templates for SIP Phones 4.1 The maximum number of templates that can be configured
was increased from 5 to 10.
3.4 Voice-register templates were introduced for SIP IP phones
directly connected to a Cisco Unified CME router.

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 Modifying Cisco Unified IP Phone Options

Last Updated: May 22, 2007

This chapter describes the screen and button features available for Cisco Unified IP phones connected
to Cisco Unified Communications Manager Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Cisco Unified IP
Phone Options” section on page 917.

Contents
• Information About Cisco Unified IP Phone Options, page 893
• How to Configure Cisco Unified IP Phone Options, page 896
• Configuration Examples for Cisco Unified IP Phone Options, page 913
• Additional References, page 915
• Feature Information for Cisco Unified IP Phone Options, page 917

Information About Cisco Unified IP Phone Options

To enable IP phone options, you should understand the following concepts:
• Customized Background Images for Cisco Unified IP Phone 7970, page 894
• Fixed Line/Feature Buttons for Cisco Unified IP Phone 7931G, page 894
• Header Bar Display, page 894
• Phone Labels, page 895
• Programmable Vendor Parameters for Phones, page 895
• System Message Display, page 895
• URL Provisioning for Feature Buttons, page 896

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Customized Background Images for Cisco Unified IP Phone 7970

The Cisco Unified IP Phone 7970 and 7971 support customized background images on the phone screen.
To enable your Cisco Unified IP Phone 7970 or 7971 to display a customized background image, follow
the procedure in the technical note at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_tech_note09186a008062495a.sht
ml
Sample background images are available in the 7970-backgrounds.tar file at
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp

Fixed Line/Feature Buttons for Cisco Unified IP Phone 7931G

In Cisco Unified CME 4.0(2) and later versions, you can select from two fixed button-layout formats to
assign functionality to certain line buttons on a Cisco Unified IP Phone 7931G to support key system
phone behavior. If you do not select a button set, no fixed set of feature/line buttons are defined.
The line button layout for the Cisco Unified IP Phone 7931G is a bottom-up array. Button 1 is at the
bottom right of the array and button 24 is at the top left of the array.
Button set 1 includes two predefined feature buttons: button 24 is Menu and button 23 is Headset.
Button set 2 includes four predefined feature buttons: button 24 is Menu; button 23 is Headset; button
22 is Directories; and button 21 is Messages.
For configuration, see the “SCCP: Selecting Button Layout for a Cisco Unified IP Phone 7931G”
section on page 897.

Header Bar Display

You can customize the content of an IP phone header bar, which is the top line of the IP phone display.
The IP phone header bar, or top line, of a Cisco Unified IP Phone normally replicates the text that
appears next to the first line button. The header bar is shown in Figure 50. 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.

Figure 50 Cisco Unified 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

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Phone Labels
Pone labels are configurable text strings that can be displayed instead of extension numbers next to line
buttons on a Cisco Unified IP phone. By default, the number that is associated to a directory number,
and assigned to a phone, is displayed next to the applicable button. The label feature allows you to enter
a meaningful text string for each directory number so that a phone user with multiple lines can select a
line by label instead of by phone number, thus eliminating the need to consult in-house phone directories.
For configuration information, see the “SCCP: Creating Labels for Directory Numbers” section on
page 901 or the “SIP: Creating Labels for Directory Numbers” section on page 903.

Programmable Vendor Parameters for Phones

The vendorConfig section of the configuration file contains phone and display parameters that are read
and implemented by a phone's firmware when that phone is booted. Only the parameters supported by
the currently loaded firmware are available. The number and type of parameters may vary from one
firmware version to the next.
The IP phone that downloads the configuration file will implement only those parameters that it can
support and ignore configured parameters that it cannot implement. For example, a Cisco Unified
IP Phone 7970G does not have a backlit display and cannot implement Backlight parameters regardless
of whether they are configured. The following text shows the format of an entry in the configuration file:
<vendorConfig>
<parameter-name>parameter-value</parameter-name>
</vendorConfig>

For configuration information at the system level, see the “SCCP: Modifying Vendor Parameters for All
Phones” section on page 910. For configuration information for individual phones, see the “SCCP:
Modifying Vendor Parameters For a Specific Phone” section on page 911.

System Message Display

The System Message Display feature allows you to specify a custom text or display message to appear
in the lower part of the display window on display-capable IP phones. If you do not set a custom text or
display message, the default message “Cisco Unified CME” is displayed.
When you specify a text message, 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.
The file-display feature allows you to specify a file to display on display-capable IP phones 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 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.

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URL Provisioning for Feature Buttons

URL provisioning for customized feature buttons allows you to specify alternative XML files to access
using the feature buttons on IP phones.
The Cisco Unified IP Phones 7940, 7940G, 7960, and 7960G have customized feature buttons that
invoke noncall-related programmable services. The four buttons—Services, Directories, Messages, and
Information (the i button)—are linked to appropriate feature operations through programmable URLs.
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.
The feature buttons are provisioned with specific URLs. The URLs link to XML web pages formatted
with XML tags that the Cisco Unified IP phone understands and uses. When you press a feature button,
the Cisco Unified IP phone uses the configured URL to access the appropriate XML web page for
instructions. The web page sends instructions to the Cisco Unified IP phone to display information on
the screen for users to navigate. Phone users can select options and enter information by using soft keys
and the scroll button.

How to Configure Cisco Unified IP Phone Options

This section contains the following tasks:

Button Layout for Cisco Unified IP Phone 7931G

• SCCP: Selecting Button Layout for a Cisco Unified IP Phone 7931G, page 897 (required)

Header Bar Display

• SCCP: Modifying Header Bar Display, page 898 (required)
• SIP: Modifying Header Bar Display, page 899 (required)
• Verifying Header Bar Display, page 901 (optional)
• Troubleshooting Header Bar Display, page 901 (optional)

Labels for Directory Numbers

• SCCP: Creating Labels for Directory Numbers, page 901 (required)
• SIP: Creating Labels for Directory Numbers, page 903 (required)
• Verifying Labels, page 904 (optional)

System Message Display

• SCCP: Modifying System Message Display, page 904 (required)
• Verifying System Message Display, page 906 (optional)
• Troubleshooting System Message Display, page 906 (optional)

URLs for Feature Buttons

• SCCP: Provisioning URLs for Feature Buttons, page 907 (required)
• SIP: Provisioning URLs for Feature Buttons, page 908 (required)
• Troubleshooting URL Provisioning for Feature Buttons, page 909 (optional)

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Programmable VendorConfig Parameters

• SCCP: Modifying Vendor Parameters for All Phones, page 910 (optional)
• SCCP: Modifying Vendor Parameters For a Specific Phone, page 911 (optional)
• Troubleshooting Vendor Parameter Configuration, page 913 (optional)

SCCP: Selecting Button Layout for a Cisco Unified IP Phone 7931G

To select a fixed-button layout for a Cisco Unified IP Phone 7931G, perform the following steps.

Prerequisites
Cisco Unified CME 4.0(2) or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone template template-tag
4. button-layout set phone-type [1 | 2]
5. exit
6. ephone phone-tag
7. ephone-template template-tag
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-template template-tag Enters ephone-template configuration mode to create an
ephone template.
Example:
Router(config)# ephone-template 15

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Command or Action Purpose

Step 4 button-layout phone-type {1 | 2} Specifies which fixed set of feature buttons appears on a
Cisco Unified IP Phone 7931G that uses a template in
which this is configured.
Example:
Router(config-ephone-template)# button-layout • 1—Includes two predefined feature buttons: button
7931 2 24 is Menu and button 23 is Headset.
• 2—Includes four predefined feature buttons: button
24 is Menu; button 23 is Headset; button 22 is
Directories; and button 21 is Messages.
Step 5 exit Exits from this command mode to the next highest mode
in the configuration mode hierarchy.
Example:
Router(config-ephone-template)# exit
Step 6 ephone phone-tag Enters ephone configuration mode.

Example:
Router(config)# ephone 1
Step 7 ephone-template template-tag Applies an ephone template to the ephone that is being
configured.
Example:
Router(config-ephone)# ephone-template 15
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone)# end

What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.

SCCP: Modifying Header Bar Display

To modify the phone header bar display, perform the following steps.

Prerequisites
Directory number to be modified is already configured. For configuration information, see “SCCP:
Creating Directory Numbers” on page 158.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag

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4. description display-text
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode.

Example:
Router(config)# ephone-dn 55
Step 4 description display-text Defines a description for the header bar of a display-capable IP
phone on which this ephone-dn appears as the first line.
Example: • display-text—Alphanumeric character string, up to
Router(config-ephone-dn)# description 40 characters. String is truncated to 14 characters in the
408-555-0134 display.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.

SIP: Modifying Header Bar Display

To modify the phone header bar display on supported SIP phones, perform the following steps.

Prerequisites
• Cisco CME 3.4 or a a later version.

Restrictions
• This feature is supported only on Cisco Unified IP Phone 7940, 7940G, 7960, and 7960G.

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SUMMARY STEPS

1. enable
2. configure terminal
3. voice register pool pool-tag
4. description string
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for a SIP phone in
Cisco Unified CME.
Example:
Router(config)# voice register pool 3
Step 4 description string Defines a customized description that appears in the header
bar of supported Cisco Unified IP phones
Example: • Truncated to 14 characters in the display.
Router(config-register-pool)# description
408-555-0100
• If string contains spaces, enclose the string in quotation
marks.
Step 5 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-pool)# end

What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “SIP: Generating Configuration Profiles for SIP Phones” on
page 250.

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Verifying Header Bar Display

Step 1 Use the show running-config command to verify your configuration. Descriptions for directory
numbers are listed in the ephone-dn and voice-register dn portions of the output.
Router# show running-config

ephone-dn 1 dual-line
number 150 secondary 151
description 555-0150
call-forward busy 160
call-forward noan 160 timeout 10
huntstop channel
no huntstop
!
!
!
voice-register dn 1
number 1101
description 555-0101

Troubleshooting Header Bar Display

Step 1 show telephony-service ephone
Use this command to ensure that the ephone-dn to which you applied the description appears on the first
button on the ephone. In the example below, ephone-dn 22 has the description in the phone display
header bar.
Router# show telephony-service ephone

ephone-dn 22
number 2149
description 408-555-0149

ephone 34
mac-address 0030.94C3.F96A
button 1:22 2:23 3:24
speed-dial 1 5004
speed-dial 2 5001

SCCP: Creating Labels for Directory Numbers

To create a label to display in place of the number next to a line button, perform the following steps.

Prerequisites
Directory number for which the label is to be created is already configured. For configuration
information, see “SCCP: Creating Directory Numbers” on page 158.

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SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag
4. label label-string
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 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 4 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 5 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “Generating Configuration Files for Phones” on page 245.

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SIP: Creating Labels for Directory Numbers

To create label to be displayed in place of a directory number for a SIP phone, intercom line, voice port,
or a message-waiting indicator (MWI), perform the following steps for each label to be created.

Prerequisites
• Cisco CME 3.4 or a later version.
• Directory number for which the label is to be created is already configured and must already have a
number assigned by using the number (voice register dn) command. For configuration
information, see “SIP: Creating Directory Numbers” on page 162.

Restrictions
• Only one label is permitted per directory number.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register dn dn-tag
4. label string
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or a message-waiting indicator (MWI).
Example:
Router(config-register-global)# voice register
dn 17
Step 4 number number Defines a valid number for a directory number.

Example:
Router(config-register-dn)# number 7001

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Command or Action Purpose

Step 5 label string Creates a text identifier, instead of a phone-number display,
for a directory number that appears on a SIP phone console.
Example:
Router(config-register-dn)# label user01
Step 6 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-dn)# end

What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “SIP: Generating Configuration Profiles for SIP Phones” on
page 250.

Verifying Labels
Step 1 Use the show running-config command to verify your configuration. Descriptions for directory
numbers are listed in the ephone-dn and voice-register dn portions of the output.
Router# show running-config

ephone-dn 1 dual-line
number 150 secondary 151
label MyLine
call-forward busy 160
call-forward noan 160 timeout 10
huntstop channel
no huntstop
!
!
!
voice-register dn 1
number 1101
label MyLine

SCCP: Modifying System Message Display

To modify the system message display on phone screen, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. system message text-message

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5. url idle url idle-timeout seconds

6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)#
Step 4 system message text-message Defines a text message to display when a phone is idle.
• text-message—Alphanumeric string to display. Display uses
Example: proportional-width font, so the number of characters that are
Router(config-telephony)# system message displayed varies based on the width of the characters that are
ABC Company used. The maximum number of displayed characters is
approximately 30.
Step 5 url idle url idle-timeout seconds Defines the location of a file to display on phones that are 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—Time interval between display refreshes, in
seconds. Range is 0 to 300.
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

What to Do Next
After configuring the url idle command, you must reset phones. See “SCCP: Using the reset Command”
on page 259.

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Verifying System Message Display

Step 1 Use the show running-config command to verify your configuration. System message display is listed
in the telephony-service portion of the output.
Router# show running-config

telephony-service
fxo hook-flash
load 7960-7940 P00307020300
load 7914 S00104000100
max-ephones 100
max-dn 500
ip source-address 10.153.13.121 port 2000
max-redirect 20
timeouts ringing 100
system message XYZ Company
voicemail 7189
max-conferences 8 gain -6
call-forward pattern .T
moh flash:music-on-hold.au
multicast moh 239.10.10.1 port 2000
web admin system name server1 password server1
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 92......
transfer-pattern 91..........
transfer-pattern 93......
transfer-pattern 94......
transfer-pattern 95......
transfer-pattern 96......
transfer-pattern 97......
transfer-pattern 98......
transfer-pattern 99......
transfer-pattern .T
secondary-dialtone 9
create cnf-files version-stamp Jan 01 2002 00:00:00

Troubleshooting System Message Display

Step 1 Ensure that the HTTP server is enabled.

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SCCP: Provisioning URLs for Feature Buttons

To customize URLs for feature buttons in the Sep*.conf.xml configuration file for SCCP IP phones,
perform the following steps.

Restrictions
• Operation of these services is determined by the Cisco Unified IP phone capabilities and the content
of the specified URL.
• Provisioning a URL to access help screens using the i or ? buttons on a phone is not supported.
• Provisioning the directory URL to select an external directory resource disables the
Cisco Unified CME local directory service.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. url {directories | information | messages | services} url
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)#

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Command or Action Purpose

Step 4 url {directories | information | messages Provisions URLs for the four feature buttons on an IP phone:
| services} url Directories, Information, Messages, and Services.
• To use a Cisco Unified Communications Manager directory
Example: as an external directory source, you must list the MAC
Router(config-telephony)# url directories addresses of the phones in Cisco Unified Communications
http://10.4.212.4/localdirectory
Manager and reset the phones from
Cisco Unified Communications Manager. You do not need to
assign ephone-dns to the phones or for the phones to register
with Cisco Unified Communications Manager.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “Generating Configuration Files for Phones” on page 245.

SIP: Provisioning URLs for Feature Buttons

To customize URLs for feature buttons SEPDEFAULT.cnf configuration profile for SIP IP phones,
perform the following steps.

Prerequisites
• Cisco CME 3.4 or a later version.

Restrictions
• Operation of these services is determined by the Cisco Unified IP phone capabilities and the content
of the specified URL.
• Provisioning a URL is supported only for Services and Directories feature buttons on SIP phones.
• Programmable Directories and Services feature buttons are supported only on the Cisco Unified IP
Phone 7960, 7960G, 7940, and 7940G.
• Provisioning the directory URL to select an external directory resource disables the
Cisco Unified CME local directory service.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global

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4. url {directory | service} url

5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register global Enters telephony-service configuration mode.

Example:
Router(config)#
Step 4 url {directory | service} url Associates a URL with the programmable feature buttons on SIP
phones.
Example:
Router(config-register-global)# url
directory http://10.0.0.11/localdirectory
Router(config-register-global)# url
service
http://10.0.0.4/CCMUser/123456/urltest.ht
ml
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-register-global)# end

What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “SIP: Generating Configuration Profiles for SIP Phones” on
page 250.

Troubleshooting URL Provisioning for Feature Buttons

Step 1 Ensure the HTTP server is enabled and that there is communication between the Cisco Unified CME
router and the server.

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SCCP: Modifying Vendor Parameters for All Phones

To configure programmable phone and display parameters in the vendorConfig section of the
SepDefault.conf.xml configuration file for all phones, perform the following steps.

Restrictions
• Only the parameters supported by the currently loaded firmware are available.
• The number and type of parameters may vary from one firmware version to the next.
• Only those parameters that are supported by a Cisco Unified IP phone and firmware version are
implemented. Parameters that are not supported are ignored.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. service phone parameter-name parameter-value
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 4 service phone parameter-name parameter-value Sets display and phone functionality for all IP phones that
support the configured parameters and to which this
template is applied.
Example:
Router(config-telephony)# service phone • The parameter name is word and case-sensitive. See the
daysBacklightNotActive 1,2,3,4,5,6,7 Cisco Unified CME Command Reference for a list of
Router(config-telephony)# service phone
parameters.
backlightOnTime 07:30
Router(config-telephony)# service phone • This command can also be configured in ephone-
backlightOnDuration 10:00 template configuration mode and applied to one or
Router(config-telephony)# service phone
backlightIdleTimeout 00.01
more phones.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “Generating Configuration Files for Phones” on page 245.

SCCP: Modifying Vendor Parameters For a Specific Phone

To configure parameters in the vendorConfig section of the Sep*.conf.xml configuration file for an
individual SCCP phone , perform the following steps.

Restrictions
• Cisco Unified CME 4.0 or a later version.
• System must be configured to for per-phone configuration files. For configuration information, see
“SCCP: Defining Per-Phone Configuration Files and Alternate Location” on page 129.
• Only the parameters supported by the currently loaded firmware are available.
• The number and type of parameters may vary from one firmware version to the next.
• Only those parameters that are supported by a Cisco Unified IP phone and firmware version are
implemented. Parameters that are not supported are ignored.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone template template-tag
4. service phone parameter-name parameter-value
5. exit
6. ephone phone-tag

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7. ephone-template template-tag
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-template template-tag Enters ephone-template configuration mode to create an
ephone template.
Example:
Router (config)# ephone-template 15
Step 4 service phone parameter-name parameter-value Sets parameters for all IP phones that support the
configured functionality and to which this template is
applied.
Example:
Router(config-ephone-template)# service phone • The parameter name is word and case-sensitive. See
daysBacklightNotActive 1,2,3,4,5,6,7 the Cisco Unified CME Command Reference for a
Router(config-ephone-template)# service phone
list of parameters.
backlightOnTime 07:30
Router(config-ephone-template)# service phone • This command can also be configured in
backlightOnDuration 10:00 telephony-service configuration mode. For
Router(config-ephone-template)# service phone
backlightIdleTimeout 00.01
individual phones, the template configuration for
this command overrides the system-level
configuration for this command.
Step 5 exit Exits from this command mode to the next highest mode
in the configuration mode hierarchy.
Example:
Router(config-ephone-template)# exit
Step 6 ephone phone-tag Enters ephone configuration mode.

Example:
Router(config)# ephone 1
Step 7 ephone-template template-tag Applies an ephone template to the ephone that is being
configured.
Example:
Router(config-ephone)# ephone-template 15
Step 8 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone)# end

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What to Do Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See the “Generating Configuration Files for Phones” on page 245.

Troubleshooting Vendor Parameter Configuration

Step 1 Ensure that the templates have been properly applied to the phones.
Step 2 Ensure that you use the create cnf-files command to regenerate configuration files and reset the phones
after you apply the templates.
Step 3 Use the show telephony-service tftp-bindings command to display the configuration files that are
associated with individual phones
Router# show telephony-service tftp-binding

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 4 Use the debug tftp events command to verify that the phone is accessing the file when you reboot the
phone.

Configuration Examples for Cisco Unified IP Phone Options

This section contains the following examples:
• Text Labels for Ephone-dns: Example, page 914
• Phone Header Bar Display: Example, page 914
• System Text Message Display: Example, page 914
• System File Display: Example, page 914
• URL Provisioning for Directories, Services, and Messages Buttons: Example, page 914
• Programmable VendorConfig Parameters: Example, page 915

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Text Labels for Ephone-dns: 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

Phone Header Bar Display: Example

The following example provides the full E.164 number for a phone line in the phone header bar:
ephone-dn 55
number 2149
description 408-555-0149

ephone-dn 56
number 2150

ephone 12
button 1:55 2:56

System Text Message Display: Example

The following example specifies text that should display on IP phones when they are not in use:
telephony-service
system message ABC Company

System File Display: Example

The following example specifies that a file called logo.htm should be displayed on IP phones when they
are not in use:
telephony-service
url idle http://www.abcwrecking.com/public/logo.htm idle-timeout 35

URL Provisioning for Directories, Services, and Messages Buttons: Example

The following example provisions the Directories, Services, and Messages buttons.
telephony-service
url directories http://10.4.212.4/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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Additional References

Programmable VendorConfig Parameters: Example

The following partial output shows a template in which programmable parameters for phone and display
functionality have been configured by using the service phone command.
ephone-template 1
button-layout 7931 1
service phone daysBacklightNotActive 1,2,3,4,5,6,7
service phone backlightOnTime 07:30
service phone backlightOnDuration 10:00
service phone backlightIdleTimeout 00.01

In the following example, the PC port is disabled on phones 26 and 27. All other phones have the PC
port enabled.
ephone-template 8
service phone pcPort 1
!
!
ephone 26
mac-address 1111.1111.1001
ephone-template 8
type 7960
button 1:26
!
!
ephone 27
mac-address 1111.2222.2002
ephone-template 8
type 7960
button 1:27

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

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Additional References

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Cisco Unified IP Phone Options

Feature Information for Cisco Unified IP Phone Options

Table 54 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 54 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 54 Feature Information for Cisco Unified IP Phone Options

Cisco Unified CME

Feature Name Version Feature Information
Fixed Line/Feature Buttons 4.0(2) Provides two preconfigured fixed sets of feature buttons for
provisioning a Cisco Unified IP Phone 7931G.
Header Bar Display 3.4 Added support for modifying header bar display on SIP
phones.
2.01 Phone header bar display is introduced.
Labels for Directory Numbers 3.4 Added support for label display on SIP phones.
3.0 Ephone-dn labels were introduced.
Programmable Vendor Parameters 4.0 Added support for configuring programmable phone and
display functionality at a phone level for SCCP phones.
3.4 Added support for configuring programmable phone and
display functionality for SIP phones.
3.2.1 Added support for programmable phone and display
functionality in vendorConfig portion of configuration file.
Implementation of configuration is firmware version
dependent.
System Message Display 3.0 System message display on idle phones using text
messages was introduced.
2.1 System message display on idle phones using HTML files
was introduced.
URL Provisioning for Feature Buttons 3.4 Added support for provisioning customized URLs for
feature buttons on supported SIP phones.
2.0 Provisioning customized URLs for feature buttons was
introduced.

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Services

Last Updated: July 5, 2007

First published: June 18, 2007

This chapter describes features in Cisco Unified Communications Manager Express

(Cisco Unified CME) that provide support for interoperability between Cisco Unified CME and
Cisco Customer Response Solutions (CRS) with Cisco Unified Contact Center Express (Unified CCX).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Interoperability
Feature” section on page 941.

Contents
• Information About Interoperability with External Services, page 919
• How to Configure Interoperability with External Services, page 921
• Configuration Examples for Interoperability with Unified CCX, page 930
• Where to Go Next, page 939
• Additional References, page 940
• Feature Information for Interoperability with External Services, page 941

Information About Interoperability with External Services

To configure interoperability, you should understand the following concepts:
• Interoperability with Unified CCX, page 920

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Interoperability with Unified CCX

Cisco Unified CME 4.2and Cisco IOS Release 12.4(11)XW2 and later versions supports interoperability
between Cisco Unified CME and Cisco Customer Response Solutions (CRS) 5.0 and later versions with
Unified CCX, including enhanced call processing, device and call monitoring, unattended call transfers
to multiple call center agents and basic extension mobility, and IP IVR applications.
The Unified CCX application uses the CRS platform to provide a multimedia (voice, data, and web).
Cisco IP IVR functionality is available with Unified CCX and includes prompt-and-collect and call
treatment.
The following functions are provided in Cisco Unified CME 4.2 and later versions:
• Support of Unified CCX Cisco Agent Desktop for use with Cisco Unified CME
• Configuration query and update between Unified CCX and Cisco Unified CME.
• SIP-based simple and supplementary call control services including:
– Call routing between Cisco Unified CME and Unified CCX using SIP-based route point
– First-party call control for SIP-based simple and supplementary calls
– Call monitoring and device monitoring based on SIP presence and dialog event package
• Unified CCX session management of Cisco Unified CME
• Unified CCX device and call monitoring of agent lines and call activities in Cisco Unified CME
Provisioning and configuration information in Unified CCX is automatically provided to
Cisco United CME. If the configuration from Unified CCX is deleted or must be modified, you can
configure the same information in Cisco Unified CME by using Cisco IOS commands.
For first party call control, a route point for Cisco CRS is a peer device to Cisco Unified CME through
a SIP trunk. An incoming call to Cisco Unified CME that is targeted to a call center phone is routed to
Unified CCX through the route point. The call is placed in a queue and redirected to the most suitable
agent by Unified CCX.
Supplementary services such as call hold, blind transfer, and semi-attended transfer are initiated by
Unified CCX. Existing SIP-based simple and supplementary service call flow applies except for blind
transfers. For blind transfers with Unified CCX as the transferrer, Unified CCX will stay in the active
state until the transfer target answers. It drops out only after the transferred call is successfully answered.
If the transfer target does not answer when ringing times out, the call is pulled back by Unified CCX and
rerouted to another agent. This mechanism also applies when the transfer target is configured with
call-forward all or forward no-answer. The forward configuration is ignored during blind transfer.
When a call moves between Unified CCX and Cisco Unified CME because of redirect, transfer, and
conference, the SIP Call-ID continues to change. For call control purposes, Cisco Unified CME issues
a unique Global Call ID (Gcid) for every outbound call leg. A Gcid remains the same for all legs of the
same call in the system, and is valid for redirect, transfer, and conference events, including 3-party
conferencing when a call center phone acts as a conference host.
Table 55 contains a list of tasks required to enable operability between Cisco Unified CME and
Unified CCX, presented in the order in which the tasks are to be completed. This section contains
information about performing tasks in the first 2 steps in this table and procedures for completing step 3.
For configuration information, see the “How to Configure Interoperability with External Services”
section on page 921.

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Table 55 Tasks to Configure Interoperability between Cisco CRS and Cisco Unified CME

Step Task Name of Document

1 Verify that Cisco Unified —
Communications Manager Express
(Cisco Unified CME) 4.2 or a later version is installed on
the router.
2 Configure the Cisco Unified CME router. Prerequisites, page 922
Tip Note the AXL user ID, password, and router’s IP
address.
3 Configure Cisco Unified CME to enable interoperability How to Configure Interoperability
with Unified CCX. with External Services, page 921
4 Install Cisco Unified Contact Center Express Cisco CRS Installation Guide at
(Unified CCX) for Cisco Unified CME. http://www.cisco.com/en/US/prod
5 Perform the initial setup of Cisco CRS for ucts/sw/custcosw/ps1846/prod_in
Cisco Unified CME. stallation_guides_list.html.

Tip When setup launches, you are asked for the AXL
user ID and password that you created in
Cisco Unified CME. You also need to enter the
router IP address.
6 Configure CME Telephony Subsystem to enable Cisco CRS Administration Guide
interoperability with Unified CCX. at
7 Create users and assign the agent capability in Cisco http://www.cisco.com/en/US/prod
CRS. ucts/sw/custcosw/ps1846/prod_m
aintenance_guides_list.html.

How to Configure Interoperability with External Services

This section contains the following procedures:
• Configuring Cisco Unified CME to Enable Interoperability with Unified CCX, page 922 (required)
• Identifying Agent Directory Numbers in Cisco Unified CME for Session Manager, page 924
(required)
• Verifying Registrations and Subscriptions in Cisco Unified CME, page 926 (optional)
• Re-creating a Session Manager in Cisco Unified CME, page 926 (optional)
• Reconfiguring a Cisco CRS Route Point as a SIP Endpoint in Cisco Unified CME, page 928
(optional)

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Configuring Cisco Unified CME to Enable Interoperability with Unified CCX

To configure Cisco Unified CME to enable interoperability between Cisco Unified CME and
Unified CCX, perform the following steps.

Note A single Cisco Unified CME can support multiple session managers.

Prerequisites
• Cisco Unified CME 4.2 using Cisco IOS Release 12.4(11)XW2 or a later version.
• XML API must be configured to create a username for Unified CCX access. For configuration
information, see “Configuring the XML API” on page 959. Make note of the user ID, password, and
router’s IP address for using during the initial setup of Cisco CRS for Cisco Unified CME.
• Phones to be connected in Cisco Unified CME must be configured. When configuring a
Unified CCX agent phone, use the keep-conference endcall command to enable conference
initiators to exit from conference calls and end the conference for the remaining parties. For
configuration information, see “Configuring Conferencing” on page 647.
• The Cisco Unified CME router must be configured to accept incoming presence requests. For
configuration information, see “Configuring Presence Service” on page 797.

Restrictions
• Interoperability between Cisco Unified CME and Unified CCX is restricted to one Unified CCX per
Cisco Unified CME.
• Support for Multi-Party Ad Hoc and Meet-Me Conferencing features is not provided.
• Only incoming calls from PSTN trunk are supported for deployment of the interoperability feature.
Other trunks, such as SIP and H.323, are supported as usual in Cisco Unified CME, however, not for
customer calls to Unified CCX.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice call send-alert
4. voice service voip
5. callmonitor
6. gcid
7. allow-connections sip-to-sip
8. no supplementary-service sip moved-temporary
9. no supplementary-service sip refer
10. sip
11. registrar server [expires [max sec] [min sec]
12. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice call send-alert Enables the terminating gateway to send an alert message
instead of a progress message after it receives a call setup
message.
Example:
Router(config)# voice call send-alert
Step 4 voice service voip Enters voice-service configuration mode and specifies
voice-over-IP encapsulation.
Example:
Router(config)# voice service voip
Step 5 callmonitor Enables call monitoring messaging functionality.
• Used by Unified CCX for processing and reporting.
Example:
Router(config-voi-serv)# callmonitor
Step 6 gcid Enables Global Call-ID (Gcid) for call control purposes.
• Used by Unified CCX for tracking call.
Example:
Router(config-voi-serv)# gcid
Step 7 allow-connections sip-to-sip Allows connections between specific types of endpoints in
a VoIP network.
Example:
Router(config-voi-serv)# allow-connections
sip-to-sip
Step 8 no supplementary-service sip moved-temporary Prevents the router from sending a redirect response to the
destination for call forwarding.
Example:
Router(config-voi-serv)# no
supplementary-service sip moved-temporary
Step 9 no supplementary-service sip refer Prevents the router from forwarding a REFER message to
the destination for call transfers.
Example:
Router(config-voi-serv)# no
supplementary-service sip refer
Step 10 sip Enters SIP configuration mode.

Example:
Router(config-voi-srv)# sip

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Command or Action Purpose

Step 11 registrar server [expires [max sec][min sec]] Enables SIP registrar functionality in Cisco Unified CME.
• expires—(Optional) Sets the active time for an
Example: incoming registration.
Router(config-voi-sip)# registrar server
expires max 600 min 60
• max sec—(Optional) Maximum time for a registration
to expire, in seconds. Range: 600 to 86400.
Default: 3600. Recommended value: 600.
• min sec—(Optional) Minimum time for a registration
to expire, in seconds. Range: 60 to 3600. Default: 60.
Step 12 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-voi-serv)# end

Identifying Agent Directory Numbers in Cisco Unified CME for Session

Manager
To specify which directory numbers, associated with phone lines on Unified CCX agent phones, can be
managed by a session manager, perform the following steps.

Prerequisites
• Up to eight session managers must be configured in Cisco Unified CME.
• Directory numbers associated with Unified CCX agent phones must be configured. Directory
numbers for agent phones must be configured as dual lines to allow an agent to make two call
connections at the same time using one phone line button. For configuration information, see
“Configuring Phones to Make Basic Calls” on page 147.

Restrictions
• Only SCCP phones can be configured as agent phones in Cisco Unified CME. The Cisco VG224
Analog Phone Gateway and analog and SIP phones are supported as usual in Cisco Unified CME,
however, not as Unified CCX agent phones.
• Cisco Unified IP Phone 7931 cannot be configured as an agent phone in Cisco Unified CME.
Cisco Unified IP Phone 7931s are supported as usual in Cisco Unified CME, however, not as
Unified CCX agent phones.
• Shared-line appearance is not supported on agent phones. A directory number cannot be associated
with more than one physical agent phone at one time.
• Overlaid lines are not supported on agent phones. More than one directory number cannot be
associated with a single line button on an agent phone.
• Monitored mode for a line button is not supported on agent phones. An agent phone cannot be
monitored by another phone.
• For call forward and call pickup, the directory number of an agent cannot forward to a Cisco CRS
route point.

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SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag
4. allow watch
5. session-server {session-tag[,...session-tag]}
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—Unique ID of an already configured directory
Example: number. The tag number corresponds to a tag number
Router(config)# ephone-dn 24 created when this directory number was initially
configured.
Step 4 session-server Specifies which session managers are to monitor the
session-server-tag[,...session-server-tag] directory number being configured.
• session-server-tag—Unique ID session manager,
Example: configured in Unified CCX and automatically provided
Router(config-ephone-dn)# session-server to Cisco Unified CME. Range: 1 to 8.
1,2,3,4,6
Tip If you do not know the value for session-server-tag,
we recommend using 1.

• Can configure up to eight session-server-tags;

individual tags must be separated by commas (,).
• Each directory number can be managed by up to eight
session managers. Each session manager can monitor
more than one directory number.

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Command or Action Purpose

Step 5 allow watch Allows the phone line associated with this directory number
to be monitored by a watcher in a presence service.
Example: • This command can also be configured in ephone-dn
Router(config-ephone-dn)# allow watch template configuration mode and applied to one or
more phones. The ephone-dn configuration has priority
over the ephone-dn template configuration.
Step 6 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-ephone-dn)# end

Verifying Registrations and Subscriptions in Cisco Unified CME

Before using the system, verify registrations and subscriptions for Unified CCX endpoints.

Step 1 Use the show sip status registrar command to verify whether session manager and Cisco CRS route
points are registered.
Step 2 Use the show presence subscription summary command to verify whether Cisco CRS route points and
Unified CCX agent directory numbers are subscribed.
The following is sample output from the show presence subscription summary command. The first two
rows show the status for two route points. The next two are for logged in agent phones.
Router# show presence subscription summary

Presence Active Subscription Records Summary: 15 subscription

Watcher Presentity SubID Expires SibID Status
======================== ======================== ====== ======= ====== ======
[email protected] [email protected] 4 3600 0 idle
[email protected] [email protected] 8 3600 0 idle
[email protected] [email protected] 10 3600 0 idle
[email protected] [email protected] 12 3599 0 idle

Re-creating a Session Manager in Cisco Unified CME

Note Provisioning and configuration information in Unified CCX is automatically provided to

Cisco United CME. The following task is required only if the configuration from Unified CCX is deleted
or must be modified.

To re-create a session manager in Cisco Unified CME for Unified CCX, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal

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3. voice register session-server session-server-tag

4. register-id name
5. keepalive seconds
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register session-server Enters voice register session-server configuration mode to
session-server-tag enable and configure a session manager for an external
feature server, such as the Unified CCX application on a
Example: Cisco CRS system.
Router(config)# voice register session-server 1 • Range: 1 to 8.
• A single Cisco Unified CME can support multiple
session managers.
Step 4 register id name (Optional) Required only if the configuration from
Unified CCX is deleted or must be modified.
Example: • name—String for identifying Unified CCX. Can
Router(config-register-fs)# CRS1 contain 1 to 30 alphanumeric characters.
Step 5 keepalive seconds (Optional) Required only if the configuration from
Unified CCX is deleted or must be modified.
Example: • Keepalive duration for registration, in seconds, after
Router(config-register-fs)# keepalive 300 which the registration expires unless Unified CCX
reregisters before the registration expiry.
• Range: 60 to 3600. Default: 300.
Note Default in Unified CCX is 120.
Step 6 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-fs)# end

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Reconfiguring a Cisco CRS Route Point as a SIP Endpoint in Cisco Unified CME

Note Provisioning and configuration information in Unified CCX is automatically provided to

Cisco United CME. The following task is required only if the configuration from Unified CCX is deleted
or must be modified.

To reconfigure a Cisco CRS route point as a SIP endpoint in Cisco Unified CME, perform the following
steps.

Prerequisites
• Directory numbers associated with Cisco CRS route points must be configured in
Cisco Unified CME. For configuration information for directory numbers associated with SIP
endpoints, see “Configuring Phones to Make Basic Calls” on page 147.
• Directory numbers associated with Cisco CRS route points must be enabled to be watched. For
configuration information, see “Configuring Presence Service” on page 797.
• The mode cme command must be enabled in Cisco Unified CME.

Restrictions
• Each Cisco CRS route point can be managed by only one session manager.
• Each session manager can manage more than one Cisco CRS route point.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register dn dn-tag
4. number number
5. allow watch
6. refer target dial-peer
7. exit
8. voice register pool pool-tag
9. number tag dn dn-tag
10. session-server session-tag
11. codec codec-type [bytes]
12. dtmf-relay rtp-relay sip-notify
13. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn configuration mode to define a
directory number for a SIP phone, intercom line, voice port,
or a message-waiting indicator (MWI).
Example:
Router(config-register-global)# voice register
dn 1
Step 4 number number Defines a valid number for a directory number.

Example:
Router(config-register-dn)# number 2777
Step 5 allow watch Allows the phone line associated with this directory number
to be monitored by a watcher in a presence service.
Example:
Router(config-register-dn)# allow watch
Step 6 refer target dial-peer Enables watcher to handle SIP REFER message from this
directory number.
Example: • target dial-peer—Refer To portion of message is
Router(config-register-dn)# refer target based on address from dial peer for this directory
dial-peer number.
Step 7 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-register-dn)# exit
Step 8 voice register pool pool-tag Enters voice register pool configuration mode to set
device-specific parameters for a Cisco CRS route point.
Example: • A voice register pool in Unified CCX can contain up to
Router(config)# voice register pool 3 10 individual SIP endpoints. Subsequent pools are
created for additional SIP endpoints.
Step 9 number tag dn dn-tag Associates a directory number with the route point being
configured.
Example:
Router(config-register-pool)# number 1 dn 1

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Command or Action Purpose

Step 10 session-server session-server-tag Identifies session manager to be used to control the route
point being configured.
Example: • session-server-tag—Unique number assigned to a
Router(config-register-pool)# session-server 1 session manager. Range: 1 to 8. The tag number
corresponds to a tag number created by using the voice
register session-server command.
Step 11 codec g711ulaw Specifies the codec for the dial peer dynamically created for
the route point being configured.
Example: • codec-type—g711ulaw is required for Unified CCX.
Router(config-register-pool)# codec g711ulaw
Step 12 dtmf-relay sip-notify Specifies DTMF Relay method to be used by the route point
being configured.
Example:
Router(config-register-pool)# dtmf-relay
sip-notify
Step 13 end Exits configuration mode and enters privileged EXEC
mode.
Example:
Router(config-register-pool)# end

Configuration Examples for Interoperability with Unified CCX

The following output from the show running-configuration command shows the configuration on a
Cisco Unified CME router that will interoperate with Unified CCX.
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname sb-sj3-3845-uut1
!
boot-start-marker
boot-end-marker
!
card type t1 0 2
card type t1 0 3
logging buffered 1000000
no logging console
enable password password
!
no aaa new-model
network-clock-participate wic 2
network-clock-participate wic 3
ip cef
!
!
no ip dhcp use vrf connected
!
!
ip dhcp excluded-address 192.0.2.250 192.0.2.254
!

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ip dhcp pool ephones

network 192.0.2.0 255.255.255.0
option 150 ip 192.0.2.254
default-router 192.0.2.254
!
!
no ip domain lookup
!
isdn switch-type primary-5ess
voice-card 0
no dspfarm
!
!
!
!
voice service voip
gcid
callmonitor
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
registrar server expires max 120 min 60
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
!
!
!
!
!
!
!
!
voice register global
mode cme
source-address 192.0.2.254 port 5060
max-dn 720
max-pool 240
authenticate presence
authenticate register
dialplan-pattern 1 511.... extension-length 4
voicemail 9001
create profile sync 0000347600391314
!
voice register session-server 1
keepalive 300
register-id SB-SJ3-UCCX1_1164774025000
!
voice register dn 1
session-server 1
number 8999
allow watch
refer target dial-peer
!
voice register dn 2
session-server 1

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number 8001
allow watch
refer target dial-peer
!
voice register dn 3
session-server 1
number 8101
allow watch
refer target dial-peer
!
voice register dn 11
number 2011
name ep-sip-1-11
mwi
!
voice register dn 12
number 2012
name ep-sip-1-12
mwi
!
voice register dn 16
number 5016
name rp-sip-1-16
label SIP 511-5016
mwi
!
voice register dn 17
number 5017
name rp-sip-1-17
label SIP 511-5017
mwi
!
voice register dn 18
number 5018
name rp-sip-1-18
label SIP 511-5018
mwi
!
voice register pool 1
session-server 1
number 1 dn 1
number 2 dn 2
number 3 dn 3
dtmf-relay sip-notify
codec g711ulaw
!
voice register pool 11
id mac 1111.0711.2011
type 7970
number 1 dn 11
dtmf-relay rtp-nte
voice-class codec 1
username 5112011 password 5112011
!
voice register pool 12
id mac 1111.0711.2012
type 7960
number 1 dn 12
dtmf-relay rtp-nte
voice-class codec 1
username 5112012 password 5112012
!
voice register pool 16
id mac 0017.0EBC.1500

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type 7961GE
number 1 dn 16
dtmf-relay rtp-nte
voice-class codec 1
username rp-sip-1-16 password pool16
!
voice register pool 17
id mac 0016.C7C5.0660
type 7971
number 1 dn 17
dtmf-relay rtp-nte
voice-class codec 1
username rp-sip-1-17 password pool17
!
voice register pool 18
id mac 0015.629E.825D
type 7971
number 1 dn 18
dtmf-relay rtp-nte
voice-class codec 1
username rp-sip-1-18 password pool18
!
!
!
!
!
!
!
controller T1 0/2/0
framing esf
clock source internal
linecode b8zs
pri-group timeslots 1-4,24
!
controller T1 0/2/1
framing esf
clock source internal
linecode b8zs
pri-group timeslots 1-4,24
!
controller T1 0/3/0
framing esf
clock source internal
linecode b8zs
ds0-group 0 timeslots 1-4 type e&m-immediate-start
!
controller T1 0/3/1
framing esf
clock source internal
linecode b8zs
ds0-group 0 timeslots 1-4 type e&m-immediate-start
vlan internal allocation policy ascending
!
!
!
!
interface GigabitEthernet0/0
ip address 209.165.201.1 255.255.255.224
duplex auto
speed auto
media-type rj45
!
interface GigabitEthernet0/1
ip address 192.0.2.254 255.255.255.0

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Configuration Examples for Interoperability with Unified CCX

duplex auto
speed auto
media-type rj45
!
interface Serial0/2/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-5ess
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
!
interface Serial0/2/1:23
no ip address
encapsulation hdlc
isdn switch-type primary-5ess
isdn protocol-emulate network
isdn incoming-voice voice
no cdp enable
!
interface Service-Engine1/0
ip unnumbered GigabitEthernet0/0
service-module ip address 209.165.202.129 255.255.255.224
service-module ip default-gateway 209.165.201.1
!
ip route 192.0.0.30 255.0.0.0 192.0.0.55
ip route 209.165.202.129 255.255.255.224 Service-Engine1/0
ip route 192.0.2.56 255.255.255.0 209.165.202.2
ip route 192.0.3.74 255.255.255.0 209.165.202.3
ip route 209.165.202.158 255.255.255.224 192.0.0.55
!
!
ip http server
ip http authentication local
ip http path flash:
!
!
ixi transport http
response size 64
no shutdown
request outstanding 1
!
ixi application cme
no shutdown
!
!
!
control-plane
!
!
!
voice-port 0/0/0
!
voice-port 0/0/1
!
voice-port 0/2/0:23
!
voice-port 0/3/0:0
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/2/1:23

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Configuration Examples for Interoperability with Unified CCX

!
voice-port 0/3/1:0
!
!
!
!
!
dial-peer voice 9000 voip
description ==> This is for internal calls to CUE
destination-pattern 9...
voice-class codec 1
session protocol sipv2
session target ipv4:209.165.202.129
dtmf-relay rtp-nte sip-notify
!
dial-peer voice 9001 voip
description ==> This is for external calls to CUE
destination-pattern 5119...
voice-class codec 1
session protocol sipv2
session target ipv4:209.165.202.129
dtmf-relay rtp-nte sip-notify
!
dial-peer voice 521 voip
destination-pattern 521....
voice-class codec 1
max-redirects 5
session protocol sipv2
session target ipv4:209.165.201.2
dtmf-relay rtp-nte sip-notify
!
dial-peer voice 531 voip
destination-pattern 531....
voice-class codec 1
max-redirects 5
session protocol sipv2
session target ipv4:209.165.201.3
dtmf-relay rtp-nte sip-notify
!
!
presence
presence call-list
watcher all
allow subscribe
!
sip-ua
mwi-server ipv4:209.165.202.128 expires 3600 port 5060 transport udp
presence enable
!
!
telephony-service
no auto-reg-ephone
xml user axluser password axlpass 15
max-ephones 240
max-dn 720
ip source-address 192.0.2.254 port 2000
system message sb-sj3-3845-uut1
url services http://192.0.2.252:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp
url authentication http:192.0.2.252:6293/ipphone/jsp/sciphonexml/IPAgentAuthenticate.jsp
cnf-file perphone
dialplan-pattern 1 511.... extension-length 4
voicemail 9001
max-conferences 8 gain -6
call-forward pattern .T

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Configuration Examples for Interoperability with Unified CCX

moh flash:music-on-hold.wav
multicast moh 239.10.10.1 port 2000
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Jun 18 2007 07:44:25
!
!
ephone-dn 1 dual-line
session-server 1
number 1001
name ag-1-1
allow watch
mwi sip
!
!
ephone-dn 2 dual-line
session-server 1
number 1002
name ag-1-2
allow watch
mwi sip
!
!
ephone-dn 3 dual-line
session-server 1
number 1003
name ag-1-3
allow watch
mwi sip
!
!
ephone-dn 4 dual-line
session-server 1
number 1004
name ag-1-4
allow watch
mwi sip
!
!
ephone-dn 5
session-server 1
number 1005
name ag-1-5
allow watch
mwi sip
!
!
ephone-dn 11 dual-line
number 3011
name ep-sccp-1-11
mwi sip
!
!
ephone-dn 12
number 3012
name ep-sccp-1-12
mwi sip
!
!
ephone-dn 16 dual-line
number 4016
label SCCP 511-4016
name rp-sccp-1-16
mwi sip

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Configuration Examples for Interoperability with Unified CCX

!
!
ephone-dn 17 dual-line
number 4017
label SCCP 511-4017
name rp-sccp-1-17
mwi sip
!
!
ephone-dn 18 dual-line
number 4018
label SCCP 511-4018
name rp-sccp-1-18
mwi sip
!
!
ephone-dn 19 dual-line
number 4019
label SCCP 511-4019
name rp-sccp-1-19
mwi sip
!
!
ephone-dn 20 dual-line
number 4020
label SCCP 511-4020
name rp-sccp-1-20
mwi sip
!
!
ephone-dn 21 dual-line
number 4021
label SCCP 511-4021
name rp-sccp-1-21
mwi sip
!
!
ephone-dn 22 dual-line
number 4022
label SCCP 511-4022
name rp-sccp-1-22
mwi sip
!
!
ephone 1
mac-address 1111.0711.1001
type 7970
keep-conference endcall
button 1:1
!
!
!
ephone 2
mac-address 1111.0711.1002
type 7970
keep-conference endcall
button 1:2
!
!
!
ephone 3
mac-address 1111.0711.1003
type 7970
keep-conference endcall

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Configuration Examples for Interoperability with Unified CCX

button 1:3
!
!
!
ephone 4
mac-address 1111.0711.1004
type 7970
keep-conference endcall
button 1:4
!
!
!
ephone 5
mac-address 1111.0711.1005
type 7970
keep-conference endcall
button 1:5
!
!
!
ephone 11
mac-address 1111.0711.3011
type 7970
keep-conference endcall
button 1:11
!
!
!
ephone 12
mac-address 1111.0711.3012
type 7960
keep-conference endcall
button 1:12
!
!
!
ephone 16
mac-address 0012.D916.5AD6
type 7960
keep-conference endcall
button 1:16
!
!
!
ephone 17
mac-address 0013.1AA6.7A9E
type 7960
keep-conference endcall
button 1:17
!
!
!
ephone 18
mac-address 0012.80F3.B013
type 7960
keep-conference endcall
button 1:18
!
!
!
ephone 19
mac-address 0013.1A1F.6282
type 7970
keep-conference endcall

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Where to Go Next

button 1:19
!
!
!
ephone 20
mac-address 0013.195A.00D0
type 7970
keep-conference endcall
button 1:20
!
!
!
ephone 21
mac-address 0017.0EBC.147C
type 7961GE
keep-conference endcall
button 1:21
!
!
!
ephone 22
mac-address 0016.C7C5.0578
type 7971
keep-conference endcall
button 1:22
!
!
!
line con 0
exec-timeout 0 0
stopbits 1
line aux 0
stopbits 1
line 66
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120
line vty 0 4
password lab
login
!
scheduler allocate 20000 1000
!
end

Where to Go Next
If you are done modifying parameters for phones in Cisco Unified CME, generate a new configuration
file and restart the phones. See “Generating Configuration Files for Phones” on page 245.

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Additional References

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME documentation roadmap
Cisco Unified Contact Center Express (Unified CCX) • Cisco Unified Contact Center Express documentation road map
Cisco Customer Response Solutions (CRS) • Cisco CRS Installation Guide
• Cisco CRS Administration Guide
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for Interoperability with External Services

Feature Information for Interoperability with External

Services
Table 56 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified Communications Manager Express and Cisco IOS Software Version Compatibility Matrix
at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 56 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 56 Feature Information for Interoperability Feature

Cisco Unified CME

Feature Name Version Modification
Interoperability with Cisco Unified 4.2 Enables interoperability between Cisco Unified CME and
Contact Center Express (Cisco UCCX) Cisco Customer Response Solutions (CRS) 5.0 and later
versions with Cisco Unified Contact Center Express
(Unified CCX), including Cisco Unified IP IVR, enhanced
call processing, device and call monitoring, and unattended
call transfers to multiple call center agents and basic
extension mobility.

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Feature Information for Interoperability with External Services

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 Configuring SRST Fallback Support

Last Updated: March 26, 2007

This chapter describes SRST fallback support using Cisco Unified Communications Manager Express
(Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for SRST Fallback
Support” section on page 957.

Contents
• Prerequisites for SRST Fallback Support, page 943
• Restrictions for SRST Fallback Support, page 944
• Information About SRST Fallback Support, page 944
• How to Configure SRST Fallback Support, page 948
• Configuration Examples for SRST Fallback Support, page 953
• Additional References, page 955
• Feature Information for SRST Fallback Support, page 957

Prerequisites for SRST Fallback Support

• The IP address of the Cisco Unified CME router must be registered as the SRST reference on the
Cisco Unified Communications Manager device pool.
• Cisco Unified CME 4.0 or a later version must be installed on the Cisco Unified CME router that is
configured in SRST mode.
• Tasks in “Generating Configuration Files for Phones” on page 245 must be completed.
• Tasks in “Configuring System-Level Parameters” on page 119 must be completed.
• Tasks in “Generating Configuration Files for Phones” on page 245 must be completed.
• Tasks in “Configuring Call Transfer and Forwarding” on page 499 must be completed.

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Restrictions for SRST Fallback Support

Restrictions for SRST Fallback Support

• The call-manager-fallback command, which is used to configure Cisco Unified SRST, cannot be
used on a router that is configured for Cisco Unified CME.
• The number of phones that fall back to a Cisco Unified CME router in SRST mode cannot exceed
the maximum number of phones that is supported by the router chassis. To find the maximum
number of phones for a particular router and Cisco Unified CME version, see the appropriate
Cisco CME Supported Firmware, Platforms, Memory, and Voice Products document at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap0918
6a0080189132.html.
• The ephone-dns and ephones that are created from fallback may have less information associated
with them than appears in their original configuration on a Cisco Unified Communications Manager
or on an active Cisco Unified CME system. This situation occurs because the Cisco Unified CME
router in SRST mode is designed to learn only a limited amount of information from the fallback IP
phones. For example, if an ephone-dn has in its configuration the command number 4888 no-reg
(to keep that extension from registering under its E.164 address), after fallback the no-reg part of
this command will be lost because this information cannot be learned from the IP phones.
• The order of the SRST fallback ephone-dns and ephones will be different from the order of the active
Cisco Unified Communications Manager or Cisco Unified CME ephone-dns and ephones. For
example, ephone 1 on an active Cisco Unified Communications Manager might be numbered
ephone 5 on the Cisco Unified CME router in SRST mode, because the order of learned ephone-dns
and ephones is determined by the sequence of the ephone fallback occurrence, which is purely
random.

Information About SRST Fallback Support

To configure SRST fallback support, you should understand the following concepts:
• SRST Fallback Support Using Cisco Unified CME, page 944
• Prebuilding Cisco Unified CME Phone Configurations, page 948

SRST Fallback Support Using Cisco Unified CME

This feature enables routers to provide call-handling support for Cisco Unified IP phones if they lose
connection to remote primary, secondary, or tertiary Cisco Unified Communications Manager
installations or if the WAN connection is down. When Cisco Unified SRST functionality is provided by
Cisco Unified CME, provisioning of phones is automatic and most Cisco Unified CME features are
available to the phones during periods of fallback, including hunt-groups, call park and access to
Cisco Unity voice messaging services using SCCP protocol. The benefit is that
Cisco Unified Communications Manager users will gain access to more features during fallback without
any additional licensing costs.
This feature offers a limited telephony feature set during fallback mode. Customers who require the
following features should continue to use Cisco Unified SRST, because these features are not supported
with SRST fallback support using Cisco Unified CME.
• More than 240 phones during fallback service
• Cisco VG 248 Analog Phone Gateway support
• Secure voice fallback during SRST fallback service

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Information About SRST Fallback Support

• Simple, one-time configuration for SRST fallback service

Cisco Unified Communications Manager supports Cisco Unified IP phones at remote sites attached to
Cisco Integrated Services Routers across the WAN. This new feature combines the many features
available in Cisco Unified CME with the ability to automatically detect IP phone configurations that is
available in Cisco Unified SRST to provide seamless call handling when communication with the
Cisco Unified Communications Manager is interrupted.
When the system automatically detects a failure, Cisco Unified SRST uses Simple Network Auto
Provisioning (SNAP) technology to auto-configure a branch office router to provide call processing for
the Cisco Unified IP phones that are registered with the router. When the WAN link or connection to the
primary Cisco Unified Communications Manager is restored, call handling returns to the primary
Cisco Unified Communications Manager.
A limited number of phone features are automatically detected at the time that call processing falls back
to the Cisco Unified CME router in SRST mode, and an advantage of SRST fallback support using
Cisco Unified CME is that you can choose to prebuild a Cisco Unified CME configuration that contains
a number of extensions (ephone-dns) with additional features that you want them to have for some or all
of your extensions. The configurations will contain ephone-dn configurations but will not identify which
phones (which MAC addresses) will be associated with which ephone-dns (extension numbers).
By copying and pasting a prebuilt configuration onto Cisco Unified CME routers at several locations,
you can use the same overall configuration for sites that are identically laid out. For example, if you have
a number of retail stores, each with five to ten checkout registers, you can use the same overall
configuration in each store. You might use a range of extensions from 1101 to 1110. Stores with fewer
than ten registers will simply not use some of the ephone-dn entries you provide in the configuration.
Stores with more extensions than you have prebuilt will use the auto-provisioning feature to populate
their extra phones. The only configuration variations from store to store will be the specific MAC
addresses of the individual phones, which are added to the configurations at the time of fallback.
When a phone registers for SRST service with a Cisco Unified CME router and the router discovers that
the phone was configured with a specific extension number, the router searches for an existing prebuilt
ephone-dn with that extension number and then assigns that ephone-dn number to the phone. If there is
no prebuilt ephone-dn with that extension number, the Cisco Unified CME system automatically creates
one. In this way, extensions without prebuilt configurations are automatically populated with extension
numbers and features as the numbers and features are “learned” by the Cisco Unified CME router in
SRST mode when the phone registers to the router after a WAN link fails.
The SRST fallback support using Cisco Unified CME feature is able to interrogate phones to learn their
MAC addresses and the extension-to-ephone relationships associated with each phone. This information
is used to dynamically create and execute the Cisco Unified CME button command for each phone and
automatically provision each phone with the extensions and features you want it to have.
The following sequence describes how Cisco Unified CME provides SRST services for
Cisco Unified Communications Manager phones when they lose connectivity with the
Cisco Unified Communications Manager and fall back to the Cisco Unified CME router in SRST mode:

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Information About SRST Fallback Support

Before Fallback
1. Phones are configured as usual in Cisco Unified Communications Manager.
2. The IP address of the Cisco Unified CME router is registered as the SRST reference on the
Cisco Unified Communications Manager device pool.
3. SRST mode is enabled on the Cisco Unified CME router.
4. (Optional) Ephone-dns and features are prebuilt on the Cisco Unified CME router.

During Fallback
5. Phones that are enabled for fallback register to the default Cisco Unified CME router that has SRST
mode enabled. Each display-enabled IP phone displays the message that has been defined using the
system message command under telephony-service configuration mode. By default, this message is
“Cisco Unified CME.”
6. While the fallback phones are registering, the router in SRST mode initiates an interrogation of the
phones in order to learn their phone and extension configurations. The following information is
acquired or “learned” by the router:
– MAC address
– Number of lines or buttons
– Ephone-dn-to-button relationship
– Speed-dial numbers
7. The option defined with the srst mode auto-provision command determines whether
Cisco Unified CME adds the learned phone and extension information to its running configuration.
If the information is added, it appears in the output when you use the show running-config
command and is saved to NVRAM when you use the write command.
– Use the srst mode auto-provision none command to enable the Cisco Unified CME router to
provide SRST fallback services for Cisco Unified Communications Manager.
– If you use the srst mode auto-provision dn or srst mode auto-provision all commands, the
Cisco Unified CME router includes the phone configuration it learns from
Cisco Unified Communications Manager in its running configuration. If you then save the
configuration, the fallback phones are treated as locally configured phones on the
Cisco Unified CME-SRST router which could adversely impact the fallback behavior of those
phones.
8. While in fallback mode, Cisco Unified IP phones periodically attempt to reestablish a connection
with Cisco Unified Communications Manager every 120 seconds (default). To manually reestablish
a connection to Cisco Unified Communications Manager you can reboot the Cisco Unified IP
phone.
9. When a connection is reestablished with Cisco Unified Communications Manager, Cisco Unified IP
phones automatically cancel their registration with the Cisco Unified CME router in SRST mode.
However, if a WAN link is unstable, Cisco Unified IP phones can bounce between
Cisco Unified Communications Manager and the Cisco Unified CME router in SRST mode.
An IP phone connected to the Cisco Unified CME-SRST router over a WAN reconnects itself to
Cisco Unified Communications Manager as soon as it can establish a connection to
Cisco Unified Communications Manager over the WAN link. However, if the WAN link is unstable,
the IP phone switches back and forth between Cisco Unified CME-SRST and
Cisco Unified Communications Manager, causing temporary loss of phone service (no dial tone).
These reconnect attempts, known as WAN link flapping issues, continue until the IP phone
successfully reconnects itself back to Cisco Unified Communications Manager.

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Information About SRST Fallback Support

WAN link disruptions can be classified into two types: infrequent random outages that occur on an
otherwise stable WAN, and sporadic, frequent disruptions that last a few minutes.
To resolve WAN-link flapping issues between Cisco Unified Communications Manager and SRST,
Cisco Unified Communications Manager provides an enterprise parameter and a setting in the
Device Pool Configuration window called Connection Monitor Duration. (Depending on system
requirements, the administrator decides which parameter to use.) The value of the parameter is
delivered to the IP phone in the XML configuration file.
• Use the enterprise parameter to change the connection duration monitor value for all IP phones
in the Cisco Unified Communications Manager cluster. The default for the enterprise parameter
is 120 seconds.
• Use the Device Pool Configuration window to change the connection duration monitor value for
all IP phones in a specific device pool.
A Cisco Unified IP phone will not reestablish a connection with the primary
Cisco Unified Communications Manager at the central office if it is engaged in an active call.

After the First Fallback

Additional features can be set up, such as ephone hunt groups, which can contain learned extensions and
prebuilt extensions. The complete core set of Cisco Unified CME phone features is available to the IP
phones and extensions, whether they are learned or configured.
Figure 51 shows a branch office with several Cisco Unified IP phones connected to a
Cisco Unified CME router in SRST mode. The router provides connections to both a WAN link and the
PSTN. The Cisco Unified IP phones connect to their primary Cisco Unified Communications Manager
at the central office via this WAN link. Cisco Unified CME provides SRST services for the phones when
connectivity over the WAN link is interrupted.

Figure 51 SRST fallback support using Cisco Unified CME

Telephone Telephone

Fax

Cisco Unified CallManager

PSTN

V
Cisco Unified CME IP
router in SRST mode network

WAN
IP IP IP Cisco Unified IP phones disconnected
146571

PCs

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How to Configure SRST Fallback Support

Prebuilding Cisco Unified CME Phone Configurations

Prebuilding Cisco Unified CME ephone-dns allows you to create a set of directory numbers with
extension numbers and some features, which will provide service during fallback that is similar to the
service that is provided during normal operation. You can prebuild all of your normal extensions, a
limited set of your extensions, or none of your extensions. Directory numbers that are not prebuilt will
be populated with extension numbers and features as they are “learned” by the Cisco Unified CME
router in SRST mode at the time of fallback.
An ephone-dn is the IP equivalent of a normal phone line in most cases. It represents a potential call
connection and is associated with a virtual voice port and virtual dial peer. An ephone-dn has one or
more extension or telephone numbers associated with it, which allow call connections to be made. An
ephone-dn can be single-line, which allows one call connection to be made at a time, or dual-line, which
allows two simultaneous call connections. Dual-line ephone-dns are useful for features such as call
transfer or call waiting, in which one call is put on hold to connect to another. Single-line ephone-dns
are required for certain features such as intercom, paging, and message-waiting indication (MWI). For
more information, see “Cisco Unified CME Overview” on page 47.

How to Configure SRST Fallback Support

This section contains the following tasks:
• Enabling SRST Mode, page 948 (required)
• Verifying SRST Mode, page 950 (optional)
• Prebuilding Cisco Unified CME Phone Configurations, page 951 (optional)
• Modifying Call Pickup for Fallback Support, page 951 (optional)

Enabling SRST Mode

To enable SRST mode on the Cisco Unified CME router, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. srst mode auto-provision {all | dn | none}
5. srst dn line-mode {dual | single}
6. srst dn template template-tag
7. srst ephone template template-tag
8. srst ephone description string
9. end

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How to Configure SRST Fallback Support

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 srst mode auto-provision {all | dn | none} Enables SRST mode for a Cisco Unified CME router.
• all—Includes information for learned ephones and
Example: ephone-dns in the running configuration.
Router(config-telephony)# srst mode
auto-provision none
• dn—Includes information for learned ephone-dns in
the running configuration.
• none—Does not include information for learned
ephones or learned ephone-dns in the running
configuration. Use this keyword when you want
Cisco Unified CME to provide SRST fallback services
for Cisco Unified Communications Manager.
Step 5 srst dn line-mode {dual | single} (Optional) Specifies the line mode for ephone-dns in SRST
mode on a Cisco Unified CME router.
Example: • dual—SRST fallback ephone-dns will be dual-line
Router(config-telephony)# srst dn line-mode ephone-dns.
dual
• single—SRST fallback ephone-dns will be single-line
ephone-dns.
Note This command is used only when ephone-dns are
learned at the time of fallback. It is ignored when
you prebuild ephone-dn configurations.
Step 6 srst dn template template-tag (Optional) Specifies an ephone-dn template to be used in
SRST mode on a Cisco Unified CME router. The template
includes features that were specified when the template was
Example:
Router(config-telephony)# srst dn template 3
created. See “Configuring Templates for Fallback Support:
Example” on page 954.
• template-tag—Identifying number of an existing
ephone-dn template. Range is 1 to 15.
Step 7 srst ephone template template-tag (Optional) Specifies an ephone template to be used in SRST
mode on a Cisco Unified CME router.
Example: • template-tag—Identifying number of an existing
Router(config-telephony)# srst ephone ephone template. Range is 1 to 20.
template 5

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Command or Action Purpose

Step 8 srst ephone description string (Optional) Specifies a description to be associated with an
ephone learned in SRST mode on a Cisco Unified CME
router.
Example:
Router(config-telephony)# srst ephone • string—Description to be associated with an ephone.
description Cisco Unified CME SRST Fallback Maximum string length is 100 characters.
Step 9 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

Verifying SRST Mode

Step 1 Use the show telephony-service all or the show running-config command to verify that SRST mode
has been set on this router.
telephony-service
srst mode auto-provision all
srst ephone template 5
srst ephone description srst fallback auto-provision phone : Jul 07 2005 17:45:08
srst dn template 8
srst dn line-mode dual
load 7960-7940 P00305000600
max-ephones 30
max-dn 60
ip source-address 10.1.68.78 port 2000
max-redirect 20
system message "SRST Mode: Cisco Unified CME’
keepalive 10
max-conferences 8 gain -6
moh welcome.au
create cnf-files version-stamp Jan 01 2002 00:00:00

Step 2 Use the show telephony-service ephone-dn command during fallback to review ephone-dn
configurations. Learned ephone-dns are noted by a line stating that they were learned during SRST
fallback.
Note Learned ephone-dns do not appear in the output for the show running-config command if the
none keyword is used in the srst mode auto-provision command.

ephone-dn 1 dual-line
number 4008
name 4008
description 4008
preference 0 secondary 9
huntstop
no huntstop channel
call-waiting beep
ephone-dn-template 8
This DN is learned from srst fallback ephones

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Step 3 Use the show telephony-service ephone command during fallback to review ephone configurations.
Learned ephones are noted by a line stating that they were learned during SRST fallback.
Note Learned ephones do not appear in the output for the show running-config command if the none
keyword is used in the srst mode auto-provision command.

ephone 1
mac-address 0112.80B3.9C16
button 1:1
multicast-moh
ephone-template 5
Always send media packets to this router: No
Preferred codec: g711ulaw
user-locale JP
network-locale US
Description: "YOUR Description" : Oct 11 2005 09:58:27
This is a srst fallback phone

Prebuilding Cisco Unified CME Phone Configurations

You can optionally create a set of ephone-dns that are preconfigured with extension numbers and some
features to provide service during fallback that is similar to the service that is provided during normal
operation. Extensions that are not prebuilt are populated with extension numbers and features as they are
"learned" by the Cisco Unified CME router in SRST mode at the time of fallback.
See the following procedures to set up a few of the most common features to associate with phones in
fallback mode:
• “SCCP: Creating Directory Numbers” section on page 158
• “Enabling Call Park” section on page 490
• “SCCP: Enabling Ephone Templates” section on page 883
• “SCCP: Enabling Ephone-dn Templates” section on page 884
• “SCCP: Configuring Hunt Groups” section on page 596

Modifying Call Pickup for Fallback Support

An especially useful feature for fallback phones is modifying the behavior of the Pickup soft key in
Cisco Unified CME to match that of the Pickup soft key in Cisco Unified Communications Manager. To
modify the call pickup feature for fallback support, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. no service directed-pickup
5. create cnf-files

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6. reset all
7. exit

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 no service directed-pickup (Optional) Disables directed call pickup and changes the
behavior of the PickUp soft key so that a user pressing it
invokes local group pickup rather than directed call pickup.
Example:
Router(telephony)# no service directed-pickup
This behavior is consistent with that of the PickUp soft key
in Cisco Unified Communications Manager.
Note For changes to the service-phone settings to be
effective, the Sep*.conf.xml file must be updated
with the create cnf-files command and the phone
units must rebooted with the reset command.
Step 5 create cnf-files Builds XML configuration files for Cisco Unified IP
phones.
Example:
Router(telephony)# create cnf-files
Step 6 reset all Resets all phones.

Example:
Router(telephony)# reset all
Step 7 exit Exits dial-peer configuration mode.

Example:
Router(telephony)# exit

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Configuration Examples for SRST Fallback Support

This section contains the following examples:
• Enabling SRST Mode: Example, page 953
• Provisioning Directory Numbers for Fallback Support: Example, page 954
• Configuring Templates for Fallback Support: Example, page 954
• Enabling Hunt Groups for Fallback Support: Example, page 955
• Modifying Call Pickup for Fallback Support: Example, page 955

Enabling SRST Mode: Example

The following example enables SRST mode on the Cisco Unified CME router. It specifies that learned
fallback ephone-dns should be created in dual-line mode and use ephone-dn template 3 for their
configuration parameters. Learned ephones will use the parameters in ephone template 5 and a
description will be associated with the phones.
telephony-service
srst mode auto-provision all
srst dn line-mode dual
srst dn template 3
srst ephone description srst fallback auto-provision phone
srst ephone template 5
.
.
.
The following excerpt from the show running-config command displays the configuration of ephone 1,
which was learned during fallback; the description is stamped with the date and time that the show
running-config command was used. The configuration of ephone 2, which was prebuilt rather than
learned, is shown for comparison.
ephone 1
description srst fallback auto-provision phone : Jul 07 2005 17:45:08
ephone-template 5
mac-address 100A.7052.2AAE
button 1:1 2:2

ephone 2
mac-address 1002.CD64.A24A
type 7960
button 1:3

The following excerpt from the show running-config command displays the configuration of
ephone-dn 1 through ephone-dn 3. All three ephones are learned ephone-dns that are configured in
dual-line mode and use ephone-dn template 5, as specified in the telephony-service configuration mode
commands.
ephone-dn 1 dual-line
number 7001
description 7001
name 7001
ephone-dn-template 5
This DN is learned from srst fallback ephones
!
!
ephone-dn 2 dual-line
number 4005

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name 4005
ephone-dn-template 5
This DN is learned from srst fallback ephones
!
!
ephone-dn 3 dual-line
number 4002
label 4002
name 4002
ephone-dn-template 5
This DN is learned from srst fallback ephones

Provisioning Directory Numbers for Fallback Support: Example

The following example sets up five ephone-dns and two call-park slots that are used for fallback phones.
ephone-dn 1
number 1101
name Register 1

ephone-dn 2
number 1102
name Register 2

ephone-dn 3
number 1103
name Register 3

ephone-dn 4
number 1104
name Register 4

ephone-dn 5
number 1105
name Register 5

ephone-dn 21
number 1121
name Park Slot 1
park-slot timeout 60 limit 3 recall alternate 1100

ephone-dn 22
number 1122
name Park Slot 2
park-slot timeout 60 limit 3 recall alternate 1100

Configuring Templates for Fallback Support: Example

The following example creates ephone-dn template 3 and ephone template 5 that will be used with the
SRST fallback support using Cisco Unified CME feature. Ephone-dn template 3 adds the fallback
phones to pickup group 24 and specifies call forwarding for busy and no-answer conditions to extension
1100. Ephone template 5 defines two fastdial numbers that will appear as menu entries displayed from
the Directories > Local Services > Personal Speed Dials option on the fallback phones, and also specifies
the soft-key layouts for the fallback phones.
ephone-dn-template 3
pickup-group 24
call-forward busy 1100
call-forward noan 1100 timeout 45

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Additional References

ephone-template 5
fastdial 1 1101 name Front Register
fastdial 2 918005550111 Headquarters
softkeys idle Newcall Cfwdall Pickup
softkeys seized Endcall Cfwdall Pickup
softkeys alerting Endcall
softkeys connected Endcall Hold Park Trnsfer

Enabling Hunt Groups for Fallback Support: Example

The following example creates a peer hunt group with the pilot number 1111.
ephone-hunt 3 peer
pilot 1111
list 1101, 1102, 1103
hops 3
timeout 25
final 1100

Modifying Call Pickup for Fallback Support: Example

The following example changes the behavior of the Pickup soft key to be like the one in
Cisco Unified Communications Manager.
telephony-service
no service directed-pickup
create cnf-files

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

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Additional References

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for SRST Fallback Support

Feature Information for SRST Fallback Support

Table 57 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 57 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 57 Feature Information for SRST Fallback Support

Cisco Unified CME

Feature Name Version Feature Information
SRST Fallback Support Using Cisco 4.0 SRST fallback support using Cisco Unified CME was
Unified CME introduced.

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 Configuring the XML API

Last Updated: March 26, 2007

This chapter describes the eXtensible Markup Language (XML) Application Programming Interface
(API) support available in Cisco Unified Communications Manager Express (Cisco Unified CME).

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for XML API” section
on page 968.

Contents
• Information About XML API, page 959
• How to Configure XML API, page 960
• Configuration Examples for XML API, page 965
• Where to Go Next, page 966
• Additional References, page 966
• Feature Information for XML API, page 968

Information About XML API

To enable XML API, you should understand the following concepts:
• XML API Definition, page 959
• XML API Provision Using IXI, page 960

XML API Definition

An XML API provides an interface to Cisco Unified CME that allows an external network management
system (NMS) to configure and monitor Cisco Unified CME operations.

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How to Configure XML API

XML API Provision Using IXI

In previous versions of Cisco Unified CME, the XML interface provided configuration and monitoring
functions using the HTTP port. The XML interface ran under the HTTP server process, simultaneously
parsing incoming XML requests on demand and processing them.
In Cisco Unified CME 4.0 and later versions, the XML interface is provided through the Cisco IOS
XML Infrastructure (IXI), in which the parser and transport layers are separated from the application.
This modularity provides scalability and enables future XML support to be developed. In
Cisco Unified CME 4.0 and later versions, all Cisco Unified CME features have XML support.

How to Configure XML API

This section contains the following tasks:
• Defining XML Transport Parameters, page 960
• Defining XML Application Parameters, page 962
• Defining Authentication for XML Access, page 963
• Defining XML Event Table Parameters, page 964
• Troubleshooting the XML Interface, page 965

Note The following Cisco IOS commands that were previously used with the XML interface are no longer
valid: log password, xmltest, xmlschema, and xmlthread.

Defining XML Transport Parameters

To define the XML transport method and associated parameters, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. ip http server
4. ixi transport http
5. response size fragment- size
6. request outstanding number
7. request timeout seconds
8. no shutdown
9. end

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How to Configure XML API

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
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 local
Cisco Unified CME router.
Example:
Router(config)# ip http server
Step 4 ixi transport http Specifies the XML transport method and enters
XML-transport configuration mode.
Example: • http—HTTP transport.
Router(config)# ixi transport http
Step 5 response size fragment-size Sets the response buffer size.
• fragment-size—Size of fragment in the response buffer,
Example: in kilobytes. Range is constrained by the transport type
Router(conf-xml-trans)# response size 8 and platform. See the CLI help for the valid range of
values.
Step 6 request outstanding number Sets the maximum number of outstanding requests allowed
for the transport type.
Example: • number—Number of requests. Range is constrained by
Router(conf-xml-trans)# request outstanding 2 the transport type and platform. See the CLI help for the
valid range of values.
Step 7 request timeout seconds Sets the number of seconds to wait, while processing a
request, before timing out.
Example: • seconds—Number of seconds. Range is 0 to 60.
Router(conf-xml-trans)# request timeout 30
Step 8 no shutdown Enables HTTP transport.

Example:
Router(conf-xml-trans)# no shutdown
Step 9 end Returns to privileged EXEC mode.

Example:
Router(config-xml-app)# end

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How to Configure XML API

Defining XML Application Parameters

To set a response timeout for communication with the XML application that overrides the setting in
transport configuration mode, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. ixi application cme
4. response timeout {-1 | seconds}
5. no shutdown
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ixi application cme Specifies the Cisco Unified CME application and enters
XML-application configuration mode.
Example:
Router(config)# ixi application cme
Step 4 response timeout {-1 | seconds} Sets a timeout for responding to the XML application and
overwrites the IXI transport level timeout.
Example: • -1—No application-specific timeout is specified. This
Router(config-xml-app) response timeout 30 is the default.
• seconds—Length of timeout, in seconds. Range is
0 to 60.
Step 5 no shutdown Enables XML communication with the application.

Example:
Router(conf-xml-app)# no shutdown
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-xml-app)# end

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How to Configure XML API

Defining Authentication for XML Access

To authenticate users for XML access, perform the following steps:

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. xml user user-name password password privilege-level
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 xml user user-name password password Defines an authorized user.
privilege-level
• user-name—Username of the authorized user.
• password—Password to use for access.
Example:
Router(config-telephony)# xml user user23 • privilege-level—Level of access to Cisco IOS
password 3Rs92uzQ 15 commands to be granted to this user. Only the
commands with the same or a lower level can be
executed via XML. Range is 0 to 15.
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end

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How to Configure XML API

Defining XML Event Table Parameters

The XML event table is an internal buffer that stores captured and time-stamped events, such as phones
registering and unregistering and extension status. One event equals one entry in the table. To set the
maximum number of events or entries that can be stored in the XML event table and the length of time
that events are retained before they are deleted from the table, perform the following steps.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. log table max-size number
5. log table retain-timer minutes
6. end
7. show fb-its-log
8. clear telephony-service xml-event-log

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)#
Step 4 log table max-size number Sets the number of entries in the XML event table.
• number—Number of entries. Range is 0 to 1000.
Example: Default is 150.
Router(config-telephony)# log table max-size
100
Step 5 log table retain-timer minutes Sets the number of minutes to retain entries in the event
table before they are deleted.
Example: • minutes—Number of minutes. Range is 2 to 500.
Router(config-telephony)# log table Default is 15.
retain-timer 30

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Configuration Examples for XML API

Command or Action Purpose

Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-telephony)# end
Step 7 show fb-its-log Displays the event logs.

Example:
Router# show fb-its-log
Step 8 clear telephony-service xml-event-log Clears XML event logs.

Example:
Router# clear telephony-service xml-event-log

Troubleshooting the XML Interface

Step 1 Use the debug cme-xml command to view debug messages for the Cisco Unified CME XML interface.

Configuration Examples for XML API

This section contains the following examples:
• XML Transport Parameters: Example, page 965
• XML Application Parameters: Example, page 965
• XML Authentication: Example, page 966
• XML Event Table: Example, page 966

XML Transport Parameters: Example

The following example selects HTTP as the XML transport method:
ip http server
ixi transport http
response size 8
request outstanding 2
request timeout 30
no shutdown

XML Application Parameters: Example

The following example sets the application response timeout to 30 seconds.
ixi application cme
response timeout 30
no shutdown

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Where to Go Next

XML Authentication: Example

The following example selects HTTP as the XML transport method. It allows access for user23 with the
password 3Rs92uzQ, and sets up access list 99 that accepts requests from the IP address 192.168.146.72.
ixi transport http
ip http server
!
telephony-service
xml user user23 password 3Rs92uzQ 15

XML Event Table: Example

The following example sets the maximum number of entries in the XML event table to 100 and the
number of minutes to retain entries at 30:
telephony-service
log table max-size 100
log table retain-timer 30

Where to Go Next
For developer information on the XML API, see the XML Provisioning Guide for Cisco CME/SRST.

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • Quick Reference Cards
• User Guides

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Additional References

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies. Access to most tools
on the Cisco Support website requires a Cisco.com user
ID and password. If you have a valid service contract
but do not have a user ID or password, you can register
on Cisco.com.

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Feature Information for XML API

Feature Information for XML API

Table 58 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 58 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 58 Feature Information for XML API

Cisco Unified CME

Feature Name Version Feature Information
Call Blocking Based on Date and Time 4.0 The XML API was modified and is now provided through
the Cisco IOS XML infrastructure. It supports all
Cisco Unified CME features. The log password, xmltest,
xmlschema, and xmlthread commands were made
obsolete.
3.0 The XML API was introduced.

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 I N D EX

associate application sccp command 663

Numerics
associate ccm command 312, 663
7931 176 associate profile command 312
ATA (Cisco Analog Telephone Adapters) 154

A audio fallback, for video calls 870

audio file for MOH 72
abbreviated dialing 856 audio paging 785
account code entry 831 authenticate command 137
Acct soft key 830 authenticate credential command 112, 810
ad hoc conferencing 648 authenticate ood-refer command 111
after-hour exempt (voice register pool) command 478 authenticate presence command 810
after-hour exempt command 477 authentication, See phone authentication
after-hours block pattern command 474 authentication for HTTP server 341
after hours call blocking authentication string, entering on phone 430
configuring exception for dial peer 475 auth-mode command 418, 428
configuring exception for SIP phone 476, 477 auth-string command 428
after-hours date command 475 auto-answer, headset 745
after-hours day command 475 auto-answer command 758
after-hours toll bar 467 auto assign command 229
agent availability, ephone hunt groups 575 auto-cut-through command 777
agent status control, hunt groups 577 auto-line command 463
alerting (call stage) 830 auto logout command 578
allow-connections command 96, 532 automatic agent status not-ready, ephone hunt groups 578
allow-connections sip-to-sip command 923 automatic line selection 461
allow subscribe command 809 automatic registration blocking 129
allow watch command 803, 926, 929 auto-reg-ephone command 129
analog phones 154
Answer soft key 830
API, XML 959
B
application (voice register global) command 140 b2bua command 373
application command 222 bandwidth, for video 877
archives, downloading for Cisco Unified CME 70 bind interface command 661
associate application command 314 bit-rate, setting maximum value 876

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 Index

blf-speed-dial command 805, 807 call-forward b2bua noan command 367, 547, 713
BLF status 798 call-forward b2bua unreachable command 548
blocking call-forward busy command 525
automatic registration 129 call-forward busy command, overlaid ephone-dns 617
caller ID 467 call forwarding
call park 489 blocking local extension forwards 525
calls 467 H.450.3 standard 503
call transfer 503 selective 500
features 832 transcoding between G.726 and G.711 304
local directory 689 call-forward max-length command 525
bulk command 124 call-forward night-service command 525
bulk-loading speed-dial numbers 858 call-forward noan command 525
bulk registration call-forward noan command, overlaid ephone-dns 617
configuring 123 call-forward pattern command 524
disabling SIP proxy registration 172 call-forward system command 273
bulk-speed-dial list command 859 call hunt 565
bulk-speed-dial-prefix command 859 calling-number local command 534
busy call forwarding 500 callmon command 923
button command 188 call park
assigning dns to phones 161 alternate target 487
overlaid ephone-dns 618 blocking 489
button-layout command 898 call-park slots 486
dedicated slots 488
directed 486
C
examples 495
CA (certification authority) 392 monitoring call-park slots 486
cadence command 659 redirect 489
Callback soft key 830 reminder ring 487
call blocking timeout interval 487
based on date and time 467 call-park system redirect command 493
override 468 call pickup
call-coverage features 563 examples 621
caller-id block command 641 group numbers 566, 591
caller ID blocking 640 call routing, loopback 763
caller-id command 766 call setup, video 871
call-forward all command 525 call start slow command 872
call-forward b2bua all command 547 call transfer 550
call-forward b2bua busy command 366, 547 blocking 503
call-forward b2bua mailbox command 366, 547 consult transfer for direct station select 502

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direct station select 502 configuring voice hunt groups 605

H.450.2 standard 503 configuring voice mailbox 365
H.450 standards 502 creating and applying templates 886
transcoding between G.726 and G.711 304 creating DNs 162
call type 870 disabling SIP proxy registration 172
call waiting generating configuration profiles 250
beep 568 RFC 2833 dual tone DTMF MTP passthrough 357
overlaid ephone-dns 583 SIP MWI - QSIG translation 359
ring 569 Cisco Unified Communications Manager
call-waiting 596 interworking with Cisco Unified CME 540, 544
call-waiting beep command 594 network scenario 516, 517
call-waiting ring command 569, 594 no support for H.450 standards 504
CAPF (certificate authority proxy function) 393 Cisco Unified IP Phone 7931G 176
capf-auth-str command 417 Cisco Unified Video Advantage 867
capf-server command 427 Cisco Unity
cert-enroll-trustpoint command 427 configuring SIP dial peer for 369
certificate 391 integration 356
certificate authority proxy function, See CAPF Cisco Unity Connection integration 356
certificate revocation list, See CRL Cisco Unity Express
certificate trust list, See CTL configuring SIP dial peer for 372
certification authority, See CA integration with Cisco Unified CME 356
cert-oper (CAPF-server) command 429 transcoding between G.726 and G.711 304
cert-oper (ephone) command 418 Cisco VG 224 154
CFwdALL soft key 830 class of restriction (COR) 469
channel huntstop 566 clear telephony-service xml-event-log command 965
CIF (common intermediate format) 868 clid strip command 642
Cisco BTS Softswitch (Cisco BTS) 504, 505, 508, 510 clid strip name command 642
Cisco phone firmware client identifier command 101
troubleshooting upgrade 88 cli write command 234
upgrading SIP 77 clock summer-time command 104
verifying firmware version on an IP phone 88 clock timezone command 104
Cisco PSTN Gateway (Cisco PGW) 504, 508, 510 cnf-file command 130
Cisco Unified CME for SIP phones cnf-file location command 130
applying translation rules 281 cnf-file perphone command 416
configuring bulk registration 123 codec (dspfarm-profile) command 314
configuring call forwarding 546 codec command 175, 197, 225, 373, 662, 930
configuring call transfer 550 codec g729r8 dspfarm-assist command 156, 305
configuring dial-plan patterns 272 codec preference command 434
configuring hunt stop 589 common intermediate format, see CIF 868

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conference add-mode command 668 customer administrator GUI access setup

conference ad-hoc command 667 using the CLI 347
conference admin command 669 using the GUI 346
conference drop-mode command 668 customer administrator GUI file 345
conference hardware command 665
conference-join custom-cptone command 662
D
conference-leave custom-cptone command 662
conference meetme command 667 database level command 322, 404
conferencing 647 database url command 322, 404
end-of-conference options 649 date-format command 127, 138
examples 671 debug cch323 video command 878
gain control 649 debug dspfarm command 332
initiator dropoff control 649 debug ephone detail command 878
transcoding between G.729 and G.711 for ad hoc debug ephone message command 878
calls 304 debug ephone mtp command 332
configuration files debug ephone register command 878
externally stored 120, 246 debug ephone video command 878
GUI 345
debug h225asn1 command 878
per phone 120, 246 debug h245 asn1 command 878
per phone type 120, 246
debug sccp command 332
Confrn soft key 830 debug tftp event command 88
connected (call stage) 830
debug voip ccapi inout command 878
connection plar opx command 179 dedicated call-park slots 488
COR (class of restriction) 469
default-router command 99, 101
cor command 472 description (ephone-hunt) command 601
corlist command 470
description command 370, 372, 899, 900
CPU consumption, for video calls 869 destination-pattern command 225, 371, 372
create cnf-files command 77, 87
device-security-mode command 415, 417
create profile command 79, 83, 172, 251, 808 DHCP server
credentials command 425
changing TFTP address 109
CRL (certificate revocation list) 392 option 150 and configuration files 109
crypto pki authenticate command 326, 327, 411, 413, 437
DHCP service
crypto pki enroll command 326, 411 description 92
crypto pki server command 321, 403, 405, 408
setting up 98
crypto pki trustpoint command 323, 324, 327, 405, 407, 410, dialing plans 267
413, 436
dial-peer hunt command 581
CTL (certificate trust list) 392
dial-peer hunt for ephone-dn dial peers
ctl-client command 420, 423
dial-peer preference 565
ctl-service admin command 425
dial peers

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ephone-dns 149 DTMF integration patterns for voice mail 367

for call transfer and forwarding 512 DTMF relay
dial-peer voice command 370, 372 for H.323 networks 93
dialplan command 168 SIP NOTIFY 372
dialplan-pattern command 272, 273 SIP trunks 106
dial plans for SIP phones 157 dtmf-relay command 165, 930
dial tone, secondary 270 dtmf-relay h245-alphanumeric command 106, 225
DID (Direct Inward Dialing) 269 dtmf-relay rtp-nte command 107, 371
digital certificate 391 dtmf-relay sip-notify command 107, 373
digital certificates 391 dual-line ephone-dn 150
digital signature 391 dual-number ephone-dns 152
digit collect kpml command 170 dualtone conference command 659
directed call park 486 dynamic membership, hunt groups 577
directed call pickup, See call pickup
directories 689
E
directory command 692
directory entry command 694 E.164 number registration
directory numbers for SIP phones 162 ephone-hunt group pilot number 600
direct station select 851 SIP 93
direct station select call transfer 502 emptycapability command 433
display-logout command 601 encryption 390
distinctive ringing 819 EndCall soft key 830
DND (do not disturb) 712 endpoint capability match 870
dnd command 713 enrollment terminal command 413
DND soft key 578, 714, 830 enrollment url command 324, 405, 407, 410
dn-webedit command 344 ephone command 161
do not disturb (DND) 712 ephone-dn command 86, 159, 693
downloading files for Cisco Unified CME 70 ephone-dn dial-peer hunt
dspfarm connection interval command 333 ephone-dns
dspfarm profile command 313 assigning to phone (button command) 161
dspfarm rtp timeout command 333 definition 49
DSP farms, usage considerations 305 dual-line 150
dspfarm transcoder maximum sessions command 316, 317 dual-number 152
DSPs (digital signal processors) 303, 306 hunt groups 569
dsp services dspfarm command 311, 315, 658 overlaid 153, 581
DSS (direct station select) service 851 secondary number 152
dst auto-adjust command 138 shared 152
dst command 138 single-line 149
DTMF integration 357 two ephone-dns with one number 151

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types 149 filename command 167

ephone-hunt command 598 files, downloading for Cisco Unified CME 70
ephone hunt groups file text command 79, 83, 251
agent availability options 575 final command 599, 608
agent status control 577 firmware files
automatic agent status not-ready 578 phone 70
dynamic membership 577 video 868
examples 622 fixed line/feature button layout 894
final number 570 Flash soft key 830, 831
hops 569 flow-around mode, video stream 871
longest-idle groups 573 flow control messages 871
peer groups 572 flow-through mode, video stream 871
sequential groups 571 forwarding, See call forwarding
ephone-hunt login command 602 forwarding calls using local hairpin routing 533
ephones forward local-calls command 525
basic configuration 160 frequency command 659
definition 49 from-ring command 601
enabling video 874 fwd-final command 600
setting video bandwidth 877 fxo hook-flash command 839
extending overlaid ephone-dn calls 585 FXS ports 154
extension assigner feature 213
extension-assigner tag-type command 227
G
extension mobility 717
external-ring command 826 G.711 conference calls 648
G.729r8 codec
remote phones 156, 305
F
transcoding 303
FAC (feature access code) 729 gain control for conferences 649
fac command 732 gatekeeper, H.323
far-end camera control, see FECC 868 not registering ephone hunt-group pilot number 600
fastdial command 856, 862 number format restrictions 268
fax relay 737 gcid command 923
feature access code (FAC) 729 generating SIP profiles 250
feature blocking 832 GPickUp soft key 830
feature buttons grant auto command 322, 405, 409
fixed 894 group call pickup 566
URL provisioning for 896 group pickup, See call pickup
features blocked command 841 GUI (graphical user interface)
FECC (far-end camera control) 868 customer administrator setup 346

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how to access 344 hairpin call routing

phone user setup 348 description 507
prerequisites 339 enabling H.323-to-H.323 connections 531
restrictions 340 network requirements 509
setting up 341 restrictions 531
system administrator setup 343 headset auto-answer 745
headset auto-answer command 748
HLog soft key 578, 830
H
hold (call stage) 830
H.225 debug messages 878 hold-alert command 596, 824
H.245 debug messages 878 Hold soft key 830
H.261 video codec 868 hookflash functionality 831
H.263 video codec 868 hops command 599, 608
H.323-to-H.323 connections, enabling 531 host command 101
H.323 video endpoints 870 HTTP path, setting 341
H.450.12 supplementary services HTTP server, enabling 341
description 507 hunt, See call hunt
H.450.2 supplementary services 503 hunt-group logout command 578, 602
description 503 hunt groups 569
enabling 519 huntstop 565, 589
handling non-H.450.2 calls 507 huntstop, channel 566
network requirements 505 huntstop channel command 587
H.450.3 supplementary services 503 huntstop channel command, overlaid ephone-dns 617
description 503 huntstop command 177, 185, 565, 590
enabling 519 huntstop command, overlaid ephone-dns 581, 617
handling non-H.450.3 calls 507
network requirements 505
I
H.450 standards
call forwarding 500 id command 164, 758
call transfer 502 idle (call stage) 830
H.450 tandem gateways incoming alerting command (redundant router) 133, 134
description 510 in key systems 188
enabling H.323-to-H.323 connections 531 input gain command 777
licensing 510 intercom command 756
h225 h245-address on-connect (h323 voice-service) intercom lines 753
command 542
configuring for SCCP phones 755
h225 h245-address on-connect (voice-class) configuring for SIP phones 757
command 543
interface command 103
h323 command 432
international languages and tones

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alternative locales 288 license requirements 49

user-defined locales 288 lifetime certificate command 404, 409
interoperability with other systems 514 line buttons
intersite calling plan 268 fixed 894
ip dhcp pool command 99, 101, 110 phone labels 895
ip helper-address command 103 line selection, automatic 461
ip http authentication command 342 list command 598, 607
ip http path flash command 342, 853 live-feed music on hold 773
ip http server command 342, 853 load (telephony service) command 77, 86
IP phone load-cfg-file command 416
configuring phone options 896 load command 79, 126, 136
programmable vendor parameters 895 local directory 689
ip source-address (credentials) command 425 locales
ip source-address command 319 multiple 288
ip source-address command (redundant router) 126, 133, system-defined 288
134
user-defined 288
issuer-name command 404
locally significant certificate, See LSC
ixi application cme command 962
local speed dial 852
ixi transport http command 961
login (telephony-service) command 475
Login soft key 830

K log table max-size command 964

log table retain-timer command 964
keepalive command 131, 927
longest-idle ephone hunt groups 573
keepalive retries command 312
loopback call routing 763
keep-conference (voice register pool) command 656
loopback-dn command 767
keep-conference command 654
LSC (locally significant certificate) 393
keygen-retry command 429
keygen-timeout command 429
keypad-normalize command 161 M
keyswitch 50, 176
mac-address command 161, 187, 232
key system 50, 176
mailbox-selection (dial-peer) command 363
KPML 157
mailbox-selection (ephone-dn) command 365
kron occurrence command 235
mailbox selection policy 362, 364
kron policy-list command 234
manufacture-installed certificate, See MIC
max-conferences command 652

L max-dn command 126, 136

max-ephones command 126
label command 902, 904
maximum bit-rate command 876
language support 287
maximum conference-party command 662

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maximum sessions command 314, 662 prefix specification for SIP 380
max-pool command 136 mwi command 378, 380
max-redirect command 602 mwi-line command 188, 374, 375
max-subscription command 801 mwi prefix command 381
max-timeout command 599 mwi reg-e164 command 377
media encryption 387 mwi-server command 379
media messages 870 mwi stutter command 377
media path setup 871 mwi-type command 178, 376
media termination point, See MTP
meet-me conferencing 650
N
messages
debug 878 name command 160, 693, 699
flow control 871 network command 99
media 870 network-locale (ephone-template) command 297
MIBs 54 network-locale command 296
MIC (manufacture-installed certificate) 393 network locales
configuring 412 alternative 288
mode command 79, 136, 808 system-defined 288
mode ra command 408 user-defined 288
MOH (music on hold) network parameters 91
audio file to download 72 NewCall soft key 830
from a live feed 773 night service 579
from an audio file 774 examples 625
transcoding between G.726 and G.711 304 notification 579
moh (telephony-service) command 775 parameters 579
moh command 779 night-service bell (ephone) command 613
monitoring call-park slots 486 night-service bell (ephone-dn) command 612
monitor-line button 850 night-service call forwarding 500
MTP (media termination point) night-service code command 612
remote phones 156 night-service date command 610, 611
transcoding for video 869 night-service day command 610, 611
mtp command 197 night-service everyday command 610, 611
multicast moh command 775 night-service weekday command 612
multicast-moh command 776, 780 night-service weekend command 612
multi-party ad hoc conferencing 649 no-answer call forwarding 500
MWI no ephone command 82
configuring Subscribe notify 378 no-reg (ephone-hunt) command 600
configuring unsolicited notify 378 no-reg command 173
defining MWI outcall 377

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no supplementary-service sip moved-temporary

P
command 923
notification, night-service 579 packet voice/data modules (PVDM) 306
notification, on-hold 820 paging 785
notify telephone-event command 97, 107 paging command 788
not-ready, automatic, ephone hunt groups 578 paging-dn command 790
no upgrade command 80 paging group command 790
no vad command 373 parallel voice hunt groups 574
no voice register pool command 85 param ea-password command 222
NTP (Network Time Protocol) 93, 120 paramspace callsetup after-hours-exempt command 476
ntp server command 104 paramspace command 223
ntp-server command 139 parking calls 485, 486
number (voice register dn) command 163, 589, 758 park-slot command 492
number (voice register pool) command 164, 758 Park soft key 830
number command 159 pattern command 167
number plan 268 pattern direct command 368
pattern ext-to-ext busy command 369
pattern ext-to-ext no-answer command 369
O
pattern trunk-to-ext busy command 369
one-quarter common intermediate format, see QCIF 868 pattern trunk-to-ext no-answer command 369
on-hold notification 820 PBX model 269
OOD-R (out-of-dialog refer) 94 PBX system 50, 158
open logical channel (OLC) 871 peer ephone hunt groups 572
operation command 778 per-phone configuration files 120, 246
option 150 and configuration files, changing TFTP personal speed dial 855
address 109
phone authentication 387
option 150 ip command 99, 101, 110
authentication string entering on phone 430
overlaid ephone-dns 153, 581
CA configuration 402
call waiting 583
CAPF server configuration 426
definition 153
certificates 391
examples 626
Cisco Unified CME server function
huntstop 581 configuration 409
preference 581 configuration tasks 402
restrictions 615 CTL client and CTL provider configuration 419, 420
rollover to another button 585 examples 438
override for after-hours toll bar 467 MIC importing 412
PKI 391
RA configuration 406
telephony-service security configuration 414

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telephony-service security parameters 414 private lines to Public Switched Telephone Network
(PSTN) 154
phone-key-size command 428
phone labels 895
profile-identifier register command 664
provision-tag command 232
phone number plan 268
phone-redirect-limit command 550
phones Q
analog 154
QCIF (one-quarter common intermediate format) 868
basic configuration 147
qsig decode command 536
configuration files 245
remote teleworker 155
phone screen R
custom background images 894
RA (registration authority) 392
header bar display 894
ready/not-ready status, hunt groups 577
system message display 895
Real-Time Transport Protocol, See RTP
phone-specific parameters for individual SIP phones 163
rebooting phones 257
phone user GUI access setup 348
Redial soft key 830
using CLI 349
refer-ood enable command 111
using GUI 348
refer target dial-peer command 929
pickup, See call pickup
regenerate command 422, 424
pickup-group command 592
register id command 927
PickUp soft key 830
register support, SIP 93
pilot command 598, 607
registrar command 97, 107, 124
PIN (personal identification number) 468
registrar server command 97, 924
pin command 477
registration, blocking automatic 129
PKI (Public Key Infrastructure) 391
registration, video-enabled endpoints 868
policy-list command 235
registration authority, See RA
port (CAPF-server) command 427
relay, DTMF 93
preference (ephone-dn) command 565, 587
reminder, call-park 487
preference (ephone-hunt) command 600
reminder, on-hold 820
preference (voice hunt group) command 608
remote phones 155
preference, dial-peer 565
request outstanding command 961
preference command 177, 589
request timeout command 961
preference command, overlaid ephone-dns 581, 616
reset (voice register global) command 262, 264
presence call-list command 801, 805, 807
reset (voice register pool) command 80
presence command 801, 809
resetting all SIP phones 261
presence enable command 801
resetting phones
presence service 797
description 257
present-call command 601
reset (ephone) command 259, 261

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resolution, for video 868 sdspfarm tag command 320, 665

response size command 961 sdspfarm transcode sessions command 319
response timeout command 962 sdspfarm units command 665
restart command 808 secondary dial tone 270
restarting phones secondary-dialtone command 284
description 257 secondary numbers 152
restart (ephone) command 260 secondary start command 601
retry register command 97, 107 secure real-time transport protocol 387
revocation-check command 323, 325, 408, 411, 413, 436 secure-signaling trustpoint command 415
RFC 2833 DTMF MTP passthrough 357 secure transcoding 306, 400
ring command 822 security, See phone authentication
ringing, distinctive 819 seized (call stage) 830
ringing call stage 830 selective call forwarding 500
ring number command (redundant router) 135 sequential ephone hunt groups 571
rollover button for overlaid ephone-dns 585 serial-number command 408
routing, loopback 763 server (CTL-client) command 421
RSA key pair 392 server cme command 421, 423
rsakeypair command 323, 325, 408, 411 server command 809
RTP (Real-Time Transport Protocol) 871 server-security-mode command 416
RTP packets 872 service command 222, 224
rule command 276 service dhcp command 103
service directed-pickup command 591
service dss command 854
S
service local-directory command 692
SAST (system administrator security token) service phone command 911, 912
sast1 trustpoint command 420, 423 service phone videoCapability command 873
sast2 trustpoint command 421, 423 Session Initiation Protocol, See SIP
SCCP (Skinny Client Control Protocol) session protocol command 373
analog phones 154 session-server command 925, 930
firmware 70 session target command 225, 371, 373
security 387 session transport protocol 158
sccp blf-speed-dial retry-interval command 809 shared ephone-dns 152
sccp ccm command 315, 660 shared-line overlays 582
sccp ccm group command 312, 660, 663 shared lines 268
sccp command 661 show call active video command 878
sccp ip precedence command 333 show call prompt-mem-usage command 545
sccp local command 315, 660 show capf-server auth-string command 419
SCCP video endpoints 870 show crypto pki certificates command 409, 412
sdspfarm conference mute-on command 665 show crypto pki server command 409

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show ctl-client command 424 SIP phone firmware, upgrading 77

show dspfarm sessions active command 332 SIP-to-SIP call forwarding 546
show dspfarm sessions command 330 sip-ua command 111, 124, 379, 800
show dspfarm sessions summary command 331 slots, call-park 486
show ephone-dn park command 495 slow start, for video 872
show ephone offhook command 878 soft key
show ephone phone-load command 88, 868 display 829
show ephone registered command 867, 874, 878 DND 578
show fb-its-log command 965 feature blocking 829
show rtp connections command 878 HLog 578
show running-config command 877 softkeys alerting command 833
show sccp command 330 softkeys commands 829
show sccp connections command 330 softkeys connected (voice register template)
command 836
show sccp connections details command 332
softkeys connected command 669, 833
show sdspfarm command 330
softkeys hold command 669, 834, 836
show sdspfarm sessions command 308
softkeys idel (voice register template) command 836
show sdspfarm units command 308
softkeys idle command 669, 834
show sip-ua register status command 108, 109
softkeys ringing command 834
show sip-ua statistics command 108, 109
softkeys seized (voice register template) command 836
show sip-ua status command 108
softkeys seized command 669, 834
show sip-ua timers command 108, 109
software, downloading for Cisco Unified CME 70
show telephony-service bulk-speed-dial command 860
source-addr command 427
show telephony-service security-info command 419
source-address command 136
show telephony-service tftp-bindings command 249
speed dial 847, 860
show voice dsp command 308
bulk loading 858
show voice register profile text command 252
buttons on phones 856
show voice register tftp-bind command 252
local 852
show voip rtp connections command 545, 872
monitor-line button 850
signal immediate command 778
personal 855
signal loop-start live-feed command 778
speeddial.xml file 848
signature 391
speed-dial command 857, 861
single in-line memory module (SIMM) sockets 306
spfarm profile command 662
single-line ephone-dn 149
squeeze command 129, 247
SIP (Session Initiation Protocol)
srst dn line-mode command 949
endpoints 869
srst dn template command 949
register support 93
srst ephone description command 950
sip command 96, 923
srst ephone template command 949
SIP dial plans 157
srst mode auto-provision command 949
SIP MWI - QSIG translation 359

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SRTP 387 templates, creating and applying 886

srtp fallback command 432, 434 tftp-path command 136
supplementary-service h225-notify cid-update (dial-peer) TFTP server
command 543
changing address 109
supplementary-service h225-notify cid-update storing configuration files 120, 245
(voice-service) command 542
tftp-server-credentials trustpoint command 415
supplementary-service h450.12 (dial-peer) command 531
tftp-server flash command 126
supplementary-service h450.12 (voice-service)
command 530 three-party ad hoc conferencing 648

supplementary-service h450.2 (dial-peer) command 522 three-party conferencing, See conferencing

supplementary-service h450.2 (voice-service) time-format command 127, 138
command 522 timeout command 599, 608
supplementary-service h450.3 (dial-peer) command 523 timeouts busy command 131
supplementary-service h450.3 (voice-service) timeouts interdigit (telephony-service) command 131
command 522, 534
timers register command 98, 108
supplementary-service h450.7 command 536, 538
time-webedit command 344
supplementary-service media-renegotiate command 432
time-zone command 128
supplementary-service qsig call-forward command 536,
538 timezone command 138

supplementary-service sip command 539 TLS (transport layer security) 393

switchback interval command 313 toll bar, after-hours 467

switchback method command 313 toll bar override 467

switchover method command 313 transcoding support 303

system administrator GUI access setup 343 transfer, See call transfer
system administrator security token, See SAST transfer max-length command 503, 528

system-defined locales 288 transfer-mode command 526

system message command 905 transfer-park blocked command 493

transfer-pattern blocked command 503, 527
transfer-pattern command 521
T transfer-system command 520

TACACS authentication for HTTP server 341 translate command 276, 279

TAPI (Telephony Application Programming Interface) translate-outgoing command 282

software 73 translation-profile command 278
telephony-service ccm-compatible (h323 voice-service) translation-profile incoming command 280
command 542
transport layer security, See TLS
telephony-service ccm-compatible (voice-class)
command 543 Trnsfer soft key 830

telephony service command 76, 86 troubleshooting Cisco phone firmware upgrade 88

telephony-service security parameters 414 troubleshooting tips for upgrading, downgrading,

converting phone firmware 88
teleworker remote phones 155
trunk command 181, 185
template command 552, 887
trustpoint 392

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trustpoint (credentials) command 425 troubleshooting 878

trustpoint-label command 427 video (config-ephone) command 875
TSP (TAPI Service Provider) software 73 video (config-telephony) command 876
type command 161, 164, 167, 188, 758 virtual voice port 149
vm-integration command 368
voice call send-alert coomand 923
U
voice-card command 658
Unified CCX, interoperability 920 voice class codec command 434
upgrade command 79, 83 voice-class codec command 434
upgrading SIP firmware 77 voice class custom-cptone command 659
url command 695, 700, 908, 909 voice hunt-group command 607
url idle command 905 voice hunt groups 605
user-defined locales 288 voice mail
user GUI access setup 348 DTMF integration patterns 367
user-locale (ephone-template) command 297 integration 355
user-locale command 296 mailbox selection policy 362, 364
user locales MWI prefix specification for SIP 380
alternative 288 voicemail (voice register global) command 361, 366
system-defined 288 voice port, virtual 149
user-defined 288 voice register dialplan command 166
username command 349 voice register dn command 163, 366, 699, 757
voice register global command 79, 82, 136, 138, 140, 550, 700
voice register pool command 79, 164, 472, 758
V
voice-register session-server command 927
vad command 225 voice register template command 551, 887
vendorConfig parameters 895 voice service voip command 923
verifying firmware version on an IP phone 88 voice translation-profile command 276
verifying SIP profiles 251 voice translation-rule command 275
video VoIP-to-VoIP connections 514
call setup 871 configuring 531
codec selection process 870 H.450 tandem gateways 510
codecs supported 868 hairpin call routing 507
enabling 873
firmware version 868
W
formats supported 868
icon 869 watcher all command 809
prerequisites 872 web admin customer command 348
restrictions 872 web admin system command 343
setting parameters 876 web customize load command 346

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OL-10663-02 983
 Index

XML application programming interface 959

XML files
downloading xml.template 72
GUI configuration file 345
speeddial.xml for system speed dial 848
xml user command 963

Cisco Unified Communications Manager Express System Administrator Guide

984 OL-10663-02