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The Playback() application is used for playing a previously recorded sound file over a
channel. Input from the user is ignored, which means that you would not use Play
back() in an auto attendant, for example, unless you did not want to accept input at
that point.
‖
Asterisk comes with many professionally recorded sound files, which
should be found in the default sounds directory (usually /var/lib/aster-
isk/sounds/). When you compile Asterisk, you can choose to install
various sets of sample sounds that have been recorded in a variety of
languages and file formats. We’ll be using these files in many of our
examples. Several of the files in our examples come from the Extra
Sound Package, so please take the time to install it (see Chapter 3). You
can also have your own sound prompts recorded in the same voices as
the stock prompts by visiting http://www.theivrvoice.com/. Later in the
book we’ll talk more about how you can use a telephone and the dialplan
to create and manage your own system recordings.
To use Playback(), specify a filename (without a file extension) as the argument. For
example, Playback(filename) would play the sound file called filename.wav, assuming
it was located in the default sounds directory. Note that you can include the full path
to the file if you want, like this:
Playback(/home/john/sounds/filename)
The previous example would play filename.wav from the /home/john/sounds/ directory.
You can also use relative paths from the Asterisk sounds directory, as follows:
Playback(custom/filename)
This example would play filename.wav from the custom/ subdirectory of the default
sounds directory (probably /var/lib/asterisk/sounds/custom/filename.wav). Note that if
the specified directory contains more than one file with that filename but with different
file extensions, Asterisk automatically plays the best file.
#
The Hangup() application does exactly as its name implies: it hangs up the active chan-
nel. You should use this application at the end of a context when you want to end the
current call, to ensure that callers don’t continue on in the dialplan in a way you might
not have anticipated. The Hangup() application does not require any arguments, but
you can pass an ISDN cause code if you want (e.g., Hangup(16)).
‖ There is another application called Background() that is very similar to Playback(), except that it does allow
input from the caller. You can read more about this application in Chapter 15 and Chapter 17.
#Asterisk selects the best file based on translation cost—that is, it selects the file that is the least CPU-intensive
to convert to its native audio format. When you start Asterisk, it calculates the translation costs between the
different audio formats (they often vary from system to system). You can see these translation costs by typing
show translation at the Asterisk command-line interface. The numbers shown represent how many
milliseconds it takes Asterisk to transcode one second of audio. We’ll talk more about the different audio
formats (known as codecs) in “Codecs” on page 625.
114 | Chapter 6: Dialplan Basics
- THIRD EDITION 5
- Table of Contents 7
- Table of Contents 9
- Table of Contents 11
- Table of Contents 13
- Table of Contents 15
- Table of Contents 17
- Table of Contents 19
- Foreword 21
- Foreword 23
- Audience 25
- Organization 26
- Preface 27
- Software 28
- Conventions Used in This Book 28
- Acknowledgments 30
- Leif Madsen 32
- Jim Van Meggelen 33
- Russell Bryant 34
- Preface 35
- A Telephony Revolution 37
- Network Telephony 38
- Asterisk: The Hacker’s PBX 40
- The Asterisk Community 41
- The Asterisk Mailing Lists 42
- Asterisk Wiki Sites 43
- The IRC Channels 43
- Asterisk User Groups 43
- The Business Case 44
- Asterisk Architecture 45
- Modules 47
- Bridging Modules 51
- Call Detail Recording Modules 51
- Channel Event Logging Modules 52
- Channel Drivers 53
- Codec Translators 54
- Format Interpreters 54
- Addon Modules 59
- File Structure 60
- Hardware 62
- Asterisk Versioning 62
- The New Release Methodology 63
- Installing Asterisk 65
- Asterisk Packages 66
- Asterisk-Based Projects 66
- Installation Cheat Sheet 67
- Installation Cheat Sheet 69
- Distribution Installation 71
- Base system update 74
- Adding a system user 75
- Ubuntu Server 76
- Software Dependencies 80
- Downloading What You Need 82
- Getting the Source via wget 83
- How to Install It 84
- DAHDI-tools and DAHDI-linux 85
- Setting File Permissions 86
- Base Configuration 87
- Initial Configuration 88
- $PATH environment variable 90
- The Asterisk Shell Command 91
- Base Configuration 93
- Uses for menuselect 95
- Using menuselect 97
- Module Dependencies 98
- Scripting menuselect 99
- Updating Asterisk 100
- You don’t need to run sudo 101
- Updating Asterisk 101
- Common Issues 102
- Upgrading Asterisk 104
- Conclusion 105
- Initial Configuration Tasks 107
- The [options] Section 108
- The [files] Section 111
- The [compat] Section 111
- The [modules] Section 112
- CHANNEL(tonezone) function 114
- Getting Free Music 115
- CentOS Prerequisite 116
- User Device Configuration 119
- Telephone Naming Concepts 120
- Configuring Asterisk 123
- [general] section 124
- Configuring Asterisk 125
- IPv6 in sip.conf 128
- PBX Modules 130
- Resource Modules 130
- Base Configuration 132
- Using Code Examples 133
- Safari Books Online 133
- How to Contact Us 133
- The Asterisk CLI 135
- Analog Phones 136
- Asterisk as a B2BUA 140
- Dialplan Basics 143
- Contexts 144
- Extensions 146
- Priorities 147
- Priority labels 148
- Applications 149
- Playback(custom/filename) 150
- A Simple Dialplan 151
- Using the Dial() Application 155
- Argument 1: Destination 156
- Argument 2: Timeout 157
- Argument 3: Option 157
- Argument 4: URI 157
- Using Variables 158
- Global variables 159
- Channel variables 159
- Environment variables 160
- Pattern Matching 161
- . (period) 162
- 555-1212: 162
- The NANP and Toll Fraud 163
- Includes 165
- Outside Connectivity 167
- * But do not collect $200 168
- PSTN Circuits 169
- Traditional PSTN Trunks 170
- Digital telephony 171
- Installing PSTN Trunks 172
- Configuring digital circuits 174
- Configuring analog circuits 178
- PSTN Termination 180
- PSTN Origination 181
- VoIP to VoIP 183
- Configuring VoIP Trunks 183
- VoIP 185
- VoIP 189
- Emergency Dialing 190
- Emergency Dialing 191
- Voicemail 193
- The [general] Section 194
- 198
- The [zonemessages] Section 202
- The Contexts Section 203
- Dialplan Integration 205
- [Services] 207
- Storage Backends 208
- Voicemail Server 210
- Dialplan requirements 212
- 178 214
- Internationalization 217
- 2001/12/Glossary#I18N 218
- * key for the dots in 220
- Figure 9-3. A balun 222
- DAHDI Drivers 223
- #) to indicate a 224
- Asterisk 225
- Asterisk 227
- [zonemessages] section 228
- Asterisk 229
- CHANNEL(language) function 230
- Deeper into the Dialplan 231
- Operators 232
- Dialplan Functions 234
- Conditional Branching 235
- GotoIf() in an example: 236
- Conditional Branching 237
- Defining Macros 240
- Macros 241
- Using Arguments in Macros 242
- Defining Subroutines 243
- GoSub() 245
- Returning from a Subroutine 246
- Local Channels 247
- Dial() application, like so: 248
- Hangup() 249
- Storing Data in the AstDB 250
- Deleting Data from the AstDB 251
- BLACKLIST() function, which 252
- Handy Asterisk Features 253
- Conferencing with MeetMe() 254
- Parking and Paging 257
- The [general] section 258
- The [featuremap] Section 261
- The [applicationmap] Section 261
- *CLI> features reload 262
- Application Map Grouping 263
- Parking Lots 264
- Places to Send Your Pages 266
- Set paging 267
- VoIP paging adaptors 270
- Combination paging 270
- Zone Paging 271
- Internet Call Routing 273
- The SIP URI 274
- SRV Records 274
- Note the trailing dot 275
- Modifying sip.conf 276
- Standard dialplan 276
- DNS and SIP URIs 277
- File parsing 278
- DNS and SIP URIs 279
- Database lookup 280
- ENUM and E.164 283
- Asterisk and ENUM 285
- ISN, ITAD, and freenum.org 286
- Got ISN? 287
- Testing Your ITAD 290
- Security and Identity 292
- Toll Fraud 293
- Phishing 294
- CHAPTER 13 297
- Creating a Simple ACD Queue 298
- Queue Members 302
- Queue Members 303
- Using Pause and Unpause 304
- Queue Members 305
- Queue Members 307
- Queue Members 309
- The queues.conf File 311
- The agents.conf File 317
- Advanced Queues 319
- Queue Member Priority 320
- Advanced Queues 321
- Announcement Control 323
- Overflow 327
- Using Local Channels 329
- Advanced Queues 331
- MANAGER in the Unique ID 333
- Device States 337
- Checking Device States 338
- Extension States 339
- Checking Extension States 340
- Extension States 341
- SIP Presence 342
- Using Custom Device States 343
- An Example 344
- Distributed Device States 345
- Using OpenAIS 346
- OpenAIS configuration 347
- Asterisk configuration 348
- Testing device state changes 349
- Using XMPP 350
- Installation 351
- Creating XMPP accounts 352
- Shared Line Appearances 354
- Configuration Overview 355
- Shared Line Appearances 357
- Shared Line Appearances 359
- Shared Extension Example 361
- Additional Configuration 363
- Limitations 364
- The Automated Attendant 367
- Designing Your Auto Attendant 368
- The Greeting 369
- The Main Menu 370
- Selection # 371
- Selection 3 371
- Selection 9 371
- Selection 0 371
- Building Your Auto Attendant 372
- The Dialplan 374
- CHAPTER 16 377
- Installing MySQL for CentOS 379
- Installing MySQL for Ubuntu 379
- Configuring PostgreSQL 379
- Configuring MySQL 381
- IDENTIFIED BY 382
- PostgreSQL if all is well: 384
- Configuring ODBC for MySQL 385
- FreeTDS if all is well: 386
- Validating the ODBC Connector 387
- ODBC_STORAGE from 388
- Managing Databases 389
- IF() function there 390
- ${ARG3} 396
- Using the ARRAY() Function 398
- Using Realtime 404
- A Word About Metrics 406
- Dynamic Realtime 407
- Column name Column type 408
- Varchar 6 408
- Varchar 128 408
- [options] 411
- ODBC Voicemail 414
- Testing ODBC Voicemail 419
- ODBC Voicemail 421
- FILE permission 422
- ALL access. If you 422
- Interactive Voice Response 425
- Components of an IVR 426
- A Perfectly Tasty IVR 427
- IVR Design Considerations 428
- A Simple IVR Using CURL 429
- Text-to-Speech 431
- External Services 433
- Calendar Integration 434
- Ubuntu dependencies 435
- Calendar Integration 437
- Triggering a wakeup call 438
- Calendar Integration 441
- CALENDAR_BUSY() 443
- Calendar Integration 445
- VoiceMail IMAP Integration 447
- CentOS dependencies 448
- Compiling the IMAP library 449
- Compiling Asterisk 450
- Jabber Dialplan Commands 455
- JabberSend() 459
- Configuring gtalk.conf 461
- Skype Integration 465
- Skype Integration 467
- Skype Integration 469
- LDAP Integration 470
- Configuring OpenLDAP 471
- field, prefixed with {md5}: 472
- Configuring res_ldap.conf 473
- Configuring extconfig.conf 475
- Text-to-Speech Utilities 476
- What Is a Fax? 479
- Obtaining spandsp 480
- Digium Fax For Asterisk 482
- Incoming Fax Handling 483
- Fax Detection 484
- Outgoing Fax Handling 485
- File Format for Faxing 486
- An Experiment in Email to Fax 487
- 452 488
- Outgoing Fax Handling 489
- Fax Pass-Through 490
- Quick Start 493
- AMI over TCP 494
- AMI over HTTP 495
- Configuration 496
- [username] section 498
- Protocol Overview 501
- Message Encoding 502
- /rawman encoding 504
- /manager encoding 505
- /mxml encoding 506
- Manager events 506
- Development Frameworks 507
- Interesting Applications 508
- “Hello World”: 512
- AGI Variants 513
- DeadAGI Is Dead 514
- FastAGI—AGI over TCP 514
- Async AGI—AMI-Controlled AGI 515
- AGI Communication Overview 516
- Commands and Responses 518
- Process-based AGI/FastAGI 521
- Async AGI 521
- Ending an AGI Session 522
- Clustering 525
- Hybrid Systems 526
- Hybrid Systems 527
- Pure Asterisk, Nondistributed 528
- Single Database 529
- 494 530
- Replicated Databases 531
- 496 532
- Queue() to 533
- 498 534
- 500 536
- Distributed Universal Number 539
- Discovery (DUNDi) 539
- The dundi.conf File 541
- General Configuration 543
- Initial DUNDi Peer Definition 545
- Creating Mapping Contexts 546
- ${NUMBER} 548
- ${IPADDR} 548
- Allowing Remote Connections 550
- Using dbsecret with iax.conf 551
- Controlling Responses 552
- System Monitoring and Logging 559
- Reviewing Asterisk Logs 561
- Call Detail Records 563
- Dialplan Applications 565
- Backends 566
- [general] sec 569
- Call Detail Records 571
- Example Call Detail Records 572
- CEL (Channel Event Logging) 573
- Channel Event Contents 575
- [mytable] 577
- • userfield 578
- CELGenUserEvent() is used 580
- [radius] 581
- Example Channel Events 582
- Single-party call 583
- Two-party call 583
- Blind transfer 584
- CentOS dependency 587
- Ubuntu dependency 587
- Installing OpenNMS 588
- SNMP 589
- Enabling SNMPv3 591
- SNMP 593
- Web Interfaces 597
- Flash Operator Panel 598
- Queue Status and Reporting 598
- A2Billing 600
- Security 601
- 566 602
- Authentication Weaknesses 603
- Fail2ban 603
- 570 606
- Encrypted Media 607
- Dialplan Vulnerabilities 607
- 572 608
- IAX2 Denial of Service 609
- Other Risk Mitigation 610
- Resources 611
- Conclusion—A Better Idiot 612
- CHAPTER 27 613
- Closed Thinking 614
- Limited Standards Compliancy 614
- Slow Release Cycles 615
- Paradigm Shift 616
- Open Architecture 617
- Standards Compliance 618
- Passionate Community 618
- Low-barrier IVR 619
- Conference rooms 621
- The Future of Asterisk 622
- Speech Processing 623
- High-Fidelity Voice 624
- Wireless 625
- Unified Messaging 626
- Challenges 627
- VoIP spam 628
- Fear, uncertainty, and doubt 628
- Bottleneck engineering 628
- Regulatory wars 628
- Quality of service 629
- Opportunities 630
- The Future of Asterisk 631
- Understanding Telephony 633
- Parts of an Analog Telephone 634
- Hybrid (or network) 635
- Digital Telephony 636
- Pulse-Code Modulation 637
- 0011, and the next 638
- Figure A-6. Delineated signal 639
- Digital Telephony 639
- 11001 11000 10110 10001 640
- Digital Telephony 641
- Nyquist’s Theorem 642
- Logarithmic companding 643
- Digital Telephony 645
- Circuit Types 646
- T-carrier circuits 647
- Digital Signaling Protocols 648
- Signaling System 7 650
- Packet-Switched Networks 651
- Protocols for VoIP 653
- The Need for VoIP Protocols 654
- VoIP Protocols 655
- Security considerations 656
- IAX and NAT 656
- Figure B-1. The SIP trapezoid 657
- VoIP Protocols 657
- SIP and NAT 658
- VoIP Protocols 659
- H.323 and NAT 660
- Proprietary Protocols 661
- Codecs 663
- Quality of Service 665
- User Datagram Protocol 666
- Differentiated Service 667
- Guaranteed Service 667
- Best Effort 668
- Why Echo Occurs 668
- Hardware Echo Cancellation 669
- Asterisk and VoIP 670
- Asterisk and VoIP 671
- VoIP Security 672
- Spoofing 673
- What Can Be Done? 673
- APPENDIX C 675
- Server Hardware Selection 677
- Choosing a Processor 680
- Small systems 681
- Medium systems 681
- Choosing a Motherboard 682
- Power Supply Requirements 684
- Environment 685
- Grounding 686
- Electrical Circuits 688
- The Equipment Room 688
- Telephony Hardware 689
- Analog interface cards 690
- Digital interface cards 690
- Channel banks 691
- Echo Cancellation 692
- Types of Phones 693
- Analog telephones 694
- ISDN telephones 695
- Softphones 696
- Telephony Adaptors 697
- Communications Terminals 697
- Linux Considerations 698
- About the Authors 733
- Colophon 733
Rockwell Automation 1771-IXHR INSTL INSTR HI RES TC/MV INPUT podręczniki
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