ATech Machine LIBRA-02 M Informazioni Techniche Pagina 68
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- SEGNALIBRI
Valutato. / 5. Basato su recensioni clienti
Some additional configuration of text files is required on Ubuntu.
See “Enable NTP for accurate system time” on page 43.
3. On CentOS, add a new system user:
CentOS (32 and 64 bit) adduser asteriskpbx && passwd asteriskpbx && yum install \
sudo && visudo
See “Adding a system user” on page 39 for specific information.
For an Ubuntu install, we are assuming that the user created during
the installation process is asteriskpbx.
4. Install software dependencies:
CentOS sudo yum install gcc gcc-c++ make wget subversion \
libxml2-devel ncurses-devel openssl-devel \
vim-enhanced
CentOS 64-bit sudo yum install gcc.x86_64 gcc-c++.x86_64 \
make.x86_64 wget.x86_64 subversion.x86_64 \
libxml2-devel.x86_64 ncurses-devel.x86_64 \
openssl-devel.x86_64 vim-enhanced.x86_64
Ubuntu sudo apt-get install build-essential subversion \
libncurses5-dev libssl-dev libxml2-dev vim-nox
5. Create your directory structure:
$ mkdir -p ~/src/asterisk-complete/asterisk
$ cd ~/src/asterisk-complete/asterisk
6. Get the latest source code via Subversion:
$ svn co http://svn.asterisk.org/svn/asterisk/branches/1.8
Or alternatively, you could check out a specific tag:
$ svn co http://svn.asterisk.org/svn/asterisk/tags/1.8.1
7. Build and install the software:
$ cd ~/src/asterisk-complete/asterisk/1.8/
$ ./configure
32 | Chapter 3: Installing Asterisk
- 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
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