Asterisk - The Open Source VoIP PBX - SetCallerPres( )

Asterisk

Asterisk

Asterisk - The Open Source VoIP PBX

Asterisk: The Future of Telephony

Table of Contents

Conventions Used in This Book

Using Code Examples

Safari® Enabled

How to Contact Us

Acknowledgments

Chapter 1. A Telephony Revolution

Section 1.1. VoIP: Bridging the Gap Between Traditional Telephony and Network Telephony

Section 1.2. Massive Change Requires Flexible Technology

Section 1.3. Asterisk: The Hacker's PBX

Section 1.4. Asterisk: The Professional's PBX

Section 1.5. The Asterisk Community

Section 1.6. The Business Case

Section 1.7. This Book

Chapter 2. Preparing a System for Asterisk

Section 2.1. Server Hardware Selection

Section 2.2. Environment

Section 2.3. Telephony Hardware

Section 2.4. Types of Phone

Section 2.5. Linux Considerations

Section 2.6. Conclusion

Chapter 3. Installing Asterisk

Section 3.1. What Packages Do I Need?

Section 3.2. Obtaining the Source Code

Section 3.3. Compiling Zaptel

Section 3.4. Compiling libpri

Section 3.5. Compiling Asterisk

Section 3.6. Installing Additional Prompts

Section 3.7. Updating Your Source Code

Section 3.8. Common Compiling Issues

Section 3.9. Loading Zaptel Modules

Section 3.10. Loading libpri

Section 3.11. Loading Asterisk

Section 3.12. Directories Used by Asterisk

Section 3.13. Conclusion

Chapter 4. Initial Configuration of Asterisk

Section 4.1. What Do I Really Need?

Section 4.2. Working with Interface Configuration Files

Section 4.3. FXO and FXS Channels

Section 4.4. Configuring an FXO Channel

Section 4.5. Configuring an FXS Channel

Section 4.6. Configuring SIP

Section 4.7. Configuring Inbound IAX Connections

Section 4.8. Configuring Outbound IAX Connections

Section 4.9. Debugging

Section 4.10. Conclusion

Chapter 5. Dialplan Basics

Section 5.1. Dialplan Syntax

Section 5.2. A Simple Dialplan

Section 5.3. Adding Logic to the Dialplan

Section 5.4. Conclusion

Chapter 6. More Dialplan Concepts

Section 6.1. Expressions and Variable Manipulation

Section 6.2. Dialplan Functions

Section 6.3. Conditional Branching

Section 6.4. Voicemail

Section 6.5. Macros

Section 6.6. Using the Asterisk Database (AstDB)

Section 6.7. Handy Asterisk Features

Section 6.8. Conclusion

Chapter 7. Understanding Telephony

Section 7.1. Analog Telephony

Section 7.2. Digital Telephony

Section 7.3. The Digital Circuit-Switched Telephone Network

Section 7.4. Packet-Switched Networks

Section 7.5. Conclusion

Chapter 8. Protocols for VoIP

Section 8.1. The Need for VoIP Protocols

Section 8.2. VoIP Protocols

Section 8.3. Codecs

Section 8.4. Quality of Service

Section 8.5. Echo

Section 8.6. Asterisk and VoIP

Section 8.7. Conclusion

Chapter 9. The Asterisk Gateway Interface (AGI)

Section 9.1. Fundamentals of AGI Communication

Section 9.2. Writing AGI Scripts in Perl

Section 9.3. Creating AGI Scripts in PHP

Section 9.4. Writing AGI Scripts in Python

Section 9.5. Debugging in AGI

Section 9.6. Conclusion

Chapter 10. Asterisk for the Über-Geek

Section 10.1. Festival

Section 10.2. Call Detail Recording

Section 10.3. Customizing System Prompts

Section 10.4. Manager

Section 10.5. Call Files

Section 10.6. DUNDi

Section 10.7. Conclusion

Chapter 11. Asterisk: The Future of Telephony

Section 11.1. The Problems with Traditional Telephony

Section 11.2. Paradigm Shift

Section 11.3. The Promise of Open Source Telephony

Section 11.4. The Future of Asterisk

Appendix A. VoIP Channels

Section A.1. IAX

Section A.2. SIP

Appendix B. Application Reference

AbsoluteTimeout( )

AddQueueMember( )

AgentCallbackLogin( )

AgentMonitorOutgoing( )

AlarmReceiver( )

AppendCDRUserField( )

Authenticate( )

BackgroundDetect( )

ChangeMonitor( )

ControlPlayback( )

DigitTimeout( )

GetGroupCount( )

GetGroupMatchCount( )

HasNewVoicemail( )

HasVoicemail( )

IAX2Provision( )

LookupBlacklist( )

LookupCIDName( )

MailboxExists( )

ParkAndAnnounce( )

PauseQueueMember( )

PrivacyManager( )

RealTimeUpdate( )

RemoveQueueMember( )

ResponseTimeout( )

SetCallerPres( )

SetCDRUserField( )

SetGlobalVar( )

SetMusicOnHold( )

SIPAddHeader( )

SIPGetHeader( )

StopPlaytones( )

UnpauseQueueMember( )

VMAuthenticate( )

VoiceMailMain( )

WaitForSilence( )

WaitMusicOnHold( )

Appendix C. AGI Reference

DATABASE DELTREE

GET FULL VARIABLE

Appendix D. Configuration Files

Section D.1. modules.conf

Section D.2. adsi.conf

Section D.3. adtranvofr.conf

Section D.4. agents.conf

Section D.5. alarmreceiver.conf

Section D.6. alsa.conf

Section D.7. asterisk.conf

Section D.8. cdr.conf

Section D.9. cdr_manager.conf

Section D.10. cdr_odbc.conf

Section D.11. cdr_pgsql.conf

Section D.12. cdr_tds.conf

Section D.13. codecs.conf

Section D.14. dnsmgr.conf

Section D.15. dundi.conf

Section D.16. enum.conf

Section D.17. extconfig.conf

Section D.18. extensions.conf

Section D.19. features.conf

Section D.20. festival.conf

Section D.21. iax.conf

Section D.22. iaxprov.conf

Section D.23. indications.conf

Section D.24. logger.conf

Section D.25. manager.conf

Section D.26. meetme.conf

Section D.27. mgcp.conf

Section D.28. modem.conf

Section D.29. musiconhold.conf

Section D.30. osp.conf

Section D.31. oss.conf

Section D.32. phone.conf

Section D.33. privacy.conf

Section D.34. queues.conf

Section D.35. res_odbc.conf

Section D.36. rpt.conf

Section D.37. rtp.conf

Section D.38. sip.conf

Section D.39. sip_notify.conf

Section D.40. skinny.conf

Section D.41. voicemail.conf

Section D.42. vpb.conf

Section D.43. zapata.conf

Section D.44. zaptel.conf

Appendix E. Asterisk Command-Line Interface Reference

!

Section E.1. add

Section E.2. agi

Section E.3. database

Section E.4. iax2

Section E.5. indication

Section E.6. logger

Section E.7. meetme

Section E.8. pri

Section E.9. remove

Section E.10. restart

Section E.11. set

Section E.12. show

Section E.13. sip

Section E.14. stop

Section E.15. zap

About the Authors

Index

A

B

C

D

E

F

G

H

I

J

K

L

M

N

O

P

Q

R

S

T

U

V

W

X

Y

Z

SetCallerPres( )

Sets Caller ID presentation flags

    SetCallerPres(presentation)

Sets the Caller ID presentation flags on a Q931 PRI connection.

Valid presentations are:

allowed_not_screened: Presentation Allowed, Not Screened
allowed_passed_screen: Presentation Allowed, Passed Screen
allowed_failed_screen: Presentation Allowed, Failed Screen
allowed: Presentation Allowed, Network Number
prohib_not_screened: Presentation Prohibited, Not Screened
prohib_passed_screen: Presentation Prohibited, Passed Screen
prohib_failed_screen: Presentation Prohibited, Failed Screen
prohib: Presentation Prohibited, Network Number
unavailable: Number Unavailable

Always returns 0.

    exten => 123,1,SetCallerPres(allowed_not_screened)
    exten => 123,2,Dial(Zap/g1/8885551212)

SeeAlso

SetCallerID( )