Asterisk
Asterisk Asterisk
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Asterisk - The Open Source VoIP PBX

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Asterisk: The Future of Telephony
Table of Contents
Copyright
Foreword
Preface
Audience
Organization
Software
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( )
ADSIProg( )
AgentCallbackLogin( )
AgentLogin( )
AgentMonitorOutgoing( )
AGI( )
AlarmReceiver( )
Answer( )
AppendCDRUserField( )
Authenticate( )
Background( )
BackgroundDetect( )
Busy( )
CallingPres( )
ChangeMonitor( )
ChanIsAvail( )
CheckGroup( )
Congestion( )
ControlPlayback( )
Curl( )
Cut( )
DateTime( )
DBdel( )
DBdeltree( )
DBget( )
DBput( )
DeadAGI( )
Dial( )
DigitTimeout( )
Directory( )
DISA( )
DumpChan( )
DUNDiLookup( )
EAGI( )
Echo( )
EndWhile( )
ENUMLookup( )
Eval( )
Exec( )
ExecIf( )
FastAGI( )
Festival( )
Flash( )
ForkCDR( )
GetCPEID( )
GetGroupCount( )
GetGroupMatchCount( )
Goto( )
GotoIf( )
GotoIfTime( )
Hangup( )
HasNewVoicemail( )
HasVoicemail( )
IAX2Provision( )
ImportVar( )
LookupBlacklist( )
LookupCIDName( )
Macro( )
MailboxExists( )
Math( )
MeetMe( )
MeetMeAdmin( )
MeetMeCount( )
Milliwatt( )
Monitor( )
MP3Player( )
MusicOnHold( )
NBScat( )
NoCDR( )
NoOp( )
Park( )
ParkAndAnnounce( )
ParkedCall( )
PauseQueueMember( )
Playback( )
Playtones( )
Prefix( )
PrivacyManager( )
Progress( )
Queue( )
Random( )
Read( )
RealTime
RealTimeUpdate( )
Record( )
RemoveQueueMember( )
ResetCDR( )
ResponseTimeout( )
RetryDial( )
Ringing( )
SayAlpha( )
SayDigits( )
SayNumber( )
SayPhonetic( )
SayUnixTime( )
SendDTMF( )
SendImage( )
SendText( )
SendURL( )
Set( )
SetAccount( )
SetAMAFlags( )
SetCallerID( )
SetCallerPres( )
SetCDRUserField( )
SetCIDName( )
SetCIDNum( )
SetGlobalVar( )
SetGroup( )
SetLanguage( )
SetMusicOnHold( )
SetRDNIS( )
SetVar( )
SIPAddHeader( )
SIPDtmfMode( )
SIPGetHeader( )
SoftHangup( )
StopMonitor( )
StopPlaytones( )
StripLSD( )
StripMSD( )
SubString( )
Suffix( )
System( )
Transfer( )
TrySystem( )
TXTCIDName( )
UnpauseQueueMember( )
UserEvent( )
Verbose( )
VMAuthenticate( )
VoiceMail( )
VoiceMailMain( )
Wait( )
WaitExten( )
WaitForRing( )
WaitForSilence( )
WaitMusicOnHold( )
While( )
Zapateller( )
ZapBarge( )
ZapRAS( )
ZapScan( )
Appendix C.  AGI Reference
ANSWER
CHANNEL STATUS
DATABASE DEL
DATABASE DELTREE
DATABASE GET
DATABASE PUT
EXEC
GET DATA
GET FULL VARIABLE
GET OPTION
GET VARIABLE
HANGUP
NOOP
RECEIVE CHAR
RECORD FILE
SAY ALPHA
SAY DATE
SAY DATETIME
SAY DIGITS
SAY NUMBER
SAY PHONETIC
SAY TIME
SEND IMAGE
SEND TEXT
SET AUTOHANGUP
SET CALLERID
SET CONTEXT
SET EXTENSION
SET MUSIC ON
SET PRIORITY
SET VARIABLE
STREAM FILE
TDD MODE
VERBOSE
WAIT FOR DIGIT
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
!
abort halt
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
Colophon
About the Authors
Colophon
Index
SYMBOL
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
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About the Authors

Jim Van Meggelen is President and CTO of Core Telecom Innovations, a Canadianbased provider of open source telephony solutions. He has over 15 years of enterprise telecom experience, for such companies as Nortel, Williams, and Telus, and has extensive knowledge of both legacy and VoIP equipment from manufacturers such as Nortel, Cisco, and Avaya.

Jim was the architect of two of the world's largest managed enterprise voice networks, each solution serving roughly 20,000 users in more than 1,000 communities across Canada and providing telecommunications in 5 different languages through 6 time zones, administered completely from a central location. These networks pioneered the use of extensive automation and database control in a branch voice networkfunctionalities not generally available in proprietary telecommunications systems. Jim has now moved on from the world of proprietary telecom, and is commited to open source telephony.

Jim is one of the principal contributors to the Asterisk Documentation Project. He enjoys teaching, public speaking, improvisational acting, and writing.

Jared Smith is one of those rare individuals whose beloved hobby is the same as his profession. The son of a computer store owner, Jared wrote his first computer program at the age of 7 on his Commodore 64. The obvious choice of major for this geek-in-embryo was Computer Engineering, and Jared received his Bachelor of Science degree with a minor in Computer Science from Utah State University. He now has over a decade of professional systems administration and programming experience in the simulation, market research, and web analytics industries. As a key architect of one of the world's largest Asterisk installations, Jared has a wealth of hands-on telephony and VoIP knowledge, which he shares through users groups and various public speaking engagements. He is an active member of the Asterisk community and a co-founder of the Asterisk Documentation Project.

Jared is active in his community, donating Asterisk services to local schools and serving in his church. The greatest joy in Jared's life comes from spending time with his children, Caleb and Sydney Jo, and his wife, Jenny.

Leif Madsen is a graduate of the Telecommuncations Technology program from the Sheridan Institute of Technology and CEO of LeifMadsen Enterprise, Incorporated, a documentation and consulting firm specializing in Asterisk. He was one of the first Digium Certified Asterisk Professionals (dCAP), and assists with the Astricon conferences and trainings organized by IPsando, LLC.

Leif first took an interest in Asterisk while attempting to find a voice conferencing solution for himself and his friends. After someone suggested trying Asterisk, the obsession began. Wanting to contribute and be involved with the community, and noticing the lack of Asterisk documentation, he co-founded the Asterisk Documentation Project.


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