Asterisk
Asterisk Asterisk
Contents

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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3.2. Obtaining the Source Code

The Asterisk source code can be obtained either through FTP or CVS. We will show you how to acquire the source with both methods, although you only need to use one of them to retrieve the packages (FTP is the preferred method).

3.2.1. Obtaining Asterisk Source Code from FTP

The Asterisk source code can be obtained from the Digium FTP server, located at ftp://ftp.digium.com. The easiest way to obtain the stable release is through the use of the program wget.

Stable and Head

Asterisk comes in two different flavors, generally referred to as stable and head. Stable, as the name implies, is the established branch of Asterisk for use in production systems. The head branch is what the developers use to test new features and bug fixes .

Bug fixes (not features) are merged over to the stable branch after a reasonable period of testing. It is entirely possible that the development branch may be broken at certain points during testing; thus, the stable branch is what you will want to run your production system on, and it is what we will be using throughout this book.

You can obtain stable releases via FTP. Both the stable and head branches of Asterisk can also be obtained from CVS, as explained later in this chapter. However, it is important to note the difference between releases and CVS. Releases are snapshots from the stable CVS tree, tagged with a version number and released via the FTP server when a new stable release is deemed ready. Note that the stable CVS branch is not a releaseit's a work in a progress, and it may be buggy (i.e., not so stable after all). The FTP tarballs are the actual releases.

To summarize, use only stable releases obtained via the FTP server for production systems.


Note that we will be making use of the /usr/src/ directory to extract and compile the Asterisk source. Also be aware that you will need root access to write files to the /usr/src/ directory and to install Asterisk and its associated packages.

To obtain the latest stable source code via wget, enter the following commands on the command line:

    # cd /usr/src/
    # wget --passive-ftp ftp.digium.com/pub/asterisk/asterisk-1.*.tar.gz
    # wget --passive-ftp ftp.digium.com/pub/asterisk/asterisk-sounds-*.tar.gz
    # wget --passive-ftp ftp.digium.com/pub/zaptel/zaptel-*.tar.gz
    # wget --passive-ftp ftp.digium.com/pub/libpri/libpri-*.tar.gz

As long as Digium doesn't change the way they put things on the FTP site, the wget command will automagically get the latest version. You may also replace the wildcard mask (*) with the currently available software version.


Now that you've retrieved the files for Asterisk and the Digium hardware, you are ready to extract the code.

3.2.2. Extracting the Source Code

If you use wget to obtain the source code from the FTP server, you will need to extract it before compiling. If you didn't download the packages to /usr/src/, either move them there now, or specify the full path to their location. We will be using the GNU tar application to extract the source code from the compressed archive. This is a simple process that can be achieved through the use of the following commands:

    # cd /usr/src/
    # tar zxvf zaptel-*.tar.gz
    # tar zxvf libpri-*.tar.gz
    # tar zxvf asterisk-*.tar.gz
    # tar zxvf asterisk-sounds*.tar.gz

These commands will extract the packages and source code to their respective directories.

3.2.3. Obtaining Asterisk Source Code from CVS

The Concurrent Versioning System (CVS) is a tool that provides a central repository that large (and diverse) development teams can use to manage the multitude of files associated with a development project. When a change is made, it is committed to the CVS server, where it is immediately available for download and compilation. Another added benefit of using CVS is that the version for any particular file can be rolled back to a certain instance, so that if something was working at one point but a change causes it to break, you can easily revert to the working version. This is true for the entire tree as well. If you find that installing the latest version of Asterisk causes any part of the system to break, you can "roll back" to an earlier point in time and investigate the cause of the problem.

If you are a developer looking to obtain the latest updates to the source code, you will need to get them from the CVS servers. You can also download the stable branch via CVS:

  • Export the CVSROOT path:

        # cd /usr/src/
        # export CVSROOT=:pserver:anoncvs:anoncvs@cvs.digium.com:/usr/cvsroot
    

  • Download HEAD from CVS:

        # cvs checkout zaptel libpri asterisk
    

  • Download STABLE 1.0 from CVS:

        # cvs checkout -r v1-0 zaptel libpri asterisk
    

  • Download STABLE 1.2 from CVS:

        # cvs checkout -r v1-2 zaptel libpri asterisk
    

  • Download optional modules from CVS:

        # cvs checkout asterisk-sounds asterisk-addons
    

Again, note that the stable branch available from CVS is not a release and should not be used for production systems.


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