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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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4.7. Configuring Inbound IAX Connections

The Inter-Asterisk eXchange (IAX) protocol is usually used for server-to-server communication; more hard phones are available that talk SIP. However, there are several soft phones that support the IAX protocol, and work is progressing on several fronts for hard phone support in firmware. The primary difference between the IAX and SIP protocols is the way media (your voice) is passed between endpoints.

With SIP, the RTP (media) traffic is passed using different ports than those used by the signaling methods. For example, Asterisk receives the signaling of SIP on port 5060 and the RTP (media) traffic on ports 10,000 through 20,000, by default. The IAX protocol differs in that both the signaling and media traffic are passed via a single port: 4569. An advantage to this approach is that the IAX protocol tends to be better suited to topologies involving NAT.

An IAX user is used to authenticate and handle calls coming into the PBX system. For calls going out from the PBX, Asterisk uses an IAX peer entry in the iax.conf file to authenticate with the remote end. (IAX peers will be explored in the section "Configuring Outbound IAX Connections.")

This section explores the configuration of your system for a Free World Dialup (FWD) account via IAX. Free World Dialup is a free VoIP service provider that allows you to connect to any other member of the network, regardless of physical location, for free. FWD is also connected to over 100 other networks to which you can connect for free.

Be sure to enable IAX2 support for your FWD account before you get started by visiting http://www.fwdnet.net/index.php?section_id=112.


This section sets up iax.conf and extensions.conf to allow you to accept calls from another FWD user. The section on outgoing IAX connections deals with placing calls.

4.7.1. iax.conf Configuration

In iax.conf, sections are defined with a name enclosed in square brackets ([]). Every iax.conf file needs at least one main section: [general]. Within the [general] section, you define the settings related to the use of the IAX protocol, such as default codecs and jitter buffering. You can override the default codecs you specify in the [general] section by specifying them within the user or peer definitions.

The following [general] section is the default from the iax.conf.sample configuration file (the same file that's installed when you perform a make samples). For more information about the options, see Appendix A.

    [general]
    bandwidth=low
    disallow=lpc10
    jitterbuffer=no
    forcejitterbuffer=no
    tos=lowdelay
    autokill=yes

    register => fwd_number:password@iax2.fwdnet.net

    [iaxfwd]
    type=user
    context=incoming
    auth=rsa
    inkeys=freeworlddialup

Within the [general] section, you'll need to add a register statement. The purpose of the register statement is to tell the FWD IAX server where you are on the Internet (your IP address). When a call is placed to your FWD number, the FWD servers do a lookup in their database and forward the call to the IP address associated with the FWD number.

In the [iaxfwd] section, define the user for incoming calls with type=user. Then define where the incoming call will be handled within the dialplan , with context=incoming. To specify that the authentication for the incoming call will be done with an RSA public/private key pair, use auth=rsa. The public key is defined with inkeys=freeworlddialup. The freeworlddialup public key comes standard with Asterisk.

4.7.2. Dialplan Configuration

Handling an incoming call in the extensions.conf file is simple. First, create a context called incoming (the same context name configured for the iaxfwd user in iax.conf). The context is followed by a Dial( ) statement that will dial the SIP extension created earlier in this chapter. Replace the number 10001 with that of your FWD account:

    [incoming]
    exten => 10001,1,Dial(SIP/john)


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