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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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6.7. Handy Asterisk Features

Now that we've gone over some more of the basics, let's look at a few popular functions that have been incorporated into Asterisk.

6.7.1. Zapateller( )

Zapateller( ) is a simple Asterisk application that plays a special information tone at the beginning of a call, which causes auto-dialers (usually used by telemarketers) to think the line has been disconnected. Not only will they hang up, but their systems will flag your number as out of service, which could help you avoid all kinds of telemarketing calls. To use this functionality within your dialplan, simply call the Zapateller( ) application.

We'll also use the optional nocallerid option so that the tone will be played only when there is no Caller ID information on the incoming call. For example, you might use Zapateller( ) in the s extension of your [incoming] context, like this:

    [incomimg]
    exten => s,1,Zapateller(nocallerid)
    exten => s,2,Playback(enter-ext-of-person)

6.7.2. Call Parking

Another handy feature is called "call parking ." Call parking allows you to place a call on hold in a "parking lot," so it can be taken off hold from another extension. Parameters for call parking (such as the extensions to use, the number of spaces, and so on) are all controlled within the features.conf configuration file. The [general] section of the features.conf file contains four settings related to call parking :



parkext

This is the parking lot extension. Transfer a call to this extension, and the system will tell you which parking position the call is in. By default, the parking extension is 700.



parkpos

This option defines the number of parking slots. For example, setting it to 701-720 creates 20 parking positions, numbered 701 through 720.



context

This is the name of the parking context. To be able to park calls, you must include this context.



parkingtime

If set, this option controls how long (in seconds) a call can stay in the parking lot. If the call isn't picked up within the specified time, the extension that parked the call will be called back.

You must restart Asterisk after editing features.conf, as the file is read only on startup. Running the reload command will not cause the features.conf file to be read.


Also note that because the user needs to be able to transfer the calls to the parking lot extension, you should make sure you're using the t and/or T options to the Dial( ) application.

So, let's create a simple dialplan to show off call parking:

    [incoming]
    include => parkedcalls

    exten=103,1,Dial(SIP/Bob,,tT)
    exten=104,1,Dial(SIP/Charlie,,tT)

To illustrate how call parking works, say that Alice calls into the system and dials extension 103, to reach Bob. After a while, Bob transfers the call to extension 700, which tells him that the call from Alice has been parked in position 701. Bob then dials Charlie at extension 104, and tells him that Alice is at extension 701. Charlie then dials extension 701, and begins to talk to Alice. This is a simple and effective way of allowing callers to be transferred between users.

6.7.3. Conferencing with MeetMe( )

Last but not least, let's cover setting up an audio conference bridge with the MeetMe( ) application.[*] This application allows multiple callers to converse together, as if they were all in the same physical location. Some of the main features include:

[*] In the world of legacy PBXs, this type of functionality is very expensive. Either you have to pay big bucks for a dial-in service, or you have to add an expensive conferencing bridge to your proprietary PBX.

  • The ability to create password-protected conferences

  • Conference administration (mute conference, lock conference, kick participants)

  • The option of muting all but one participant (useful for company announcements, broadcasts, etc.)

  • Static or dynamic conference creation

Let's walk through setting up a basic conference room. The configuration options for the MeetMe conferencing system are found in meetme.conf. Inside the configuration file, you define conference rooms and optional numeric passwords. (If a password is defined here, it will be required to enter all conferences using that room.) For our example, let's set up a conference room at extension 600. First, we'll set up the conference room in meetme.conf. We'll call it 600, and we won't assign a password at this time:

    [rooms]
    conf => 600

Now that the configuration file is complete, we'll need to restart Asterisk so that it can re-read the meetme.conf file. Next, we'll add support for the conference room to our dialplan with the MeetMe( ) application. MeetMe( ) takes three arguments: the name of the conference room (as defined in meetme.conf), a set of options, and the password the user must enter to join this conference. Let's set up a simple conference using room 600, the i option (which announces when people enter and exit the conference), and a password of 54321:

    exten => 600,1,MeetMe(600,i,54321)

That's all there is to it! When callers enter extension 600, they will be prompted for the password. If they correctly enter 54321, they will be added to the conference. See Appendix B for a list of all the options supported by the MeetMe( ) application.

Another useful application is MeetMeCount( ). As its name suggests, this application counts the number of users in a particular conference room. It takes up to two arguments: the conference room in which to count the number of participants, and optionally a variable name to assign the count to. If the variable name is not passed as the second argument, the count is read to the caller:

    exten => 601,1,Playback(conf-thereare)
    exten => 601,2,MeetMeCount(600)
    exten => 601,3,Playback(conf-peopleinconf)

If you pass a variable as the second argument to MeetMeCount( ), the count is assigned to the variable and playback of the count is skipped. You might use this to limit the number of participants, like this:

    ; limit the conference room to 10 participants
    exten => 600,1,MeetMeCount(600,CONFCOUNT)
    exten => 600,2,GotoIf($[${CONFCOUNT} <= 10]?3:100)
    exten => 600,3,MeetMe(600,i,54321)
    exten => 600,100,Playback(conf-full)

Isn't Asterisk fun?


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