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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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D.34. queues.conf

Asterisk provides basic call center functionality via its queueing system, but those who are using it in more mission-critical environments often report that their solutions required customization. You can do this customization in the queues.conf file.

The [general] section of queues.conf contains settings that will apply to all queues. Currently, the only parameter that is supported is persistentmembers. If this parameter is set to yes, a member that is added to the system via the AddQueueMember( ) application will be stored in the AstDB, and therefore retained across a restart.

You can define a queue by placing its name inside of square brackets ([]). Within each queue, the following parameters are available:



musiconhold

This parameter allows you to configure which Music on Hold class (configured in musiconhold.conf) to use for the queue.



announce

When a call is presented to a member of the queue, the prompt specified by announce will be played to that agent before the caller is connected. This can be useful for agents who are logged into more than one queue. You can specify either the full path to the file, or a path relative to /var/lib/asterisk/sounds/.



strategy

Asterisk can use six strategies to distribute calls to agents:



ringall

The queue rings every available agent and connects the call to whichever agent answers first (this is the default).



roundrobin

The queue cycles through the agents until it finds one who is available to take the call. roundrobin does not take into account the workload of the agents. Also, because roundrobin always starts with the first agent in the queue, this strategy is suitable only in an environment where you want your higher-ranked agents to handle all calls unless they are busy, in which case the lower-ranked agents may get a call.



leastrecent

The call is presented to the agent who has not been presented a call for the longest period of time.



fewestcalls

The call is presented to the agent who has received the least amount of calls. This strategy does not take into account the actual agent workloads; it only considers the number of calls they have taken (for example, an agent who has had 3 calls that each lasted for 10 minutes will be preferred over an agent who has had 5 calls each lasting 2 minutes).



random

As its name suggests, the random strategy chooses an agent at random. In a small call center, this strategy may prove to be the most fair.



rrmemory

The queue cycles through each agent, keeping track of which agent last received a call (this strategy is known as round-robin memory). This ensures that call presentation cycles through the agents as fairly as possible.



servicelevel

In a call center, the service level represents the maximum amount of time a caller should ideally have to wait before being presented to an agent. For example, if servicelevel is set to 60 and the service level percentage is 80%, that means 80% of the calls that came into the queue were presented to an agent in less than 60 seconds.



context

If a context is assigned to a queue, the caller will be able to press a single digit to exit to the corresponding extension within the configured context, if it exists. This action takes the caller out of the queue, which means that she will lose her place in the queuebe aware of this when you use this feature.



timeout

The timeout value defines the maximum amount of time (in seconds) to let an agent's phone ring before deeming the agent unavailable and placing the call back into the queue.



retry

When a timeout occurs, the retry value specifies how many seconds to wait before presenting the call again to an available agent.



weight

The weight parameter assigns a rank to the queue. If calls are waiting in multiple queues, those queues with the highest weight values will be presented to agents first. When you are designing your queues, be aware that this strategy can prevent a call in a lower-weighted queue from ever being answered. Always ensure that calls in lower-weighted queues eventually get promoted to higher-weighted queues to ensure that they don't have to hold forever.



wrapuptime

You can configure this parameter to allow agents a few seconds of downtime after completing a call before the queue presents them with another call.



maxlen

maxlen is the maximum number of calls that can be added to the queue before the call goes to the next priority of the current extension.



announce-frequency

The announce-frequency value (defined in seconds) determines how often to announce to the caller his place in the queue and estimated hold time.



announce-holdtime

There are three possible values for this parameter: yes, no, and once. The announce-holdtime parameter determines whether or not to include the estimated hold time within the position announcement. If set to once, it will be played to the caller only once.



monitor-format

This parameter accepts three possible values: wav, gsm, and wav49. By enabling this option, you are telling Asterisk that you wish to record all completed calls in the queue in the format specified. If this option is not specified, no calls will be recorded.



monitor-join

The Monitor( ) application in Asterisk normally records either end of the conversation in a separate file. Setting monitor-join to yes instructs Asterisk to merge the files at the end of the call.



joinempty

This parameter accepts three values: yes, no, and strict. It allows you to determine whether callers can be added to a queue based on the status of the members of the queue. The strict option will not allow callers to join the queue if all members are unavailable.



leavewhenempty

This parameter determines whether you want your holding callers to be removed from the queue when the conditions preventing a caller from joining exist (i.e., when all of your agents log out and go home).



eventwhencalled

Set eventwhencalled to yes if you wish to have queue events presented on the Manager interface.



eventmemberstatusoff

Setting this parameter to no will generate extra information pertaining to each queue member.



reportholdtime

If you set this parameter to yes, the amount of time the caller held before being connected will be announced to the answering agent.



memberdelay

This parameter defines whether a delay will be inserted between the time when the queue identifies a free agent and the time when the call is connected to that agent.



member => member_name

Members of a queue can be either channel types or agents. Any agents you list here must be defined in the agents.conf file.


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