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Asterisk: The Future of Telephony, 2nd Edition

by Jim Van Meggelen, Jared Smith, Leif Madsen

August 2007

Intermediate to advanced

608 pages

20h 33m

English

O'Reilly Media, Inc.

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Audience

Jim Van MeggelenLeif MadsenJared Smith
VoIP: Bridging the Gap Between Traditional and Network TelephonyThe Zapata Telephony Project
The Asterisk Mailing ListsThe Asterisk WikiThe IRC ChannelsAsterisk User GroupsThe Asterisk Documentation Project
Server Hardware SelectionPerformance IssuesChoosing a ProcessorSmall systemsMedium systemsLarge systemsChoosing a MotherboardPower Supply RequirementsComputer power suppliesRedundant power supplies
Power Conditioning and Uninterruptible Power SuppliesPower-conditioned UPSesGroundingElectrical CircuitsThe Equipment RoomHumidityTemperatureDustSecurity
Connecting to the PSTNAnalog interface cardsDigital interface cardsChannel banksOther types of PSTN interfacesConnecting Exclusively to a Packet-Based Telephone NetworkEcho Cancellation
Physical TelephonesAnalog telephonesProprietary digital telephonesISDN telephonesIP telephonesSoftphonesTelephony AdaptorsCommunications Terminals
What Packages Do I Need?Linux Package Requirements
Obtaining Asterisk Source CodeExtracting the Source Code
The ztdummy DriverThe Zapata Telephony DriversUsing ztcfg and zttool
Standard InstallationAlternative make Argumentsmake cleanmake distcleanmake updatemake webvmailmake progdocsmake configUsing Precompiled Binaries
Asteriskconfigure: error: no acceptable C compiler found in $PATHconfigure: error: C++ preprocessor "/lib/cpp" fails sanity checkconfigure: error: *** termcap support not foundZaptelmake: cc: Command not foundFATAL: Module wctdm/fxs/fxo not foundUnresolved symbol link when loading ztdummyDepmod errors during compilation
Systems Running udevdLoading ZaptelLoading ztdummy
Console Commands
/etc/asterisk//usr/lib/asterisk/modules//var/lib/asterisk/var/spool/asterisk//var/run//var/log/asterisk//var/log/asterisk/cdr-csv
What Is AsteriskNOW?Before You BeginWhat You Will NeedInstallationQuick installationExtended procedureAccessing the GUIAlternate InstallationsFor More Information
What Do I Really Need?
Determining the FXO and FXS Ports on Your TDM400P
Zaptel Hardware ConfigurationZapata Hardware ConfigurationDialplan ConfigurationDialing In
Zaptel Hardware ConfigurationZapata Hardware ConfigurationDialplan Configuration
Basic SIP Telephone Configuration in AsteriskDefining the SIP device in AsteriskConfiguring the Device ItselfEssential Server ComponentsDHCP serverFTP serverCounterPath’s X-Lite SoftphonePolycom’s IP 430DHCP serverProtocol to use for downloadingFTPThe Polycom configuration filesThe bootROMThe application imageThe sip.cfg fileThe master config file for each phoneThe set-specific config fileGotchasCisco 7960 TelephoneLinksys SPA-942Logging in to the phoneRegistering your phone to AsteriskConfiguring the Dialplan for Testing
Configuring Our Asterisk BoxesSIP Phone ConfigurationConfiguring the Dialplan
Configuring the Channel Configuration File (iax.conf)Configure the SoftphoneConfiguring the Dialplan for Testing
Configuring Our Asterisk BoxesIAX Phone ConfigurationConfiguring the Dialplan
Connecting to the ConsoleEnabling Verbosity and Debugging
Dialplan SyntaxContextsExtensionsPrioritiesUnnumbered prioritiesPriority labelsApplications
The s ExtensionThe Answer(), Playback(), and Hangup() ApplicationsOur First Dialplan
The Background(), WaitExten(), and Goto() ApplicationsHandling Invalid Entries and TimeoutsUsing the Dial() ApplicationAdding a Context for Internal CallsUsing VariablesGlobal variablesChannel variablesEnvironment variablesAdding variables to our dialplanPattern MatchingPattern-matching syntaxPattern-matching examplesUsing the ${EXTEN} channel variableEnabling Outbound DialingIncludes
Expressions and Variable ManipulationBasic ExpressionsOperators
SyntaxExamples of Dialplan Functions
The GotoIf() ApplicationTime-Based Conditional Branching with GotoIfTime()
Creating MailboxesAdding Voicemail to the DialplanAccessing VoicemailCreating a Dial-by-Name Directory
Defining MacrosCalling Macros from the DialplanUsing Arguments in Macros
Storing Data in the AstDBRetrieving Data from the AstDBDeleting Data from the AstDBUsing the AstDB in the Dialplan
Zapateller()Call ParkingConferencing with MeetMe()
Analog TelephonyParts of an Analog TelephoneRingerDial padHybrid (or network)Hook switch (or switch hook)HandsetTip and Ring
Pulse-Code ModulationDigitally encoding an analog waveformIncreasing the sampling resolution and rateNyquist’s TheoremLogarithmic compandingAliasing
Circuit TypesThe humble DS-0―the foundation of it allT-carrier circuitsSONET and OC circuitsDigital Signaling ProtocolsChannel Associated Signaling (CAS)ISDNISDN-BRI/BRAISDN-PRI/PRASignaling System 7
The Need for VoIP Protocols
IAX (The “Inter-Asterisk eXchange” Protocol)HistoryFutureSecurity considerationsIAX and NATSIPHistoryFutureSecurity considerationsSIP and NATH.323HistoryFutureSecurity considerationsH.323 and NATMGCPProprietary ProtocolsSkinny/SCCPUNISTIM
G.711G.726G.729AGSMiLBCSpeexMP3
TCP, UDP, and SCTPTransmission Control ProtocolUser Datagram ProtocolStream Control Transmission ProtocolDifferentiated ServiceGuaranteed ServiceMPLSRSVPBest Effort
Why Echo OccursManaging Echo on Zaptel ChannelsHardware Echo Cancellation
Users and Peers and Friends—Oh My!UsersPeersFriendsregister Statements
Spam over Internet Telephony (SPIT)Encrypting Audio with Secure RTPSpoofingWhat Can Be Done?Basic network securitySegregating voice and data trafficDMZServer hardeningEncryptionPhysical security
Fundamentals of AGI CommunicationWhat Are STDIN, STDOUT, and STDERR?The Standard Pattern of AGI CommunicationCalling an AGI Script from the Dialplan
The Perl AGI Library
The PHP AGI Library
The Python AGI Library
Debugging from the Operating SystemUsing Asterisk’s agi debug Command
The Manager InterfaceConnecting to the Manager InterfaceSending CommandsTransferring a callReading a configuration fileUpdating configuration files
A New Approach to DialplansAsterisk Development with AdhearsionInstalling AdhearsionInstalling Ruby/RubyGems on AsteriskNOWInstalling Ruby/RubyGems on LinuxInstalling Ruby/RubyGems on Mac OS XRuby/RubyGems on WindowsInstalling Adhearsion from RubyGemsExploring a New Adhearsion ProjectAdhearsion dialplan writingDatabase integrationDistributing and reusing codeIntegrate with Your Desk Phone Using MicromenusIntegrating with a Web ApplicationUsing JavaMore Information
Why a GUI for Asterisk?
Mark Spencer Talks About the GUIUsing the GUIGUI elements
Components of the Asterisk GUIAsterisk Manager InterfaceManager over HTTP and the Asterisk web serverAJAM and JavaScript
Setting up http.conf and manager.conf
Issuing Manager Commands over HTTPLOGINTransferring a callReading a configuration fileUpdating configuration files using UPDATECONFIGError responseAjax, AJAM, and AsteriskForm processing in a traditional web applicationForm processing in an Ajax applicationThe Prototype frameworkCustomization of the GUIAdding a new tab to the GUIExposing configuration settings in the GUIFor More Information
Introduction
Configuring res_odbc for Access to Our Database
Static RealtimeDynamic Realtime
Creating the Large Object TypeConfiguring voicemail.conf for ODBC StorageTesting ODBC Voicemail
Call Detail Recording
FestivalGetting Festival Set Up and Ready for AsteriskConfiguring Asterisk for FestivalStarting the Festival ServerCalling Festival from the Dialplan
How Does DUNDi Work?Configuring Asterisk for Use with DUNDiThe General Peering AgreementGeneral configurationCreating mapping contextsDefining DUNDi peersAllowing remote connectionsConfiguring the dialplan
Storing Voicemail in an IMAP ServerStoring Voicemail in an ODBC Database
The Problems with Traditional TelephonyClosed ThinkingLimited Standards CompliancySlow Release CyclesRefusing to Let Go of the Past and Embrace the Future
The Itch That Asterisk ScratchesOpen ArchitectureStandards ComplianceLightning-Fast Response to New TechnologiesPassionate CommunitySome Things That Are Now PossibleLegacy PBX migration gatewayLow-barrier IVRConference roomsHome automation
Speech ProcessingFestivalSpeech recognitionHigh-Fidelity VoiceVideoThe challenge of video-conferencingWhy we love video-conferencingWhy video-conferencing may never totally replace voiceWirelessWi-FiWi-MAXUnified MessagingPeeringE.164ENUMe164.orgDUNDiChallengesToo much change, too few standardsVoIP spamFear, uncertainty, and doubtBottleneck engineeringRegulatory warsQuality of serviceComplexityOpportunitiesTailor-made private telecommunications networksLow barrier to entryHosted solutions of similar complexity to corporate web sitesProper integration of communications technologies
IAXGeneral IAX SettingsRegistering to Other Servers with register StatementsIAX Channel DefinitionsChannel-specific parameters
General SIP ParametersSIP Channel Definitions
modules.conf
[general][logfiles]
General Voicemail SettingsVoicemail ZonesDefining Voicemail Contexts and Mailboxes

Content preview from Asterisk: The Future of Telephony, 2nd Edition

Name

Page() — Opens one-way audio to multiple phones

Synopsis

Page(Tech/chan1[&Tech/chan2]&[...][&Tech/chanN][|options])

Places outbound calls to the given technology/resource and dumps them in to a conference bridge as muted participants. The original caller is dumped in to the conference as a speaker, and the room is destroyed when the original caller leaves. The following options may be specified:

d: Full duplex audio. Allow the paged persons to respond to the caller.
q: Quiet. Do not play a beep to the caller.
r: Record the page. See the r option to MeetMe for more information.

exten => 123,1,Page(SIP/101&SIP/102&IAX2/iaxy123)

Asterisk: The Future of Telephony, 2nd Edition

by Jim Van Meggelen, Jared Smith, Leif Madsen

Name

Synopsis

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