book

Asterisk: The Future of Telephony

by Jim Van Meggelen, Jared Smith, Leif Madsen

September 2005

Intermediate to advanced

408 pages

16h 20m

English

O'Reilly Media, Inc.

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Audience
Leif MadsenJared SmithJim Van Meggelen

VoIP: Bridging the Gap Between Traditional Telephony and Network TelephonyThe Zapata Telephony Project
The Asterisk Mailing ListsThe Asterisk WikiThe IRC ChannelsThe 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 UPSsGroundingElectrical CircuitsThe Equipment RoomHumidityTemperatureDustSecurity
Connecting to the PSTNAnalog interface cardsDigital interface cardsChannel banksOther types of PSTN interfacesConnecting Exclusively to a Packet-Based Telephone Network
Physical TelephonesAnalog telephonesProprietary digital telephonesISDN telephonesIP telephonesSoft PhonesTelephony AdaptorsCommunications Terminals
What Packages Do I Need?Package Requirements
Obtaining Asterisk Source Code from FTPExtracting the Source CodeObtaining Asterisk Source Code from CVS
The ztdummy DriverThe Zapata Telephony DriversUsing ztcfg and zttoolThe zconfig.h FileBoost ringerDisable μ-law/A-law precomputationEnable MMX optimizationChoose echo cancellation methodEnable aggressive suppressionDisable echo cancellationEnable HDLCEnable ZapRASEnable Zaptel’s watchdogSet default tone zoneEnable CAC ground start signalingTDM400P Revision H PCI ID workaroundPassing Module Parameters to Configure Zaptel
Standard InstallationAlternative make Argumentsmake cleanmake updatemake upgrademake webvmailmake progdocsmake mpg123make configEditing the MakefileEnabling GSM optimizationsDisabling configuration file overwritesEnabling debug profiling informationSpecifying where to install Asterisk after compilingChanging the staging directoryCompiling on VIA motherboardsUsing Precompiled Binaries
AsteriskC compiler cannot create executablesbison: command not found/usr/bin/ld: cannot find -lsslrpmbuild: command 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
CLI CommandsRed Hat-Style Initialization ScriptThe safe_asterisk Script
/etc/asterisk//usr/lib/asterisk/modules//var/lib/asterisk/var/spool/asterisk//var/run//var/log/asterisk//var/log/asterisk/cdr-csv
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
SIP and RTPSIP ConfigurationClient ConfigurationDialplan Configuration
iax.conf ConfigurationDialplan Configuration
iax.conf ConfigurationDialplan Configuration
Connecting to the ConsoleEnabling Verbosity and Debugging
Dialplan SyntaxContextsExtensionsPrioritiesApplications
The s ExtensionThe Answer(), Playback(), and Hangup() ApplicationsOur First Dialplan
The Background() 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.723.1G.729AGSMiLBCSpeexMP3
TCP, UDP, and SCTPTransmission Control ProtocolUser Datagram ProtocolStream Control Transmission ProtocolDifferentiated ServiceGuaranteed ServiceMPLSRSVPBest Effort
Why Echo OccursManaging Echo
Users and Peers and Friends—Oh My!UsersPeersFriendsregister Statements
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
FestivalGetting Festival Set Up and Ready for AsteriskConfiguring Asterisk for FestivalStarting the Festival ServerCalling Festival from the Dialplan
CDR Challenges
Manager CommandsThe Flash Operator Panel
How Does DUNDi Work?Configuring Asterisk for Use with DUNDiThe General Peering AgreementGeneral configurationCreating mapping contextsDefining DUNDi peersAllowing remote connectionsConfiguring the dialplan
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 ProcessingFestivalSphinxHigh-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 Settingsregister StatementsIAX Channel DefinitionsChannel-specific parameters
General SIP ParametersSIP Channel Definitions
modules.confResourcesApplicationsDatabase-Stored Call Detail RecordsChannelsCodecsFormatsPBX Core ModulesStandalone Functions
[general][logfiles]
General Voicemail SettingsVoicemail ZonesDefining Voicemail Contexts and Mailboxes
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Content preview from Asterisk: The Future of Telephony

Name

sip debug peer

Synopsis

    sip [no] debug peer peer_name

Debugs (or disables debugging of) SIP messages from an individual peer, referenced by the peer name configured in sip.conf. Debugging information can be displayed for a dynamic host only if that host is registered with you. If you are trying to debug a registration issue, see sip debug ip.

    sip debug peer john

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Publisher Resources

ISBN: 0596009623Supplemental Content Catalog Page Errata