mirror of https://github.com/asterisk/asterisk
these chanes will only be done in the trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58955 65c4cc65-6c06-0410-ace0-fbb531ad65f31.4
Russell Bryant
18 years ago
parent
2fc431b373
commit
1b8da31130
@ -0,0 +1,74 @@
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Files in the /doc directory:
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----------------------------
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In addition to these files, there is a lot of documentation of various
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configuration options in the sample configuration files, in the /configs
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directory of your source code
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Start here
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----------
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security.txt IMPORTANT INFORMATION ABOUT ASTERISK SECURITY
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hardware.txt Hardware supported by Asterisk
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Configuration
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-------------
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configuration.txt Features in the configuration parser
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extensions.txt Basics about the dialplan
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extconfig.txt How to use databases for configuration of Asterisk (ARA)
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ip-tos.txt About the IP Type Of Service settings
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realtime.txt The Asterisk Realtime Architecture - database support
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freetds.txt Information about the FreeTDS support
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ael.txt Information about the Asterisk Extension Language
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Misc
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----
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PEERING The General Peering Agreement for Dundi
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ajam.txt About the HTTP-based manager interface
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app_sms.txt How to configure the SMS application
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asterisk.conf.txt Documentation of various options in asterisk.conf
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callingpres.txt Settings for Caller ID presentation
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billing.txt Call Data Record information
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cliprompt.txt How to change the Asterisk CLI prompt
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dundi.txt Dundi - a discovery protocol
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enum.txt Enum support in Asterisk
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ices.txt Integrating ICEcast streaming in Asterisk
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jitterbuffer.txt About the IAX2 jitterbuffer implementation
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math.txt About the math() application
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mp3.txt About MP3 support in Asterisk
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musiconhold-fpm.txt Free Music On Hold music
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mysql.txt About MYSQL support in Asterisk
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odbcstorage.txt Voicemail storage of messages in UnixODBC
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privacy.txt Privacy enhancements in Asterisk
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queuelog.txt Agent and queue logging
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channelvariables.txt Channel variables
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cdrdrivers.txt About CDR storage in various databases (needs update)
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asterisk-mib.txt SNMP mib for Asterisk (net-snmp)
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digium-mib.txt SNMP mib for Asterisk (net-snmp)
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Channel drivers
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---------------
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misdn.txt The mISDN channel driver for ISDN BRI cards
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h323.txt How to compile and configure the H.323 channel
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chaniax.txt About the IAX2 protocol support in Asterisk
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localchannel.txt The local channel is a "gosub" in the dialplan
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Portability
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-----------
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cygwin.txt Compiling Asterisk on CygWin platforms (Windows)
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For developers
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--------------
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See http://www.asterisk.org/developers for more information
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manager.txt About the AMI - Asterisk Manager Interface
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for third party call control and PBX management
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backtrace.txt How to produce a backtrace when Asterisk crashes
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CODING-GUIDELINES Guidelines for developers
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channels.txt What is a channel?
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externalivr.txt Documentation of the protocol used in externalivr()
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linkedlists.txt How to develop linked lists in Asterisk (old)
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iax.txt About the IAX protocol
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apps.txt About application development
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model.txt About the call model in Asterisk (old)
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modules.txt How Asterisk modules work
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datastores.txt About channel data stores
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speechrec.txt The Generic Speech Recognition API
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File diff suppressed because it is too large
Load Diff
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Asterisk applications register themselves with ast_application_register.
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They should have a short, unique name, and an exec function which takes
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as its arguments a channel and some data that might be useful for callback
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stuff. Remember to keep track of how many and which channels are using
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your application so that should the module need to be unloaded
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(particularly force unloaded), you will be able to ast_softhangup all the
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channels. An application should *never* call ast_hangup on the channel
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that it is running on (although it could conceivably hang up other
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channels that it allocates). See app_playback.c as an example of a simple
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application.
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File diff suppressed because it is too large
Load Diff
@ -1,130 +0,0 @@
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\subsubsection{Asterisk Main Configuration File}
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Below is a sample of the main Asterisk configuration file,
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asterisk.conf. Note that this file is not provided in
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sample form, because the Makefile creates it when needed
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and does not touch it when it already exists.
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\begin{verbatim}
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[directories]
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; Make sure these directories have the right permissions if not
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; running Asterisk as root
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; Where the configuration files (except for this one) are located
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astetcdir => /etc/asterisk
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; Where the Asterisk loadable modules are located
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astmoddir => /usr/lib/asterisk/modules
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; Where additional 'library' elements (scripts, etc.) are located
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astvarlibdir => /var/lib/asterisk
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; Where AGI scripts/programs are located
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astagidir => /var/lib/asterisk/agi-bin
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; Where spool directories are located
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; Voicemail, monitor, dictation and other apps will create files here
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; and outgoing call files (used with pbx_spool) must be placed here
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astspooldir => /var/spool/asterisk
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; Where the Asterisk process ID (pid) file should be created
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astrundir => /var/run/asterisk
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; Where the Asterisk log files should be created
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astlogdir => /var/log/asterisk
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[options]
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;Under "options" you can enter configuration options
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;that you also can set with command line options
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; Verbosity level for logging (-v)
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verbose = 0
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; Debug: "No" or value (1-4)
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debug = 3
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; Background execution disabled (-f)
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nofork=yes | no
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; Always background, even with -v or -d (-F)
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alwaysfork=yes | no
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; Console mode (-c)
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console= yes | no
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; Execute with high priority (-p)
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highpriority = yes | no
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; Initialize crypto at startup (-i)
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initcrypto = yes | no
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; Disable ANSI colors (-n)
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nocolor = yes | no
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; Dump core on failure (-g)
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dumpcore = yes | no
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; Run quietly (-q)
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quiet = yes | no
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; Force timestamping in CLI verbose output (-T)
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timestamp = yes | no
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; User to run asterisk as (-U) NOTE: will require changes to
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; directory and device permissions
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runuser = asterisk
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; Group to run asterisk as (-G)
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rungroup = asterisk
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|
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; Enable internal timing support (-I)
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internal_timing = yes | no
|
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|
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; These options have no command line equivalent
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|
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; Cache record() files in another directory until completion
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cache_record_files = yes | no
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||||||
record_cache_dir = <dir>
|
|
||||||
|
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; Build transcode paths via SLINEAR
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transcode_via_sln = yes | no
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|
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; send SLINEAR silence while channel is being recorded
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transmit_silence_during_record = yes | no
|
|
||||||
|
|
||||||
; The maximum load average we accept calls for
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maxload = 1.0
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||||||
|
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||||||
; The maximum number of concurrent calls you want to allow
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maxcalls = 255
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||||||
|
|
||||||
; Allow #exec entries in configuration files
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||||||
execincludes = yes | no
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||||||
|
|
||||||
; Don't over-inform the Asterisk sysadm, he's a guru
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dontwarn = yes | no
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|
||||||
|
|
||||||
; System name. Used to prefix CDR uniqueid and to fill \${SYSTEMNAME}
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||||||
systemname = <a_string>
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|
||||||
|
|
||||||
; Should language code be last component of sound file name or first?
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||||||
; when off, sound files are searched as <path>/<lang>/<file>
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||||||
; when on, sound files are search as <lang>/<path>/<file>
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||||||
; (only affects relative paths for sound files)
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||||||
languageprefix = yes | no
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||||||
|
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||||||
|
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||||||
[files]
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|
||||||
; Changing the following lines may compromise your security
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||||||
; Asterisk.ctl is the pipe that is used to connect the remote CLI
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|
||||||
; (asterisk -r) to Asterisk. Changing these settings change the
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||||||
; permissions and ownership of this file.
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||||||
; The file is created when Asterisk starts, in the "astrundir" above.
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||||||
|
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;astctlpermissions = 0660
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||||||
;astctlowner = root
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||||||
;astctlgroup = asterisk
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||||||
;astctl = asterisk.ctl
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||||||
|
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||||||
\end{verbatim}
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||||||
@ -0,0 +1,83 @@
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|||||||
|
Asterisk Main Configuration File
|
||||||
|
-----------------------------------------------------
|
||||||
|
Below is a sample of the main Asterisk configuration file,
|
||||||
|
asterisk.conf. Note that this file is _not_ provided in
|
||||||
|
sample form, because the Makefile creates it when needed
|
||||||
|
and does not touch it when it already exists.
|
||||||
|
|
||||||
|
---------------
|
||||||
|
|
||||||
|
[directories]
|
||||||
|
; Make sure these directories have the right permissions if not
|
||||||
|
; running Asterisk as root
|
||||||
|
|
||||||
|
; Where the configuration files (except for this one) are located
|
||||||
|
astetcdir => /etc/asterisk
|
||||||
|
|
||||||
|
; Where the Asterisk loadable modules are located
|
||||||
|
astmoddir => /usr/lib/asterisk/modules
|
||||||
|
|
||||||
|
; Where additional 'library' elements (scripts, etc.) are located
|
||||||
|
astvarlibdir => /var/lib/asterisk
|
||||||
|
|
||||||
|
; Where AGI scripts/programs are located
|
||||||
|
astagidir => /var/lib/asterisk/agi-bin
|
||||||
|
|
||||||
|
; Where spool directories are located
|
||||||
|
; Voicemail, monitor, dictation and other apps will create files here
|
||||||
|
; and outgoing call files (used with pbx_spool) must be placed here
|
||||||
|
astspooldir => /var/spool/asterisk
|
||||||
|
|
||||||
|
; Where the Asterisk process ID (pid) file should be created
|
||||||
|
astrundir => /var/run/asterisk
|
||||||
|
|
||||||
|
; Where the Asterisk log files should be created
|
||||||
|
astlogdir => /var/log/asterisk
|
||||||
|
|
||||||
|
|
||||||
|
[options]
|
||||||
|
;Under "options" you can enter configuration options
|
||||||
|
;that you also can set with command line options
|
||||||
|
|
||||||
|
verbose = 0 ; Verbosity level for logging (-v)
|
||||||
|
debug = 3 ; Debug: "No" or value (1-4)
|
||||||
|
nofork=yes | no ; Background execution disabled (-f)
|
||||||
|
alwaysfork=yes | no ; Always background, even with -v or -d (-F)
|
||||||
|
console= yes | no ; Console mode (-c)
|
||||||
|
highpriority = yes | no ; Execute with high priority (-p)
|
||||||
|
initcrypto = yes | no ; Initialize crypto at startup (-i)
|
||||||
|
nocolor = yes | no ; Disable ANSI colors (-n)
|
||||||
|
dumpcore = yes | no ; Dump core on failure (-g)
|
||||||
|
quiet = yes | no ; Run quietly (-q)
|
||||||
|
timestamp = yes | no ; Force timestamping in CLI verbose output (-T)
|
||||||
|
runuser = asterisk ; User to run asterisk as (-U) NOTE: will require changes to
|
||||||
|
; directory and device permissions
|
||||||
|
rungroup = asterisk ; Group to run asterisk as (-G)
|
||||||
|
internal_timing = yes | no ; Enable internal timing support (-I)
|
||||||
|
|
||||||
|
;These options have no command line equivalent
|
||||||
|
cache_record_files = yes | no ; Cache record() files in another directory until completion
|
||||||
|
record_cache_dir = <dir>
|
||||||
|
transcode_via_sln = yes | no ; Build transcode paths via SLINEAR
|
||||||
|
transmit_silence_during_record = yes | no ; send SLINEAR silence while channel is being recorded
|
||||||
|
maxload = 1.0 ; The maximum load average we accept calls for
|
||||||
|
maxcalls = 255 ; The maximum number of concurrent calls you want to allow
|
||||||
|
execincludes = yes | no ; Allow #exec entries in configuration files
|
||||||
|
dontwarn = yes | no ; Don't over-inform the Asterisk sysadm, he's a guru
|
||||||
|
systemname = <a_string> ; System name. Used to prefix CDR uniqueid and to fill ${SYSTEMNAME}
|
||||||
|
languageprefix = yes | no ; Should language code be last component of sound file name or first?
|
||||||
|
; when off, sound files are searched as <path>/<lang>/<file>
|
||||||
|
; when on, sound files are search as <lang>/<path>/<file>
|
||||||
|
; (only affects relative paths for sound files)
|
||||||
|
|
||||||
|
[files]
|
||||||
|
; Changing the following lines may compromise your security
|
||||||
|
; Asterisk.ctl is the pipe that is used to connect the remote CLI
|
||||||
|
; (asterisk -r) to Asterisk. Changing these settings change the
|
||||||
|
; permissions and ownership of this file.
|
||||||
|
; The file is created when Asterisk starts, in the "astrundir" above.
|
||||||
|
|
||||||
|
;astctlpermissions = 0660
|
||||||
|
;astctlowner = root
|
||||||
|
;astctlgroup = asterisk
|
||||||
|
;astctl = asterisk.ctl
|
||||||
@ -1,134 +0,0 @@
|
|||||||
% To generate a PDF from this, install the "rubber" tool, and the LaTeX
|
|
||||||
% dependencies for it. Then, run:
|
|
||||||
%
|
|
||||||
% rubber asterisk.tex
|
|
||||||
%
|
|
||||||
% http://www.pps.jussieu.fr/~beffara/soft/rubber/
|
|
||||||
|
|
||||||
\documentclass[12pt,a4]{report}
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|
||||||
\usepackage{hyperref}
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|
||||||
|
|
||||||
\author{Asterisk Development Team \\ Asterisk.org}
|
|
||||||
\title{Asterisk Reference Information \\ Version ASTERISKVERSION}
|
|
||||||
|
|
||||||
\begin{document}
|
|
||||||
\maketitle
|
|
||||||
|
|
||||||
\tableofcontents
|
|
||||||
|
|
||||||
\chapter{Introduction}
|
|
||||||
|
|
||||||
This document contains various pieces of information that are useful for
|
|
||||||
reference purposes.
|
|
||||||
|
|
||||||
\section{License Information}
|
|
||||||
\input{../LICENSE}
|
|
||||||
\subsection{Hold Music}
|
|
||||||
Digium has licensed the music included with
|
|
||||||
the Asterisk distribution From FreePlayMusic
|
|
||||||
for use and distribution with Asterisk. It
|
|
||||||
is licensed ONLY for use as hold music within
|
|
||||||
an Asterisk based PBX.
|
|
||||||
\section{Security}
|
|
||||||
\input{security.tex}
|
|
||||||
\section{Hardware}
|
|
||||||
\input{hardware.tex}
|
|
||||||
|
|
||||||
\chapter{Configuration}
|
|
||||||
\section{General Configuration Information}
|
|
||||||
\subsection{Configuration Parser}
|
|
||||||
\input{configuration.tex}
|
|
||||||
\subsection{Asterisk.conf}
|
|
||||||
\input{asterisk-conf.tex}
|
|
||||||
\subsection{CLI Prompt}
|
|
||||||
\input{cliprompt.tex}
|
|
||||||
\subsection{Extensions}
|
|
||||||
\input{extensions.tex}
|
|
||||||
\subsection{IP Type of Service}
|
|
||||||
\input{ip-tos.tex}
|
|
||||||
\subsection{MP3 Support}
|
|
||||||
\input{mp3.tex}
|
|
||||||
\subsection{ICES}
|
|
||||||
\input{ices.tex}
|
|
||||||
\section{Database Support}
|
|
||||||
\subsection{Realtime Database Configuration}
|
|
||||||
\input{realtime.tex}
|
|
||||||
\subsection{FreeTDS}
|
|
||||||
\input{freetds.tex}
|
|
||||||
\section{Privacy}
|
|
||||||
\input{privacy.tex}
|
|
||||||
|
|
||||||
\chapter{Channel Variables}
|
|
||||||
\input{channelvariables.tex}
|
|
||||||
|
|
||||||
\chapter{AEL, Asterisk Extension Language}
|
|
||||||
\input{ael.tex}
|
|
||||||
|
|
||||||
\chapter{SLA (Shared Line Appearances)}
|
|
||||||
\input{sla.tex}
|
|
||||||
|
|
||||||
\chapter{Channel Drivers}
|
|
||||||
\section{IAX2}
|
|
||||||
\input{chaniax.tex}
|
|
||||||
\subsection{IAX2 Jitterbuffer}
|
|
||||||
\input{jitterbuffer.tex}
|
|
||||||
\section{mISDN}
|
|
||||||
\input{misdn.tex}
|
|
||||||
\section{Local}
|
|
||||||
\input{localchannel.tex}
|
|
||||||
|
|
||||||
\chapter{Distributed Universal Number Discovery (DUNDi)}
|
|
||||||
\section{Introduction}
|
|
||||||
\input{dundi.tex}
|
|
||||||
\section{Peering Agreement}
|
|
||||||
\input{PEERING}
|
|
||||||
|
|
||||||
\chapter{ENUM}
|
|
||||||
\input{enum.tex}
|
|
||||||
|
|
||||||
\chapter{AMI: Asterisk Manager Interface}
|
|
||||||
\input{manager.tex}
|
|
||||||
\input{ajam.tex}
|
|
||||||
|
|
||||||
\chapter{CDR: Call Detail Records}
|
|
||||||
\input{billing.tex}
|
|
||||||
\input{cdrdriver.tex}
|
|
||||||
|
|
||||||
\chapter{Voicemail}
|
|
||||||
\section{ODBC Storage}
|
|
||||||
\input{odbcstorage.tex}
|
|
||||||
\section{IMAP Storage}
|
|
||||||
\input{imapstorage.tex}
|
|
||||||
|
|
||||||
\chapter{SMS}
|
|
||||||
\input{app-sms.tex}
|
|
||||||
|
|
||||||
\chapter{Queues}
|
|
||||||
\input{queues-with-callback-members.tex}
|
|
||||||
\section{Queue Logs}
|
|
||||||
\input{queuelog.tex}
|
|
||||||
|
|
||||||
% Generate this using the "core dumpappdocs" CLI command that is present
|
|
||||||
% when Asterisk is built with dev-mode enabled.
|
|
||||||
\chapter{Application Reference}
|
|
||||||
\input{ast_appdocs.tex}
|
|
||||||
|
|
||||||
% This is a list of files not yet integrated into this document:
|
|
||||||
%
|
|
||||||
%Misc
|
|
||||||
%----
|
|
||||||
%asterisk-mib.txt SNMP mib for Asterisk (net-snmp)
|
|
||||||
%digium-mib.txt SNMP mib for Asterisk (net-snmp)
|
|
||||||
%
|
|
||||||
%For developers
|
|
||||||
%--------------
|
|
||||||
%See http://www.asterisk.org/developers for more information
|
|
||||||
%
|
|
||||||
%backtrace.txt How to produce a backtrace when Asterisk crashes
|
|
||||||
%CODING-GUIDELINES Guidelines for developers
|
|
||||||
%externalivr.txt Documentation of the protocol used in externalivr()
|
|
||||||
%modules.txt How Asterisk modules work
|
|
||||||
%datastores.txt About channel data stores
|
|
||||||
%speechrec.txt The Generic Speech Recognition API
|
|
||||||
|
|
||||||
\enddocument
|
|
||||||
@ -1,87 +0,0 @@
|
|||||||
\section{Applications}
|
|
||||||
|
|
||||||
\begin{itemize}
|
|
||||||
\item SetAccount - Set account code for billing
|
|
||||||
\item SetAMAFlags - Sets AMA flags
|
|
||||||
\item NoCDR - Make sure no CDR is saved for a specific call
|
|
||||||
\item ResetCDR - Reset CDR
|
|
||||||
\item ForkCDR - Save current CDR and start a new CDR for this call
|
|
||||||
\item Authenticate - Authenticates and sets the account code
|
|
||||||
\item SetCDRUserField - Set CDR user field
|
|
||||||
\item AppendCDRUserField - Append data to CDR User field
|
|
||||||
\end{itemize}
|
|
||||||
|
|
||||||
For more information, use the "core show application <application>" command.
|
|
||||||
You can set default account codes and AMA flags for devices in
|
|
||||||
channel configuration files, like sip.conf, iax.conf etc.
|
|
||||||
|
|
||||||
|
|
||||||
\section{Fields of the CDR in Asterisk}
|
|
||||||
|
|
||||||
\begin{itemize}
|
|
||||||
\item accountcode: What account number to use, (string, 20 characters)
|
|
||||||
\item src: Caller*ID number (string, 80 characters)
|
|
||||||
\item dst: Destination extension (string, 80 characters)
|
|
||||||
\item dcontext: Destination context (string, 80 characters)
|
|
||||||
\item clid: Caller*ID with text (80 characters)
|
|
||||||
\item channel: Channel used (80 characters)
|
|
||||||
\item dstchannel: Destination channel if appropriate (80 characters)
|
|
||||||
\item lastapp: Last application if appropriate (80 characters)
|
|
||||||
\item lastdata: Last application data (arguments) (80 characters)
|
|
||||||
\item start: Start of call (date/time)
|
|
||||||
\item answer: Answer of call (date/time)
|
|
||||||
\item end: End of call (date/time)
|
|
||||||
\item duration: Total time in system, in seconds (integer), from dial to hangup
|
|
||||||
\item billsec: Total time call is up, in seconds (integer), from answer to hangup
|
|
||||||
\item disposition: What happened to the call: ANSWERED, NO ANSWER, BUSY
|
|
||||||
\item amaflags: What flags to use: DOCUMENTATION, BILL, IGNORE etc,
|
|
||||||
specified on a per channel basis like accountcode.
|
|
||||||
\item user field: A user-defined field, maximum 255 characters
|
|
||||||
\end{itemize}
|
|
||||||
|
|
||||||
In some cases, uniqueid is appended:
|
|
||||||
|
|
||||||
\begin{itemize}
|
|
||||||
\item uniqueid: Unique Channel Identifier (32 characters)
|
|
||||||
This needs to be enabled in the source code at compile time
|
|
||||||
\end{itemize}
|
|
||||||
|
|
||||||
NOTE: If you use IAX2 channels for your calls, and allow 'full' transfers
|
|
||||||
(not media-only transfers), then when the calls is transferred the server
|
|
||||||
in the middle will no longer be involved in the signaling path, and thus
|
|
||||||
will not generate accurate CDRs for that call. If you can, use media-only
|
|
||||||
transfers with IAX2 to avoid this problem, or turn off transfers completely
|
|
||||||
(although this can result in a media latency increase since the media packets
|
|
||||||
have to traverse the middle server(s) in the call).
|
|
||||||
|
|
||||||
\section{CDR Variables}
|
|
||||||
|
|
||||||
If the channel has a cdr, that cdr record has its own set of variables which
|
|
||||||
can be accessed just like channel variables. The following builtin variables
|
|
||||||
are available.
|
|
||||||
|
|
||||||
\begin{verbatim}
|
|
||||||
${CDR(clid)} Caller ID
|
|
||||||
${CDR(src)} Source
|
|
||||||
${CDR(dst)} Destination
|
|
||||||
${CDR(dcontext)} Destination context
|
|
||||||
${CDR(channel)} Channel name
|
|
||||||
${CDR(dstchannel)} Destination channel
|
|
||||||
${CDR(lastapp)} Last app executed
|
|
||||||
${CDR(lastdata)} Last app's arguments
|
|
||||||
${CDR(start)} Time the call started.
|
|
||||||
${CDR(answer)} Time the call was answered.
|
|
||||||
${CDR(end)} Time the call ended.
|
|
||||||
${CDR(duration)} Duration of the call.
|
|
||||||
${CDR(billsec)} Duration of the call once it was answered.
|
|
||||||
${CDR(disposition)} ANSWERED, NO ANSWER, BUSY
|
|
||||||
${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc
|
|
||||||
${CDR(accountcode)} The channel's account code.
|
|
||||||
${CDR(uniqueid)} The channel's unique id.
|
|
||||||
${CDR(userfield)} The channels uses specified field.
|
|
||||||
\end{verbatim}
|
|
||||||
|
|
||||||
In addition, you can set your own extra variables by using Set(CDR(name)=value).
|
|
||||||
These variables can be output into a text-format CDR by using the cdr\_custom
|
|
||||||
CDR driver; see the cdr\_custom.conf.sample file in the configs directory for
|
|
||||||
an example of how to do this.
|
|
||||||
@ -0,0 +1,105 @@
|
|||||||
|
Asterisk billing support - Call Detail Records
|
||||||
|
----------------------------------------------
|
||||||
|
Asterisk generates Call Detail Records in a database or in a comma
|
||||||
|
separated text file.
|
||||||
|
|
||||||
|
* cdr_csv supports comma separated text file storage, this is the
|
||||||
|
default driver
|
||||||
|
* cdr_manager supports CDR information via the AMI, The Asterisk Manager
|
||||||
|
interface
|
||||||
|
* cdr_odbc supports UnixODBC databases, see http://www.unixodbc.org
|
||||||
|
for an updated list of supported databases, from MySQL to MsSQL
|
||||||
|
and text files.
|
||||||
|
* cdr_tds supports FreeTDS databases (Among them MS SQL)
|
||||||
|
NOTE: Please read doc/freetds.txt for information on possible
|
||||||
|
problems with the FreeTDS driver
|
||||||
|
* cdr_sqlite supports SQlite
|
||||||
|
* cdr_pgsql supports PostgreSQL
|
||||||
|
|
||||||
|
In the asterisk-addons subversion repository, there's a cdr_mysql driver for
|
||||||
|
MySQL.
|
||||||
|
|
||||||
|
Applications
|
||||||
|
------------
|
||||||
|
|
||||||
|
* SetAccount Set account code for billing
|
||||||
|
* SetAMAFlags Sets AMA flags
|
||||||
|
* NoCDR Make sure no CDR is saved for a specific call
|
||||||
|
* ResetCDR Reset CDR
|
||||||
|
* ForkCDR Save current CDR and start a new CDR for this call
|
||||||
|
* Authenticate Authenticates and sets the account code
|
||||||
|
* SetCDRUserField Set CDR user field
|
||||||
|
* AppendCDRUserField Append data to CDR User field
|
||||||
|
|
||||||
|
For more information, use the "show application" command.
|
||||||
|
You can set default account codes and AMA flags for devices in
|
||||||
|
channel configuration files, like sip.conf, iax.conf etc.
|
||||||
|
|
||||||
|
|
||||||
|
Fields of the CDR in Asterisk
|
||||||
|
-----------------------------
|
||||||
|
|
||||||
|
1. accountcode: What account number to use, (string, 20 characters)
|
||||||
|
2. src: Caller*ID number (string, 80 characters)
|
||||||
|
3. dst: Destination extension (string, 80 characters)
|
||||||
|
4. dcontext: Destination context (string, 80 characters)
|
||||||
|
5. clid: Caller*ID with text (80 characters)
|
||||||
|
6. channel: Channel used (80 characters)
|
||||||
|
7. dstchannel: Destination channel if appropriate (80 characters)
|
||||||
|
8. lastapp: Last application if appropriate (80 characters)
|
||||||
|
9. lastdata: Last application data (arguments) (80 characters)
|
||||||
|
10. start: Start of call (date/time)
|
||||||
|
11. answer: Answer of call (date/time)
|
||||||
|
12. end: End of call (date/time)
|
||||||
|
13. duration: Total time in system, in seconds (integer), from dial to hangup
|
||||||
|
14. billsec: Total time call is up, in seconds (integer), from answer to hangup
|
||||||
|
15. disposition: What happened to the call: ANSWERED, NO ANSWER, BUSY
|
||||||
|
16. amaflags: What flags to use: DOCUMENTATION, BILL, IGNORE etc,
|
||||||
|
specified on a per channel basis like accountcode.
|
||||||
|
17. user field: A user-defined field, maximum 255 characters
|
||||||
|
|
||||||
|
In some cases, uniqueid is appended:
|
||||||
|
|
||||||
|
* uniqueid: Unique Channel Identifier (32 characters)
|
||||||
|
This needs to be enabled in the source code at compile time
|
||||||
|
|
||||||
|
|
||||||
|
NOTE: If you use IAX2 channels for your calls, and allow 'full' transfers
|
||||||
|
(not media-only transfers), then when the calls is transferred the server
|
||||||
|
in the middle will no longer be involved in the signaling path, and thus
|
||||||
|
will not generate accurate CDRs for that call. If you can, use media-only
|
||||||
|
transfers with IAX2 to avoid this problem, or turn off transfers completely
|
||||||
|
(although this can result in a media latency increase since the media packets
|
||||||
|
have to traverse the middle server(s) in the call).
|
||||||
|
|
||||||
|
____________________________________
|
||||||
|
CDR Variables
|
||||||
|
------------------------------------
|
||||||
|
|
||||||
|
If the channel has a cdr, that cdr record has its own set of variables which
|
||||||
|
can be accessed just like channel variables. The following builtin variables
|
||||||
|
are available.
|
||||||
|
|
||||||
|
${CDR(clid)} Caller ID
|
||||||
|
${CDR(src)} Source
|
||||||
|
${CDR(dst)} Destination
|
||||||
|
${CDR(dcontext)} Destination context
|
||||||
|
${CDR(channel)} Channel name
|
||||||
|
${CDR(dstchannel)} Destination channel
|
||||||
|
${CDR(lastapp)} Last app executed
|
||||||
|
${CDR(lastdata)} Last app's arguments
|
||||||
|
${CDR(start)} Time the call started.
|
||||||
|
${CDR(answer)} Time the call was answered.
|
||||||
|
${CDR(end)} Time the call ended.
|
||||||
|
${CDR(duration)} Duration of the call.
|
||||||
|
${CDR(billsec)} Duration of the call once it was answered.
|
||||||
|
${CDR(disposition)} ANSWERED, NO ANSWER, BUSY
|
||||||
|
${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc
|
||||||
|
${CDR(accountcode)} The channel's account code.
|
||||||
|
${CDR(uniqueid)} The channel's unique id.
|
||||||
|
${CDR(userfield)} The channels uses specified field.
|
||||||
|
|
||||||
|
In addition, you can set your own extra variables by using Set(CDR(name)=value).
|
||||||
|
These variables can be output into a text-format CDR by using the cdr_custom
|
||||||
|
CDR driver; see the cdr_custom.conf.sample file in the configs directory for
|
||||||
|
an example of how to do this.
|
||||||
@ -0,0 +1,18 @@
|
|||||||
|
Caller ID presentation values
|
||||||
|
-----------------------------
|
||||||
|
|
||||||
|
In some channels it is possible to set Caller ID presentation for a device. It is
|
||||||
|
also possible to set the presentation for an active channel in the dial plan
|
||||||
|
with the setcallerpres() application.
|
||||||
|
|
||||||
|
Valid values are:
|
||||||
|
|
||||||
|
allowed_not_screened : Presentation Allowed, Not Screened
|
||||||
|
allowed_passed_screen : Presentation Allowed, Passed Screen
|
||||||
|
allowed_failed_screen : Presentation Allowed, Failed Screen
|
||||||
|
allowed : Presentation Allowed, Network Number
|
||||||
|
prohib_not_screened : Presentation Prohibited, Not Screened
|
||||||
|
prohib_passed_screen : Presentation Prohibited, Passed Screen
|
||||||
|
prohib_failed_screen : Presentation Prohibited, Failed Screen
|
||||||
|
prohib : Presentation Prohibited, Network Number
|
||||||
|
unavailable : Number Unavailable
|
||||||
@ -1,431 +0,0 @@
|
|||||||
Call data records can be stored in many different databases or even CSV text.
|
|
||||||
|
|
||||||
\section{MSSQL}
|
|
||||||
|
|
||||||
Asterisk can currently store CDRs into an MSSQL database in
|
|
||||||
two different ways: cdr\_odbc or cdr\_tds
|
|
||||||
|
|
||||||
Call Data Records can be stored using unixODBC (which requires
|
|
||||||
the FreeTDS package) [cdr\_odbc] or directly by using just the
|
|
||||||
FreeTDS package [cdr\_tds] The following provide some
|
|
||||||
examples known to get asterisk working with mssql.
|
|
||||||
|
|
||||||
NOTE: Only choose one db connector.
|
|
||||||
|
|
||||||
\subsection{ODBC using cdr\_odbc}
|
|
||||||
Compile, configure, and install the latest unixODBC package:
|
|
||||||
\begin{verbatim}
|
|
||||||
tar -zxvf unixODBC-2.2.9.tar.gz &&
|
|
||||||
cd unixODBC-2.2.9 &&
|
|
||||||
./configure --sysconfdir=/etc --prefix=/usr --disable-gui &&
|
|
||||||
make &&
|
|
||||||
make install
|
|
||||||
\end{verbatim}
|
|
||||||
|
|
||||||
Compile, configure, and install the latest FreeTDS package:
|
|
||||||
\begin{verbatim}
|
|
||||||
tar -zxvf freetds-0.62.4.tar.gz &&
|
|
||||||
cd freetds-0.62.4 &&
|
|
||||||
./configure --prefix=/usr --with-tdsver=7.0 \
|
|
||||||
--with-unixodbc=/usr/lib &&
|
|
||||||
make && make install
|
|
||||||
\end{verbatim}
|
|
||||||
|
|
||||||
Compile, or recompile, asterisk so that it will now add support
|
|
||||||
for cdr\_odbc.
|
|
||||||
\begin{verbatim}
|
|
||||||
make clean && ./configure --with-odbc &&
|
|
||||||
make update &&
|
|
||||||
make &&
|
|
||||||
make install
|
|
||||||
\end{verbatim}
|
|
||||||
|
|
||||||
Setup odbc configuration files. These are working examples
|
|
||||||
from my system. You will need to modify for your setup.
|
|
||||||
You are not required to store usernames or passwords here.
|
|
||||||
|
|
||||||
\begin{verbatim}
|
|
||||||
/etc/odbcinst.ini
|
|
||||||
[FreeTDS]
|
|
||||||
Description = FreeTDS ODBC driver for MSSQL
|
|
||||||
Driver = /usr/lib/libtdsodbc.so
|
|
||||||
Setup = /usr/lib/libtdsS.so
|
|
||||||
FileUsage = 1
|
|
||||||
|
|
||||||
/etc/odbc.ini
|
|
||||||
[MSSQL-asterisk]
|
|
||||||
description = Asterisk ODBC for MSSQL
|
|
||||||
driver = FreeTDS
|
|
||||||
server = 192.168.1.25
|
|
||||||
port = 1433
|
|
||||||
database = voipdb
|
|
||||||
tds_version = 7.0
|
|
||||||
language = us_english
|
|
||||||
\end{verbatim}
|
|
||||||
|
|
||||||
Only install one database connector. Do not confuse asterisk
|
|
||||||
by using both ODBC (cdr\_odbc) and FreeTDS (cdr\_tds).
|
|
||||||
This command will erase the contents of cdr\_tds.conf
|
|
||||||
\begin{verbatim}
|
|
||||||
[ -f /etc/asterisk/cdr_tds.conf ] > /etc/asterisk/cdr_tds.conf
|
|
||||||
\end{verbatim}
|
|
||||||
NOTE: unixODBC requires the freeTDS package, but asterisk does
|
|
||||||
not call freeTDS directly.
|
|
||||||
|
|
||||||
Now set up cdr\_odbc configuration files. These are working samples
|
|
||||||
from my system. You will need to modify for your setup. Define
|
|
||||||
your usernames and passwords here, secure file as well.
|
|
||||||
\begin{verbatim}
|
|
||||||
/etc/asterisk/cdr_odbc.conf
|
|
||||||
[global]
|
|
||||||
dsn=MSSQL-asterisk
|
|
||||||
username=voipdbuser
|
|
||||||
password=voipdbpass
|
|
||||||
loguniqueid=yes
|
|
||||||
\end{verbatim}
|
|
||||||
And finally, create the 'cdr' table in your mssql database.
|
|
||||||
\begin{verbatim}
|
|
||||||
CREATE TABLE cdr (
|
|
||||||
[calldate] [datetime] NOT NULL ,
|
|
||||||
[clid] [varchar] (80) NOT NULL ,
|
|
||||||
[src] [varchar] (80) NOT NULL ,
|
|
||||||
[dst] [varchar] (80) NOT NULL ,
|
|
||||||
[dcontext] [varchar] (80) NOT NULL ,
|
|
||||||
[channel] [varchar] (80) NOT NULL ,
|
|
||||||
[dstchannel] [varchar] (80) NOT NULL ,
|
|
||||||
[lastapp] [varchar] (80) NOT NULL ,
|
|
||||||
[lastdata] [varchar] (80) NOT NULL ,
|
|
||||||
[duration] [int] NOT NULL ,
|
|
||||||
[billsec] [int] NOT NULL ,
|
|
||||||
[disposition] [varchar] (45) NOT NULL ,
|
|
||||||
[amaflags] [int] NOT NULL ,
|
|
||||||
[accountcode] [varchar] (20) NOT NULL ,
|
|
||||||
[uniqueid] [varchar] (32) NOT NULL ,
|
|
||||||
[userfield] [varchar] (255) NOT NULL
|
|
||||||
)
|
|
||||||
\end{verbatim}
|
|
||||||
Start asterisk in verbose mode, you should see that asterisk
|
|
||||||
logs a connection to the database and will now record every
|
|
||||||
call to the database when it's complete.
|
|
||||||
|
|
||||||
\subsection{TDS, using cdr\_tds}
|
|
||||||
Compile, configure, and install the latest FreeTDS package:
|
|
||||||
\begin{verbatim}
|
|
||||||
tar -zxvf freetds-0.62.4.tar.gz &&
|
|
||||||
cd freetds-0.62.4 &&
|
|
||||||
./configure --prefix=/usr --with-tdsver=7.0
|
|
||||||
make &&
|
|
||||||
make install
|
|
||||||
\end{verbatim}
|
|
||||||
Compile, or recompile, asterisk so that it will now add support
|
|
||||||
for cdr\_tds.
|
|
||||||
\begin{verbatim}
|
|
||||||
make clean && ./configure --with-tds &&
|
|
||||||
make update &&
|
|
||||||
make &&
|
|
||||||
make install
|
|
||||||
\end{verbatim}
|
|
||||||
Only install one database connector. Do not confuse asterisk
|
|
||||||
by using both ODBC (cdr\_odbc) and FreeTDS (cdr\_tds).
|
|
||||||
This command will erase the contents of cdr\_odbc.conf
|
|
||||||
\begin{verbatim}
|
|
||||||
[ -f /etc/asterisk/cdr_odbc.conf ] > /etc/asterisk/cdr_odbc.conf
|
|
||||||
\end{verbatim}
|
|
||||||
Setup cdr\_tds configuration files. These are working samples
|
|
||||||
from my system. You will need to modify for your setup. Define
|
|
||||||
your usernames and passwords here, secure file as well.
|
|
||||||
\begin{verbatim}
|
|
||||||
/etc/asterisk/cdr_tds.conf
|
|
||||||
[global]
|
|
||||||
hostname=192.168.1.25
|
|
||||||
port=1433
|
|
||||||
dbname=voipdb
|
|
||||||
user=voipdbuser
|
|
||||||
password=voipdpass
|
|
||||||
charset=BIG5
|
|
||||||
\end{verbatim}
|
|
||||||
And finally, create the 'cdr' table in your mssql database.
|
|
||||||
\begin{verbatim}
|
|
||||||
CREATE TABLE cdr (
|
|
||||||
[accountcode] [varchar] (20) NULL ,
|
|
||||||
[src] [varchar] (80) NULL ,
|
|
||||||
[dst] [varchar] (80) NULL ,
|
|
||||||
[dcontext] [varchar] (80) NULL ,
|
|
||||||
[clid] [varchar] (80) NULL ,
|
|
||||||
[channel] [varchar] (80) NULL ,
|
|
||||||
[dstchannel] [varchar] (80) NULL ,
|
|
||||||
[lastapp] [varchar] (80) NULL ,
|
|
||||||
[lastdata] [varchar] (80) NULL ,
|
|
||||||
[start] [datetime] NULL ,
|
|
||||||
[answer] [datetime] NULL ,
|
|
||||||
[end] [datetime] NULL ,
|
|
||||||
[duration] [int] NULL ,
|
|
||||||
[billsec] [int] NULL ,
|
|
||||||
[disposition] [varchar] (20) NULL ,
|
|
||||||
[amaflags] [varchar] (16) NULL ,
|
|
||||||
[uniqueid] [varchar] (32) NULL
|
|
||||||
)
|
|
||||||
\end{verbatim}
|
|
||||||
Start asterisk in verbose mode, you should see that asterisk
|
|
||||||
logs a connection to the database and will now record every
|
|
||||||
call to the database when it's complete.
|
|
||||||
|
|
||||||
|
|
||||||
\section{MYSQL}
|
|
||||||
|
|
||||||
Using MySQL for CDR records is supported by using ODBC and the cdr\_odbc module.
|
|
||||||
|
|
||||||
\section{PGSQL}
|
|
||||||
If you want to go directly to postgresql database, and have the cdr\_pgsql.so
|
|
||||||
compiled you can use the following sample setup.
|
|
||||||
On Debian, before compiling asterisk, just install libpqxx-dev.
|
|
||||||
Other distros will likely have a similiar package.
|
|
||||||
|
|
||||||
Once you have the compile done,
|
|
||||||
copy the sample cdr\_pgsql.conf file or create your own.
|
|
||||||
|
|
||||||
Here is a sample:
|
|
||||||
\begin{verbatim}
|
|
||||||
/etc/asterisk/cdr_pgsql.conf
|
|
||||||
; Sample Asterisk config file for CDR logging to PostgresSQL
|
|
||||||
[global]
|
|
||||||
hostname=localhost
|
|
||||||
port=5432
|
|
||||||
dbname=asterisk
|
|
||||||
password=password
|
|
||||||
user=postgres
|
|
||||||
table=cdr
|
|
||||||
\end{verbatim}
|
|
||||||
Now create a table in postgresql for your cdrs
|
|
||||||
|
|
||||||
\begin{verbatim}
|
|
||||||
CREATE TABLE cdr (
|
|
||||||
calldate time NOT NULL ,
|
|
||||||
clid varchar (80) NOT NULL ,
|
|
||||||
src varchar (80) NOT NULL ,
|
|
||||||
dst varchar (80) NOT NULL ,
|
|
||||||
dcontext varchar (80) NOT NULL ,
|
|
||||||
channel varchar (80) NOT NULL ,
|
|
||||||
dstchannel varchar (80) NOT NULL ,
|
|
||||||
lastapp varchar (80) NOT NULL ,
|
|
||||||
lastdata varchar (80) NOT NULL ,
|
|
||||||
duration int NOT NULL ,
|
|
||||||
billsec int NOT NULL ,
|
|
||||||
disposition varchar (45) NOT NULL ,
|
|
||||||
amaflags int NOT NULL ,
|
|
||||||
accountcode varchar (20) NOT NULL ,
|
|
||||||
uniqueid varchar (32) NOT NULL ,
|
|
||||||
userfield varchar (255) NOT NULL
|
|
||||||
);
|
|
||||||
\end{verbatim}
|
|
||||||
|
|
||||||
\section{SQLLITE}
|
|
||||||
|
|
||||||
SQLite version 2 is supported in cdr\_sqlite.
|
|
||||||
|
|
||||||
\section{RADIUS}
|
|
||||||
|
|
||||||
\subsection{What is needed}
|
|
||||||
|
|
||||||
\begin{itemize}
|
|
||||||
\item FreeRADIUS server
|
|
||||||
\item Radiusclient-ng library
|
|
||||||
\item Asterisk PBX
|
|
||||||
\end{itemize}
|
|
||||||
|
|
||||||
\begin{verbatim}
|
|
||||||
+--------------------+
|
|
||||||
| Asterisk PBX |
|
|
||||||
| |
|
|
||||||
|********************|
|
|
||||||
| | +---------------+
|
|
||||||
| RADIUS client |------->| RADIUS server |
|
|
||||||
| |<-------| (FreeRADIUS) |
|
|
||||||
+--------------------+ +---------------+
|
|
||||||
\end{verbatim}
|
|
||||||
|
|
||||||
|
|
||||||
|
|
||||||
\subsection{Steps to follow in order to have RADIUS support}
|
|
||||||
|
|
||||||
\subsubsection{Installation of the Radiusclient library}
|
|
||||||
Installation:
|
|
||||||
\begin{verbatim}
|
|
||||||
Download the sources from:
|
|
||||||
|
|
||||||
http://developer.berlios.de/projects/radiusclient-ng/
|
|
||||||
|
|
||||||
Untar the source tarball.
|
|
||||||
root@localhost:/usr/local/src# tar xvfz radiusclient-ng-0.5.2.tar.gz
|
|
||||||
|
|
||||||
Compile and install the library.
|
|
||||||
root@localhost:/usr/local/src# cd radiusclient-ng-0.5.2
|
|
||||||
root@localhost:/usr/local/src/radiusclient-ng-0.5.2# ./configure
|
|
||||||
root@localhost:/usr/local/src/radiusclient-ng-0.5.2# make
|
|
||||||
root@localhost:/usr/local/src/radiusclient-ng-0.5.2# make install
|
|
||||||
\end{verbatim}
|
|
||||||
|
|
||||||
\subsubsection{Configuration of the Radiusclient library}
|
|
||||||
|
|
||||||
By default all the configuration files of the radiusclient library will
|
|
||||||
be in /usr/local/etc/radiusclient-ng directory.
|
|
||||||
|
|
||||||
File "radiusclient.conf"
|
|
||||||
Open the file and find lines containing the following:
|
|
||||||
|
|
||||||
authserver localhost
|
|
||||||
|
|
||||||
This is the hostname or IP address of the RADIUS server used for
|
|
||||||
authentication. You will have to change this unless the server is
|
|
||||||
running on the same host as your Asterisk PBX.
|
|
||||||
|
|
||||||
acctserver localhost
|
|
||||||
This is the hostname or IP address of the RADIUS server used for
|
|
||||||
accounting. You will have to change this unless the server is running
|
|
||||||
on the same host as your Asterisk PBX.
|
|
||||||
|
|
||||||
File "servers"
|
|
||||||
|
|
||||||
RADIUS protocol uses simple access control mechanism based on shared
|
|
||||||
secrets that allows RADIUS servers to limit access from RADIUS clients.
|
|
||||||
|
|
||||||
A RADIUS server is configured with a secret string and only RADIUS
|
|
||||||
clients that have the same secret will be accepted.
|
|
||||||
|
|
||||||
You need to configure a shared secret for each server you have
|
|
||||||
configured in radiusclient.conf file in the previous step. The shared
|
|
||||||
secrets are stored in /usr/local/etc/radiusclient-ng/servers file.
|
|
||||||
|
|
||||||
Each line contains hostname of a RADIUS server and shared secret
|
|
||||||
used in communication with that server. The two values are separated
|
|
||||||
by white spaces. Configure shared secrets for every RADIUS server you
|
|
||||||
are going to use.
|
|
||||||
|
|
||||||
File "dictionary"
|
|
||||||
|
|
||||||
Asterisk uses some attributes that are not included in the
|
|
||||||
dictionary of radiusclient library, therefore it is necessary to add
|
|
||||||
them. A file called dictionary.digium (kept in the contrib dir)
|
|
||||||
was created to list all new attributes used by Asterisk.
|
|
||||||
Add to the end of the main dictionary file
|
|
||||||
/usr/local/etc/radiusclient-ng/dictionary
|
|
||||||
the line:
|
|
||||||
\begin{verbatim}
|
|
||||||
\$INCLUDE /path/to/dictionary.digium
|
|
||||||
\end{verbatim}
|
|
||||||
|
|
||||||
\subsubsection{Install FreeRADIUS Server (Version 1.1.1)}
|
|
||||||
|
|
||||||
Download sources tarball from:
|
|
||||||
|
|
||||||
http://freeradius.org/
|
|
||||||
|
|
||||||
Untar, configure, build, and install the server:
|
|
||||||
|
|
||||||
\begin{verbatim}
|
|
||||||
root@localhost:/usr/local/src# tar xvfz freeradius-1.1.1.tar.gz
|
|
||||||
root@localhost:/usr/local/src# cd freeradius-1.1.1
|
|
||||||
root@localhost"/usr/local/src/freeradius-1.1.1# ./configure
|
|
||||||
root@localhost"/usr/local/src/freeradius-1.1.1# make
|
|
||||||
root@localhost"/usr/local/src/freeradius-1.1.1# make install
|
|
||||||
\end{verbatim}
|
|
||||||
|
|
||||||
All the configuration files of FreeRADIUS server will be in
|
|
||||||
/usr/local/etc/raddb directory.
|
|
||||||
|
|
||||||
|
|
||||||
\subsubsection{Configuration of the FreeRADIUS Server}
|
|
||||||
|
|
||||||
There are several files that have to be modified to configure the
|
|
||||||
RADIUS server. These are presented next.
|
|
||||||
|
|
||||||
File "clients.conf"
|
|
||||||
|
|
||||||
File /usr/local/etc/raddb/clients.conf contains description of
|
|
||||||
RADIUS clients that are allowed to use the server. For each of the
|
|
||||||
clients you need to specify its hostname or IP address and also a
|
|
||||||
shared secret. The shared secret must be the same string you configured
|
|
||||||
in radiusclient library.
|
|
||||||
|
|
||||||
Example:
|
|
||||||
\begin{verbatim}
|
|
||||||
client myhost {
|
|
||||||
secret = mysecret
|
|
||||||
shortname = foo
|
|
||||||
}
|
|
||||||
\end{verbatim}
|
|
||||||
|
|
||||||
This fragment allows access from RADIUS clients on "myhost" if they use
|
|
||||||
"mysecret" as the shared secret.
|
|
||||||
The file already contains an entry for localhost (127.0.0.1), so if you
|
|
||||||
are running the RADIUS server on the same host as your Asterisk server,
|
|
||||||
then modify the existing entry instead, replacing the default password.
|
|
||||||
|
|
||||||
File "dictionary"
|
|
||||||
|
|
||||||
Note : as of version 1.1.2, the dictionary.digium file ships with FreeRADIUS.
|
|
||||||
The following procedure brings the dictionary.digium file to previous versions
|
|
||||||
of FreeRADIUS.
|
|
||||||
|
|
||||||
File /usr/local/etc/raddb/dictionary contains the dictionary of
|
|
||||||
FreeRADIUS server. You have to add the same dictionary file
|
|
||||||
(dictionary.digium), which you added to the dictionary of radiusclient-ng
|
|
||||||
library. You can include it into the main file, adding the following line at the
|
|
||||||
end of file '/usr/local/etc/raddb/dictionary':
|
|
||||||
|
|
||||||
\$INCLUDE /path/to/dictionary.digium
|
|
||||||
|
|
||||||
That will include the same new attribute definitions that are used
|
|
||||||
in radiusclient-ng library so the client and server will understand each
|
|
||||||
other.
|
|
||||||
|
|
||||||
|
|
||||||
\subsubsection{Asterisk Accounting Configuration}
|
|
||||||
|
|
||||||
Compilation and installation:
|
|
||||||
|
|
||||||
The module will be compiled as long as the radiusclient-ng
|
|
||||||
library has been detected on your system.
|
|
||||||
|
|
||||||
By default FreeRADIUS server will log all accounting requests into
|
|
||||||
/usr/local/var/log/radius/radacct directory in form of plain text files.
|
|
||||||
The server will create one file for each hostname in the directory. The
|
|
||||||
following example shows how the log files look like.
|
|
||||||
|
|
||||||
Asterisk now generates Call Detail Records. See /include/asterisk/cdr.h
|
|
||||||
for all the fields which are recorded. By default, records in comma
|
|
||||||
separated values will be created in /var/log/asterisk/cdr-csv.
|
|
||||||
|
|
||||||
The configuration file for cdr\_radius.so module is :
|
|
||||||
|
|
||||||
/etc/asterisk/cdr.conf
|
|
||||||
This is where you can set CDR related parameters as well as the path to
|
|
||||||
the radiusclient-ng library configuration file.
|
|
||||||
|
|
||||||
|
|
||||||
\section{Logged Values}
|
|
||||||
\begin{verbatim}
|
|
||||||
"Asterisk-Acc-Code", The account name of detail records
|
|
||||||
"Asterisk-Src",
|
|
||||||
"Asterisk-Dst",
|
|
||||||
"Asterisk-Dst-Ctx", The destination context
|
|
||||||
"Asterisk-Clid",
|
|
||||||
"Asterisk-Chan", The channel
|
|
||||||
"Asterisk-Dst-Chan", (if applicable)
|
|
||||||
"Asterisk-Last-App", Last application run on the channel
|
|
||||||
"Asterisk-Last-Data", Argument to the last channel
|
|
||||||
"Asterisk-Start-Time",
|
|
||||||
"Asterisk-Answer-Time",
|
|
||||||
"Asterisk-End-Time",
|
|
||||||
"Asterisk-Duration", Duration is the whole length that the entire
|
|
||||||
call lasted. ie. call rx'd to hangup
|
|
||||||
"end time" minus "start time"
|
|
||||||
"Asterisk-Bill-Sec", The duration that a call was up after other
|
|
||||||
end answered which will be <= to duration
|
|
||||||
"end time" minus "answer time"
|
|
||||||
"Asterisk-Disposition", ANSWERED, NO ANSWER, BUSY
|
|
||||||
"Asterisk-AMA-Flags", DOCUMENTATION, BILL, IGNORE etc, specified on
|
|
||||||
a per channel basis like accountcode.
|
|
||||||
"Asterisk-Unique-ID", Unique call identifier
|
|
||||||
"Asterisk-User-Field" User field set via SetCDRUserField
|
|
||||||
\end{verbatim}
|
|
||||||
@ -0,0 +1,215 @@
|
|||||||
|
Call data records can be stored in many different databases or even CSV text.
|
||||||
|
|
||||||
|
MSSQL: Asterisk can currently store CDRs into an MSSQL database in
|
||||||
|
two different ways: cdr_odbc.c or cdr_tds.c
|
||||||
|
|
||||||
|
Call Data Records can be stored using unixODBC (which requires
|
||||||
|
the FreeTDS package) [cdr_odbc.c] or directly by using just the
|
||||||
|
FreeTDS package [cdr_tds.c] The following provide some
|
||||||
|
examples known to get asterisk working with mssql.
|
||||||
|
NOTE: Only choose one db connector.
|
||||||
|
|
||||||
|
ODBC [cdr_odbc.c]:
|
||||||
|
Compile, configure, and install the latest unixODBC package:
|
||||||
|
tar -zxvf unixODBC-2.2.9.tar.gz &&
|
||||||
|
cd unixODBC-2.2.9 &&
|
||||||
|
./configure --sysconfdir=/etc --prefix=/usr --disable-gui &&
|
||||||
|
make &&
|
||||||
|
make install
|
||||||
|
|
||||||
|
Compile, configure, and install the latest FreeTDS package:
|
||||||
|
tar -zxvf freetds-0.62.4.tar.gz &&
|
||||||
|
cd freetds-0.62.4 &&
|
||||||
|
./configure --prefix=/usr --with-tdsver=7.0 \
|
||||||
|
--with-unixodbc=/usr/lib &&
|
||||||
|
make &&
|
||||||
|
make install
|
||||||
|
|
||||||
|
Compile, or recompile, asterisk so that it will now add support
|
||||||
|
for cdr_odbc.c
|
||||||
|
|
||||||
|
make clean &&
|
||||||
|
make update &&
|
||||||
|
make &&
|
||||||
|
make install
|
||||||
|
|
||||||
|
Setup odbc configuration files. These are working examples
|
||||||
|
from my system. You will need to modify for your setup.
|
||||||
|
You are not required to store usernames or passwords here.
|
||||||
|
|
||||||
|
/etc/odbcinst.ini
|
||||||
|
[FreeTDS]
|
||||||
|
Description = FreeTDS ODBC driver for MSSQL
|
||||||
|
Driver = /usr/lib/libtdsodbc.so
|
||||||
|
Setup = /usr/lib/libtdsS.so
|
||||||
|
FileUsage = 1
|
||||||
|
|
||||||
|
/etc/odbc.ini
|
||||||
|
[MSSQL-asterisk]
|
||||||
|
description = Asterisk ODBC for MSSQL
|
||||||
|
driver = FreeTDS
|
||||||
|
server = 192.168.1.25
|
||||||
|
port = 1433
|
||||||
|
database = voipdb
|
||||||
|
tds_version = 7.0
|
||||||
|
language = us_english
|
||||||
|
|
||||||
|
Only install one database connector. Do not confuse asterisk
|
||||||
|
by using both ODBC (cdr_odbc.c) and FreeTDS (cdr_tds.c).
|
||||||
|
This command will erase the contents of cdr_tds.conf
|
||||||
|
|
||||||
|
[ -f /etc/asterisk/cdr_tds.conf ] > /etc/asterisk/cdr_tds.conf
|
||||||
|
|
||||||
|
NOTE: unixODBC requires the freeTDS package, but asterisk does
|
||||||
|
not call freeTDS directly.
|
||||||
|
|
||||||
|
Setup cdr_odbc configuration files. These are working samples
|
||||||
|
from my system. You will need to modify for your setup. Define
|
||||||
|
your usernames and passwords here, secure file as well.
|
||||||
|
|
||||||
|
/etc/asterisk/cdr_odbc.conf
|
||||||
|
[global]
|
||||||
|
dsn=MSSQL-asterisk
|
||||||
|
username=voipdbuser
|
||||||
|
password=voipdbpass
|
||||||
|
loguniqueid=yes
|
||||||
|
|
||||||
|
And finally, create the 'cdr' table in your mssql database.
|
||||||
|
|
||||||
|
CREATE TABLE cdr (
|
||||||
|
[calldate] [datetime] NOT NULL ,
|
||||||
|
[clid] [varchar] (80) NOT NULL ,
|
||||||
|
[src] [varchar] (80) NOT NULL ,
|
||||||
|
[dst] [varchar] (80) NOT NULL ,
|
||||||
|
[dcontext] [varchar] (80) NOT NULL ,
|
||||||
|
[channel] [varchar] (80) NOT NULL ,
|
||||||
|
[dstchannel] [varchar] (80) NOT NULL ,
|
||||||
|
[lastapp] [varchar] (80) NOT NULL ,
|
||||||
|
[lastdata] [varchar] (80) NOT NULL ,
|
||||||
|
[duration] [int] NOT NULL ,
|
||||||
|
[billsec] [int] NOT NULL ,
|
||||||
|
[disposition] [varchar] (45) NOT NULL ,
|
||||||
|
[amaflags] [int] NOT NULL ,
|
||||||
|
[accountcode] [varchar] (20) NOT NULL ,
|
||||||
|
[uniqueid] [varchar] (32) NOT NULL ,
|
||||||
|
[userfield] [varchar] (255) NOT NULL
|
||||||
|
)
|
||||||
|
|
||||||
|
Start asterisk in verbose mode, you should see that asterisk
|
||||||
|
logs a connection to the database and will now record every
|
||||||
|
call to the database when it's complete.
|
||||||
|
|
||||||
|
TDS [cdr_tds.c]:
|
||||||
|
Compile, configure, and install the latest FreeTDS package:
|
||||||
|
tar -zxvf freetds-0.62.4.tar.gz &&
|
||||||
|
cd freetds-0.62.4 &&
|
||||||
|
./configure --prefix=/usr --with-tdsver=7.0
|
||||||
|
make &&
|
||||||
|
make install
|
||||||
|
|
||||||
|
Compile, or recompile, asterisk so that it will now add support
|
||||||
|
for cdr_tds.c
|
||||||
|
|
||||||
|
make clean &&
|
||||||
|
make update &&
|
||||||
|
make &&
|
||||||
|
make install
|
||||||
|
|
||||||
|
Only install one database connector. Do not confuse asterisk
|
||||||
|
by using both ODBC (cdr_odbc.c) and FreeTDS (cdr_tds.c).
|
||||||
|
This command will erase the contents of cdr_odbc.conf
|
||||||
|
|
||||||
|
[ -f /etc/asterisk/cdr_odbc.conf ] > /etc/asterisk/cdr_odbc.conf
|
||||||
|
|
||||||
|
Setup cdr_tds configuration files. These are working samples
|
||||||
|
from my system. You will need to modify for your setup. Define
|
||||||
|
your usernames and passwords here, secure file as well.
|
||||||
|
|
||||||
|
/etc/asterisk/cdr_tds.conf
|
||||||
|
[global]
|
||||||
|
hostname=192.168.1.25
|
||||||
|
port=1433
|
||||||
|
dbname=voipdb
|
||||||
|
user=voipdbuser
|
||||||
|
password=voipdpass
|
||||||
|
charset=BIG5
|
||||||
|
|
||||||
|
And finally, create the 'cdr' table in your mssql database.
|
||||||
|
|
||||||
|
CREATE TABLE cdr (
|
||||||
|
[accountcode] [varchar] (20) NULL ,
|
||||||
|
[src] [varchar] (80) NULL ,
|
||||||
|
[dst] [varchar] (80) NULL ,
|
||||||
|
[dcontext] [varchar] (80) NULL ,
|
||||||
|
[clid] [varchar] (80) NULL ,
|
||||||
|
[channel] [varchar] (80) NULL ,
|
||||||
|
[dstchannel] [varchar] (80) NULL ,
|
||||||
|
[lastapp] [varchar] (80) NULL ,
|
||||||
|
[lastdata] [varchar] (80) NULL ,
|
||||||
|
[start] [datetime] NULL ,
|
||||||
|
[answer] [datetime] NULL ,
|
||||||
|
[end] [datetime] NULL ,
|
||||||
|
[duration] [int] NULL ,
|
||||||
|
[billsec] [int] NULL ,
|
||||||
|
[disposition] [varchar] (20) NULL ,
|
||||||
|
[amaflags] [varchar] (16) NULL ,
|
||||||
|
[uniqueid] [varchar] (32) NULL
|
||||||
|
)
|
||||||
|
|
||||||
|
Start asterisk in verbose mode, you should see that asterisk
|
||||||
|
logs a connection to the database and will now record every
|
||||||
|
call to the database when it's complete.
|
||||||
|
|
||||||
|
|
||||||
|
MYSQL:
|
||||||
|
|
||||||
|
|
||||||
|
PGSQL:
|
||||||
|
If you want to go directly to postgresql database, and have the cdr_pgsql.so
|
||||||
|
compiled you can use the following sample setup.
|
||||||
|
On Debian, before compiling asterisk, just install libpqxx-dev.
|
||||||
|
Other distros will likely have a similiar package.
|
||||||
|
|
||||||
|
Once you have the compile done,
|
||||||
|
copy the sample cdr_pgsql.conf file or create your own.
|
||||||
|
|
||||||
|
Here is a sample:
|
||||||
|
|
||||||
|
/etc/asterisk/cdr_pgsql.conf
|
||||||
|
; Sample Asterisk config file for CDR logging to PostgresSQL
|
||||||
|
[global]
|
||||||
|
hostname=localhost
|
||||||
|
port=5432
|
||||||
|
dbname=asterisk
|
||||||
|
password=password
|
||||||
|
user=postgres
|
||||||
|
table=cdr
|
||||||
|
|
||||||
|
;Now create a table in postgresql for your cdrs
|
||||||
|
|
||||||
|
;SQL table where CDRs will be inserted
|
||||||
|
;Copy and paste this into your postgresql prompt.
|
||||||
|
CREATE TABLE cdr (
|
||||||
|
calldate time NOT NULL ,
|
||||||
|
clid varchar (80) NOT NULL ,
|
||||||
|
src varchar (80) NOT NULL ,
|
||||||
|
dst varchar (80) NOT NULL ,
|
||||||
|
dcontext varchar (80) NOT NULL ,
|
||||||
|
channel varchar (80) NOT NULL ,
|
||||||
|
dstchannel varchar (80) NOT NULL ,
|
||||||
|
lastapp varchar (80) NOT NULL ,
|
||||||
|
lastdata varchar (80) NOT NULL ,
|
||||||
|
duration int NOT NULL ,
|
||||||
|
billsec int NOT NULL ,
|
||||||
|
disposition varchar (45) NOT NULL ,
|
||||||
|
amaflags int NOT NULL ,
|
||||||
|
accountcode varchar (20) NOT NULL ,
|
||||||
|
uniqueid varchar (32) NOT NULL ,
|
||||||
|
userfield varchar (255) NOT NULL
|
||||||
|
);
|
||||||
|
|
||||||
|
|
||||||
|
SQLLITE:
|
||||||
|
|
||||||
|
|
||||||
|
RADIUS: See doc/radius.txt for more information on cdr_radius
|
||||||
@ -1,84 +0,0 @@
|
|||||||
\subsection{Introduction}
|
|
||||||
|
|
||||||
This section is intended as an introduction to the Inter-Asterisk
|
|
||||||
eXchange v2 (or simply IAX2) protocol. It provides both a theoretical
|
|
||||||
background and practical information on its use.
|
|
||||||
|
|
||||||
\subsection{Why IAX2?}
|
|
||||||
|
|
||||||
The first question most people are thinking at this point is "Why do you
|
|
||||||
need another VoIP protocol? Why didn't you just use SIP or H.323?"
|
|
||||||
|
|
||||||
Well, the answer is a fairly complicated one, but in a nutshell it's like
|
|
||||||
this... Asterisk is intended as a very flexible and powerful
|
|
||||||
communications tool. As such, the primary feature we need from a VoIP
|
|
||||||
protocol is the ability to meet our own goals with Asterisk, and one with
|
|
||||||
enough flexibility that we could use it as a kind of laboratory for
|
|
||||||
inventing and implementing new concepts in the field. Neither H.323 or
|
|
||||||
SIP fit the roles we needed, so we developed our own protocol, which,
|
|
||||||
while not standards based, provides a number of advantages over both SIP
|
|
||||||
and H.323, some of which are:
|
|
||||||
|
|
||||||
\begin{itemize}
|
|
||||||
\item Interoperability with NAT/PAT/Masquerade firewalls
|
|
||||||
\begin{itemize}
|
|
||||||
\item IAX seamlessly interoperates through all sorts of NAT and PAT
|
|
||||||
and other firewalls, including the ability to place and
|
|
||||||
receive calls, and transfer calls to other stations.
|
|
||||||
\end{itemize}
|
|
||||||
\item High performance, low overhead protocol
|
|
||||||
\begin{itemize}
|
|
||||||
\item When running on low-bandwidth connections, or when running
|
|
||||||
large numbers of calls, optimized bandwidth utilization is
|
|
||||||
imperative. IAX uses only 4 bytes of overhead
|
|
||||||
\end{itemize}
|
|
||||||
\item Internationalization support
|
|
||||||
\begin{itemize}
|
|
||||||
\item IAX transmits language information, so that remote PBX
|
|
||||||
content can be delivered in the native language of the
|
|
||||||
calling party.
|
|
||||||
\end{itemize}
|
|
||||||
\item Remote dialplan polling
|
|
||||||
\begin{itemize}
|
|
||||||
\item IAX allows a PBX or IP phone to poll the availability of a
|
|
||||||
number from a remote server. This allows PBX dialplans to
|
|
||||||
be centralized.
|
|
||||||
\end{itemize}
|
|
||||||
\item Flexible authentication
|
|
||||||
\begin{itemize}
|
|
||||||
\item IAX supports cleartext, md5, and RSA authentication,
|
|
||||||
providing flexible security models for outgoing calls and
|
|
||||||
registration services.
|
|
||||||
\end{itemize}
|
|
||||||
\item Multimedia protocol
|
|
||||||
\begin{itemize}
|
|
||||||
\item IAX supports the transmission of voice, video, images, text,
|
|
||||||
HTML, DTMF, and URL's. Voice menus can be presented in both
|
|
||||||
audibly and visually.
|
|
||||||
\end{itemize}
|
|
||||||
\item Call statistic gathering
|
|
||||||
\begin{itemize}
|
|
||||||
\item IAX gathers statistics about network performance (including
|
|
||||||
latency and jitter, as well as providing end-to-end latency
|
|
||||||
measurement.
|
|
||||||
\end{itemize}
|
|
||||||
\item Call parameter communication
|
|
||||||
\begin{itemize}
|
|
||||||
\item Caller*ID, requested extension, requested context, etc are
|
|
||||||
all communicated through the call.
|
|
||||||
\end{itemize}
|
|
||||||
\item Single socket design
|
|
||||||
\begin{itemize}
|
|
||||||
\item IAX's single socket design allows up to 32768 calls to be
|
|
||||||
multiplexed.
|
|
||||||
\end{itemize}
|
|
||||||
\end{itemize}
|
|
||||||
|
|
||||||
While we value the importance of standards based (i.e. SIP) call handling,
|
|
||||||
hopefully this will provide a reasonable explanation of why we developed
|
|
||||||
IAX rather than starting with SIP.
|
|
||||||
|
|
||||||
\subsection{Configuration}
|
|
||||||
|
|
||||||
For examples of a configuration, please see the iax.conf.sample in
|
|
||||||
your the /configs directory of you source code distribution.
|
|
||||||
@ -0,0 +1,369 @@
|
|||||||
|
Inter-Asterisk eXchange Protocol
|
||||||
|
================================
|
||||||
|
|
||||||
|
INTRODUCTION
|
||||||
|
------------
|
||||||
|
|
||||||
|
This document is intended as an introduction to the Inter-Asterisk
|
||||||
|
eXchange (or simply IAX) protocol. It provides both a theoretical
|
||||||
|
background and practical information on its use.
|
||||||
|
|
||||||
|
WHY IAX
|
||||||
|
-------
|
||||||
|
The first question most people are thinking at this point is "Why do you
|
||||||
|
need another VoIP protocol? Why didn't you just use SIP or H.323?"
|
||||||
|
|
||||||
|
Well, the answer is a fairly complicated one, but in a nutshell it's like
|
||||||
|
this... Asterisk is intended as a very flexible and powerful
|
||||||
|
communications tool. As such, the primary feature we need from a VoIP
|
||||||
|
protocol is the ability to meet our own goals with Asterisk, and one with
|
||||||
|
enough flexibility that we could use it as a kind of laboratory for
|
||||||
|
inventing and implementing new concepts in the field. Neither H.323 or
|
||||||
|
SIP fit the roles we needed, so we developed our own protocol, which,
|
||||||
|
while not standards based, provides a number of advantages over both SIP
|
||||||
|
and H.323, some of which are:
|
||||||
|
|
||||||
|
* Interoperability with NAT/PAT/Masquerade firewalls
|
||||||
|
IAX seamlessly interoperates through all sorts of NAT and PAT
|
||||||
|
and other firewalls, including the ability to place and
|
||||||
|
receive calls, and transfer calls to other stations.
|
||||||
|
|
||||||
|
* High performance, low overhead protocol
|
||||||
|
When running on low-bandwidth connections, or when running
|
||||||
|
large numbers of calls, optimized bandwidth utilization is
|
||||||
|
imperative. IAX uses only 4 bytes of overhead
|
||||||
|
|
||||||
|
* Internationalization support
|
||||||
|
IAX transmits language information, so that remote PBX
|
||||||
|
content can be delivered in the native language of the
|
||||||
|
calling party.
|
||||||
|
|
||||||
|
* Remote dialplan polling
|
||||||
|
IAX allows a PBX or IP phone to poll the availability of a
|
||||||
|
number from a remote server. This allows PBX dialplans to
|
||||||
|
be centralized.
|
||||||
|
|
||||||
|
* Flexible authentication
|
||||||
|
IAX supports cleartext, md5, and RSA authentication,
|
||||||
|
providing flexible security models for outgoing calls and
|
||||||
|
registration services.
|
||||||
|
|
||||||
|
* Multimedia protocol
|
||||||
|
IAX supports the transmission of voice, video, images, text,
|
||||||
|
HTML, DTMF, and URL's. Voice menus can be presented in both
|
||||||
|
audibly and visually.
|
||||||
|
|
||||||
|
* Call statistic gathering
|
||||||
|
IAX gathers statistics about network performance (including
|
||||||
|
latency and jitter, as well as providing end-to-end latency
|
||||||
|
measurement.
|
||||||
|
|
||||||
|
* Call parameter communication
|
||||||
|
Caller*ID, requested extension, requested context, etc are
|
||||||
|
all communicated through the call.
|
||||||
|
|
||||||
|
* Single socket design
|
||||||
|
IAX's single socket design allows up to 32768 calls to be
|
||||||
|
multiplexed.
|
||||||
|
|
||||||
|
While we value the importance of standards based (i.e. SIP) call handling,
|
||||||
|
hopefully this will provide a reasonable explanation of why we developed
|
||||||
|
IAX rather than starting with SIP.
|
||||||
|
|
||||||
|
CONFIG FILE CONVENTIONS
|
||||||
|
-----------------------
|
||||||
|
Lines beginning with '>' represent lines which might appear in an actual
|
||||||
|
configuration file. The '>' is used to help separate them from the
|
||||||
|
descriptive text and should not actually be included in the file itself.
|
||||||
|
|
||||||
|
Lines within []'s by themselves represent section labels within the
|
||||||
|
configuration file. like this:
|
||||||
|
|
||||||
|
> [mysection]
|
||||||
|
|
||||||
|
Options are set using the "=" sign, for example
|
||||||
|
|
||||||
|
> myoption = value
|
||||||
|
|
||||||
|
Sometimes an option will have a number of discrete values which it can
|
||||||
|
take. In that case, in the documentation, the options will be listed
|
||||||
|
within square brackets (the "[" and "]" ones) separated by the pipe symbol
|
||||||
|
("|"). For example:
|
||||||
|
|
||||||
|
> myoption = [value1|value2|value3]
|
||||||
|
|
||||||
|
means the option "myoption" can be assigned a value of "value1", "value2",
|
||||||
|
or "value3".
|
||||||
|
|
||||||
|
Objects, or pseudo-objects are instantiated using the "=>" construct. For
|
||||||
|
example:
|
||||||
|
|
||||||
|
> myobject => parameter
|
||||||
|
|
||||||
|
creates an object called "myobject" with some parameter whose definition
|
||||||
|
would be specific to that object. Note that the config file parser
|
||||||
|
considers "=>" and "=" to be equivalent and their use is purely to make
|
||||||
|
configuration files more readable and easier to "humanly parse".
|
||||||
|
|
||||||
|
The comment character in Asterisk configuration files is the semicolon
|
||||||
|
";". The reason it is not "#" is because the "#" symbol can be used as
|
||||||
|
parts of extensions and it didn't seem like a good idea to have to escape
|
||||||
|
it.
|
||||||
|
|
||||||
|
IAX CONFIGURATION IN ASTERISK
|
||||||
|
-----------------------------
|
||||||
|
|
||||||
|
Like everything else in Asterisk, IAX's configuration lies in
|
||||||
|
/etc/asterisk -- specifically /etc/asterisk/iax.conf
|
||||||
|
|
||||||
|
The IAX configuration file is a collection of sections, each of which
|
||||||
|
(with the exception of the "general" section) represents an entity within
|
||||||
|
the IAX scope.
|
||||||
|
|
||||||
|
------------
|
||||||
|
|
||||||
|
The first section is typically the "general" section. In this area,
|
||||||
|
a number of parameters which affect the entire system are configured.
|
||||||
|
Specifically, the default codecs, port and address, jitter behavior, TOS
|
||||||
|
bits, and registrations.
|
||||||
|
|
||||||
|
The first line of the "general" section is always:
|
||||||
|
|
||||||
|
> [general]
|
||||||
|
|
||||||
|
Following the first line are a number of other possibilities:
|
||||||
|
|
||||||
|
> bindport = <portnum>
|
||||||
|
|
||||||
|
This sets the port that IAX will bind to. The default IAX version 1
|
||||||
|
port number is 5036. For IAX version 2, that is now the default in
|
||||||
|
Asterisk, the default port is 4569.
|
||||||
|
It is recommended that this value not be altered in general.
|
||||||
|
|
||||||
|
> bindaddr = <ipaddr>
|
||||||
|
|
||||||
|
This allows you to bind IAX to a specific local IP address instead of
|
||||||
|
binding to all addresses. This could be used to enhance security if, for
|
||||||
|
example, you only wanted IAX to be available to users on your LAN.
|
||||||
|
|
||||||
|
> bandwidth = [low|medium|high]
|
||||||
|
|
||||||
|
The bandwidth selection initializes the codec selection to appropriate
|
||||||
|
values for given bandwidths. The "high" selection enables all codecs and
|
||||||
|
is recommended only for 10Mbps or higher connections. The "medium"
|
||||||
|
bandwidth eliminates signed linear, Mu-law and A-law codecs, leaving only
|
||||||
|
the codecs which are 32kbps and smaller (with MP3 as a special case). It
|
||||||
|
can be used with broadband connections if desired. "low" eliminates ADPCM
|
||||||
|
and MP3 formats, leaving only the G.723.1, GSM, and LPC10.
|
||||||
|
|
||||||
|
> allow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all]
|
||||||
|
> disallow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all]
|
||||||
|
|
||||||
|
The "allow" and "disallow" allow you to fine tune the codec selection
|
||||||
|
beyond the initial bandwidth selection on a codec-by-codec basis.
|
||||||
|
|
||||||
|
The recommended configuration is to select "low" bandwidth and then
|
||||||
|
disallow the LPC10 codec just because it doesn't sound very good.
|
||||||
|
|
||||||
|
> jitterbuffer = [yes|no]
|
||||||
|
> dropcount = <dropamount>
|
||||||
|
> maxjitterbuffer = <max>
|
||||||
|
> maxexcessbuffer = <max>
|
||||||
|
|
||||||
|
These parameters control the operation of the jitter buffer. The
|
||||||
|
jitterbuffer should always be enabled unless you expect all your
|
||||||
|
connections to be over a LAN.
|
||||||
|
* drop count is the maximum number of voice packets to allow to drop
|
||||||
|
(out of 100). Useful values are 3-10.
|
||||||
|
* maxjitterbuffer is the maximum amount of jitter buffer to permit to be
|
||||||
|
used.
|
||||||
|
* maxexcessbuffer is the maximum amount of excess jitter buffer
|
||||||
|
that is permitted before the jitter buffer is slowly shrunk to eliminate
|
||||||
|
latency.
|
||||||
|
* minexcessbuffer is the minimum amount of excess jitter buffer
|
||||||
|
|
||||||
|
> accountcode = <code>
|
||||||
|
> amaflags = [default|omit|billing|documentation]
|
||||||
|
|
||||||
|
These parameters affect call detail record generation. The first sets the
|
||||||
|
account code for records received with IAX. The account code can be
|
||||||
|
overridden on a per-user basis for incoming calls (see below). The
|
||||||
|
amaflags controls how the record is labeled ("omit" causes no record to be
|
||||||
|
written. "billing" and "documentation" label the records as billing or
|
||||||
|
documentation records respectively, and "default" selects the system
|
||||||
|
default.
|
||||||
|
|
||||||
|
> tos = [lowdelay|throughput|reliability|mincost|none]
|
||||||
|
|
||||||
|
IAX can optionally set the TOS (Type of Service) bits to specified values
|
||||||
|
to help improve performance in routing. The recommended value is
|
||||||
|
"lowdelay", which many routers (including any Linux routers with 2.4
|
||||||
|
kernels that have not been altered with ip tables) will give priority to
|
||||||
|
these packets, improving voice quality.
|
||||||
|
|
||||||
|
> register => <name>[:<secret>]@<host>[:port]
|
||||||
|
|
||||||
|
Any number of registry entries may be instantiated in the general
|
||||||
|
section. Registration allows Asterisk to notify a remote Asterisk server
|
||||||
|
(with a fixed address) what our current address is. In order for
|
||||||
|
registration to work, the remote Asterisk server will need to have a
|
||||||
|
dynamic peer entry with the same name (and secret if provided).
|
||||||
|
|
||||||
|
The name is a required field, and is the remote peer name that we wish to
|
||||||
|
identify ourselves as. A secret may be provided as well. The secret is
|
||||||
|
generally a shared password between the local server and the remote
|
||||||
|
server. However, if the secret is in square brackets ([]'s) then it is
|
||||||
|
interpreted as the name of a RSA key to use. In that case, the local Asterisk
|
||||||
|
server must have the *private* key (/var/lib/asterisk/keys/<name>.key) and
|
||||||
|
the remote server will have to have the corresponding public key.
|
||||||
|
|
||||||
|
The "host" is a required field and is the hostname or IP address of the
|
||||||
|
remote Asterisk server. The port specification is optional and is by
|
||||||
|
default 4569 for iax2 if not specified.
|
||||||
|
|
||||||
|
> notransfer = yes | no
|
||||||
|
|
||||||
|
If an IAX phone calls another IAX phone by using a Asterisk server,
|
||||||
|
Asterisk will transfer the call to go peer to peer. If you do not
|
||||||
|
want this, turn on notransfer with a "yes". This is also settable
|
||||||
|
for peers and users.
|
||||||
|
|
||||||
|
-------------
|
||||||
|
|
||||||
|
The following sections, after "general" define either users, peers or
|
||||||
|
friends. A "user" is someone who connects to us. A "peer" is someone
|
||||||
|
that we connect to. A "friend" is simply shorthand for creating a "user"
|
||||||
|
and "peer" with identical parameters (i.e. someone who can contact us and
|
||||||
|
who we contact).
|
||||||
|
|
||||||
|
> [identifier]
|
||||||
|
|
||||||
|
The section begins with the identifier in square brackets. The identifier
|
||||||
|
should be an alphanumeric string.
|
||||||
|
|
||||||
|
> type = [user|peer|friend]
|
||||||
|
|
||||||
|
This line tells Asterisk how to interpret this entity. Users are things
|
||||||
|
that connect to us, while peers are phones we connect to, and a friend is
|
||||||
|
shorthand for creating a user and a peer with identical information
|
||||||
|
|
||||||
|
----------------
|
||||||
|
User fields:
|
||||||
|
|
||||||
|
> context = <context>
|
||||||
|
|
||||||
|
One or more context lines may be specified in a user, thus giving the user
|
||||||
|
access to place calls in the given contexts. Contexts are used by
|
||||||
|
Asterisk to divide dialing plans into logical units each with the ability
|
||||||
|
to have numbers interpreted differently, have their own security model,
|
||||||
|
auxiliary switch handling, and include other contexts. Most users are
|
||||||
|
given access to the default context. Trusted users could be given access
|
||||||
|
to the local context for example.
|
||||||
|
|
||||||
|
> permit = <ipaddr>/<netmask>
|
||||||
|
> deny = <ipaddr>/<netmask>
|
||||||
|
|
||||||
|
Permit and deny rules may be applied to users, allowing them to connect
|
||||||
|
from certain IP addresses and not others. The permit and deny rules are
|
||||||
|
interpreted in sequence and all are evaluated on a given IP address, with
|
||||||
|
the final result being the decision. For example:
|
||||||
|
|
||||||
|
> permit = 0.0.0.0/0.0.0.0
|
||||||
|
> deny = 192.168.0.0/255.255.255.0
|
||||||
|
|
||||||
|
would deny anyone in 192.168.0.0 with a netmask of 24 bits (class C),
|
||||||
|
whereas:
|
||||||
|
|
||||||
|
> deny = 192.168.0.0/24
|
||||||
|
> permit = 0.0.0.0/0
|
||||||
|
|
||||||
|
would not deny anyone since the final rule would permit anyone, thus
|
||||||
|
overriding the denial.
|
||||||
|
|
||||||
|
If no permit/deny rules are listed, it is assumed that someone may connect
|
||||||
|
from anywhere.
|
||||||
|
|
||||||
|
> callerid = <callerid>
|
||||||
|
|
||||||
|
You may override the Caller*ID information passed by a user to you (if
|
||||||
|
they choose to send it) in order that it always be accurate from the
|
||||||
|
perspective of your server.
|
||||||
|
|
||||||
|
> auth = [md5|plaintext|rsa]
|
||||||
|
|
||||||
|
You may select which authentication methods are permitted to be used by
|
||||||
|
the user to authenticate to us. Multiple methods may be specified,
|
||||||
|
separated by commas. If md5 or plaintext authentication is selected, a
|
||||||
|
secret must be provided. If RSA authentication is specified, then one or
|
||||||
|
more key names must be specified with "inkeys"
|
||||||
|
|
||||||
|
If no secret is specified and no authentication method is specified, then
|
||||||
|
no authentication will be required.
|
||||||
|
|
||||||
|
> secret = <secret>
|
||||||
|
|
||||||
|
The "secret" line specifies the shared secret for md5 and plaintext
|
||||||
|
authentication methods. It is never suggested to use plaintext except in
|
||||||
|
some cases for debugging.
|
||||||
|
|
||||||
|
> inkeys = key1[:key2...]
|
||||||
|
|
||||||
|
The "inkeys" line specifies which keys we can use to authenticate the
|
||||||
|
remote peer. If the peer's challenge passes with any of the given keys,
|
||||||
|
then we accept its authentication. The key files live in
|
||||||
|
/var/lib/asterisk/keys/<name>.pub and are *public keys*. Public keys are
|
||||||
|
not typically DES3 encrypted and thus do not usually need initialization.
|
||||||
|
|
||||||
|
---------------
|
||||||
|
Peer configuration
|
||||||
|
|
||||||
|
> allow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all]
|
||||||
|
> disallow = [gsm|lpc10|g723.1|adpcm|ulaw|alaw|mp3|slinear|all]
|
||||||
|
|
||||||
|
The "allow" and "disallow" may be used to enable or disable specific codec
|
||||||
|
support on a per-peer basis.
|
||||||
|
|
||||||
|
> host = [<ipaddr>|dynamic]
|
||||||
|
|
||||||
|
The host line specifies the hostname or IP address of the remote host, or
|
||||||
|
may be the word "dynamic" signifying that the host will register with us
|
||||||
|
(see register => in the general section above).
|
||||||
|
|
||||||
|
> defaultip = <ipaddr>
|
||||||
|
|
||||||
|
If the host uses dynamic registration, Asterisk may still be given a
|
||||||
|
default IP address to use when dynamic registration has not been performed
|
||||||
|
or has timed out.
|
||||||
|
|
||||||
|
> peercontext = <context>
|
||||||
|
|
||||||
|
Specifies the context name to be passed to the peer for it to use when routing
|
||||||
|
the call through its dial plan. This entry will be used only if a context
|
||||||
|
is not included in the IAX2 channel name passed to the Dial command.
|
||||||
|
|
||||||
|
> qualify = [yes | no | <value>]
|
||||||
|
|
||||||
|
Qualify turns on checking of availability of the remote peer. If the
|
||||||
|
peer becomes unavailable, no calls are placed to the peer until
|
||||||
|
it is reachable again. This is also helpful in certain NAT situations.
|
||||||
|
|
||||||
|
> jitterbuffer = [yes | no]
|
||||||
|
|
||||||
|
Turns on or off the jitterbuffer for this peer
|
||||||
|
|
||||||
|
> mailbox = <mailbox>[@mailboxcontext]
|
||||||
|
|
||||||
|
Specifies a mailbox to check for voicemail notification.
|
||||||
|
|
||||||
|
> permit = <ipaddr>/<netmask>
|
||||||
|
> deny = <ipaddr>/<netmask>
|
||||||
|
|
||||||
|
Permit and deny rules may be applied to users, allowing them to connect
|
||||||
|
from certain IP addresses and not others. The permit and deny rules are
|
||||||
|
interpreted in sequence and all are evaluated on a given IP address, with
|
||||||
|
the final result being the decision. See the user section above
|
||||||
|
for examples.
|
||||||
|
|
||||||
|
----------------------------------------------------------------------
|
||||||
|
For more examples of a configuration, please see the iax.conf.sample in
|
||||||
|
your the /configs directory of you source code distribution
|
||||||
@ -0,0 +1,44 @@
|
|||||||
|
Implementing a Channel
|
||||||
|
======================
|
||||||
|
|
||||||
|
* What is a channel?
|
||||||
|
|
||||||
|
A channel is a unit which brings in a call to the Asterisk PBX. A channel
|
||||||
|
could be connected to a real telephone (like the Internet Phone Jack) or
|
||||||
|
to a logical call (like an Internet phone call). Asterisk makes no
|
||||||
|
distinction between "FXO" and "FXS" style channels (that is, it doesn't
|
||||||
|
distinguish between telephone lines and telephones).
|
||||||
|
|
||||||
|
Every call is placed or received on a distinct channel. Asterisk uses a
|
||||||
|
channel driver (typically named chan_xxx.so) to support each type of
|
||||||
|
hardware.
|
||||||
|
|
||||||
|
* What do I need to create a channel?
|
||||||
|
|
||||||
|
In order to support a new piece of hardware you need to write a channel
|
||||||
|
driver. The easiest way to do so is to look at an existing channel driver
|
||||||
|
and model your own code after it.
|
||||||
|
|
||||||
|
* What's the general architecture?
|
||||||
|
|
||||||
|
Typically, a channel reads a configuration file on startup which tells it
|
||||||
|
something about the hardware it's going to be servicing. Then, it
|
||||||
|
launches a thread which monitors all the idle channels (See the chan_modem
|
||||||
|
or the chan_ixj for an example of this). When a "RING" or equivalent is
|
||||||
|
detected, the monitoring thread should allocate a channel structure and
|
||||||
|
assign all the callbacks to it (see ixj_new, for example), and then call
|
||||||
|
ast_pbx_start on that channel. ast_pbx_start will launch a new thread to
|
||||||
|
handle the channel as long as the call is up, so once pbx_start has
|
||||||
|
successfully been run, the monitor should no longer monitor that channel.
|
||||||
|
The PBX thread will use the channel, reading, writing, calling, etc., and
|
||||||
|
multiplexing that channel with others using select() on the channel's
|
||||||
|
file descriptor (if your channel doesn't have an associated file
|
||||||
|
descriptor, you'll need to emulate one somehow, perhaps along the lines of
|
||||||
|
what the translator API does with its channel.
|
||||||
|
|
||||||
|
When the PBX is finished with the line, it will hang up the line, at which
|
||||||
|
point it the hardware should again be monitored by the monitoring thread.
|
||||||
|
|
||||||
|
---------------
|
||||||
|
For more information, please consult the Asterisk Developer's Documentation
|
||||||
|
on http://www.asterisk.org
|
||||||
@ -0,0 +1,9 @@
|
|||||||
|
Cygwin support is completely experimental and unsupported at this time. The
|
||||||
|
current state of cygwin support is that it will compile, and start the cli,
|
||||||
|
but will not yet take calls properly.
|
||||||
|
|
||||||
|
To compile with cygwin, you will need at least a standard base cygwin install plus the following packages:
|
||||||
|
|
||||||
|
minires
|
||||||
|
minires-devel
|
||||||
|
|
||||||
@ -0,0 +1,91 @@
|
|||||||
|
Asterisk external configuration
|
||||||
|
===============================
|
||||||
|
|
||||||
|
The Asterisk external configuration engine is the result of work by
|
||||||
|
Anthony Minessale II, Mark Spencer and Constantine Filin.
|
||||||
|
|
||||||
|
It is designed to provide a flexible, seamless integration between
|
||||||
|
Asterisk's internal configuration structure and external SQL other other
|
||||||
|
databases (maybe even LDAP one day).
|
||||||
|
|
||||||
|
The external configuration engine is the basis for the ARA, the
|
||||||
|
Asterisk Realtime Architecture (see doc/realtime.txt for more
|
||||||
|
information).
|
||||||
|
|
||||||
|
* Configuration
|
||||||
|
|
||||||
|
External configuration is configured in /etc/asterisk/extconfig.conf
|
||||||
|
allowing you to map any configuration file (static mappings) to
|
||||||
|
be pulled from the database, or to map special runtime entries which
|
||||||
|
permit the dynamic creation of objects, entities, peers, etc. without
|
||||||
|
the necessity of a reload.
|
||||||
|
|
||||||
|
Generally speaking, the columns in your tables should line up with the
|
||||||
|
fields you would specify in the given entity declaration. If an entry
|
||||||
|
would appear more than once, in the column it should be separated by a
|
||||||
|
semicolon. For example, an entity that looks like:
|
||||||
|
|
||||||
|
[foo]
|
||||||
|
host=dynamic
|
||||||
|
secret=bar
|
||||||
|
context=default
|
||||||
|
context=local
|
||||||
|
|
||||||
|
could be stored in a table like this:
|
||||||
|
|
||||||
|
+------+--------+-------+--------------+----------+-----+-----------+
|
||||||
|
| name | host | secret| context | ipaddr | port| regseconds|
|
||||||
|
+------+--------+-------+--------------+----------+-----+-----------+
|
||||||
|
| foo | dynamic| bar | default;local| 127.0.0.1| 4569| 1096954152|
|
||||||
|
+------+--------+-------+--------------+----------+-----+-----------+
|
||||||
|
|
||||||
|
Note that for use with IAX or SIP, the table will also need the "name",
|
||||||
|
"ipaddr", "port", "regseconds" columns. If you wanted to be able to
|
||||||
|
configure the callerid, you could just add a callerid column to the
|
||||||
|
table, for example.
|
||||||
|
|
||||||
|
A SIP table would look more like this:
|
||||||
|
|
||||||
|
+------+--------+-------+----------+-----+------------+----------+
|
||||||
|
| name | host | secret| ipaddr | port| regseconds | username |
|
||||||
|
+------+--------+-------+----------+-----+------------+----------+
|
||||||
|
| foo | dynamic| bar | 127.0.0.1| 4569| 1096954152 | 1234 |
|
||||||
|
+------+--------+-------+----------+-----+------------+----------+
|
||||||
|
|
||||||
|
in order to store appropriate parameters required for SIP.
|
||||||
|
|
||||||
|
In addition to this, if you add a field named "regserver" to the
|
||||||
|
SIP peers table and have the system name set in asterisk.conf,
|
||||||
|
Asterisk will store the system name that the user registered on in
|
||||||
|
the database. This can be used to direct calls to go through the server
|
||||||
|
that holds the registration (for NAT traversal purposes).
|
||||||
|
|
||||||
|
A Voicemail table would look more like this:
|
||||||
|
|
||||||
|
+----------+---------+----------+----------+-----------+---------------+
|
||||||
|
| uniqueid | mailbox | context | password |email | fullname |
|
||||||
|
+----------+---------+----------+----------+-----------+---------------+
|
||||||
|
| 1 | 1234 | default | 4242 | a@b.com | Joe Schmoe |
|
||||||
|
+----------+---------+----------+----------+-----------+---------------+
|
||||||
|
|
||||||
|
The uniqueid should be unique to each voicemail user and can be
|
||||||
|
autoincrement. It need not have any relation to the mailbox or context.
|
||||||
|
|
||||||
|
An extension table would look more like this:
|
||||||
|
|
||||||
|
+----------+---------+----------+-------+-----------+
|
||||||
|
| context | exten | priority | app | appdata |
|
||||||
|
+----------+---------+----------+-------+-----------+
|
||||||
|
| default | 1234 | 1 | Dial | Zap/1 |
|
||||||
|
+----------+---------+----------+-------+-----------+
|
||||||
|
|
||||||
|
In the dialplan you just use the Realtime switch:
|
||||||
|
|
||||||
|
[foo]
|
||||||
|
switch => Realtime
|
||||||
|
|
||||||
|
or:
|
||||||
|
|
||||||
|
[bar]
|
||||||
|
switch => Realtime/bar@extensions
|
||||||
|
|
||||||
@ -1,10 +1,12 @@
|
|||||||
The cdr\_tds module is NOT compatible with version 0.63 of FreeTDS.
|
PLEASE NOTE
|
||||||
|
|
||||||
The cdr\_tds module is known to work with FreeTDS version 0.62.1;
|
The cdr_tds module is NOT compatible with version 0.63 of FreeTDS.
|
||||||
|
|
||||||
|
The cdr_tds module is known to work with FreeTDS version 0.62.1;
|
||||||
it should also work with 0.62.2, 0.62.3 and 0.62.4, which are bug
|
it should also work with 0.62.2, 0.62.3 and 0.62.4, which are bug
|
||||||
fix releases.
|
fix releases.
|
||||||
|
|
||||||
The cdr\_tds module uses the raw "libtds" API of FreeTDS. It appears
|
The cdr_tds module uses the raw "libtds" API of FreeTDS. It appears
|
||||||
that from 0.63 onwards, this is not considered a published API
|
that from 0.63 onwards, this is not considered a published API
|
||||||
of FreeTDS and is subject to change without notice.
|
of FreeTDS and is subject to change without notice.
|
||||||
|
|
||||||
@ -0,0 +1,22 @@
|
|||||||
|
The Asterisk PBX supports H.323 via two totally separate
|
||||||
|
channel drivers.
|
||||||
|
|
||||||
|
You can find more information Asterisk's native H.323
|
||||||
|
support in /path/to/asterisk/channels/h323/README or
|
||||||
|
you can download a third party driver at
|
||||||
|
http://www.inaccessnetworks.com/projects/asterisk-oh323
|
||||||
|
|
||||||
|
Asterisk's native H.323 is supported and maintained by
|
||||||
|
Jeremy McNamara (JerJer in irc). Support for the third
|
||||||
|
party H.323 driver is supplied by inAccessNetworks.
|
||||||
|
|
||||||
|
If you have trouble with either driver you should direct
|
||||||
|
your debug and comments to the appropriate party, making
|
||||||
|
sure to be specific in exactly which H.323 driver you are
|
||||||
|
running.
|
||||||
|
|
||||||
|
Please, read all supplied documentation before contacting
|
||||||
|
either party for support. Many issues can be quickly
|
||||||
|
resolved by simply following the instructions that are
|
||||||
|
provided.
|
||||||
|
|
||||||
@ -1,100 +0,0 @@
|
|||||||
\subsection{Introduction}
|
|
||||||
|
|
||||||
A PBX is only really useful if you can get calls into it. Of course, you
|
|
||||||
can use Asterisk with VoIP calls (SIP, H.323, IAX), but you can also talk
|
|
||||||
to the real PSTN through various cards.
|
|
||||||
|
|
||||||
Supported Hardware is divided into two general groups: Zaptel devices and
|
|
||||||
non-zaptel devices. The Zaptel compatible hardware supports pseudo-TDM
|
|
||||||
conferencing and all call features through chan\_zap, whereas non-zaptel
|
|
||||||
compatible hardware may have different features.
|
|
||||||
|
|
||||||
\subsection{Zaptel compatible hardware}
|
|
||||||
|
|
||||||
\begin{itemize}
|
|
||||||
\item Digium, Inc. (Primary Developer of Asterisk)
|
|
||||||
http://www.digium.com
|
|
||||||
\begin{itemize}
|
|
||||||
\item Analog Interfaces
|
|
||||||
\begin{itemize}
|
|
||||||
\item TDM400P - The TDM400P is a half-length PCI 2.2-compliant card that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC.
|
|
||||||
\item TDM800P - The TDM800P is a half-length PCI 2.2-compliant, 8 port card using Digium's VoiceBus technology that supports FXS and FXO station interfaces for connecting analog telephones and analog POTS lines through a PC.
|
|
||||||
\item TDM2400P - The TDM2400P is a full-length PCI 2.2-compliant card for connecting analog telephones and analog POTS lines through a PC. It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines.
|
|
||||||
\end{itemize}
|
|
||||||
\item Digital Interfaces
|
|
||||||
\begin{itemize}
|
|
||||||
\item TE412P - The TE412P offers an on-board DSP-based echo cancellation module. It supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis.
|
|
||||||
\item TE410P - The TE410P improves performance and scalability through bus mastering architecture. It supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis.
|
|
||||||
\item TE407P - The TE407P offers an on-board DSP-based echo cancellation module. It supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis.
|
|
||||||
\item TE405P - The TE405P improves performance and scalability through bus mastering architecture. It supports both E1, T1, J1 environments and is selectable on a per-card or per-port basis.
|
|
||||||
\item TE212P - The TE212P offers an on-board DSP-based echo cancellation module. It supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis.
|
|
||||||
\item TE210P - The TE210P improves performance and scalability through bus mastering architecture. It supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis.
|
|
||||||
\item TE207P - The TE207P offers an on-board DSP-based echo cancellation module. It supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis.
|
|
||||||
\item TE205P - The TE205P improves performance and scalability through bus mastering architecture. It supports both E1 and T1/J1 environments and is selectable on a per-card or per-port basis.
|
|
||||||
\item TE120P - The TE120P is a single span, selectable T1, E1, or J1 card and utilizes Digium's VoiceBus™ technology. It supports both voice and data modes.
|
|
||||||
\item TE110P - The TE110P brings a high-performance, cost-effective, and flexible single span togglable T1, E1, J1 interface to the Digium line-up of telephony interface devices.
|
|
||||||
\end{itemize}
|
|
||||||
\end{itemize}
|
|
||||||
\end{itemize}
|
|
||||||
|
|
||||||
\subsection{Non-zaptel compatible hardware}
|
|
||||||
|
|
||||||
\begin{itemize}
|
|
||||||
\item QuickNet, Inc.
|
|
||||||
http://www.quicknet.net
|
|
||||||
\begin{itemize}
|
|
||||||
\item Internet PhoneJack - Single FXS interface. Supports Linux telephony
|
|
||||||
interface. DSP compression built-in.
|
|
||||||
|
|
||||||
\item Internet LineJack - Single FXS or FXO interface. Supports Linux
|
|
||||||
telephony interface.
|
|
||||||
\end{itemize}
|
|
||||||
\end{itemize}
|
|
||||||
|
|
||||||
\subsection{mISDN compatible hardware}
|
|
||||||
|
|
||||||
mISDN homepage: http://www.isdn4linux.de/mISDN/
|
|
||||||
|
|
||||||
Any adapter with an mISDN driver should be compatible with
|
|
||||||
chan\_misdn. See the mISDN section for more information.
|
|
||||||
|
|
||||||
\begin{itemize}
|
|
||||||
\item Digium, Inc. (Primary Developer of Asterisk)
|
|
||||||
http://www.digium.com
|
|
||||||
\begin{itemize}
|
|
||||||
\item B410P - 4 Port BRI card (TE/NT)
|
|
||||||
\end{itemize}
|
|
||||||
\end{itemize}
|
|
||||||
|
|
||||||
\begin{itemize}
|
|
||||||
\item beroNet
|
|
||||||
http://www.beronet.com
|
|
||||||
\begin{itemize}
|
|
||||||
\item BN4S0 - 4 Port BRI card (TE/NT)
|
|
||||||
|
|
||||||
\item BN8S0 - 8 Port BRI card (TE/NT)
|
|
||||||
|
|
||||||
\item Billion Card - Single Port BRI card (TE (/NT with crossed cable) )
|
|
||||||
\end{itemize}
|
|
||||||
\end{itemize}
|
|
||||||
|
|
||||||
\subsection{Miscellaneous other interfaces}
|
|
||||||
|
|
||||||
\begin{itemize}
|
|
||||||
\item Digium, Inc. (Primary Developer of Asterisk)
|
|
||||||
\begin{itemize}
|
|
||||||
\item TC400B - The TC400B is a half-length, low-profile PCI 2.2-compliant card for transforming complex VoIP codecs (G.729) into simple codecs.
|
|
||||||
\end{itemize}
|
|
||||||
|
|
||||||
\item ALSA
|
|
||||||
http://www.alsa-project.org
|
|
||||||
\begin{itemize}
|
|
||||||
\item Any ALSA compatible full-duplex sound card
|
|
||||||
\end{itemize}
|
|
||||||
|
|
||||||
\item OSS
|
|
||||||
http://www.opensound.com
|
|
||||||
\begin{itemize}
|
|
||||||
\item Any OSS compatible full-duplex sound card
|
|
||||||
\end{itemize}
|
|
||||||
\end{itemize}
|
|
||||||
@ -0,0 +1,74 @@
|
|||||||
|
A PBX is only really useful if you can get calls into it. Of course, you
|
||||||
|
can use Asterisk with VoIP calls (SIP, H.323, IAX), but you can also talk
|
||||||
|
to the real PSTN through various cards.
|
||||||
|
|
||||||
|
Supported Hardware is divided into two general groups: Zaptel devices and
|
||||||
|
non-zaptel devices. The Zaptel compatible hardware supports pseudo-TDM
|
||||||
|
conferencing and all call features through chan_zap, whereas non-zaptel
|
||||||
|
compatible hardware may have different features.
|
||||||
|
|
||||||
|
Zaptel compatible hardware
|
||||||
|
==========================
|
||||||
|
|
||||||
|
-- Digium (Primary author of Asterisk)
|
||||||
|
http://www.digium.com, http://store.digium.com
|
||||||
|
|
||||||
|
* Wildcard T400P (obsolete) - Quad T1 interface connects to four T1/PRI
|
||||||
|
interfaces. Supports RBS and PRI voice and PPP, FR, and HDLC data.
|
||||||
|
|
||||||
|
* Wildcard E400P (obsolete)- Quad E1 interface connects to four E1/PRI
|
||||||
|
(or PRA) interfaces. Supports PRA/PRI, EuroISDN voice and data.
|
||||||
|
|
||||||
|
* Wildcard T100P - Single T1 interface connects to a single T1/PRI
|
||||||
|
interface. Supports RBS and PRI voice and PPP, FR, and HDLC data.
|
||||||
|
|
||||||
|
* Wildcard E100P - Single E1 interface connects to a single E1/PRI (or PRA)
|
||||||
|
interface. Supports PRA/PRI, EuroISDN voice and PPP, FR, HDLC data.
|
||||||
|
|
||||||
|
* Wildcard TDM400P - Quad Modular FXS interface connects to standard
|
||||||
|
analog telephones.
|
||||||
|
|
||||||
|
* Wildcard TE410P - Quad T1/E1 switchable interface. Supports PRI and
|
||||||
|
RBS signalling, as well as PPP, FR, and HDLC data modes.
|
||||||
|
|
||||||
|
Non-zaptel compatible hardware
|
||||||
|
==============================
|
||||||
|
|
||||||
|
-- QuickNet, Inc.
|
||||||
|
http://www.quicknet.net
|
||||||
|
|
||||||
|
* Internet PhoneJack - Single FXS interface. Supports Linux telephony
|
||||||
|
interface. DSP compression built-in.
|
||||||
|
|
||||||
|
* Internet LineJack - Single FXS or FXO interface. Supports Linux
|
||||||
|
telephony interface.
|
||||||
|
|
||||||
|
mISDN compatible hardware
|
||||||
|
=========================
|
||||||
|
mISDN homepage: http://www.isdn4linux.de/mISDN/
|
||||||
|
|
||||||
|
Any adapter with an mISDN driver should be compatible with
|
||||||
|
chan_misdn. See misdn.txt for information.
|
||||||
|
|
||||||
|
-- beroNet
|
||||||
|
http://www.beronet.com
|
||||||
|
|
||||||
|
* BN4S0 - 4 Port BRI card (TE/NT)
|
||||||
|
|
||||||
|
* BN8S0 - 8 Port BRI card (TE/NT)
|
||||||
|
|
||||||
|
* Billion Card - Single Port BRI card (TE (/NT with crossed cable) )
|
||||||
|
|
||||||
|
|
||||||
|
Miscellaneous other interfaces
|
||||||
|
==============================
|
||||||
|
|
||||||
|
-- ALSA
|
||||||
|
http://www.alsa-project.org
|
||||||
|
|
||||||
|
* Any ALSA compatible full-duplex sound card
|
||||||
|
|
||||||
|
-- OSS
|
||||||
|
http://www.opensound.com
|
||||||
|
|
||||||
|
* Any OSS compatible full-duplex sound card
|
||||||
@ -0,0 +1,67 @@
|
|||||||
|
Inter-Asterisk eXchange Protocol
|
||||||
|
================================
|
||||||
|
|
||||||
|
Usage:
|
||||||
|
======
|
||||||
|
The format for the dialing string on Asterisk is:
|
||||||
|
IAX/[user@]peer[:exten[@context]]
|
||||||
|
|
||||||
|
(Note, []'s denote optional fields). The peer is either an IP address
|
||||||
|
or a peer as specified in the /etc/asterisk/iax.conf file. Exten is
|
||||||
|
an optional requested extension (otherwise "s" will be used), and
|
||||||
|
"context" is an optional context to request. The user is an optional
|
||||||
|
username specified in the peer's iax.conf. If the user is not specified,
|
||||||
|
the peer will select one.
|
||||||
|
|
||||||
|
The peer uses a score to determine the best user entry to match against if
|
||||||
|
one is not specified:
|
||||||
|
|
||||||
|
1. User entry last specified in iax.conf (this is the baseline).
|
||||||
|
2. User entry with secret specified and ACL specified.
|
||||||
|
3. User entry with no secret specified and no ACL specified.
|
||||||
|
4. User entry with no secret specified and ACL specified.
|
||||||
|
5. User entry matched via username.
|
||||||
|
|
||||||
|
The higher the score the better it is with 5 being an exact match and the maximum
|
||||||
|
score possible.
|
||||||
|
|
||||||
|
Protocol and rationale:
|
||||||
|
=======================
|
||||||
|
IAX is a simple, low overhead and low bandwidth VoIP protocol designed to
|
||||||
|
allow multiple Asterisk PBX's to communicate with one another without
|
||||||
|
the overhead of more complex protocols like H.323. Payload is sent with
|
||||||
|
a header overhead of only 4 octets. Control functions (and one payload packet
|
||||||
|
per minute or so) is sent with a more complex header of 12 octets.
|
||||||
|
|
||||||
|
IAX is slightly stateful.
|
||||||
|
|
||||||
|
IAX contains two kinds of packets: The full header packet type, which
|
||||||
|
contains much information about the frame, in addition to its contents,
|
||||||
|
and the mini header type, which is used only for non-reliable voice
|
||||||
|
packet delivery.
|
||||||
|
|
||||||
|
All packets are immediately transmitted. Packets are received, but not
|
||||||
|
delivered to the actual channels until a given time quantum has passed, in
|
||||||
|
order to try to eliminate jitter.
|
||||||
|
|
||||||
|
All full header packets must be ackd (except, obviously for the ACK packets
|
||||||
|
themselves and not so obviously for hangup packets). The "timestamp" field of
|
||||||
|
ack packets is not the normal offset, but rather a quote of the timestamp as
|
||||||
|
included with the original packet that you're acking, and likewise the
|
||||||
|
seqno field is the seqno of the packet you're acking, not your own seqno,
|
||||||
|
and you do not increment your own sequence number. ACKing is based on the
|
||||||
|
sequence number.
|
||||||
|
|
||||||
|
See iax.h for a description of the frame formats.
|
||||||
|
|
||||||
|
IAX internal frames use the AST_FRAME_IAX type. The subclass of these
|
||||||
|
frames is the IAX control number, as seen in iax.h. The first frame sent
|
||||||
|
must be an AST_FRAME_IAX with the control AST_IAX_CONTROL_NEW.
|
||||||
|
|
||||||
|
The AST_IAX_CONTROL_NEW establishes a new connection.
|
||||||
|
|
||||||
|
The first frame sent MUST be an AST_CONTROL_NEW to start a connection.
|
||||||
|
|
||||||
|
IAX connnections may require authentication using either simple plaintext
|
||||||
|
passwords or an md5 challenge/response pair.
|
||||||
|
|
||||||
@ -1,7 +1,12 @@
|
|||||||
|
Icecast + Asterisk
|
||||||
|
==================
|
||||||
The advent of icecast into Asterisk allows you to do neat things like have
|
The advent of icecast into Asterisk allows you to do neat things like have
|
||||||
a caller stream right into an ice-cast stream as well as using chan\_local
|
a caller stream right into an ice-cast stream as well as using chan_local
|
||||||
to place things like conferences, music on hold, etc. into the stream.
|
to place things like conferences, music on hold, etc. into the stream.
|
||||||
|
|
||||||
You'll need to specify a config file for the ices encoder. An example is
|
You'll need to specify a config file for the ices encoder. An example is
|
||||||
included in contrib/asterisk-ices.xml.
|
included in contrib/asterisk-ices.xml
|
||||||
|
|
||||||
|
Anyway hope you like it.
|
||||||
|
|
||||||
|
Mark
|
||||||
@ -0,0 +1,98 @@
|
|||||||
|
As of 2004-12-23, this documentation is no longer maintained. The doxygen documentation
|
||||||
|
generated from linkedlists.h should be referred to in its place, as it is more complete
|
||||||
|
and better maintained.
|
||||||
|
|
||||||
|
2nd version, implemented as macros.
|
||||||
|
|
||||||
|
include <asterisk/linkedlists.h>
|
||||||
|
|
||||||
|
AST_LIST_ENTRY declares pointers inside the object structure :
|
||||||
|
|
||||||
|
struct ast_var_t {
|
||||||
|
char *name;
|
||||||
|
char *value;
|
||||||
|
AST_LIST_ENTRY(ast_var_t) listpointers;
|
||||||
|
};
|
||||||
|
|
||||||
|
AST_LIST_HEAD declares a head structure, which is initialized
|
||||||
|
to AST_LIST_HEAD_NULL :
|
||||||
|
|
||||||
|
AST_LIST_HEAD(head, ast_var_t) head
|
||||||
|
|
||||||
|
Next, we declare a pointer to this structure :
|
||||||
|
|
||||||
|
struct headtype *headp = head;
|
||||||
|
|
||||||
|
AST_LIST_INIT initializes the head pointer to a null value
|
||||||
|
|
||||||
|
AST_LIST_INIT(headp);
|
||||||
|
|
||||||
|
AST_LIST_INSERT_HEAD inserts an element to the head of the list :
|
||||||
|
|
||||||
|
struct ast_var_t *node;
|
||||||
|
|
||||||
|
node=malloc(sizeof(struct ast_var_t));
|
||||||
|
(...we fill data in struct....)
|
||||||
|
data->name=malloc(100);
|
||||||
|
strcpy(data->name,"lalalalaa");
|
||||||
|
etc etc
|
||||||
|
|
||||||
|
(then we insert the node in the head of the list :)
|
||||||
|
|
||||||
|
AST_LIST_INSERT_HEAD(headp,node,listpointers);
|
||||||
|
|
||||||
|
AST_LIST_INSERT_HEAD_AFTER inserts an element after another :
|
||||||
|
|
||||||
|
struct ast_var_t *node1;
|
||||||
|
...
|
||||||
|
AST_LIST_INSERT_AFTER(node,node1,listpointers);
|
||||||
|
|
||||||
|
AST_LIST_REMOVE removes an arbitrary element from the head:
|
||||||
|
|
||||||
|
AST_LIST_REMOVE(headp,node1,ast_var_t,listpointers);
|
||||||
|
|
||||||
|
AST_LIST_REMOVE_HEAD removes the entry at the head of the list and
|
||||||
|
returns a pointer to the removed entry:
|
||||||
|
|
||||||
|
AST_LIST_REMOVE_HEAD(headp,node,listpointers);
|
||||||
|
|
||||||
|
AST_LIST_FIRST returns a pointer to the first element of the list;
|
||||||
|
|
||||||
|
struct ast_var_t *firstnode;
|
||||||
|
firstnode=AST_LIST_FIRST(headp);
|
||||||
|
|
||||||
|
AST_LIST_NEXT returns a pointer to the next element :
|
||||||
|
|
||||||
|
struct ast_var_t *nextnode;
|
||||||
|
nextnode=AST_LIST_NEXT(firstnode,listpointers);
|
||||||
|
|
||||||
|
AST_LIST_TRAVERSE traverses all elements of the list :
|
||||||
|
|
||||||
|
struct ast_var_t *node;
|
||||||
|
|
||||||
|
AST_LIST_TRAVERSE(headp,node,listpointers) {
|
||||||
|
printf("%s\n",node->name);
|
||||||
|
}
|
||||||
|
|
||||||
|
AST_LIST_EMPTY evaluates to a true condition if there are no elements on
|
||||||
|
the list.
|
||||||
|
|
||||||
|
To completely delete a list :
|
||||||
|
|
||||||
|
struct ast_var_t *vardata;
|
||||||
|
|
||||||
|
while (!AST_LIST_EMPTY(headp)) { /* List Deletion. */
|
||||||
|
vardata = AST_LIST_REMOVE_HEAD(head, ast_var_t, listpointers);
|
||||||
|
free(vardata->name);
|
||||||
|
free(vardata->value);
|
||||||
|
}
|
||||||
|
|
||||||
|
AST_LIST_LOCK returns true if it can lock the list, AST_LIST_UNLOCK unlocks
|
||||||
|
the list :
|
||||||
|
|
||||||
|
if (AST_LIST_LOCK(headp)) {
|
||||||
|
...do all list operations here...
|
||||||
|
AST_LIST_UNLOCK(headp);
|
||||||
|
} else {
|
||||||
|
ast_log(LOG_WARNING,"List locked bla bla bla\n");
|
||||||
|
}
|
||||||
@ -0,0 +1,69 @@
|
|||||||
|
|
||||||
|
Mathematical dialplan function
|
||||||
|
|
||||||
|
Yeah, I thought it was a little insane too..
|
||||||
|
|
||||||
|
adds:
|
||||||
|
|
||||||
|
Sum, Multiply, Divide, Subtract, Modulus, GT, LT, GTE, LTE, EQ functions to Asterisk
|
||||||
|
|
||||||
|
All functions follow the same basic pattern for parameters:
|
||||||
|
|
||||||
|
parameter 1 = the math expression
|
||||||
|
parameter 2 = the type of result
|
||||||
|
|
||||||
|
Perform calculation on number 1 to number 2. Valid ops are:
|
||||||
|
+,-,/,*,%,<,>,>=,<=,==
|
||||||
|
and behave as their C equivalents.
|
||||||
|
|
||||||
|
<type_of_result> - wanted type of result:
|
||||||
|
f, float - float(default)
|
||||||
|
i, int - integer,
|
||||||
|
h, hex - hex,
|
||||||
|
c, char - char
|
||||||
|
|
||||||
|
Each math expression is performed as
|
||||||
|
|
||||||
|
Action param1 on param2
|
||||||
|
|
||||||
|
eg:
|
||||||
|
|
||||||
|
Action = Divide
|
||||||
|
Param1 = 10
|
||||||
|
Param2 = 2
|
||||||
|
|
||||||
|
Results in
|
||||||
|
|
||||||
|
Divide 10 by 2
|
||||||
|
|
||||||
|
|
||||||
|
Example dialplan:
|
||||||
|
|
||||||
|
exten => 11099,1,Set(RV=${MATH(1+20)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
|
exten => 11099,n,Set(RV=${MATH(10*2)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
|
exten => 11099,n,Set(RV=${MATH(10*2)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
|
exten => 11099,n,Set(RV=${MATH(10-2)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
|
exten => 11099,n,Set(RV=${MATH(2%10)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
|
exten => 11099,n,Set(RV=${MATH(10/0)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
|
exten => 11099,n,Set(RV=${MATH(10-200)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
|
exten => 11099,n,Set(RV=${MATH(1-20)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
|
exten => 11099,n,Set(RV=${MATH(1<20)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
|
exten => 11099,n,Set(RV=${MATH(1>=20)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
|
exten => 11099,n,Set(RV=${MATH(101>20)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
|
exten => 11099,n,Set(RV=${MATH(1==20)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
|
exten => 11099,n,Set(RV=${MATH(20<=20)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
|
exten => 11099,n,Set(RV=${MATH(123%16,int)})
|
||||||
|
exten => 11099,n,NOOP(${RV})
|
||||||
@ -0,0 +1,15 @@
|
|||||||
|
Description of call model:
|
||||||
|
|
||||||
|
Incoming Call:
|
||||||
|
|
||||||
|
Channel backend waits for a RING or equivalent on some sort of
|
||||||
|
interface. Typically this is done in its own thread. When a RING is
|
||||||
|
detected, the backend should create a channel structure and then call
|
||||||
|
ast_pbx_start() on that channel, which will create a thread to monitor
|
||||||
|
that interface. At this point, the PBX and/or applications it launches
|
||||||
|
will manage the interface, and it need not be monitored by the
|
||||||
|
aforementioned thread. When the applications are finished, the requisite
|
||||||
|
hangup function will be called, at which the channel can be considered to
|
||||||
|
be no longer valid, and the thread that controls it will imminently be
|
||||||
|
terminated.
|
||||||
|
|
||||||
@ -1,5 +1,7 @@
|
|||||||
\subsubsection{MP3 Music On Hold}
|
* Asterisk MP3 Support
|
||||||
|
======================
|
||||||
|
|
||||||
|
* MP3 Music On Hold
|
||||||
Use of the mpg123 for your music on hold is no longer recommended and is now
|
Use of the mpg123 for your music on hold is no longer recommended and is now
|
||||||
officially deprecated. You should now use one of the native formats for your
|
officially deprecated. You should now use one of the native formats for your
|
||||||
music on hold selections.
|
music on hold selections.
|
||||||
@ -0,0 +1,8 @@
|
|||||||
|
About Hold Music
|
||||||
|
================
|
||||||
|
Digium has licensed the music included with
|
||||||
|
the Asterisk distribution From FreePlayMusic
|
||||||
|
for use and distribution with Asterisk. It
|
||||||
|
is licensed ONLY for use as hold music within
|
||||||
|
an Asterisk based PBX.
|
||||||
|
|
||||||
@ -0,0 +1,15 @@
|
|||||||
|
MYSQL LICENSING UPDATE
|
||||||
|
======================
|
||||||
|
We were recently contacted by MySQL and informed that the MySQL client
|
||||||
|
libraries are now under GPL license and not LGPL license as before.
|
||||||
|
|
||||||
|
Since Asterisk does allow exceptions to GPL, we are removing MySQL support
|
||||||
|
from standard Asterisk. We will, where appropriate, make it available via
|
||||||
|
a separate package which will only be usable when Asterisk is used completely
|
||||||
|
within GPL (i.e. not in conjunction with G.729, OpenH.323, etc). We
|
||||||
|
apologize for the confusion.
|
||||||
|
|
||||||
|
You may find this in the new "asterisk-addons" package.
|
||||||
|
|
||||||
|
Mark Spencer
|
||||||
|
Digium
|
||||||
@ -0,0 +1,203 @@
|
|||||||
|
Call Detail Recording to RADIUS Server
|
||||||
|
======================================
|
||||||
|
|
||||||
|
|
||||||
|
Configuration of Asterisk to send CDRs to (Free)RADIUS servers.
|
||||||
|
|
||||||
|
|
||||||
|
A. What is needed :
|
||||||
|
* FreeRADIUS server
|
||||||
|
* Radiusclient-ng library
|
||||||
|
* Asterisk PBX
|
||||||
|
|
||||||
|
|
||||||
|
+--------------------+
|
||||||
|
| Asterisk PBX |
|
||||||
|
| |
|
||||||
|
|********************|
|
||||||
|
| | +---------------+
|
||||||
|
| RADIUS client |------->| RADIUS server |
|
||||||
|
| |<-------| (FreeRADIUS) |
|
||||||
|
+--------------------+ +---------------+
|
||||||
|
|
||||||
|
|
||||||
|
|
||||||
|
|
||||||
|
B. Steps to follow in order to have RADIUS support:
|
||||||
|
|
||||||
|
1.Radiusclient library
|
||||||
|
1.a Installation
|
||||||
|
|
||||||
|
Download the sources from:
|
||||||
|
|
||||||
|
http://developer.berlios.de/projects/radiusclient-ng/
|
||||||
|
|
||||||
|
Untar the source tarball.
|
||||||
|
root@localhost:/usr/local/src# tar xvfz radiusclient-ng-0.5.2.tar.gz
|
||||||
|
|
||||||
|
Compile and install the library.
|
||||||
|
root@localhost:/usr/local/src# cd radiusclient-ng-0.5.2
|
||||||
|
root@localhost:/usr/local/src/radiusclient-ng-0.5.2# ./configure
|
||||||
|
root@localhost:/usr/local/src/radiusclient-ng-0.5.2# make
|
||||||
|
root@localhost:/usr/local/src/radiusclient-ng-0.5.2# make install
|
||||||
|
|
||||||
|
1.b Configuration
|
||||||
|
|
||||||
|
By default all the configuration files of the radiusclient library will
|
||||||
|
be in /usr/local/etc/radiusclient-ng directory.
|
||||||
|
|
||||||
|
File "radiusclient.conf"
|
||||||
|
Open the file and find lines containing the following:
|
||||||
|
|
||||||
|
authserver localhost
|
||||||
|
|
||||||
|
This is the hostname or IP address of the RADIUS server used for
|
||||||
|
authentication. You will have to change this unless the server is
|
||||||
|
running on the same host as your Asterisk PBX.
|
||||||
|
|
||||||
|
acctserver localhost
|
||||||
|
|
||||||
|
This is the hostname or IP address of the RADIUS server used for
|
||||||
|
accounting. You will have to change this unless the server is running
|
||||||
|
on the same host as your Asterisk PBX.
|
||||||
|
|
||||||
|
File "servers"
|
||||||
|
|
||||||
|
RADIUS protocol uses simple access control mechanism based on shared
|
||||||
|
secrets that allows RADIUS servers to limit access from RADIUS clients.
|
||||||
|
|
||||||
|
A RADIUS server is configured with a secret string and only RADIUS
|
||||||
|
clients that have the same secret will be accepted.
|
||||||
|
|
||||||
|
You need to configure a shared secret for each server you have
|
||||||
|
configured in radiusclient.conf file in the previous step. The shared
|
||||||
|
secrets are stored in /usr/local/etc/radiusclient-ng/servers file.
|
||||||
|
|
||||||
|
Each line contains hostname of a RADIUS server and shared secret
|
||||||
|
used in communication with that server. The two values are separated
|
||||||
|
by white spaces. Configure shared secrets for every RADIUS server you
|
||||||
|
are going to use.
|
||||||
|
|
||||||
|
File "dictionary"
|
||||||
|
|
||||||
|
Asterisk uses some attributes that are not included in the
|
||||||
|
dictionary of radiusclient library, therefore it is necessary to add
|
||||||
|
them. A file called dictionary.digium (kept in the contrib dir)
|
||||||
|
was created to list all new attributes used by Asterisk.
|
||||||
|
Add to the end of the main dictionary file
|
||||||
|
/usr/local/etc/radiusclient-ng/dictionary
|
||||||
|
the line:
|
||||||
|
|
||||||
|
$INCLUDE /path/to/dictionary.digium
|
||||||
|
|
||||||
|
2.FreeRADIUS Server (Version 1.1.1)
|
||||||
|
2.a Installation
|
||||||
|
|
||||||
|
Download sources tarball from:
|
||||||
|
|
||||||
|
http://freeradius.org/
|
||||||
|
|
||||||
|
Untar, configure, build, and install the server:
|
||||||
|
|
||||||
|
root@localhost:/usr/local/src# tar xvfz freeradius-1.1.1.tar.gz
|
||||||
|
root@localhost:/usr/local/src# cd freeradius-1.1.1
|
||||||
|
root@localhost"/usr/local/src/freeradius-1.1.1# ./configure
|
||||||
|
root@localhost"/usr/local/src/freeradius-1.1.1# make
|
||||||
|
root@localhost"/usr/local/src/freeradius-1.1.1# make install
|
||||||
|
|
||||||
|
All the configuration files of FreeRADIUS server will be in
|
||||||
|
/usr/local/etc/raddb directory.
|
||||||
|
|
||||||
|
|
||||||
|
2.b Configuration
|
||||||
|
|
||||||
|
There are several file that have to be modified to configure the
|
||||||
|
RADIUS server. These are presented next.
|
||||||
|
|
||||||
|
File "clients.conf"
|
||||||
|
|
||||||
|
File /usr/local/etc/raddb/clients.conf contains description of
|
||||||
|
RADIUS clients that are allowed to use the server. For each of the
|
||||||
|
clients you need to specify its hostname or IP address and also a
|
||||||
|
shared secret. The shared secret must be the same string you configured
|
||||||
|
in radiusclient library.
|
||||||
|
|
||||||
|
Example:
|
||||||
|
client myhost {
|
||||||
|
secret = mysecret
|
||||||
|
shortname = foo
|
||||||
|
}
|
||||||
|
|
||||||
|
This fragment allows access from RADIUS clients on "myhost" if they use
|
||||||
|
"mysecret" as the shared secret.
|
||||||
|
The file already contains an entry for localhost (127.0.0.1), so if you
|
||||||
|
are running the RADIUS server on the same host as your Asterisk server,
|
||||||
|
then modify the existing entry instead, replacing the default password.
|
||||||
|
|
||||||
|
File "dictionary"
|
||||||
|
|
||||||
|
Note : as of version 1.1.2, the dictionary.digium file ships with FreeRADIUS.
|
||||||
|
The following procedure brings the dictionary.digium file to previous versions
|
||||||
|
of FreeRADIUS.
|
||||||
|
|
||||||
|
File /usr/local/etc/raddb/dictionary contains the dictionary of
|
||||||
|
FreeRADIUS server. You have to add the same dictionary file
|
||||||
|
(dictionary.digium), which you added to the dictionary of radiusclient-ng
|
||||||
|
library. You can include it into the main file, adding the following line at the
|
||||||
|
end of file '/usr/local/etc/raddb/dictionary':
|
||||||
|
|
||||||
|
$INCLUDE /path/to/dictionary.digium
|
||||||
|
|
||||||
|
That will include the same new attribute definitions that are used
|
||||||
|
in radiusclient-ng library so the client and server will understand each
|
||||||
|
other.
|
||||||
|
|
||||||
|
|
||||||
|
3. Asterisk Accounting Configuration
|
||||||
|
|
||||||
|
Compilation and installation:
|
||||||
|
The module will be compiled as long as the radiusclient-ng
|
||||||
|
library has been detected on your system.
|
||||||
|
|
||||||
|
By default FreeRADIUS server will log all accounting requests into
|
||||||
|
/usr/local/var/log/radius/radacct directory in form of plain text files.
|
||||||
|
The server will create one file for each hostname in the directory. The
|
||||||
|
following example shows how the log files look like.
|
||||||
|
|
||||||
|
Asterisk now generates Call Detail Records. See /include/asterisk/cdr.h
|
||||||
|
for all the fields which are recorded. By default, records in comma
|
||||||
|
separated values will be created in /var/log/asterisk/cdr-csv.
|
||||||
|
|
||||||
|
The configuration file for cdr_radius.so module is :
|
||||||
|
|
||||||
|
/etc/asterisk/cdr.conf
|
||||||
|
This is where you can set CDR related parameters as well as the path to
|
||||||
|
the radiusclient-ng library configuration file.
|
||||||
|
|
||||||
|
|
||||||
|
4. Logged Values
|
||||||
|
|
||||||
|
"Asterisk-Acc-Code", The account name of detail records
|
||||||
|
"Asterisk-Src",
|
||||||
|
"Asterisk-Dst",
|
||||||
|
"Asterisk-Dst-Ctx", The destination context
|
||||||
|
"Asterisk-Clid",
|
||||||
|
"Asterisk-Chan", The channel
|
||||||
|
"Asterisk-Dst-Chan", (if applicable)
|
||||||
|
"Asterisk-Last-App", Last application run on the channel
|
||||||
|
"Asterisk-Last-Data", Argument to the last channel
|
||||||
|
"Asterisk-Start-Time",
|
||||||
|
"Asterisk-Answer-Time",
|
||||||
|
"Asterisk-End-Time",
|
||||||
|
"Asterisk-Duration", Duration is the whole length that the entire
|
||||||
|
call lasted. ie. call rx'd to hangup
|
||||||
|
"end time" minus "start time"
|
||||||
|
"Asterisk-Bill-Sec", The duration that a call was up after other
|
||||||
|
end answered which will be <= to duration
|
||||||
|
"end time" minus "answer time"
|
||||||
|
"Asterisk-Disposition", ANSWERED, NO ANSWER, BUSY
|
||||||
|
"Asterisk-AMA-Flags", DOCUMENTATION, BILL, IGNORE etc, specified on
|
||||||
|
a per channel basis like accountcode.
|
||||||
|
"Asterisk-Unique-ID", Unique call identifier
|
||||||
|
"Asterisk-User-Field" User field set via SetCDRUserField
|
||||||
|
|
||||||
Binary file not shown.
Loading…
Reference in new issue