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Update templates for asterisk-13.18.4
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2 changed files with 80 additions and 45 deletions
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@ -1,14 +1,15 @@
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[directories]
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[directories](!)
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astetcdir => /etc/asterisk
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astmoddir => /usr/lib/asterisk/modules
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astvarlibdir => /usr/lib/asterisk
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astdbdir => /usr/lib/asterisk
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astkeydir => /usr/lib/asterisk
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astdatadir => /usr/lib/asterisk
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astagidir => /usr/lib/asterisk/agi-bin
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astvarlibdir => /var/lib/asterisk
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astdbdir => /var/lib/asterisk
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astkeydir => /var/lib/asterisk
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astdatadir => /usr/share/asterisk
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astagidir => /usr/share/asterisk/agi-bin
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astspooldir => /var/spool/asterisk
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astrundir => /var/run/asterisk
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astlogdir => /var/log/asterisk
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astsbindir => /usr/sbin
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[options]
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;verbose = 3
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@ -17,7 +18,7 @@ astlogdir => /var/log/asterisk
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;nofork = yes ; Same as -f at startup.
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;quiet = yes ; Same as -q at startup.
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;timestamp = yes ; Same as -T at startup.
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execincludes = yes ; Support #exec in config files.
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;execincludes = yes ; Support #exec in config files.
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;console = yes ; Run as console (same as -c at startup).
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;highpriority = yes ; Run realtime priority (same as -p at
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; startup).
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@ -27,12 +28,14 @@ execincludes = yes ; Support #exec in config files.
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;dontwarn = yes ; Disable some warnings.
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;dumpcore = yes ; Dump core on crash (same as -g at startup).
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;languageprefix = yes ; Use the new sound prefix path syntax.
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;internal_timing = yes
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;systemname = my_system_name ; Prefix uniqueid with a system name for
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; Global uniqueness issues.
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;autosystemname = yes ; Automatically set systemname to hostname,
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; uses 'localhost' on failure, or systemname if
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; set.
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;mindtmfduration = 80 ; Set minimum DTMF duration in ms (default 80 ms)
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; If we get shorter DTMF messages, these will be
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; changed to the minimum duration
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;maxcalls = 10 ; Maximum amount of calls allowed.
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;maxload = 0.9 ; Asterisk stops accepting new calls if the
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; load average exceed this limit.
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@ -73,7 +76,35 @@ documentation_language = en_US ; Set the language you want documentation
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;lockconfdir = no ; Protect the directory containing the
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; configuration files (/etc/asterisk) with a
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; lock.
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mindtmfduration=60
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;stdexten = gosub ; How to invoke the extensions.conf stdexten.
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; macro - Invoke the stdexten using a macro as
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; done by legacy Asterisk versions.
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; gosub - Invoke the stdexten using a gosub as
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; documented in extensions.conf.sample.
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; Default gosub.
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;live_dangerously = no ; Enable the execution of 'dangerous' dialplan
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; functions from external sources (AMI,
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; etc.) These functions (such as SHELL) are
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; considered dangerous because they can allow
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; privilege escalation.
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; Default no
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;entityid=00:11:22:33:44:55 ; Entity ID.
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; This is in the form of a MAC address.
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; It should be universally unique.
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; It must be unique between servers communicating
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; with a protocol that uses this value.
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; This is currently is used by DUNDi and
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; Exchanging Device and Mailbox State
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; using protocols: XMPP, Corosync and PJSIP.
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;rtp_pt_dynamic = 96 ; Normally the Dynamic RTP Payload Type numbers
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; are 96-127, which allow 32 formats. When you
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; use more and receive the message "No Dynamic
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; RTP mapping available", extend the dynamic
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; range by going for 35 (or 0) instead of 96.
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; This allows 29 (or 64) more formats. 96 is the
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; default because any number below might be
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; rejected by a remote implementation; although
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; no such broken implementation is known, yet.
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; Changing the following lines may compromise your security.
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;[files]
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@ -81,8 +112,3 @@ mindtmfduration=60
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;astctlowner = root
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;astctlgroup = apache
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;astctl = asterisk.ctl
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[compat]
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pbx_realtime=1.6
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res_agi=1.6
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app_set=1.6
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@ -1,36 +1,45 @@
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;
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; Asterisk configuration file
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;
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; Module Loader configuration file
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;
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[modules]
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autoload=yes
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load => chan_brcm.so
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load => res_jabber.so ; Used for Gtalk
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load => res_clioriginate.so ; originate calls from commandline
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load => res_rtp_asterisk.so ; rtp "engine" is now a loadable module in asterisk 1.8
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load => pbx_config.so ; Text Extension Configuration Requires N/A
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load => func_callerid.so ; Gets or sets Caller*ID data on the channel. - Requires ?
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load => func_channel.so
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load => func_logic.so ; Logic functions (if, etc.)
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load => func_strings.so ; string manipulation functions
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load => cdr_manager.so ; Asterisk Call Manager CDR Backend - Requires N/A
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load => chan_local.so ; Show status of local channels- Requires N/A
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load => chan_gtalk.so ; Use gtalk
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load => chan_sip.so ; Session Initiation Protocol (SIP) - Requires res_features.so
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load => codec_alaw.so ; A-law Coder/Decoder - Requires N/A
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load => codec_a_mu.so ; A-law and Mulaw direct Coder/Decoder - Requires N/A
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load => codec_gsm.so ; GSM/PCM16 (signed linear) Codec Translat - Requires N/A
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load => codec_ulaw.so ; Mu-law Coder/Decoder - Requires N/A
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load => format_gsm.so ; Raw GSM data - Requires N/A
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load => format_pcm.so ; Raw uLaw 8khz Audio support (PCM) - Requires N/A
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load => app_dial.so ; Dialing Application - Requires res_features.so, res_musiconhold.so
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load => app_parkandannounce.so ; Call Parking and Announce Application - Requires res_features.so
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load => app_playback.so ; Sound File Playback Application - Requires N/A
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load => app_system.so ; Execute a system command - Requires N/A
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load => app_disa.so ; Direct Inward System Access
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load => app_authenticate.so ; Authenticate via pin
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load => app_senddtmf.so ; Ability to send DTMF tones on the line.
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load => func_cut.so ; To manipulate strings
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load => res_musiconhold.so
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load => func_db.so
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;
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; Any modules that need to be loaded before the Asterisk core has been
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; initialized (just after the logger has been initialized) can be loaded
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; using 'preload'. This will frequently be needed if you wish to map all
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; module configuration files into Realtime storage, since the Realtime
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; driver will need to be loaded before the modules using those configuration
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; files are initialized.
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;
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; An example of loading ODBC support would be:
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;preload => res_odbc.so
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;preload => res_config_odbc.so
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;
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; Uncomment the following if you wish to use the Speech Recognition API
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;preload => res_speech.so
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;
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; If you want Asterisk to fail if a module does not load, then use
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; the "require" keyword. Asterisk will exit with a status code of 2
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; if a required module does not load.
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;
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; require = chan_sip.so
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; If you want you can combine with preload
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; preload-require = res_odbc.so
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;
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; If you want, load the GTK console right away.
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;
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noload => pbx_gtkconsole.so
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;load => pbx_gtkconsole.so
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;
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load => res_musiconhold.so
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;
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; Load one of: chan_oss, alsa, or console (portaudio).
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; By default, load chan_oss only (automatically).
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;
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noload => chan_alsa.so
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;noload => chan_oss.so
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noload => chan_console.so
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[global]
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chan_modem.so=no
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;
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