Asterisk - The Open Source Telephony Project  GIT-master-1f78ee9
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk
astsbindir => /usr/sbin

;verbose = 3
;debug = 3
;trace = 0              ; Set the trace level.
;refdebug = yes			; Enable reference count debug logging.
;alwaysfork = yes		; Same as -F at startup.
;nofork = yes			; Same as -f at startup.
;quiet = yes			; Same as -q at startup.
;timestamp = yes		; Same as -T at startup.
;execincludes = yes		; Support #exec in config files.
;console = yes			; Run as console (same as -c at startup).
;highpriority = yes		; Run realtime priority (same as -p at
				; startup).
;initcrypto = yes		; Initialize crypto keys (same as -i at
				; startup).
;nocolor = yes			; Disable console colors.
;dontwarn = yes			; Disable some warnings.
;dumpcore = yes			; Dump core on crash (same as -g at startup).
;languageprefix = yes		; Use the new sound prefix path syntax.
;systemname = my_system_name	; Prefix uniqueid with a system name for
				; Global uniqueness issues.
;autosystemname = yes		; Automatically set systemname to hostname,
				; uses 'localhost' on failure, or systemname if
				; set.
;mindtmfduration = 80		; Set minimum DTMF duration in ms (default 80 ms)
				; If we get shorter DTMF messages, these will be
				; changed to the minimum duration
;maxcalls = 10			; Maximum amount of calls allowed.
;maxload = 0.9			; Asterisk stops accepting new calls if the
				; load average exceed this limit.
;maxfiles = 1000		; Maximum amount of openfiles.
;minmemfree = 1			; In MBs, Asterisk stops accepting new calls if
				; the amount of free memory falls below this
				; watermark.
;cache_media_frames = yes	; Cache media frames for performance
				; Disable this option to help track down media frame
				; mismanagement when using valgrind or MALLOC_DEBUG.
				; The cache gets in the way of determining if the
				; frame is used after being freed and who freed it.
				; NOTE: This option has no effect when Asterisk is
				; compiled with the LOW_MEMORY compile time option
				; enabled because the cache code does not exist.
				; Default yes
;cache_record_files = yes	; Cache recorded sound files to another
				; directory during recording.
;record_cache_dir = /tmp	; Specify cache directory (used in conjunction
				; with cache_record_files).
;transmit_silence = yes		; Transmit silence while a channel is in a
				; waiting state, a recording only state, or
				; when DTMF is being generated.  Note that the
				; silence internally is generated in raw signed
				; linear format. This means that it must be
				; transcoded into the native format of the
				; channel before it can be sent to the device.
				; It is for this reason that this is optional,
				; as it may result in requiring a temporary
				; codec translation path for a channel that may
				; not otherwise require one.
;transcode_via_sln = yes	; Build transcode paths via SLINEAR, instead of
				; directly.
;runuser = asterisk		; The user to run as.
;rungroup = asterisk		; The group to run as.
;lightbackground = yes		; If your terminal is set for a light-colored
				; background.
;forceblackbackground = yes     ; Force the background of the terminal to be
                                ; black, in order for terminal colors to show
                                ; up properly.
;defaultlanguage = en           ; Default language
documentation_language = en_US	; Set the language you want documentation
				; displayed in. Value is in the same format as
				; locale names.
;hideconnect = yes		; Hide messages displayed when a remote console
				; connects and disconnects.
;lockconfdir = no		; Protect the directory containing the
				; configuration files (/etc/asterisk) with a
				; lock.
;stdexten = gosub		; How to invoke the extensions.conf stdexten.
				; macro - Invoke the stdexten using a macro as
				;         done by legacy Asterisk versions.
				; gosub - Invoke the stdexten using a gosub as
				;         documented in extensions.conf.sample.
				; Default gosub.
;live_dangerously = no		; Enable the execution of 'dangerous' dialplan
				; functions from external sources (AMI,
				; etc.) These functions (such as SHELL) are
				; considered dangerous because they can allow
				; privilege escalation.
				; Default no
;entityid=00:11:22:33:44:55	; Entity ID.
				; This is in the form of a MAC address.
				; It should be universally unique.
				; It must be unique between servers communicating
				; with a protocol that uses this value.
				; This is currently is used by DUNDi and
				; Exchanging Device and Mailbox State
				; using protocols: XMPP, Corosync and PJSIP.
;rtp_use_dynamic = yes          ; When set to "yes" RTP dynamic payload types
                                ; are assigned dynamically per RTP instance vs.
                                ; allowing Asterisk to globally initialize them
                                ; to pre-designated numbers (defaults to "yes").
;rtp_pt_dynamic = 35		; Normally the Dynamic RTP Payload Type numbers
				; are 96-127, which allow just 32 formats. The
				; starting point 35 enables the range 35-63 and
				; allows 29 additional formats. When you use
				; more than 32 formats in the dynamic range and
				; calls are not accepted by a remote
				; implementation, please report this and go
				; back to value 96.
;hide_messaging_ami_events = no;  This option, if enabled, will
                ; suppress all of the Message/ast_msg_queue channel's
                ; housekeeping AMI and ARI channel events.  This can
                ; reduce the load on the manager and ARI applications
                ; when the Digium Phone Module for Asterisk is in use.

; Changing the following lines may compromise your security.
;astctlpermissions = 0660
;astctlowner = root
;astctlgroup = apache
;astctl = asterisk.ctl