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Sophie

distrib > Mandriva > 2010.0 > i586 > media > contrib-release > by-pkgid > d84165eece04d09102359b21b401a991 > files > 8

asterisk-1.6.1.8-1mdv2010.0.i586.rpm

[directories](!) ; remove the (!) to enable this
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

;[options]
;verbose = 3
;debug = 3
;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
;internal_timing = yes
;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
;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_record_files = yes ; Cache recorded sound files to another directory during recording
;record_cache_dir = /tmp ; Specify cache directory (used in cnjunction with cache_record_files)
;transmit_silence_during_record = yes ; Transmit SLINEAR silence while a channel is being recorded
;transmit_silence = yes ; Transmit SLINEAR silence while a channel is being recorded or DTMF is being generated
;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

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

[compat]
pbx_realtime=1.6
res_agi=1.6
app_set=1.6