Dear it's the default setting of asterisk.conf, your config is not complete. -f in the ps output, shows your asterisk have been run in fork mode, disable it.
[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 conjunction with cache_record_files) ;transmit_silence_during_record = yes ; Transmit SLINEAR silence while a channel is being recorded ;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 ;sendfullybooted = yes ; Send the FullyBooted AMI event on AMI login and when all modules are finished loading ;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 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 ; 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 On Sat, Jan 29, 2011 at 4:32 PM, Gilles <codecompl...@free.fr> wrote: > On Sat, 29 Jan 2011 15:47:53 +0330, Pezhman Lali <l...@lopl.net> > wrote: > >check your /etc/asterisk/asterisk.conf and post it here > > Here goes: > > root:/var/tmp> cat /etc/asterisk/asterisk.conf > [directories] > astetcdir => /etc/asterisk > astmoddir => /usr/lib/asterisk/modules > astvarlibdir => /var/lib/asterisk > astagidir => /usr/share/asterisk/agi-bin > astspooldir => /var/spool/asterisk > astlogdir => /var/log/asterisk > > Thank you. > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users