OpenPKG Specs: asterisk/asterisk.txt@a2c6460cfb16 (annotated)

asterisk/asterisk.txt@a2c6460cfb16 (annotated)

asterisk/asterisk.txt

Mon, 28 Jan 2013 17:37:18 +0100

author: Michael Schloh von Bennewitz <michael@schloh.com>
date: Mon, 28 Jan 2013 17:37:18 +0100
changeset 758: a2c6460cfb16
parent 399: 9360a38b5c9f
permissions: -rw-r--r--

Correct socket error reporting improvement with IPv6 portable code,
after helpful recommendation by Saúl Ibarra Corretgé on OSips devlist.

 <file name="asterisk.conf">
 ;;
 ;;  asterisk.conf -- Asterisk master configuration
 ;;
 [directories]
 astetcdir          = @l_prefix@/etc/asterisk
 astmoddir          = @l_prefix@/lib/asterisk/modules
 astagidir          = @l_prefix@/lib/asterisk/agi-bin
 astvarlibdir       = @l_prefix@/share/asterisk
 astspooldir        = @l_prefix@/var/asterisk/spool
 astrundir          = @l_prefix@/var/asterisk/run
 astlogdir          = @l_prefix@/var/asterisk/log
 astdbdir           = @l_prefix@/var/asterisk/db
 [files]
 astctlowner        = @l_rusr@
 astctlgroup        = @l_rgrp@
 astctlpermissions  = 700
 astctl             = asterisk.ctl
 [options]
 systemname         = openpkg-pbx
 runuser            = @l_rusr@
 rungroup           = @l_rgrp@
 verbose            = 0
 alwaysfork         = yes
 dumpcore           = no
 quiet              = yes
 highpriority       = no
 initcrypto         = no
 nocolor            = yes
 execincludes       = no
 ;timestamp         = yes
 ;optiondebug       = no
 ;nofork            = no
 ;console           = no
 ;dontwarn          = no
 </file>
 <file name="amd.conf">
 ;;
 ;;  amd.conf -- Answering Machine Detection configuration
 ;;
 [general]
 initial_silence        = 2500 ; Maximum silence duration before the greeting.
                               ; If exceeded then MACHINE.
 greeting               = 1500 ; Maximum length of a greeting.
                               ;If exceeded then MACHINE.
 after_greeting_silence = 800  ; Silence after detecting a greeting.
                               ; If exceeded then HUMAN.
 total_analysis_time    = 5000 ; Maximum time allowed for the algorithm to
                               ; decide on a HUMAN or MACHINE.
 min_word_length        = 100  ; Minimum duration of Voice to considered a word.
 between_words_silence  = 50   ; Minimum duration of silence after a word to
                               ; consider the audio what follows as a new word.
 maximum_number_of_words = 3   ; Maximum number of words in the greeting.
                               ; If exceeded then MACHINE.
 silence_threshold       = 256
 </file>
 <file name="modules.conf">
 ;;
 ;;  modules.conf -- Asterisk functionality module configuration
 ;;
 [modules]
 autoload           = yes
 [global]
 </file>
 <file name="logger.conf">
 ;;
 ;;  logger.conf -- Asterisk logging configuration
 ;;
 [general]
 dateformat         = %F %T
 queue_log          = no
 event_log          = no
 [logfiles]
 console            = error,warning,notice,verbose
 asterisk.log       = error,warning,notice ; verbose,debug
 </file>
 <file name="manager.conf">
 ;;
 ;;  manager.conf -- Asterisk internal manager API configuration
 ;;
 [general]
 enabled            = yes
 webenabled         = no
 bindaddr           = 127.0.0.1
 port               = 5038
 ; You can open a TLS connection to this socket with:
 ;
 ;   openssl s_client -connect my_host:5039
 ;
 tlsenable          = no
 tlsbindaddr        = 127.0.0.1
 tlsbindport        = 5039
 tlscertfile        = @l_prefix@/etc/asterisk/asterisk.pem
 ; if tlsprivatekey is not specified search tlscertfile for key
 ;tlsprivatekey      = @l_prefix@/etc/asterisk/asterkey.pem
 ;tlscipher         = ALL:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL
 displayconnects    = yes
 allowmultiplelogin = yes
 timestampevents    = yes
 [asterisk]
 secret             = asterisk
 deny               = 0.0.0.0/0.0.0.0
 permit             = 127.0.0.1/255.0.0.0
 read               = system,call,agent,user,config,log,verbose,dtmf,reporting,cdr,dialplan
 write              = system,call,agent,user,config,command,reporting
 </file>
 <file name="http.conf">
 ;;
 ;;  http.conf -- Asterisk HTTP server interface
 ;;
 [general]
 enabled            = no
 bindaddr           = 127.0.0.1
 bindport           = 8088
 tlsenable          = no
 tlsbindport        = 8089
 tlsbindaddr        = 127.0.0.1
 tlscertfile        = @l_prefix@/etc/asterisk/asterisk.pem
 enablestatic       = yes
 prefix             = asterisk
 redirect           = / /asterisk/static/docs/index.html
 [post_mappings]
 uploads            = @l_prefix@/var/asterisk/spool/uploads/
 </file>
 <file name="sip.conf">
 ;;
 ;;  sip.conf -- Asterisk SIP configuration
 ;;
 [general]
 useragent          = OpenPKG Asterisk PBX
 realm              = example
 bindaddr           = 127.0.0.1
 bindport           = 5060
 tcpenable          = yes
 tcpbindaddr        = 127.0.0.1:5060
 tlsenable          = no
 tlsbindaddr        = 127.0.0.1:5061
 tlscipher          = ALL:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:+eNULL
 tlscertfile        = asterisk.pem
 tlscafile          = asterisk.pem
 srvlookup          = yes
 useclientcode      = yes
 allowguest         = yes
 canreinvite        = no
 nat                = no
 disallow           = all
 allow              = speex
 allow              = g726
 allow              = ulaw
 allow              = alaw
 allow              = gsm
 videosupport       = no
 ;allow             = h263
 ;allow             = h263p
 notifyhold         = yes
 notifyringing      = yes
 limitonpeer        = yes
 call-limit         = 1
 incominglimit      = 1
 context            = external
 ;register          = NNNNNNN:XXXXXX:NNNNNNN@sipgate.de/s
 ;tos               = 0x18
 ;[sipgate]
 ;type              = peer
 ;defaultuser       = NNNNNNN
 ;host              = sipgate.de
 ;fromuser          = NNNNNNN
 ;fromdomain        = sipgate.de
 ;canreinvite       = no
 ;disallow          = all
 ;allow             = speex
 ;allow             = g726
 ;allow             = ulaw
 ;allow             = alaw
 ;allow             = gsm
 ;context           = external
 ;[gw]
 ;type              = friend
 ;defaultuser       = gw
 ;callerid          = "ISDN-to-SIP" <gw>
 ;fromdomain        = example.com
 ;secret            = asterisk
 ;host              = dynamic
 ;canreinvite       = no
 ;disallow          = all
 ;allow             = g726
 ;allow             = ulaw
 ;allow             = alaw
 ;allow             = gsm
 ;dtmfmode          = rfc2833
 ;qualify           = yes
 ;insecure          = yes
 ;context           = external
 ;nat               = no
 [std-user](!)
 type               = friend
 context            = internal
 host               = dynamic
 dtmfmode           = rfc2833
 qualify            = yes
 disallow           = all
 allow              = speex
 allow              = g726
 allow              = ulaw
 allow              = alaw
 [behind-nat](!)
 nat                = yes
 [with-mailbox](!)
 hasvoicemail       = yes
 subscribemwi       = yes
 subscribecontext   = internal
 vmexten            = voicemail
 [foo](std-user,with-mailbox)
 secret             = asterisk
 callerid           = "Mr. Foo" <11>
 mailbox            = 11@internal
 [bar](std-user,with-mailbox)
 secret             = asterisk
 callerid           = "Mr. Bar" <12>
 mailbox            = 12@internal
 </file>
 <file name="iax.conf">
 ;;
 ;;  iax.conf -- Asterisk IAX configuration
 ;;
 ;; This configuration is reread at reload
 ;; or with the CLI command
 ;;   reload chan_iax2.so
 ;;
 ;; General settings, like port number to bind to, and
 ;; an option address (the default is to bind to all
 ;; local addresses).
 ;;
 ;[general]
 ;bindport=4569       ; bindport and bindaddr may be specified
 ;                    ; NOTE: bindport must be specified BEFORE
 ;                    ; bindaddr or may be specified on a specific
 ;                    ; bindaddr if followed by colon and port
 ;                    ; (e.g. bindaddr=192.168.0.1:4569)
 ;bindaddr=127.0.0.1  ; more than once to bind to multiple
 ;                    ; addresses, but the first will be the
 ;                    ; default
 </file>
 <file name="iaxprov.conf">
 ;;
 ;;  iaxprov.conf -- IAX2 provisioning information
 ;;
 ; Contains provisioning information for templates and for specific service
 ; entries.
 ;
 ; Templates provide a group of settings from which provisioning takes place.
 ; A template may be based upon any template that has been specified before
 ; it.  If the template that an entry is based on is not specified then it is
 ; presumed to be 'default' (unless it is the first of course).
 ;
 ; Templates which begin with 'si-' are used for provisioning units with
 ; specific service identifiers.  For example the entry "si-000364000126"
 ; would be used when the device with the corresponding service identifier of
 ; "000364000126" attempts to register or make a call.
 ;
 [default]
 port=4569  ; the port number the device should bind to (default 4569)
 server=127.0.0.1  ; our PRIMARY server for registration and placing calls
 ; altserver is the BACKUP server for registration and placing calls in the
 ; event the primary server is unavailable.
 ;
 altserver=127.0.0.2
 ; port is the port number to use for IAX2 outbound.  The connections to the
 ; server and altserver (default 4569)
 ;
 serverport=4569
 language=es  ; the preferred language for the device
 codec=ulaw  ; requested codec, the iaxy supports ulaw and adpcm
 ; flags is a comma separated list of flags which the device should
 ; use and may contain any of the following keywords:
 ;
 ;  "register"  - Register with server
 ;  "secure"    - Do not accept calls / provisioning not originated by server
 ;  "heartbeat" - Generate status packets on port 9999 sent to 255.255.255.255
 ;  "debug"     - Output extra debugging to port 9999
 ;
 ;  Note that use can use += and -= to adjust parameters
 ;
 flags=register
 tos=ef  ; see doc/ip-tos.txt
 ; Example iaxy provisioning
 ;
 ;[si-000364000126]
 ;user=iaxy
 ;pass=bitsy
 ;flags += debug
 ;[si-000364000127]
 ;user=iaxy2
 ;pass=bitsy2
 ;template=si-000364000126
 ;flags += debug
 ;  If specified, the '*' provisioning is used for all devices which do not
 ;  have another provisioning entry within the file.  If unspecified, no
 ;  provisioning will take place for devices which have no entry.  DO NOT
 ;  USE A '*' PROVISIONING ENTRY UNLESS YOU KNOW WHAT YOU'RE DOING.
 ;
 ;[*]
 ;template=default
 </file>
 <file name="rtp.conf">
 ;;
 ;;  rtp.conf -- Asterisk RTP configuration
 ;;
 [general]
 rtpstart           = 7070
 rtpend             = 7089
 </file>
 <file name="sip_notify.conf">
 ;;
 ;;  sip_notify.conf -- Asterisk NOTIFY automation from command line
 ;;
 ; rfc3842
 ; put empty "Content=>" at the end to have CRLF after last body line
 [clear-mwi]
 Event=>message-summary
 Content-type=>application/simple-message-summary
 Content=>Messages-Waiting: no
 Content=>Message-Account: sip:asterisk@127.0.0.1
 Content=>Voice-Message: 0/0 (0/0)
 Content=>
 ; Aastra
 [aastra-check-cfg]
 Event=>check-sync
 [aastra-xml]
 Event=>aastra-xml
 ; Linksys
 [linksys-cold-restart]
 Event=>reboot_now
 [linksys-warm-restart]
 Event=>restart_now
 ; Polycom
 [polycom-check-cfg]
 Event=>check-sync
 ; Sipura
 [sipura-check-cfg]
 Event=>resync
 [sipura-get-report]
 Event=>report
 ; Snom
 [snom-check-cfg]
 Event=>check-sync\;reboot=false
 [snom-reboot]
 Event=>reboot
 ; Cisco
 [cisco-check-cfg]
 Event=>check-sync
 </file>
 <file name="extconfig.conf">
 ;;
 ;;  extconfig.conf -- Static and realtime external configuration engine
 ;;
 [settings]
 ;
 ; Static configuration files:
 ;
 ; file.conf => driver,database[,table[,priority]]
 ;
 ; maps a particular configuration file to the given
 ; database driver, database and table (or uses the
 ; name of the file as the table if not specified)
 ;
 ;uncomment to load queues.conf via the odbc engine.
 ;
 ;queues.conf => odbc,asterisk,ast_config
 ;extensions.conf => sqlite,asterisk,ast_config
 ;
 ; The following files CANNOT be loaded from Realtime storage:
 ;	asterisk.conf
 ;	extconfig.conf (this file)
 ;	logger.conf
 ;
 ; Additionally, the following files cannot be loaded from
 ; Realtime storage unless the storage driver is loaded
 ; early using 'preload' statements in modules.conf:
 ;	manager.conf
 ;	cdr.conf
 ;	rtp.conf
 ;
 ;
 ; Realtime configuration engine
 ;
 ; maps a particular family of realtime
 ; configuration to a given database driver,
 ; database and table (or uses the name of
 ; the family if the table is not specified
 ;
 ;example => odbc,asterisk,alttable,1
 ;example => mysql,asterisk,alttable,2
 ;example2 => ldap,"dc=oxymium,dc=net",example2
 ;
 ; Additionally, priorities are now supported for use as failover methods
 ; for retrieving realtime data.  If one connection fails to retrieve any
 ; information, the next sequential priority will be tried next.  This
 ; especially works well with ODBC connections, since res_odbc now caches
 ; when connection failures occur and prevents immediately retrying those
 ; connections until after a specified timeout.  Note:  priorities must
 ; start at 1 and be sequential (i.e. if you have only priorities 1, 2,
 ; and 4, then 4 will be ignored, because there is no 3).
 ;
 ; "odbc" is shown in the examples below, but is not the only valid realtime
 ; engine.  There is:
 ;    odbc ... res_config_odbc
 ;    sqlite ... res_config_sqlite
 ;    pgsql ... res_config_pgsql
 ;    curl ... res_config_curl
 ;    ldap ... res_config_ldap
 ;
 ;iaxusers => odbc,asterisk
 ;iaxpeers => odbc,asterisk
 ;sippeers => odbc,asterisk
 ;sipregs => odbc,asterisk ; (avoid sipregs if possible, e.g. by using a view)
 ;voicemail => odbc,asterisk
 ;extensions => odbc,asterisk
 ;meetme => mysql,general
 ;queues => odbc,asterisk
 ;queue_members => odbc,asterisk
 ;musiconhold => mysql,general
 ;queue_log => mysql,general
 ;
 ;
 ; While most dynamic realtime engines are automatically used when defined in
 ; this file, 'extensions', distinctively, is not.  To activate dynamic realtime
 ; extensions, you must turn them on in each respective context within
 ; extensions.conf with a switch statement.  The syntax is:
 ;      switch => Realtime/[[db_context@]tablename]/<opts>
 ; The only option available currently is the 'p' option, which disallows
 ; extension pattern queries to the database.  If you have no patterns defined
 ; in a particular context, this will save quite a bit of CPU time.  However,
 ; note that using dynamic realtime extensions is not recommended anymore as a
 ; best practice; instead, you should consider writing a static dialplan with
 ; proper data abstraction via a tool like func_odbc.
 </file>
 <file name="extensions.conf">
 ;;
 ;;  extensions.conf -- Asterisk inbound & outbound call configuration
 ;;
 [general]
 static             = yes
 writeprotect       = yes
 autofallthrough    = yes
 [globals]
 MEETME_SPOOLDIR    = @l_prefix@/var/asterisk/spool/meetme
 STAFF              = SIP/foo&SIP/bar
 CONSOLE            = Console/dsp
 DOLLAR             = $
 ;;
 ;;  SPECIAL CONTEXTS
 ;;
 [macro-dial]
 exten              = s,1,Dial(${ARG1},${ARG2},${ARG3})
 exten              = s,n,Goto(s-${DIALSTATUS},1)
 exten              = s-ANSWER,1,Hangup
 exten              = s-BUSY,1,GotoIf($["${ARG4}" == ""]?novm)
 exten              = s-BUSY,n,GotoIf($[${MAILBOX_EXISTS(${ARG4})} == 0]?novm)
 exten              = s-BUSY,n,VoiceMail(${ARG4},b)
 exten              = s-BUSY,n,Playback(vm-goodbye)
 exten              = s-BUSY,n(novm),Hangup
 exten              = s-NOANSWER,1,GotoIf($["${ARG4}" == ""]?novm)
 exten              = s-NOANSWER,n,MailboxExists(${ARG4})
 exten              = s-NOANSWER,n,GotoIf($[${MAILBOX_EXISTS(${ARG4})} == 0]?novm)
 exten              = s-NOANSWER,n,VoiceMail(${ARG4},u)
 exten              = s-NOANSWER,n,Playback(vm-goodbye)
 exten              = s-NOANSWER,n(novm),Hangup
 exten              = _s-.,1,Goto(s-NOANSWER,1)
 [default]
 ;   currently empty
 ;;
 ;;  EXTERNAL DIAL CONTEXT
 ;;
 [external]
 include            = default
 ;   external incoming SIP connection
 exten              = example,hint,${STAFF}
 exten              = example,1,Goto(s,1)
 exten              = s,n,Ringing
 exten              = s,n,Wait(1)
 exten              = s,n,Answer
 exten              = s,n,Macro(dial,${STAFF},30,gTtr,1@external)
 ;   external to internal mapping
 exten              = foo,hint,SIP/foo
 exten              = foo,1,Goto(internal,foo,1)
 exten              = bar,hint,SIP/bar
 exten              = bar,1,Goto(internal,bar,1)
 ;;
 ;;  INTERNAL DIAL CONTEXT
 ;;
 [internal]
 include            = default
 ;include           = parkedcalls
 ;   internal to external mapping
 exten              = example,1,Goto(external,example,1)
 ;   internal user <foo> #11
 exten              = foo,hint,SIP/foo
 exten              = foo,1,Goto(11,1)
 exten              = 11,hint,SIP/foo
 exten              = 11,1,Macro(dial,SIP/foo,30,gTtr,11@internal)
 ;   internal user <bar> #12
 exten              = bar,hint,SIP/bar
 exten              = bar,1,Goto(12,1)
 exten              = 12,hint,SIP/bar
 exten              = 12,1,Macro(dial,SIP/bar,30,gTtr,12@internal)
 ;   internal group <all> #20
 exten              = all,1,Goto(20,1)
 exten              = 20/foo,1,Macro(dial,SIP/bar,60,)
 exten              = 20/bar,1,Macro(dial,SIP/foo,60,)
 ;   internal service <conference> #7<n>
 exten              = conference,1,Goto(70,1)
 exten              = _7[0-9],1,Set(confno=${EXTEN:1})
 exten              = _7[0-9],n,Goto(7,enter)
 exten              = 7,1,Set(TIMEOUT(digit)=3)
 exten              = 7,n,Set(TIMEOUT(response)=6)
 exten              = 7,n(repeat),Read(confno,conf-getconfno,3)
 exten              = 7,n,GotoIf($[${confno} >= 0 & ${confno} <= 9]?enter)
 exten              = 7,n,Playback(conf-invalid)
 exten              = 7,n,Goto(repeat)
 exten              = 7,n(enter),Playback(conf-placeintoconf)
 exten              = 7,n,SayNumber(${confno})
 exten              = 7,n,Set(SPYGROUP=conference-${confno})
 exten              = 7,n,Set(confopt=cCpsMvio)
 exten              = 7,n,GotoIf($[${confno} >= 4 & ${confno} <= 9]?l1:l2)
 exten              = 7,n(l1),Set(confopt=${confopt}i)
 exten              = 7,n(l2),GotoIf($[${confno} >= 7 & ${confno} <= 9]?l3:l4)
 exten              = 7,n(l3),Set(confopt=${confopt}r)
 exten              = 7,n,Set(MEETME_RECORDINGFILE=${MEETME_SPOOLDIR}/meetme-conference-${confno}-${STRFTIME(${EPOCH},UTC,%Y%m%d%H%M)})
 exten              = 7,n,Set(MEETME_RECORDINGFORMAT=wav49)
 exten              = 7,n,Playback(this-call-may-be-monitored-or-recorded)
 exten              = 7,n(l4),MeetMe(${confno},${confopt})
 exten              = 7,n,Playback(beep)
 exten              = 7,n,Wait(1)
 exten              = 7,n,Playback(vm-goodbye)
 exten              = 7,n,Hangup
 ;   internal service <voicemail> #80/#*80<n>
 exten              = voicemail,1,Goto(80,1)
 exten              = 80,1,GotoIf($[${MAILBOX_EXISTS(${CALLERID(num)}@internal)} == 0]?novm)
 exten              = 80,n,VoiceMailMain(${CALLERID(num)}@internal,s)
 exten              = 80,n,Hangup
 exten              = 80,n(novm),Playback(invalid)
 exten              = 80,n,Hangup
 exten              = _*80.,1,GotoIf($[${MAILBOX_EXISTS(${EXTEN:3}@internal)} == 0]?novm)
 exten              = _*80.,n,VoiceMailMain(${EXTEN:3}@internal)
 exten              = _*80.,n,Hangup
 exten              = _*80.,n(novm),Playback(invalid)
 exten              = _*80.,n,Hangup
 ;   internal service <echo> #81
 exten              = echo,1,Goto(81,1)
 exten              = 81,1,Answer
 exten              = 81,n,Playback(demo-echotest)
 exten              = 81,n,Wait(1)
 exten              = 81,n,Playback(beep)
 exten              = 81,n,Echo
 exten              = 81,n,Wait(1)
 exten              = 81,n,Playback(demo-echodone)
 exten              = 81,n,Wait(1)
 exten              = 81,n,Playback(vm-goodbye)
 exten              = 81,n,Hangup
 ;   internal service <reload> #82
 exten              = reload,1,Goto(82,1)
 exten              = 82,1,Answer
 exten              = 82,n,Read(pin,conf-getpin,4)
 exten              = 82,n,GotoIf($[${pin} = 1234]?ok)
 exten              = 82,n,Playback(conf-invalidpin)
 exten              = 82,n,Hangup
 exten              = 82,n(ok),Playback(beep)
 exten              = 82,n,Wait(1)
 exten              = 82,n,Playback(beep)
 exten              = 82,n,Wait(1)
 exten              = 82,n,Playback(beep)
 exten              = 82,n,Wait(1)
 exten              = 82,n,System(@l_prefix@/sbin/asterisk -rx reload)
 exten              = 82,n,Hangup
 ;   external outgoing ISDN (via SIP-to-ISDN gateway call-through)
 ;exten             = _0.,1,Set(number=${EXTEN:1})
 ;exten             = _0.,n,Set(enum=${ENUMLOOKUP(+${number},ALL)})
 ;exten             = _0.,n,Set(enum_is_sip_url=${REGEX("^SIP/.+" ${enum})})
 ;exten             = _0.,n,GotoIf($["${enum_is_sip_url}" = "1"]?sip:isdn)
 ;exten             = _0.,n(sip),Dial(${enum},60,o)
 ;exten             = _0.,n,Goto(_0.,7)
 ;exten             = _0.,n(isdn),Dial(SIP/gw,60,D(w1234w0#31#${number}#))
 ;exten             = _0.,n,Hangup
 ;   internal outgoing SIP call (part 1/2)
 ;   (notice sort-order trickery!)
 include            = internal-siponly
 [internal-siponly]
 ;   internal outgoing SIP call (part 2/2)
 ;   (notice sort-order trickery!)
 exten              = _.[@].,1,Dial(SIP/${EXTEN}@${SIPDOMAIN},60,o)
 exten              = _.[@].,n,Hangup
 exten              = _.[@].,102,Busy
 </file>
 <file name="enum.conf">
 ;;
 ;;  enum.conf -- Asterisk ENUM configuration
 ;;
 [general]
 search             = e164.arpa
 search             = e164.org
 </file>
 <file name="musiconhold.conf">
 ;;
 ;;  musiconhold.conf -- Asterisk music on hold configuration
 ;;
 [default]
 mode               = files
 directory          = @l_prefix@/share/asterisk/moh
 </file>
 <file name="voicemail.conf">
 ;;
 ;;  voicemail.conf -- Asterisk voice mail configuration
 ;;
 [general]
 format             = wav49
 serveremail        = example@example.com
 attach             = yes
 maxmsg             = 20
 maxsecs            = 180
 minsecs            = 3
 maxgreet           = 60
 skipms             = 3000
 maxsilence         = 10
 silencethreshold   = 128
 maxlogins          = 3
 charset            = ISO-8859-1
 pbxskip            = yes
 fromstring         = Asterisk PBX
 usedirectory       = yes
 emailsubject       = [PBX]: New voice message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
 emailbody          = Dear ${VM_NAME},\n\njust wanted to let you know you were left a ${VM_DUR} long\nvoice message (number ${VM_MSGNUM}) in voice mailbox ${VM_MAILBOX}\nfrom caller ${VM_CALLERID},\non ${VM_DATE}.\nYou might want to check it when you get a chance. Thanks!\n\n\t\t\t\t-- OpenPKG Asterisk PBX\n
 pagerfromstring    = Asterisk PBX
 pagersubject       = New VM
 pagerbody          = New ${VM_DUR} long msg in box ${VM_MAILBOX}\nfrom ${VM_CALLERID}, on ${VM_DATE}
 emaildateformat    = %A, %d %B %Y %H:%M:%S %r
 mailcmd            = @l_prefix@/sbin/sendmail -t
 [default]
 [external]
 = 1234,Example,example@example.com,,delete=yes
 [internal]
 = 1234,Mr. Foo,foo@example.com,,delete=no
 = 1234,Mr. Bar,bar@example.com,,delete=no
 </file>
 <file name="cdr.conf">
 ;;
 ;;  cdr.conf -- Asterisk Call Detail Record (CDR) configuration
 ;;
 [general]
 enable             = yes
 unanswered         = no
 batch              = no
 size               = 100
 time               = 300
 scheduleronly      = no
 safeshutdown       = yes
 endbeforehexten    = yes
 </file>
 <file name="cdr_custom.conf">
 ;;
 ;;  cdr_custom.conf -- Asterisk Call Detail Record (CDR) via Comma Separated Value (CSV) format configuration
 ;;
 [mappings]
 master.csv         = "${CDR(start)}", "${CDR(answer)}", "${CDR(end)}", "${CDR(duration)}", "${CDR(billsec)}", "${CDR(clid)}", "${CDR(src)}", "${CDR(dst)}", "${CDR(dcontext)}", "${CDR(channel)}", "${CDR(dstchannel)}", "${CDR(lastapp)}", "${CDR(lastdata)}", "${CDR(disposition)}", "${CDR(amaflags)}", "${CDR(accountcode)}", "${CDR(uniqueid)}", "${CDR(userfield)}"
 </file>
 <file name="cdr_sqlite3_custom.conf">
 ;;
 ;;  cdr_sqlite3_custom.conf -- Asterisk Call Detail Record (CDR) via SQLite RDBMS format configuration
 ;;
 [master]
 table              = cdr
 columns            = start, answer, end, duration, billsec, clid, src, dst, dcontext, channel, dstchannel, lastapp, lastdata, disposition, amaflags, accountcode, uniqueid, userfield
 values             = "${CDR(start)}", "${CDR(answer)}", "${CDR(end)}", "${CDR(duration)}", "${CDR(billsec)}", "${CDR(clid)}", "${CDR(src)}", "${CDR(dst)}", "${CDR(dcontext)}", "${CDR(channel)}", "${CDR(dstchannel)}", "${CDR(lastapp)}", "${CDR(lastdata)}", "${CDR(disposition)}", "${CDR(amaflags)}", "${CDR(accountcode)}", "${CDR(uniqueid)}", "${CDR(userfield)}"
 </file>
 <file name="cdr_manager.conf">
 ;;
 ;;  cdr_manager.conf -- Asterisk Call Detail Record (CDR) via Asterisk Manager Interface (AMI) configuration
 ;;
 [general]
 enabled            = yes
 </file>
 <file name="meetme.conf">
 ;;
 ;;  meetme.conf -- Asterisk conference configuration
 ;;
 [general]
 audiobuffers       = 32
 ;schedule          = yes
 ;logmembercount    = yes
 ;fuzzystart        = 300
 ;earlyalert        = 3600
 ;endalert          = 120
 [rooms]
 conf               = 0
 conf               = 1
 conf               = 2
 conf               = 3
 conf               = 4
 conf               = 5
 conf               = 6
 conf               = 7
 conf               = 8
 conf               = 9,1234,1234
 </file>
 <file name="codecs.conf">
 ;;
 ;;  codecs.conf -- Asterisk codec configuration
 ;;
 [speex]
 quality            = 6
 complexity         = 4
 enhancement        = true
 vad                = true
 vbr                = true
 abr                = 8000
 vbr_quality        = 5
 dtx                = false
 preprocess         = false
 pp_vad             = false
 pp_agc             = false
 pp_agc_level       = 8000
 pp_denoise         = false
 pp_dereverb        = false
 pp_dereverb_decay  = 0.4
 pp_dereverb_level  = 0.3
 [plc]
 genericplc         = true
 </file>
 <file name="chan_dahdi.conf">
 ;;
 ;;  chan_dahdi.conf -- Asterisk DAHDI channel configuration
 ;;
 ;   (an empty configuration is ok, but required even for DAHDI "dahdidummy" only)
 [trunkgroups]
 [channels]
 </file>
 <file name="capi.conf">
 ;;
 ;;  capi.conf -- Asterisk ISDN/CAPI channel configuration
 ;;
 [general]
 nationalprefix     = 0
 internationalprefix= 00
 rxgain             = 1.0
 txgain             = 1.0
 ulaw               = no
 debug              = yes
 [ISDN1]
 isdnmode           = msn
 incomingmsn        = *
 controller         = 0
 group              = 1
 ;prefix            = 0
 softdtmf           = off
 relaxdtmf          = off
 accountcode        =
 context            = external
 holdtype           = local
 ;immediate         = yes
 echocancel         = no
 echosquelch        = no
 ;echotail          = 64
 ;bridge            = yes
 ;callgroup         = 1
 ;deflect           = 1234567
 devices            = 2
 ;dtmf_generate     = yes
 </file>
 <file name="features.conf">
 ;;
 ;;  features.conf -- Asterisk call features configuration
 ;;
 [general]
 ;parkext           = 700
 ;parkpos           = 701-720
 ;context           = parkedcalls
 </file>
 <file name="festival.conf">
 ;;
 ;;  festival.conf -- Asterisk festival configuration
 ;;
 [general]
 host      = localhost  ; default 'localhost'
 port      = 1314       ; default '1314'
 usecache  = no         ; default 'no'
 ; If usecache=yes, a directory to store waveform cache files.
 ; The cache is never cleared (yet), so you must take care of cleaning it
 ; yourself (just delete any or all files from the cache).
 ; THIS DIRECTORY *MUST* EXIST and must be writable from the asterisk process.
 ; Defaults to /tmp/
 ;
 ;cachedir  = /opsw/var/asterisk/festivalcache/
 ;
 ; Festival command to send to the server.
 ; Defaults to: (tts_textasterisk "%s" 'file)(quit)\n
 ; %s is replaced by the desired text to say. The command MUST end with a
 ; (quit) directive, or the cache handling mechanism will hang. Do not
 ; forget the \n at the end.
 ;
 festivalcommand = (tts_textasterisk "%s" 'file)(quit)\n
 </file>
 <file name="gtalk.conf">
 ;;
 ;;  gtalk.conf -- Asterisk GTalk configuration
 ;;
 [general]
 ;context     = default     ; Context to dump call into
 ;bindaddr    = 0.0.0.0     ; Address to bind to
 ;externip    = 127.0.0.1   ; Set your external ip if you are behind a NAT.
 ;stunaddr    = <hostname>  ; Get your external ip from a STUN server.
 ;                          ; Note, if the STUN query is successful, this
 ;                          ; will replace any value placed in externip;
 ;allowguest  = yes         ; Allow calls from people not in list of peers
 [guest]                   ; special account for options on guest account
 ;disallow    = all
 ;allow       = ulaw
 ;context     = guest
 [ogorman]
 ;username    = <person>@gmail.com  ; username of the peer you're
 ;                                  ; calling or accepting calls from
 ;disallow    = all
 ;allow       = ulaw
 ;context     = default
 ;connection  = asterisk            ; client or component in jabber.conf
 ;                                  ; for the call to leave on.
 </file>
 <file name="jabber.conf">
 ;;
 ;;  jabber.conf -- Asterisk Jabber configuration
 ;;
 [general]
 ;debug             = yes
 ;autoprune         = yes
 ;autoregister      = yes
 ;[asterisk]
 ;type              = client
 ;serverhost        = jabber.example.com
 ;username          = asterisk@example.com/asterisk
 ;secret            = asterisk
 ;priority          = 1
 ;port              = 5222
 ;usetls            = no
 ;usesasl           = no
 ;buddy             = buddy@example.com
 ;status            = available
 ;timeout           = 100
 </file>
 <file name="indications.conf">
 ;;
 ;;  indications.conf -- Asterisk tone indications
 ;;
 [general]
 country            = us
 ;   United States
 ;   (according to tones in North America)
 [us]
 description        = United States (US)
 ringcadence        = 2000,4000
 dial               = 350+440
 busy               = 480+620/500,0/500
 ring               = 440+480/2000,0/4000
 congestion         = 480+620/250,0/250
 callwaiting        = 440/300,0/10000
 dialrecall         = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
 record             = 1400/500,0/15000
 info               = !950/330,!1400/330,!1800/330,0
 stutter            = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
 ;   United Kingdom
 ;   (partly according to BT SIN350)
 [uk]
 description        = United Kingdom (UK)
 ringcadence        = 400,200,400,2000
 dial               = 350+440
 busy               = 400/375,0/375
 ring               = 400+450/400,0/200,400+450/400,0/2000
 congestion         = 400/400,0/350,400/225,0/525
 callwaiting        = 400/100,0/4000
 dialrecall         = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
 record             = 1400/500,0/60000
 info               = 950/330,0/15,1400/330,0/15,1800/330,0/1000
 stutter            = 350+440/750,440/750
 ;   Germany
 ;   (according to http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf)
 [de]
 description        = Germany (DE)
 ringcadence        = 1000,4000
 dial               = 425
 busy               = 425/480,0/480
 ring               = 425/1000,0/4000
 congestion         = 425/240,0/240
 callwaiting        = !425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,0
 dialrecall         = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
 record             = 1400/80,0/15000
 info               = 950/330,1400/330,1800/330,0/1000
 stutter            = 425+400
 </file>
 <file name="ccss.conf">
 ;;
 ;;  ccss.conf -- Asterisk Call Completion Supplementary Services configuration
 ;;
 ; Warning! The CCSS implementation brings several changes to the
 ; channel components of Asterisk. To use CCSS, specify the number
 ; of maximum requests in this file but do not forget to specify
 ; the new CCSS related channel specific options in other config
 ; files as well! Some are 'cc_offer_timer', 'ccbs_available_timer',
 ; 'cc_agent_policy=never' and many more (in other files.)
 [general]
 ; There is only a single option that may be defined in this file.
 ; The cc_max_requests option is a global limit on the number of
 ; CC requests that may be in the Asterisk system at any time.
 ;
 cc_max_requests = 20
 </file>
 <file name="res_fax.conf">
 ;;
 ;;  res_fax.conf -- Asterisk fax resource configuration
 ;;
 [general]
 ; Maximum Transmission Rate
 ; Possible values are { 2400 | 4800 | 7200 | 9600 | 12000 | 14400 }
 ; Set this value to the maximum desired transfer rate.  Default: 14400
 maxrate=14400
 ; Minimum Transmission Rate
 ; Possible values are { 2400 | 4800 | 7200 | 9600 | 12000 | 14400 }
 ; Set this value to the minimum desired transfer rate.  Default: 2400
 minrate=2400
 ; Send Progress/Status events to manager session
 ; Manager events with 'call' class permissions will receive events indicating the
 ; steps to initiate a fax session.  Fax completion events are always sent to manager
 ; sessions with 'call' class permissions, regardless of the value of this option.
 ; Default: no
 statusevents=yes
 ; modem capabilities
 ; Possible values are { v17 | v27 | v29 }
 ; Set this value to modify the default modem options.  Default: v17,v27,v29
 modems=v17,v27,v29
 ; Enable/disable T.30 ECM (error correction mode) by default.
 ; Default: Enabled
 ecm=yes
 </file>
 <file name="res_odbc.conf">
 ;;
 ;;  res_odbc.conf -- Asterisk ODBC resource configuration
 ;;
 [ENV]
 [asterisk-sqlite]
 enabled            = no
 dsn                = asterisk-sqlite
 username           =
 password           =
 pre-connect        = no
 sanitysql          = SELECT 1
 ;idlecheck         = 3600
 backslash_is_escape= yes
 share_connections  = yes
 limit              = 10
 </file>
 <file name="func_odbc.conf">
 ;;
 ;;  func_odbc.conf -- Asterisk ODBC dialplan function configuration
 ;;
 ;   SQLite-based Asterisk Database Access (random SQL access)
 ;   Set(<variable_name>=${ASTDB_SQL(SELECT [...])})
 ;   Set(ASTDB_SQL(UPDATE [...]))
 [SQL]
 prefix             = ASTDB
 dsn                = asterisk-sqlite
 readsql            = ${ARG1}
 writesql           = ${ARG1}
 ;   SQLite-based Asterisk Database Access (fixed key/value access)
 ;   Set(<variable_name>=${ASTDB_MAP(<key>)})
 ;   Set(ASTDB_MAP(<key>)=<value>)
 [MAP]
 prefix             = ASTDB
 dsn                = asterisk-sqlite
 readsql            = SELECT val FROM map WHERE key='${SQL_ESC(${ARG1})}'
 writesql           = UPDATE map SET val='${SQL_ESC(${VAL1})}' WHERE key='${SQL_ESC(${ARG1})}'
 escapecommas       = no
 </file>
 <file name="asterisk.pem">
 -----BEGIN CERTIFICATE-----
 MIIDNjCCAp+gAwIBAgIBATANBgkqhkiG9w0BAQQFADCBqTELMAkGA1UEBhMCWFkx
 FTATBgNVBAgTDFNuYWtlIERlc2VydDETMBEGA1UEBxMKU25ha2UgVG93bjEXMBUG
 A1UEChMOU25ha2UgT2lsLCBMdGQxHjAcBgNVBAsTFUNlcnRpZmljYXRlIEF1dGhv
 cml0eTEVMBMGA1UEAxMMU25ha2UgT2lsIENBMR4wHAYJKoZIhvcNAQkBFg9jYUBz
 bmFrZW9pbC5kb20wHhcNOTkxMDIxMTgyMTUxWhcNMDExMDIwMTgyMTUxWjCBpzEL
 MAkGA1UEBhMCWFkxFTATBgNVBAgTDFNuYWtlIERlc2VydDETMBEGA1UEBxMKU25h
 a2UgVG93bjEXMBUGA1UEChMOU25ha2UgT2lsLCBMdGQxFzAVBgNVBAsTDldlYnNl
 cnZlciBUZWFtMRkwFwYDVQQDExB3d3cuc25ha2VvaWwuZG9tMR8wHQYJKoZIhvcN
 AQkBFhB3d3dAc25ha2VvaWwuZG9tMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKB
 gQC554Ro+VH0dJONqljPBW+C72MDNGNy9eXnzejXrczsHs3Pc92Vaat6CpIEEGue
 yG29xagb1o7Gj2KRgpVYcmdx6tHd2JkFW5BcFVfWXL42PV4rf9ziYon8jWsbK2aE
 +L6hCtcbxdbHOGZdSIWZJwc/1Vs70S/7ImW+Zds8YEFiAwIDAQABo24wbDAbBgNV
 HREEFDASgRB3d3dAc25ha2VvaWwuZG9tMDoGCWCGSAGG+EIBDQQtFittb2Rfc3Ns
 IGdlbmVyYXRlZCBjdXN0b20gc2VydmVyIGNlcnRpZmljYXRlMBEGCWCGSAGG+EIB
 AQQEAwIGQDANBgkqhkiG9w0BAQQFAAOBgQB6MRsYGTXUR53/nTkRDQlBdgCcnhy3
 hErfmPNl/Or5jWOmuufeIXqCvM6dK7kW/KBboui4pffIKUVafLUMdARVV6BpIGMI
 LmVFK3sgwuJ01v/90hCt4kTWoT8YHbBLtQh7PzWgJoBAY7MJmjSguYCRt91sU4K
 s0dfWsdItkw4uQ==
 -----END CERTIFICATE-----
 -----BEGIN RSA PRIVATE KEY-----
 MIICXgIBAAKBgQC554Ro+VH0dJONqljPBW+C72MDNGNy9eXnzejXrczsHs3Pc92V
 aat6CpIEEGueyG29xagb1o7Gj2KRgpVYcmdx6tHd2JkFW5BcFVfWXL42PV4rf9zi
 Yon8jWsbK2aE+L6hCtcbxdbHOGZdSIWZJwc/1Vs70S/7ImW+Zds8YEFiAwIDAQAB
 AoGBAKTvnFGKSkUJnNQGe66I0wunGgCA3W7kbarAzEF2qKYhGlZhJQnn68RmVnAW
 pXUFvB+vmtu/+4J9OmWBJsGHFvC9xH32a0PWNr7APjAKrjAD8GWS7Z6BjuxN8QhD
 WlFMmpYhYIjT1jt7RNfs2gJGS2Ryu3zutUQGwtUB9Pou03dJAkEA6yttwVINFqQP
 utgUZ1JUHrN/rE73FzYsF/CwJp5d3rLHenZzLT0iW+kNDLUw/VpzYxK7bF2Qrt/3
 QIUWwm2InQJBAMpe+jhNMJeLDLc3tG3zeithT0mFkuzWWmT2PJgQ0V78UWhw/fSn
 Qqnq7KBY/DNjlfhezrozLDD73/ccmha0Ax8CQQCBaBlyOtNm9QqO116K6HvPlRiZ
 Wa6QQEgNOG3GInknFZu9ILcKWsywZNLAfmgh0gcSqnkmDWqTQD0PbOz0Ok/lAkEA
 g24JrfUbwOASww9PhDUju/a36rTwhhZ0oKt3EP+jKsBOErmHhZP3bKlhQoZoTOu5
 Y5QXSMChS7LZcwDFZkdE2wJATRgMbhErif+ZRwt9XJRdCo5Sx6ewyGyxjc5gvUyK
 KegHcgru/ZC3pGlujRD2LqxgJNAn5QTdW4LK8xVPFySTYg==
 -----END RSA PRIVATE KEY-----
 </file>

OpenPKG Specs / annotate

asterisk/asterisk.txt@a2c6460cfb16 (annotated)

asterisk/asterisk.txt