OpenPKG Specs: asterisk/asterisk.txt@fd611cde817f (annotated)

asterisk/asterisk.txt@fd611cde817f (annotated)

asterisk/asterisk.txt

Sat, 24 Mar 2012 21:40:49 +0100

author: Michael Schloh von Bennewitz <michael@schloh.com>
date: Sat, 24 Mar 2012 21:40:49 +0100
changeset 414: fd611cde817f
parent 399: 9360a38b5c9f
permissions: -rw-r--r--

Introduce many changes to the buildconf and source code including:
(01) clean up, update, and partially update default config files,
(02) seems that Melware is unable to perform release engineering so
update chan_capi to new daily snapshot to solve echo problems,
(03) correct Asterisk inadequate hard coded gmime version check,
(04) force postgresql pthreads linkage to solve build problem,
(05) remove buggy hard coded LibXML configure definitions,
(06) remove local architecture specification to allow GCC
internal logic to determine proper CPU type instead,
(07) remove vendor sound install target causing uncontrolled
downloads and non RPM managed file installation,
(08) solve long outstanding bug in tcptls causing Asterisk
to ignore any intermediate CA certificate signatures,
(09) back out Digium engineering team's bright idea of replacing the
very portable and pervasive POSIX rand(1) with ast_random(), and
then not even implementing it causing all references to fail in
platforms not providing the very new POSIX.1-2008 mkdtemp(3)
function only distributed by BSD and some Linux,
(10) withdraw advanced linker symbol manipulations from SVR5 builds
until either Binutils supports hybrid versioned and anonymous
linker scripts or GCC stops hard coding versioned linker scripts,
(11) correct missing library linkage, some tailored to a specific OS,
(12) remove outdated logic for the no longer distributed gmime-config(1),
(13) remove local gmime buildconf hacks now that Asterisk has corrected
their own build configuration to almost portably support gmime,
(14) solve build problems relating to undetected LibXML paths,
(15) correct erroneous out of tree include definitions,
(16) improve some variable and comment naming,
(17) simplify sound language path hierarchy creation,
and correct australian english installation logic.

 <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
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 MAkGA1UEBhMCWFkxFTATBgNVBAgTDFNuYWtlIERlc2VydDETMBEGA1UEBxMKU25h
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 gQC554Ro+VH0dJONqljPBW+C72MDNGNy9eXnzejXrczsHs3Pc92Vaat6CpIEEGue
 yG29xagb1o7Gj2KRgpVYcmdx6tHd2JkFW5BcFVfWXL42PV4rf9ziYon8jWsbK2aE
 +L6hCtcbxdbHOGZdSIWZJwc/1Vs70S/7ImW+Zds8YEFiAwIDAQABo24wbDAbBgNV
 HREEFDASgRB3d3dAc25ha2VvaWwuZG9tMDoGCWCGSAGG+EIBDQQtFittb2Rfc3Ns
 IGdlbmVyYXRlZCBjdXN0b20gc2VydmVyIGNlcnRpZmljYXRlMBEGCWCGSAGG+EIB
 AQQEAwIGQDANBgkqhkiG9w0BAQQFAAOBgQB6MRsYGTXUR53/nTkRDQlBdgCcnhy3
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 LmVFK3sgwuJ01v/90hCt4kTWoT8YHbBLtQh7PzWgJoBAY7MJmjSguYCRt91sU4K
 s0dfWsdItkw4uQ==
 -----END CERTIFICATE-----
 -----BEGIN RSA PRIVATE KEY-----
 MIICXgIBAAKBgQC554Ro+VH0dJONqljPBW+C72MDNGNy9eXnzejXrczsHs3Pc92V
 aat6CpIEEGueyG29xagb1o7Gj2KRgpVYcmdx6tHd2JkFW5BcFVfWXL42PV4rf9zi
 Yon8jWsbK2aE+L6hCtcbxdbHOGZdSIWZJwc/1Vs70S/7ImW+Zds8YEFiAwIDAQAB
 AoGBAKTvnFGKSkUJnNQGe66I0wunGgCA3W7kbarAzEF2qKYhGlZhJQnn68RmVnAW
 pXUFvB+vmtu/+4J9OmWBJsGHFvC9xH32a0PWNr7APjAKrjAD8GWS7Z6BjuxN8QhD
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 Qqnq7KBY/DNjlfhezrozLDD73/ccmha0Ax8CQQCBaBlyOtNm9QqO116K6HvPlRiZ
 Wa6QQEgNOG3GInknFZu9ILcKWsywZNLAfmgh0gcSqnkmDWqTQD0PbOz0Ok/lAkEA
 g24JrfUbwOASww9PhDUju/a36rTwhhZ0oKt3EP+jKsBOErmHhZP3bKlhQoZoTOu5
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 KegHcgru/ZC3pGlujRD2LqxgJNAn5QTdW4LK8xVPFySTYg==
 -----END RSA PRIVATE KEY-----
 </file>

OpenPKG Specs / annotate

asterisk/asterisk.txt@fd611cde817f (annotated)

asterisk/asterisk.txt