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80
acl.conf Normal file
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;
; Named Access Control Lists (ACLs)
;
; A convenient way to share acl definitions
;
; This configuration file is read on startup
;
; CLI Commands
; -----------------------------------------------------------
; acl show Show all named ACLs configured
; acl show <name> Show contents of a particular named ACL
; reload acl Reload configuration file
;
; Any configuration that uses ACLs which has been made to be able to use named
; ACLs will specify a named ACL with the 'acl' option in its configuration in
; a similar fashion to the usual 'permit' and 'deny' options. Example:
; acl=my_named_acl
;
; Multiple named ACLs can be applied by either comma separating the arguments or
; just by adding additional ACL lines. Example:
; acl=my_named_acl
; acl=my_named_acl2
;
; or
;
; acl=my_named_acl,my_named_acl2
;
; ACLs specified by name are evaluated independently from the ACL specified via
; permit/deny. In order for an address to pass a given ACL, it must pass both
; the ACL specified by permit/deny for a given item as well as any named ACLs
; that were specified.
;
;[example_named_acl1]
;deny=0.0.0.0/0.0.0.0
;permit=209.16.236.0
;permit=209.16.236.1
;
;[example_named_acl2]
;permit=0.0.0.0/0.0.0.0
;deny=10.24.20.171
;deny=10.24.20.103
;deny=209.16.236.1
;
; example_named_acl1 above shows an example of whitelisting. When whitelisting, the
; named ACLs should follow a deny that blocks everything (like deny=0.0.0.0/0.0.0.0)
; The following example explains how combining the ACLs works:
; <in another configuration>
; [example_item_with_acl]
; acl=example_named_acl1
; acl=example_named_acl2
;
; Suppose 209.16.236.0 tries to communicate and the ACL for that example is applied to it...
; First, example_named_acl1 is evaluated. The address is allowed by that ACL.
; Next, example_named_acl2 is evaluated. The address isn't blocked by example_named_acl2
; either, so it passes.
;
; Suppose instead 209.16.236.1 tries to communicate and the same ACL is applied.
; First, example_named_acl1 is evaluated and the address is allowed.
; However, it is blocked by example_named_acl2, so the address is blocked from the combined
; ACL.
;
; Similarly, the permits/denies in specific configurations that make up an ACL definition
; are also treated as a separate ACL for evaluation. So if we change the example above to:
; <in another configuration>
; [example_item_with_acl]
; acl=example_named_acl1
; acl=example_named_acl2
; deny=209.16.236.0
;
; Then 209.16.236.0 will be rejected by the non-named component of the combined ACL even
; though it passes the two named components.
;
;
; Named ACLs can use ipv6 addresses just like normal ACLs.
;[ipv6_example_1]
;deny = ::
;permit = ::1/128
;
;[ipv6_example_2]
;permit = fe80::21d:bad:fad:2323

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adsi.conf Normal file
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;
; Sample ADSI Configuration file
;
[intro]
alignment = center
greeting => Welcome to the
greeting => Asterisk
greeting => Open Source PBX

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;
; Agent pool configuration
;
[general]
; The general section of this config is not currently used, but reserved
; for future use.
;[agent-id]
; Define ackcall to require the agent to give a DTMF acknowledgement
; when the agent receives a call.
; The channel variable AGENTACKCALL overrides on agent login.
; Default is "no".
;ackcall=no
;
; Set what DTMF key sequence the agent should use to acknowledge a call.
; The channel variable AGENTACCEPTDTMF overrides on agent login.
; This option is ignored unless ackcall is enabled.
; Default is "#".
;acceptdtmf=##
;
; Set how many seconds a call for the agent has to wait for the agent to
; acknowledge the call before the agent is automatically logged off. If
; set to zero then the call will wait forever for the agent to acknowledge.
; The channel variable AGENTAUTOLOGOFF overrides on agent login.
; This option is ignored unless ackcall is enabled.
; Default is 0.
;autologoff=15
;
; Set the minimum amount of time after disconnecting a call before
; the agent can receive a new call in milliseconds.
; The channel variable AGENTWRAPUPTIME overrides on agent login.
; Default is 0.
;wrapuptime=5000
;
; Set the musiconhold class for the agent.
; Default is "default".
;musiconhold=default
;
; Enable recording calls the agent takes automatically by invoking the
; DTMF automixmon feature when the agent connects to a caller.
; See features.conf.sample for information about the automixmon feature.
; Default is "no".
;recordagentcalls=yes
;
; The sound file played to alert the agent when a call is present.
; Default is "beep".
;custom_beep=beep
;
; A friendly name for the agent used in log messages.
; Default is "".
;fullname=Mark Spencer
;
; --------------------------------------------------
;
; This section contains example agent definitions:
;
; Define a template called my-agents:
;[my-agents](!)
;autologoff=15
;ackcall=yes
;acceptdtmf=##
;
; Define agent 1001 using the my-agents template:
;[1001](my-agents)
;fullname=Mark Spencer
;
; Define agent 1002 using the my-agents template:
;[1002](my-agents)
;fullname=Will Meadows

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;
; Answering Machine Detection Configuration
;
[general]
total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide
; on whether the audio represents a HUMAN, or a MACHINE
silence_threshold = 256 ; If the average level of noise in a sample does not reach
; this value, from a scale of 0 to 32767, then we will consider
; it to be silence.
; Greeting ;
initial_silence = 2500 ; Maximum silence duration before the greeting.
; If exceeded, then the result is detection as a MACHINE.
after_greeting_silence = 800 ; Silence after detecting a greeting.
; If exceeded, then the result is detection as a HUMAN
greeting = 1500 ; Maximum length of a greeting. If exceeded, then the
; result is detection as a MACHINE.
; Word detection ;
min_word_length = 100 ; Minimum duration of Voice to considered as a word
maximum_word_length = 5000 ; Maximum duration of a single Voice utterance allowed.
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 = 2 ; Maximum number of words in the greeting
; If exceeded, then the result is detection as a MACHINE

27
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[general]
games=3
cheat=no
[sounds]
prompt=please-enter-your,number,queue-less-than
wrong_guess=vm-pls-try-again
right_guess=auth-thankyou
too_high=high
too_low=low
lose=vm-goodbye
[easy]
max_number=10
max_guesses=4
[medium]
max_number=100
max_guesses=6
[hard]
max_number=1000
max_guesses=7
[nightmare]
max_number=1000
max_guesses=1

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[general]
enabled = yes ; When set to no, ARI support is disabled.
;pretty = no ; When set to yes, responses from ARI are
; ; formatted to be human readable.
;allowed_origins = ; Comma separated list of allowed origins, for
; ; Cross-Origin Resource Sharing. May be set to * to
; ; allow all origins.
;auth_realm = ; Realm to use for authentication. Defaults to Asterisk
; ; REST Interface.
;
; Default write timeout to set on websockets. This value may need to be adjusted
; for connections where Asterisk must write a substantial amount of data and the
; receiving clients are slow to process the received information. Value is in
; milliseconds; default is 100 ms.
;websocket_write_timeout = 100
;
; Display certain channel variables every time a channel-oriented
; event is emitted:
;
;channelvars = var1,var2,var3
;[username]
;type = user ; Specifies user configuration
;read_only = no ; When set to yes, user is only authorized for
; ; read-only requests.
;
;password = ; Crypted or plaintext password (see password_format).
;
; password_format may be set to plain (the default) or crypt. When set to crypt,
; crypt(3) is used to validate the password. A crypted password can be generated
; using mkpasswd -m sha-512.
;
; When set to plain, the password is in plaintext.
;
;password_format = plain

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;
; Asterisk default ADSI script
;
;
; Begin with the preamble requirements
;
DESCRIPTION "Asterisk PBX" ; Name of vendor
VERSION 0x00 ; Version of stuff
;SECURITY "_AST" ; Security code
SECURITY 0X9BDBF7AC ; Security code
FDN 0x0000000F ; Descriptor number
;
; Flags
;
FLAG "nocallwaiting"
;
; Predefined strings
;
DISPLAY "titles" IS "** Asterisk PBX **"
DISPLAY "talkingto" IS "Call active." JUSTIFY LEFT
DISPLAY "callname" IS "$Call1p" JUSTIFY LEFT
DISPLAY "callnum" IS "$Call1s" JUSTIFY LEFT
DISPLAY "incoming" IS "Incoming call!" JUSTIFY LEFT
DISPLAY "ringing" IS "Calling... " JUSTIFY LEFT
DISPLAY "callended" IS "Call ended." JUSTIFY LEFT
DISPLAY "missedcall" IS "Missed call." JUSTIFY LEFT
DISPLAY "busy" IS "Busy." JUSTIFY LEFT
DISPLAY "reorder" IS "Reorder." JUSTIFY LEFT
DISPLAY "cwdisabled" IS "Callwait disabled"
DISPLAY "empty" IS "asdf"
;
; Begin soft key definitions
;
KEY "callfwd" IS "CallFwd" OR "Call Forward"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "*60"
GOTO "offHook"
ENDKEY
KEY "vmail_OH" IS "VMail" OR "Voicemail"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "8500"
ENDKEY
KEY "vmail" IS "VMail" OR "Voicemail"
SENDDTMF "8500"
ENDKEY
KEY "backspace" IS "BackSpc" OR "Backspace"
BACKSPACE
ENDKEY
KEY "cwdisable" IS "CWDsble" OR "Disable Call Wait"
SENDDTMF "*70"
SETFLAG "nocallwaiting"
SHOWDISPLAY "cwdisabled" AT 4
TIMERCLEAR
TIMERSTART 1
ENDKEY
KEY "cidblock" IS "CIDBlk" OR "Block Callerid"
SENDDTMF "*67"
SETFLAG "nocallwaiting"
ENDKEY
;
; Begin main subroutine
;
SUB "main" IS
IFEVENT NEARANSWER THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "talkingto" AT 2 NOUPDATE
SHOWDISPLAY "callname" AT 3
SHOWDISPLAY "callnum" AT 4
GOTO "stableCall"
ENDIF
IFEVENT OFFHOOK THEN
CLEAR
CLEARFLAG "nocallwaiting"
CLEARDISPLAY
SHOWDISPLAY "titles" AT 1
SHOWKEYS "vmail"
SHOWKEYS "cidblock"
SHOWKEYS "cwdisable" UNLESS "nocallwaiting"
GOTO "offHook"
ENDIF
IFEVENT IDLE THEN
CLEAR
SHOWDISPLAY "titles" AT 1
SHOWKEYS "vmail_OH"
ENDIF
IFEVENT CALLERID THEN
CLEAR
; SHOWDISPLAY "titles" AT 1 NOUPDATE
; SHOWDISPLAY "incoming" AT 2 NOUPDATE
SHOWDISPLAY "callname" AT 3 NOUPDATE
SHOWDISPLAY "callnum" AT 4
ENDIF
IFEVENT RING THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "incoming" AT 2
ENDIF
IFEVENT ENDOFRING THEN
SHOWDISPLAY "missedcall" AT 2
CLEAR
SHOWDISPLAY "titles" AT 1
SHOWKEYS "vmail_OH"
ENDIF
IFEVENT TIMER THEN
CLEAR
SHOWDISPLAY "empty" AT 4
ENDIF
ENDSUB
SUB "offHook" IS
IFEVENT FARRING THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "ringing" AT 2 NOUPDATE
SHOWDISPLAY "callname" at 3 NOUPDATE
SHOWDISPLAY "callnum" at 4
ENDIF
IFEVENT FARANSWER THEN
CLEAR
SHOWDISPLAY "talkingto" AT 2
GOTO "stableCall"
ENDIF
IFEVENT BUSY THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "busy" AT 2 NOUPDATE
SHOWDISPLAY "callname" at 3 NOUPDATE
SHOWDISPLAY "callnum" at 4
ENDIF
IFEVENT REORDER THEN
CLEAR
SHOWDISPLAY "titles" AT 1 NOUPDATE
SHOWDISPLAY "reorder" AT 2 NOUPDATE
SHOWDISPLAY "callname" at 3 NOUPDATE
SHOWDISPLAY "callnum" at 4
ENDIF
ENDSUB
SUB "stableCall" IS
IFEVENT REORDER THEN
SHOWDISPLAY "callended" AT 2
ENDIF
ENDSUB

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asterisk.conf Normal file
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[directories](!)
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /usr/share/asterisk
astagidir => /usr/share/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk
astsbindir => /usr/sbin
[options]
;verbose = 3
;debug = 3
;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.
; Changing the following lines may compromise your security.
;[files]
;astctlpermissions = 0660
;astctlowner = root
;astctlgroup = apache
;astctl = asterisk.ctl

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;[calendar1]
;type = ical ; type of calendar--currently supported: ical, caldav, exchange, or ews
;url = https://example.com/home/jdoe/Calendar/ ; URL to shared calendar (Zimbra example)
;user = jdoe ; web username
;secret = supersecret ; web password
;refresh = 15 ; refresh calendar every n minutes
;timeframe = 60 ; number of minutes of calendar data to pull for each refresh period
; ; should always be >= refresh
;fetch_again_at_reload = no ; to reload the calendar content when the module is reloaded
;
;
; You can set up res_calendar to execute a call upon an upcoming busy status
; The following fields are available from the ${CALENDAR_EVENT(<field>)} dialplan function:
;
; summary : The VEVENT Summary property or Exchange subject
; description : The text description of the vent
; organizer : The organizer of the event
; location : The location field of the event
; calendar : The name of the calendar tied to the event
; uid : The unique ID for this event
; start : Start time of the event
; end : The end time of the event
; busystate : 0=FREE, 1=TENTATIVE, 2=BUSY
;
;autoreminder = 10 ; Override event-defined reminder before each busy status (in mins)
;
;channel = SIP/60001 ; Channel to dial
;context = default ; Context to connect to on answer
;extension = 123 ; Extension to connect to on answer
;
; or
;
;app = Playback ; Application to execute on answer (instead of context/extension)
;appdata = tt-weasels ; Data part of application to execute on answer
;
;waittime = 30 ; How long to wait for an answer, defaults to 30 seconds
;
; Channel variables can be set on the notification channel. The format is
; setvar=name=value. Variable subsitution is done on the value to allow the use of dialplan
; functions like CALENDAR_EVENT. The variables are set in order, so one can use the value
; of earlier variables in the definition of later ones.
;
;setvar = CALLERID(name)=${CALENDAR_EVENT(summary)}
;[calendar2]
; Note: Support for Exchange Server 2003
;
;type = exchange ; type of calendar--currently supported: ical, caldav, exchange, or ews
;url = https://example.com/exchange/jdoe ; URL to MS Exchange OWA for user (usually includes exchange/user)
;user = jdoe ; Exchange username
;secret = mysecret ; Exchange password
;refresh = 15 ; refresh calendar every n minutes
;timeframe = 60 ; number of minutes of calendar data to pull for each refresh period
; ; should always be >= refresh
;
; You can set up res_calendar to execute a call upon an upcoming busy status
;autoreminder = 10 ; Override event-defined reminder before each busy status (in mins)
;
;channel = SIP/1234 ; Channel to dial
;context = default ; Context to connect to on answer
;extension = 1234 ; Extension to connect to on answer
;
; or
;
;[calendar3]
; Note: Support for Exchange Server 2007+
;
;type = ews ; type of calendar--currently supported: ical, caldav, exchange, or ews
;url = https://example.com/ews/Exchange.asmx ; URL to MS Exchange EWS
;user = jdoe ; Exchange username
;secret = mysecret ; Exchange password
;refresh = 15 ; refresh calendar every n minutes
;timeframe = 60 ; number of minutes of calendar data to pull for each refresh period
; ; should always be >= refresh
;
; You can set up res_calendar to execute a call upon an upcoming busy status
;autoreminder = 10 ; Override event-defined reminder before each busy status (in mins)
;
;channel = SIP/1234 ; Channel to dial
;context = default ; Context to connect to on answer
;extension = 1234 ; Extension to connect to on answer
;
; or
;
;app = Playback ; Application to execute on answer (instead of context/extension)
;appdata = tt-weasels ; Data part of application to execute on answer
;
;waittime = 30 ; How long to wait for an answer, defaults to 30 seconds
;[calendar4]
;type = caldav ; type of calendar--currently supported: ical, caldav, exchange, or ews
;url = https://www.google.com/calendar/dav/username@gmail.com/events/ ; Main GMail calendar (the trailing slash is significant!)
;user = jdoe@gmail.com ; username
;secret = mysecret ; password
;refresh = 15 ; refresh calendar every n minutes
;timeframe = 60 ; number of minutes of calendar data to pull for each refresh period
; ; should always be >= refresh
;
; You can set up res_calendar to execute a call upon an upcoming busy status
;autoreminder = 10 ; Override event-defined reminder before each busy status (in mins)
;
;channel = SIP/1234 ; Channel to dial
;context = default ; Context to connect to on answer
;extension = 1234 ; Extension to connect to on answer
;
; or
;
;app = Playback ; Application to execute on answer (instead of context/extension)
;appdata = tt-weasels ; Data part of application to execute on answer
;
;waittime = 30 ; How long to wait for an answer, defaults to 30 seconds

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;
; --- Call Completion Supplementary Services ---
;
; For more information about CCSS, see the CCSS user documentation
; https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+(CCSS)
;
[general]
; 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
;
; The cc_STATE_devstate variables listed below can be used to change the
; default mapping of the internal state machine tracking the state of
; call completion to an Asterisk Device State value. The acceptable values
; that can be provided are as follows, with a description of what the
; equivalent device BLF that this maps to:
;
; UNKNOWN ; Device is valid but channel didn't know state
; NOT_INUSE ; Device is not used
; INUSE ; Device is in use
; BUSY ; Device is busy
; INVALID ; Device is invalid
; UNAVAILABLE ; Device is unavailable
; RINGING ; Device is ringing
; RINGINUSE ; Device is ringing *and* in use
; ONHOLD ; Device is on hold
;
; These states are used to generate DEVICE_STATE information that can be
; included with Asterisk hints for phones to subscribe to the state information
; or dialplan to check the state using the EXTENSION_STATE() function or
; the DEVICE_STATE() function.
;
; The states are in the format of: "ccss:TECH/ID" so an example of device
; SIP/3000 making a CallCompletionRequest() could be checked by looking at
; DEVICE_STATE(ccss:SIP/3000) or an Asterisk Hint could be generated such as
;
; [hint-context]
; exten => *843000,hint,ccss:SIP/3000
;
; and then accessed with EXTENSION_STATE(*843000@hint-context)
; or subscribed to with a BLF button on a phone.
;
; The available state mapping and default values are:
;
; cc_available_devstate = NOT_INUSE
; cc_offered_devstate = NOT_INUSE
; cc_caller_requested_devstate = NOT_INUSE
; cc_active_devstate = INUSE
; cc_callee_ready_devstate = INUSE
; cc_caller_busy_devstate = ONHOLD
; cc_recalling_devstate = RINGING
; cc_complete_devstate = NOT_INUSE
; cc_failed_devstate = NOT_INUSE
;
;============================================
; PLEASE READ THIS!!!
; The options described below should NOT be
; set in this file. Rather, they should be
; set per-device in a channel driver
; configuration file.
; PLEASE READ THIS!!!
;===========================================
;
; --------------------------------------------------------------------
; Timers
; --------------------------------------------------------------------
;There are three configurable timers for all types of CC: the
;cc_offer_timer, the ccbs_available_timer, and the ccnr_available_timer.
;In addition, when using a generic agent, there is a fourth timer,
;the cc_recall_timer. All timers are configured in seconds, and the
;values shown below are the defaults.
;
;When a caller is offered CCBS or CCNR, the cc_offer_timer will
;be started. If the caller does not request CC before the
;cc_offer_timer expires, then the caller will be unable to request
;CC for this call.
;
;cc_offer_timer = 20
;
;Once a caller has requested CC, then either the ccbs_available_timer
;or the ccnr_available_timer will run, depending on the service
;requested. The reason why there are two separate timers for CCBS
;and CCNR is that it is reasonable to want to have a shorter timeout
;configured for CCBS than for CCNR. If the available timer expires
;before the called party becomes available, then the CC attempt
;will have failed and monitoring of the called party will stop.
;
;ccbs_available_timer = 4800
;ccnr_available_timer = 7200
;
; When using a generic agent, the original caller is called back
; when one of the original called parties becomes available. The
; cc_recall_timer tells Asterisk how long it should let the original
; caller's phone ring before giving up. Please note that this parameter
; only affects operation when using a generic agent.
;
;cc_recall_timer = 20
; --------------------------------------------------------------------
; Policies
; --------------------------------------------------------------------
; Policy settings tell Asterisk how to behave and what sort of
; resources to allocate in order to facilitate CC. There are two
; settings to control the actions Asterisk will take.
;
; The cc_agent_policy describes the behavior that Asterisk will
; take when communicating with the caller during CC. There are
; three possible options.
;
;never: Never offer CC to the caller. Setting the cc_agent_policy
; to this value is the way to disable CC for a call.
;
;generic: A generic CC agent is one which uses no protocol-specific
; mechanisms to offer CC to the caller. Instead, the caller
; requests CC using a dialplan function. Due to internal
; restrictions, you should only use a generic CC agent on
; phones (i.e. not "trunks"). If you are using phones which
; do not support a protocol-specific method of using CC, then
; generic CC agents are what you should use.
;
;native: A native CC agent is one which uses protocol-specific
; signaling to offer CC to the caller and accept CC requests
; from the caller. The supported protocols for native CC
; agents are SIP, ISDN ETSI PTP, ISDN ETSI PTMP, and Q.SIG
;cc_agent_policy=never
;
; The cc_monitor_policy describes the behavior that Asterisk will
; take when communicating with the called party during CC. There
; are four possible options.
;
;never: Analogous to the cc_agent_policy setting. We will never
; attempt to request CC services on this interface.
;
;generic: Analogous to the cc_agent_policy setting. We will monitor
; the called party's progress using protocol-agnostic
; capabilities. Like with generic CC agents, generic CC
; monitors should only be used for phones.
;
;native: Analogous to the cc_agent_policy setting. We will use
; protocol-specific methods to request CC from this interface
; and to monitor the interface for availability.
;
;always: If an interface is set to "always," then we will accept
; protocol-specific CC offers from the caller and use
; a native CC monitor for the remainder of the CC transaction.
; However, if the interface does not offer protocol-specific
; CC, then we will fall back to using a generic CC monitor
; instead. This is a good setting to use for phones for which
; you do not know if they support protocol-specific CC
; methodologies.
;cc_monitor_policy=never
;
;
; --------------------------------------------------------------------
; Limits
; --------------------------------------------------------------------
;
; The use of CC requires Asterisk to potentially use more memory than
; some administrators would like. As such, it is a good idea to limit
; the number of CC requests that can be in the system at a given time.
; The values shown below are the defaults.
;
; The cc_max_agents setting limits the number of outstanding CC
; requests a caller may have at any given time. Please note that due
; to implementation restrictions, this setting is ignored when using
; generic CC agents. Generic CC agents may only have one outstanding
; CC request.
;
;cc_max_agents = 5
;
; The cc_max_monitors setting limits the number of outstanding CC
; requests can be made to a specific interface at a given time.
;
;cc_max_monitors = 5
;
; --------------------------------------------------------------------
; Other
; --------------------------------------------------------------------
;
; When using a generic CC agent, the caller who requested CC will be
; called back when a called party becomes available. When the caller
; answers his phone, the administrator may opt to have a macro run.
; What this macro does is up to the administrator. By default there
; is no callback macro configured.
;
;cc_callback_macro=
;
; Alternatively, the administrator may run a subroutine. By default
; there is no callback subroutine configured. The subroutine should
; be specified in the format: [[context,]exten,]priority
;
;cc_callback_sub=
;
; When using an ISDN phone and a generic CC agent, Asterisk is unable
; to determine the dialstring that should be used when calling back
; the original caller. Furthermore, if you desire to use any dialstring-
; specific options, such as distinctive ring, you must set this
; configuration option. For non-ISDN phones, it is not necessary to
; set this, since Asterisk can determine the dialstring to use since
; it is identical to the name of the calling device. By default, there
; is no cc_agent_dialstring set.
;
;cc_agent_dialstring=

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;
; Asterisk Channel Event Logging (CEL)
;
; Channel Event Logging is a mechanism to provide fine-grained event information
; that can be used to generate billing information. Such event information can
; be recorded to various backend modules.
;
[general]
; CEL Activation
;
; Use the 'enable' keyword to turn CEL on or off.
;
; Accepted values: yes and no
; Default value: no
;enable=yes
; Application Tracking
;
; Use the 'apps' keyword to specify the list of applications for which you want
; to receive CEL events. This is a comma separated list of Asterisk dialplan
; applications, such as Dial, Queue, and Park.
;
; Accepted values: A comma separated list of Asterisk dialplan applications
; Default value: none
;
; Note: You may also use 'all' which will result in CEL events being reported
; for all Asterisk applications. This may affect Asterisk's performance
; significantly.
apps=dial,park
; Event Tracking
;
; Use the 'events' keyword to specify the list of events which you want to be
; raised when they occur. This is a comma separated list of the values in the
; table below.
;
; Accepted values: A comma separated list of one or more of the following:
; ALL -- Generate entries on all events
; CHAN_START -- The time a channel was created
; CHAN_END -- The time a channel was terminated
; ANSWER -- The time a channel was answered (ie, phone taken off-hook)
; HANGUP -- The time at which a hangup occurred
; BRIDGE_ENTER -- The time a channel was connected into a conference room
; BRIDGE_EXIT -- The time a channel was removed from a conference room
; APP_START -- The time a tracked application was started
; APP_END -- the time a tracked application ended
; PARK_START -- The time a call was parked
; PARK_END -- Unpark event
; BLINDTRANSFER -- When a blind transfer is initiated
; ATTENDEDTRANSFER -- When an attended transfer is initiated
; PICKUP -- This channel picked up the specified channel
; FORWARD -- This channel is being forwarded somewhere else
; LINKEDID_END -- The last channel with the given linkedid is retired
; USER_DEFINED -- Triggered from the dialplan, and has a name given by the
; user
; LOCAL_OPTIMIZE -- A local channel pair is optimizing away.
;
; Default value: none
; (Track no events)
events=APP_START,CHAN_START,CHAN_END,ANSWER,HANGUP,BRIDGE_ENTER,BRIDGE_EXIT
; Date Format
;
; Use the 'dateformat' keyword to specify the date format used when CEL events
; are raised.
;
; Accepted values: A strftime format string (see man strftime)
;
; Example: "%F %T"
; -> This gives the date and time in the format "2009-06-23 17:02:35"
;
; If this option is not specified, the default format is "<seconds>.<microseconds>"
; since epoch. The microseconds field will always be 6 digits in length, meaning it
; may have leading zeros.
;
;dateformat = %F %T
;
; Asterisk Manager Interface (AMI) CEL Backend
;
[manager]
; AMI Backend Activation
;
; Use the 'enable' keyword to turn CEL logging to the Asterisk Manager Interface
; on or off.
;
; Accepted values: yes and no
; Default value: no
;enabled=yes
; Use 'show_user_defined' to put "USER_DEFINED" in the EventName header,
; instead of (by default) just putting the user defined event name there.
; When enabled the UserDefType header is added for user defined events to
; provide the user defined event name.
;
;show_user_defined=yes
;
; RADIUS CEL Backend
;
[radius]
;
; Log date/time in GMT
;usegmtime=yes
;
; Set this to the location of the radiusclient-ng configuration file
; The default is /etc/radiusclient-ng/radiusclient.conf
;radiuscfg => /usr/local/etc/radiusclient-ng/radiusclient.conf
;

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;
; Asterisk CLI configuration
;
[startup_commands]
;
; Any commands listed in this section will get automatically executed
; when Asterisk starts as a daemon or foreground process (-c).
;
;sip set debug on = yes
;core set verbose 3 = yes
;core set debug 1 = yes

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;
; CLI Aliases configuration
;
; This module also registers a "cli show aliases" CLI command to list
; configured CLI aliases.
[general]
; Here you define what alias templates you want to use. You can also define
; multiple templates to use as well. If you do, and there is a conflict, then
; the first alias defined will win.
;
template = friendly ; By default, include friendly aliases
;template = asterisk_1dot2 ; Asterisk 1.2 style syntax
;template = asterisk_1dot4 ; Asterisk 1.4 style syntax
;template = individual_custom ; see [individual_custom] example below which
; includes a list of aliases from an external
; file
; Because the Asterisk CLI syntax follows a "module verb argument" syntax,
; sometimes we run into an issue between being consistant with this format
; in the core system, and maintaining system friendliness. In order to get
; around this we're providing some useful aliases by default.
;
[friendly]
hangup request=channel request hangup
originate=channel originate
help=core show help
pri intense debug span=pri set debug intense span
reload=module reload
pjsip reload=module reload res_pjsip.so res_pjsip_authenticator_digest.so res_pjsip_endpoint_identifier_ip.so res_pjsip_mwi.so res_pjsip_notify.so res_pjsip_outbound_publish.so res_pjsip_publish_asterisk.so res_pjsip_outbound_registration.so
; CLI Alias Templates
; -------------------
;
; You can define several alias templates.
; It works with context templates like all other configuration files
;
;[asterisk](!)
; To create an alias you simply set the variable name as the alias and variable
; value as the real CLI command you want executed
;
;die die die=stop now
;[asterisk_1dot6](asterisk)
; Alias for making voicemail reload actually do module reload app_voicemail.so
;voicemail reload=module reload app_voicemail.so
; This will make the CLI command "mr" behave as though it is "module reload".
;mr=module reload
;
;
; In addition, you could also include a flat file of aliases which is loaded by
; the [individual_custom] template in the [general] section.
;
;[individual_custom]
;#include "/etc/asterisk/aliases"
; Implemented CLI Alias Templates
; -------------------------------
;
; Below here we have provided you with some templates, easily allowing you to
; utilize previous Asterisk CLI commands with any version of Asterisk. In this
; way you will be able to use Asterisk 1.2 and 1.4 style CLI syntax with any
; version Asterisk going forward into the future.
;
; We have also separated out the vanilla syntax into a context template which
; allows you to keep your custom changes separate of the standard templates
; we have provided you. In this way you can clearly see your custom changes,
; and also allowing you to combine various templates as you see fit.
;
; The naming scheme we have used is recommended, but certainly is not enforced
; by Asterisk. If you wish to use the provided templates, simply define the
; context name which does not utilize the '_tpl' at the end. For example,
; if you would like to use the Asterisk 1.2 style syntax, define in the
; [general] section
[asterisk_1dot2_tpl](!)
show channeltypes=core show channeltypes
show channeltype=core show channeltype
show manager command=manager show command
show manager commands=manager show commands
show manager connected=manager show connected
show manager eventq=manager show eventq
rtp no debug=rtp set debug off
rtp rtcp debug ip=rtcp debug ip
rtp rtcp debug=rtcp debug
rtp rtcp no debug=rtcp debug off
rtp rtcp stats=rtcp stats
rtp rtcp no stats=rtcp stats off
stun no debug=stun debug off
udptl no debug=udptl debug off
show image formats=core show image formats
show file formats=core show file formats
show applications=core show applications
show functions=core show functions
show switches=core show switches
show hints=core show hints
show globals=core show globals
show function=core show function
show application=core show application
set global=core set global
show dialplan=dialplan show
show codecs=core show codecs
show audio codecs=core show audio codecs
show video codecs=core show video codecs
show image codecs=core show image codecs
show codec=core show codec
moh classes show=moh show classes
moh files show=moh show files
agi no debug=agi debug off
show agi=agi show
dump agihtml=agi dumphtml
show features=feature show
show indications=indication show
answer=console answer
hangup=console hangup
flash=console flash
dial=console dial
mute=console mute
unmute=console unmute
transfer=console transfer
send text=console send text
autoanswer=console autoanswer
oss boost=console boost
console=console active
save dialplan=dialplan save
add extension=dialplan add extension
remove extension=dialplan remove extension
add ignorepat=dialplan add ignorepat
remove ignorepat=dialplan remove ignorepat
include context=dialplan add include
dont include=dialplan remove include
extensions reload=dialplan reload
show translation=core show translation
convert=file convert
show queue=queue show
add queue member=queue add member
remove queue member=queue remove member
ael no debug=ael nodebug
sip debug=sip set debug
sip no debug=sip set debug off
show voicemail users=voicemail show users
show voicemail zones=voicemail show zones
iax2 trunk debug=iax2 set debug trunk
iax2 jb debug=iax2 set debug jb
iax2 no debug=iax2 set debug off
iax2 no trunk debug=iax2 set debug trunk off
iax2 no jb debug=iax2 set debug jb off
show agents=agent show
show agents online=agent show online
show memory allocations=memory show allocations
show memory summary=memory show summary
show version=core show version
show version files=core show file version
show profile=core show profile
clear profile=core clear profile
soft hangup=channel request hangup
[asterisk_1dot2](asterisk_1dot2_tpl)
; add any additional custom commands you want below here, for example:
;die quickly=stop now
[asterisk_1dot4_tpl](!)
cdr status=cdr show status
rtp debug=rtp set debug on
rtcp debug=rtcp set debug on
rtcp stats=rtcp set stats on
stun debug=stun set debug on
udptl debug=udptl set debug on
core show globals=dialplan show globals
core set global=dialplan set global
core set chanvar=dialplan set chanvar
agi dumphtml=agi dump html
ael debug=ael set debug
funcdevstate list=devstate list
sip history=sip set history on
skinny debug=skinny set debug on
mgcp set debug=mgcp set debug on
abort shutdown=core abort shutdown
stop now=core stop now
stop gracefully=core stop gracefully
stop when convenient=core stop when convenient
restart now=core restart now
restart gracefully=core restart gracefully
restart when convenient=core restart when convenient
soft hangup=channel request hangup
[asterisk_1dot4](asterisk_1dot4_tpl)
; add any additional custom commands you want below here.
[asterisk_11_tpl](!)
jabber list nodes=xmpp list nodes
jabber purge nodes=xmpp purge nodes
jabber delete node=xmpp delete node
jabber create collection=xmpp create collection
jabber create leaf=xmpp create leaf
jabber set debug=xmpp set debug
jabber show connections=xmpp show connections
jabber show buddies=xmpp show buddies
features reload=module reload features
[asterisk_11](asterisk_11_tpl)
; add any additional custom commands you want below here.

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;
; CLI permissions configuration example for Asterisk
;
; All the users that you want to connect with asterisk using
; rasterisk, should have write/read access to the
; asterisk socket (asterisk.ctl). You could change the permissions
; of this file in 'asterisk.conf' config parameter: 'astctlpermissions' (0666)
; found on the [files] section.
;
; general options:
;
; default_perm = permit | deny
; This is the default permissions to apply for a user that
; does not has a permissions definided.
;
; user options:
; permit = <command name> | all ; allow the user to run 'command' |
; ; allow the user to run 'all' the commands
; deny = <command name> | all ; disallow the user to run 'command' |
; ; disallow the user to run 'all' commands.
;
[general]
default_perm=permit ; To leave asterisk working as normal
; we should set this parameter to 'permit'
;
; Follows the per-users permissions configs.
;
; This list is read in the sequence that is being written, so
; In this example the user 'eliel' is allow to run only the following
; commands:
; sip show peer
; core set debug
; core set verbose
; If the user is not specified, the default_perm option will be apply to
; every command.
;
; Notice that you can also use regular expressions to allow or deny access to a
; certain command like: 'core show application D*'. In this example the user will be
; allowed to view the documentation for all the applications starting with 'D'.
; Another regular expression could be: 'channel originate SIP/[0-9]* extension *'
; allowing the user to use 'channel originate' on a sip channel and with the 'extension'
; parameter and avoiding the use of the 'application' parameter.
;
; We can also use the templates syntax:
; [supportTemplate](!)
; deny=all
; permit=sip show ; all commands starting with 'sip show' will be allowed
; permit=core show
;
; You can specify permissions for a local group instead of a user,
; just put a '@' and we will know that is a group.
; IMPORTANT NOTE: Users permissions overwrite group permissions.
;
;[@adm]
;deny=all
;permit=sip
;permit=core
;
;
;[eliel]
;deny=all
;permit=sip show peer
;deny=sip show peers
;permit=core set
;
;
;User 'tommy' inherits from template 'supportTemplate':
; deny=all
; permit=sip show
; permit=core show
;[tommy](supportTemplate)
;permit=core set debug
;permit=dialplan show
;
;
;[mark]
;deny=all
;permit=all
;
;

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[speex]
; CBR encoding quality [0..10]
; used only when vbr = false
quality => 3
; codec complexity [0..10]
; tradeoff between cpu/quality
complexity => 2
; perceptual enhancement [true / false]
; improves clarity of decoded speech
enhancement => true
; voice activity detection [true / false]
; reduces bitrate when no voice detected, used only for CBR
; (implicit in VBR/ABR)
vad => true
; variable bit rate [true / false]
; uses bit rate proportionate to voice complexity
vbr => true
; available bit rate [bps, 0 = off]
; encoding quality modulated to match this target bit rate
; not recommended with dtx or pp_vad - may cause bandwidth spikes
abr => 0
; VBR encoding quality [0-10]
; floating-point values allowed
vbr_quality => 4
; discontinuous transmission [true / false]
; stops transmitting completely when silence is detected
; pp_vad is far more effective but more CPU intensive
dtx => false
; preprocessor configuration
; these options only affect Speex v1.1.8 or newer
; enable preprocessor [true / false]
; allows dsp functionality below but incurs CPU overhead
preprocess => false
; preproc voice activity detection [true / false]
; more advanced equivalent of DTX, based on voice frequencies
pp_vad => false
; preproc automatic gain control [true / false]
pp_agc => false
pp_agc_level => 8000
; preproc denoiser [true / false]
pp_denoise => false
; preproc dereverb [true / false]
pp_dereverb => false
pp_dereverb_decay => 0.4
pp_dereverb_level => 0.3
; experimental bitrate changes depending on RTCP feedback [true / false]
experimental_rtcp_feedback => false
[plc]
; for all codecs which do not support native PLC
; this determines whether to perform generic PLC
; there is a minor performance penalty for this.
; By default plc is applied only when the 2 codecs
; in a channel are different.
genericplc => true
; Apply generic plc to channels even if the 2 codecs
; are the same. This forces transcoding via slin so
; the performance impact should be considered.
; Ignored if genericplc is not also enabled.
genericplc_on_equal_codecs => false
; Generate custom formats for formats requiring attributes.
; After defining the custom format, the name used in defining
; the format can be used throughout Asterisk in the format 'allow'
; and 'disallow' options.
;
; Example: silk8 is a predefined custom format in this config file.
; Once this config file is loaded, silk8 can be used anywhere a
; peer's codec capabilities are defined.
;
; In sip.conf 'silk8' can be defined as a capability for a peer.
; [peer1]
; type=peer
; host=dynamic
; disallow=all
; allow=silk8 ;custom codec defined in codecs.conf
;
; LIMITATIONS
; Custom formats can only be defined at startup. Any changes to this
; file made after startup will not take into effect until after Asterisk
; is restarted.
;
; Default Custom SILK format definitions, only one custom SILK format per
; sample rate is allowed.
[silk8]
type=silk
samprate=8000
fec=true ; turn on or off encoding with forward error correction.
; On recommended, off by default.
packetloss_percentage=10 ; Estimated packet loss percentage in uplink direction. This
; affects how much redundancy is built in when using fec.
; The higher the percentage, the larger amount of bandwidth is
; used. Default is 0%, 10% is recommended when fec is in use.
maxbitrate=10000 ; Use the table below to make sure a useful bitrate is choosen
; for maxbitrate. If not set or value is not within the bounds
; of the encoder, a default value is chosen.
;
; sample rate | bitrate range
; 8khz | 5000 - 20000 bps
; 12khz | 7000 - 25000 bps
; 16khz | 8000 - 30000 bps
; 24khz | 20000- 40000 bps
;
;dtx=true ; Encode using discontinuous transmission mode or not. Turning this
; on will save bandwidth during periods of silence at the cost of
; increased computational complexity. Off by default.
[silk12]
type=silk
samprate=12000
maxbitrate=12000
fec=true
packetloss_percentage=10;
[silk16]
type=silk
samprate=16000
maxbitrate=20000
fec=true
packetloss_percentage=10;
[silk24]
type=silk
samprate=24000
maxbitrate=30000
fec=true
packetloss_percentage=10;
; Default custom CELT codec definitions. Only one custom CELT definition is allowed
; per a sample rate.
;[celt44]
;type=celt
;samprate=44100 ; The samplerate in hz. This option is required.
;framesize=480 ; The framesize option represents the duration of each frame in samples.
; This must be a factor of 2. This option is only advertised in an SDP
; when it is set. Otherwise a default of framesize of 480 is assumed
; internally
;[celt48]
;type=celt
;samprate=48000
;[celt32]
;type=celt
;samprate=32000
;============================ OPUS Section Options ============================
;
; NOTE: Accurate documentation corresponding to your downloaded version of
; codec_opus is available from Asterisk's CLI:
;
; *CLI> config show help codec_opus opus
;
;[opus]
;type= ; Must be of type "opus" (default: "")
;packet_loss= ; Encoder's packet loss percentage. Can be any number between 0
; and 100, inclusive. A higher value results in more loss
; resistance. (default: 0)
;complexity= ; Encoder's computational complexity. Can be any number between 0
; and 10, inclusive. Note, 10 equals the highest complexity.
; (default: 10)
;max_bandwidth= ; Encoder's maximum bandwidth allowed. Sets an upper bandwidth
; bound on the encoder. Can be any of the following: narrow,
; medium, wide, super_wide, full. (default: full)
;signal= ; Encoder's signal type. Aids in mode selection on the encoder: Can
; be any of the following: auto, voice, music. (default: auto)
;application= ; Encoder's application type. Can be any of the following: voip,
; audio, low_delay. (default: voip)
;max_playback_rate= ; Override the maximum playback rate in the offer's SDP.
; Any value between 8000 and 48000 (inclusive) is valid,
; however typically it should match one of the usual opus
; bandwidths. (default: 48000)
;bitrate= ; Override the maximum average bitrate in the offer's SDP. Any value
; between 500 and 512000 is valid. The following values are also
; allowed: auto, max. (default: auto)
;cbr= ; Override the constant bit rate parameter in the offer's SDP. A value of
; 0/false/no represents a variable bit rate whereas 1/true/yes represents
; a constant bit rate. (default: no)
;fec= ; Override the use inband fec parameter in the offer's SDP. A value of
; 0/false/no represents disabled whereas 1/true/yes represents enabled.
; (default: yes)
;dtx= ; Override the use dtx parameter in the offer's SDP. A value of 0/false/no
; represents disabled whereas 1/true/yes represents enabled. (default: no)
;=============================== OPUS Examples ================================
;
;[opus]
;type=opus
;max_playback_rate=8000 ; Limit the maximum playback rate on the encoder
;fec=no ; No inband fec
;[myopus]
;type=opus
;max_bandwidth=wide ; Maximum encoded bandwidth set to wide band (0-8000 Hz
; ; audio bandwidth at 16Khz sample rate)
;cbr=yes ; Negotiate a constant bit rate

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[general]
; The general section of this config
; is not currently used, but reserved
; for future use.
;
; --- Default Information ---
; The default_user and default_bridge sections are applied
; automatically to all ConfBridge instances invoked without
; a user, or bridge argument. No menu is applied by default.
;
; Note that while properties of the default_user or default_bridge
; profile can be overridden, if removed, they will be automatically
; added and made available to the dialplan upon module load.
;
; --- ConfBridge User Profile Options ---
[default_user]
type=user
;admin=yes ; Sets if the user is an admin or not. Off by default.
;send_events=no ; If events are enabled for this bridge and this option is
; set, users will receive events like join, leave, talking,
; etc. via text messages. For users accessing the bridge
; via chan_pjsip, this means in-dialog MESSAGE messages.
; This is most useful for WebRTC participants where the
; browser application can use the messages to alter the user
; interface.
;echo_events=yes ; If events are enabled for this user and this option is set,
; the user will receive events they trigger, talking, mute, etc.
; If not set, they will not receive their own events.
;marked=yes ; Sets if this is a marked user or not. Off by default.
;startmuted=yes; Sets if all users should start out muted. Off by default
;music_on_hold_when_empty=yes ; Sets whether MOH should be played when only
; one person is in the conference or when the
; the user is waiting on a marked user to enter
; the conference. Off by default.
;music_on_hold_class=default ; The MOH class to use for this user.
;quiet=yes ; When enabled enter/leave prompts and user intros are not played.
; There are some prompts, such as the prompt to enter a PIN number,
; that must be played regardless of what this option is set to.
; Off by default
;announce_user_count=yes ; Sets if the number of users should be announced to the
; caller. Off by default.
;announce_user_count_all=yes ; Sets if the number of users should be announced to
; all the other users in the conference when someone joins.
; This option can be either set to 'yes' or a number.
; When set to a number, the announcement will only occur
; once the user count is above the specified number.
;announce_only_user=yes ; Sets if the only user announcement should be played
; when a channel enters a empty conference. On by default.
;wait_marked=yes ; Sets if the user must wait for a marked user to enter before
; joining the conference. Off by default.
;end_marked=yes ; This option will kick every user with this option set in their
; user profile after the last Marked user exists the conference.
;dsp_drop_silence=yes ; This option drops what Asterisk detects as silence from
; entering into the bridge. Enabling this option will drastically
; improve performance and help remove the buildup of background
; noise from the conference. Highly recommended for large conferences
; due to its performance enhancements.
;dsp_talking_threshold=128 ; Average magnitude threshold to determine talking.
;
; The minimum average magnitude per sample in a frame for the
; DSP to consider talking/noise present. A value below this
; level is considered silence. This value affects several
; operations and should not be changed unless the impact on
; call quality is fully understood.
;
; What this value affects internally:
;
; 1. Audio is only mixed out of a user's incoming audio
; stream if talking is detected. If this value is set too
; high the user will hear himself talking.
;
; 2. When talk detection AMI events are enabled, this value
; determines when talking has begun which results in an
; AMI event to fire. If this value is set too low AMI
; events may be falsely triggered by variants in room
; noise.
;
; 3. The 'drop_silence' option depends on this value to
; determine when the user's audio should be mixed into the
; bridge after periods of silence. If this value is too
; high the user's speech will get discarded as they will
; be considered silent.
;
; Valid values are 1 through 2^15.
; By default this value is 160.
;dsp_silence_threshold=2000 ; The number of milliseconds of silence necessary to declare
; talking stopped.
;
; The time in milliseconds of sound falling below the
; 'dsp_talking_threshold' option when a user is considered to
; stop talking. This value affects several operations and
; should not be changed unless the impact on call quality is
; fully understood.
;
; What this value affects internally:
;
; 1. When talk detection AMI events are enabled, this value
; determines when the user has stopped talking after a
; period of talking. If this value is set too low AMI
; events indicating the user has stopped talking may get
; falsely sent out when the user briefly pauses during mid
; sentence.
;
; 2. The 'drop_silence' option depends on this value to
; determine when the user's audio should begin to be
; dropped from the conference bridge after the user stops
; talking. If this value is set too low the user's audio
; stream may sound choppy to the other participants. This
; is caused by the user transitioning constantly from
; silence to talking during mid sentence.
;
; The best way to approach this option is to set it slightly
; above the maximum amount of milliseconds of silence a user
; may generate during natural speech.
;
; Valid values are 1 through 2^31.
; By default this value is 2500ms.
;talk_detection_events=yes ; This option sets whether or not notifications of when a user
; begins and ends talking should be sent out as events over AMI.
; By default this option is off.
;denoise=yes ; Sets whether or not a denoise filter should be applied
; to the audio before mixing or not. Off by default. Requires
; func_speex to be built and installed. Do not confuse this option
; with drop_silence. Denoise is useful if there is a lot of background
; noise for a user as it attempts to remove the noise while preserving
; the speech. This option does NOT remove silence from being mixed into
; the conference and does come at the cost of a slight performance hit.
;jitterbuffer=yes ; Enabling this option places a jitterbuffer on the user's audio stream
; before audio mixing is performed. This is highly recommended but will
; add a slight delay to the audio. This option is using the JITTERBUFFER
; dialplan function's default adaptive jitterbuffer. For a more fine tuned
; jitterbuffer, disable this option and use the JITTERBUFFER dialplan function
; on the user before entering the ConfBridge application.
;pin=1234 ; Sets if this user must enter a PIN number before entering
; the conference. The PIN will be prompted for.
;announce_join_leave=yes ; When enabled, this option will prompt the user for a
; name when entering the conference. After the name is
; recorded, it will be played as the user enters and exists
; the conference. This option is off by default.
;announce_join_leave_review=yes ; When enabled, implies announce_join_leave, but the user
; will be prompted to review their recording before
; entering the conference. During this phase, the recording
; may be listened to, re-recorded, or accepted as is. This
; option is off by default.
;dtmf_passthrough=yes ; Sets whether or not DTMF should pass through the conference.
; This option is off by default.
;announcement=</path/to/file> ; Play a sound file to the user when they join the conference.
;timeout=3600 ; When set non-zero, this specifies the number of seconds that the participant
; may stay in the conference before being automatically ejected. When the user
; is ejected from the conference, the user's channel will have the CONFBRIDGE_RESULT
; variable set to "TIMEOUT". A value of 0 indicates that there is no timeout.
; Default: 0
; --- ConfBridge Bridge Profile Options ---
[default_bridge]
type=bridge
;max_members=50 ; This option limits the number of participants for a single
; conference to a specific number. By default conferences
; have no participant limit. After the limit is reached, the
; conference will be locked until someone leaves. Note however
; that an Admin user will always be alowed to join the conference
; regardless if this limit is reached or not.
;record_conference=yes ; Records the conference call starting when the first user
; enters the room, and ending when the last user exits the room.
; The default recorded filename is
; 'confbridge-<name of conference bridge>-<start time>.wav
; and the default format is 8khz slinear. This file will be
; located in the configured monitoring directory in asterisk.conf.
;record_file=</path/to/file> ; When record_conference is set to yes, the specific name of the
; record file can be set using this option. Note that since multiple
; conferences may use the same bridge profile, this may cause issues
; depending on the configuration. It is recommended to only use this
; option dynamically with the CONFBRIDGE() dialplan function. This
; allows the record name to be specified and a unique name to be chosen.
; By default, the record_file is stored in Asterisk's spool/monitor directory
; with a unique filename starting with the 'confbridge' prefix.
;record_file_append=yes ; Append record file when starting/stopping on same conference recording.
;record_file_timestamp=yes ; Append the start time to the record file name.
;record_options= ; Pass additional options to MixMonitor.
;record_command=</path/to/command> ; Command to execute when recording finishes.
;internal_sample_rate=auto ; Sets the internal native sample rate the
; conference is mixed at. This is set to automatically
; adjust the sample rate to the best quality by default.
; Other values can be anything from 8000-192000. If a
; sample rate is set that Asterisk does not support, the
; closest sample rate Asterisk does support to the one requested
; will be used.
;mixing_interval=40 ; Sets the internal mixing interval in milliseconds for the bridge. This
; number reflects how tight or loose the mixing will be for the conference.
; In order to improve performance a larger mixing interval such as 40ms may
; be chosen. Using a larger mixing interval comes at the cost of introducing
; larger amounts of delay into the bridge. Valid values here are 10, 20, 40,
; or 80. By default 20ms is used.
;video_mode = follow_talker; Sets how confbridge handles video distribution to the conference participants.
; Note that participants wanting to view and be the source of a video feed
; _MUST_ be sharing the same video codec. Also, using video in conjunction with
; with the jitterbuffer currently results in the audio being slightly out of sync
; with the video. This is a result of the jitterbuffer only working on the audio
; stream. It is recommended to disable the jitterbuffer when video is used.
;
; --- MODES ---
; none: No video sources are set by default in the conference. It is still
; possible for a user to be set as a video source via AMI or DTMF action
; at any time.
;
; follow_talker: The video feed will follow whoever is talking and providing video.
;
; last_marked: The last marked user to join the conference with video capabilities
; will be the single source of video distributed to all participants.
; If multiple marked users are capable of video, the last one to join
; is always the source, when that user leaves it goes to the one who
; joined before them.
;
; first_marked: The first marked user to join the conference with video capabilities
; is the single source of video distribution among all participants. If
; that user leaves, the marked user to join after them becomes the source.
;
; sfu: Selective Forwarding Unit - Sets multi-stream operation
; for a multi-party video conference.
;language=en ; Set the language used for announcements to the conference.
; Default is en (English).
;regcontext=conferences ; The name of the context into which to register conference names as extensions.
;video_update_discard=2000 ; Amount of time (in milliseconds) to discard video update requests after sending a video
; update request. Default is 2000. A video update request is a request for a full video
; intra-frame. Clients can request this if they require a full frame in order to decode
; the video stream. Since a full frame can be large limiting how often they occur can
; reduce bandwidth usage at the cost of increasing how long it may take a newly joined
; channel to receive the video stream.
;remb_send_interval=1000 ; Interval (in milliseconds) at which a combined REMB frame will be sent to sources of video.
; A REMB frame contains receiver estimated maximum bitrate information. By creating a combined
; frame and sending it to the sources of video the sender can be influenced on what bitrate
; they choose allowing a better experience for the receivers. This defaults to 0, or disabled.
;remb_behavior=average ; How the combined REMB report for an SFU video bridge is constructed. If set to "average" then
; the estimated maximum bitrate of each receiver is used to construct an average bitrate. If
; set to "lowest" the lowest maximum bitrate is forwarded to the sender. If set to "highest"
; the highest maximum bitrate is forwarded to the sender. This defaults to "average".
;enable_events=no ; If enabled, recipients who joined the bridge via a channel driver
; that supports Enhanced Messaging (currently only chan_pjsip) will
; receive in-dialog messages containing a JSON body describing the
; event. The Content-Type header will be
; "text/x-ast-confbridge-event".
; This feature must also be enabled in user profiles.
; All sounds in the conference are customizable using the bridge profile options below.
; Simply state the option followed by the filename or full path of the filename after
; the option. Example: sound_had_joined=conf-hasjoin This will play the conf-hasjoin
; sound file found in the sounds directory when announcing someone's name is joining the
; conference.
;sound_join ; The sound played to everyone when someone enters the conference.
;sound_leave ; The sound played to everyone when someone leaves the conference.
;sound_has_joined ; The sound played before announcing someone's name has
; joined the conference. This is used for user intros.
; Example "_____ has joined the conference"
;sound_has_left ; The sound played when announcing someone's name has
; left the conference. This is used for user intros.
; Example "_____ has left the conference"
;sound_kicked ; The sound played to a user who has been kicked from the conference.
;sound_muted ; The sound played when the mute option is toggled on using DTMF menu.
;sound_unmuted ; The sound played when the mute option is toggled off using DTMF menu.
;sound_only_person ; The sound played when the user is the only person in the conference.
;sound_only_one ; The sound played to a user when there is only one other
; person is in the conference.
;sound_there_are ; The sound played when announcing how many users there
; are in a conference.
;sound_other_in_party; ; This file is used in conjunction with 'sound_there_are"
; when announcing how many users there are in the conference.
; The sounds are stringed together like this.
; "sound_there_are" <number of participants> "sound_other_in_party"
;sound_place_into_conference ; The sound played when someone is placed into the conference
; after waiting for a marked user. This sound is now deprecated
; since it was only ever used improperly and correcting that bug
; made it completely unused.
;sound_wait_for_leader ; The sound played when a user is placed into a conference that
; can not start until a marked user enters.
;sound_leader_has_left ; The sound played when the last marked user leaves the conference.
;sound_get_pin ; The sound played when prompting for a conference pin number.
;sound_invalid_pin ; The sound played when an invalid pin is entered too many times.
;sound_locked ; The sound played to a user trying to join a locked conference.
;sound_locked_now ; The sound played to an admin after toggling the conference to locked mode.
;sound_unlocked_now; The sound played to an admin after toggling the conference to unlocked mode.
;sound_error_menu ; The sound played when an invalid menu option is entered.
;sound_begin ; The sound played to the conference when the first marked user enters the conference.
;sound_binaural_on ; The sound played when binaural audio is turned on
;sound_binaural_off ; The sound played when binaural audio is turned off
; --- ConfBridge Menu Options ---
; The ConfBridge application also has the ability to
; apply custom DTMF menus to each channel using the
; application. Like the User and Bridge profiles
; a menu is passed in to ConfBridge as an argument in
; the dialplan.
;
; Below is a list of menu actions that can be assigned
; to a DTMF sequence.
;
; To have the first DTMF digit in a sequence be the '#' character, you need to
; escape it. If it is not escaped then normal config file processing will
; think it is a directive like #include. For example:
; \#1=toggle_mute ; Pressing #1 will toggle the mute setting.
;
; A single DTMF sequence can have multiple actions associated with it. This is
; accomplished by stringing the actions together and using a ',' as the delimiter.
; Example: Both listening and talking volume is reset when '5' is pressed.
; 5=reset_talking_volume, reset_listening_volume
;
; playback(<name of audio file>&<name of audio file>)
; Playback will play back an audio file to a channel
; and then immediately return to the conference.
; This file can not be interupted by DTMF.
; Mutliple files can be chained together using the
; '&' character.
; playback_and_continue(<name of playback prompt>&<name of playback prompt>)
; playback_and_continue will
; play back a prompt while continuing to
; collect the dtmf sequence. This is useful
; when using a menu prompt that describes all
; the menu options. Note however that any DTMF
; during this action will terminate the prompts
; playback. Prompt files can be chained together
; using the '&' character as a delimiter.
; toggle_mute ; Toggle turning on and off mute. Mute will make the user silent
; to everyone else, but the user will still be able to listen in.
; toggle_binaural ; Toggle on or off binaural audio processing.
; no_op ; This action does nothing (No Operation). Its only real purpose exists for
; being able to reserve a sequence in the config as a menu exit sequence.
; decrease_listening_volume ; Decreases the channel's listening volume.
; increase_listening_volume ; Increases the channel's listening volume.
; reset_listening_volume ; Reset channel's listening volume to default level.
; decrease_talking_volume ; Decreases the channel's talking volume.
; increase_talking_volume ; Icreases the channel's talking volume.
; reset_talking_volume ; Reset channel's talking volume to default level.
;
; dialplan_exec(context,exten,priority) ; The dialplan_exec action allows a user
; to escape from the conference and execute
; commands in the dialplan. Once the dialplan
; exits the user will be put back into the
; conference. The possibilities are endless!
; leave_conference ; This action allows a user to exit the conference and continue
; execution in the dialplan.
;
; admin_kick_last ; This action allows an Admin to kick the last participant from the
; conference. This action will only work for admins which allows
; a single menu to be used for both users and admins.
;
; admin_toggle_conference_lock ; This action allows an Admin to toggle locking and
; unlocking the conference. Non admins can not use
; this action even if it is in their menu.
; set_as_single_video_src ; This action allows any user to set themselves as the
; single video source distributed to all participants.
; This will make the video feed stick to them regardless
; of what the video_mode is set to.
; release_as_single_video_src ; This action allows a user to release themselves as
; the video source. If video_mode is not set to "none"
; this action will result in the conference returning to
; whatever video mode the bridge profile is using.
;
; Note that this action will have no effect if the user
; is not currently the video source. Also, the user is
; not guaranteed by using this action that they will not
; become the video source again. The bridge will return
; to whatever operation the video_mode option is set to
; upon release of the video src.
; admin_toggle_mute_participants ; This action allows an administrator to toggle the mute
; state for all non-admins within a conference.
; Subsequent non-admins joining a muted conference will
; start muted. All admin users are unaffected by this
; option. Note that all users, regardless of their admin
; status, are notified that the conference is muted when
; the state is toggled.
; participant_count ; This action plays back the number of participants currently
; in a conference
[sample_user_menu]
type=menu
*=playback_and_continue(conf-usermenu)
*1=toggle_mute
1=toggle_mute
*4=decrease_listening_volume
4=decrease_listening_volume
*6=increase_listening_volume
6=increase_listening_volume
*7=decrease_talking_volume
7=decrease_talking_volume
*8=leave_conference
8=leave_conference
*9=increase_talking_volume
9=increase_talking_volume
[sample_admin_menu]
type=menu
*=playback_and_continue(conf-adminmenu)
*1=toggle_mute
1=toggle_mute
*2=admin_toggle_conference_lock ; only applied to admin users
2=admin_toggle_conference_lock ; only applied to admin users
*3=admin_kick_last ; only applied to admin users
3=admin_kick_last ; only applied to admin users
*4=decrease_listening_volume
4=decrease_listening_volume
*6=increase_listening_volume
6=increase_listening_volume
*7=decrease_talking_volume
7=decrease_talking_volume
*8=no_op
8=no_op
*9=increase_talking_volume
9=increase_talking_volume

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[general]
;enable=yes ; enable creation of managed DNS lookups
; default is 'no'
;refreshinterval=1200 ; refresh managed DNS lookups every <n> seconds
; default is 300 (5 minutes)

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[default]
;
; Length of sound (in milliseconds) before a period of silence is considered
; to be a change from talking to silence or a period of noise converts silence
; to talking. [default=256]
;
;silencethreshold=256
; DTMF Reverse Twist and Normal Twist is the difference in power between the row and column energies.
;
; Normal Twist is where the row energy is greater than the column energy.
; Reverse Twist is where the column energy is greater.
;
; Power level difference between frequencies for different Administrations/RPOAs
; Power Gain equiv
; normal reverse dB's
; AT&T(default) 6.31 2.51 8dB(normal), 4dB(reverse)
; NTT 3.16 3.16 Max. 5dB
; Danish 3.98 3.98 Max. 6dB
; Australian 10.0 10.0 Max. 10dB
; Brazilian 7.94 7.94 Max. 9dB
; ETSI 3.98 3.98 Max. 6dB
;previous version compatible AT&T values
; RADIO_RELAX disabled, and relaxdtmf=no
; 6.30 2.50 7.99dB(normal), 3.98dB(reverse)
; RADIO_RELAX disabled, and relaxdtmf=yes
; 6.30 4.00 7.99dB(normal), 6.02dB(reverse)
; RADIO_RELAX enabled, and relaxdtmf=no
; 6.30 2.50 7.99dB(normal), 3.984dB(reverse)
; RADIO_RELAX enabled, and relaxdtmf=yes
; 6.30 6.50 7.99dB(normal), 8.13dB(reverse)
;If you don't know what these mean, don't change them.
;dtmf_normal_twist=6.31
;dtmf_reverse_twist=2.51
;relax_dtmf_normal_twist=6.31
;relax_dtmf_reverse_twist=3.98
;successive number hits/misses of 12.75ms before a digit/nodigit is considered valid
;dtmf_hits_to_begin=2
;dtmf_misses_to_end=3

22
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;
; ENUM Configuration for resolving phone numbers over DNS
;
; Sample config for Asterisk
; This file is reloaded at "module reload enum" in the CLI
;
[general]
;
; The search list for domains may be customized. Domains are searched
; in the order they are listed here.
;
search => e164.arpa
;
; If you'd like to use the E.164.org public ENUM registry in addition
; to the official e164.arpa one, uncomment the following line
;
;search => e164.org
;
; As there are more H323 drivers available you have to select to which
; drive a H323 URI will map. Default is "H323".
;
h323driver => H323

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;
; Static and realtime external configuration
; engine configuration
;
; See https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
; for basic table formatting information.
;
[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
;
; Named ACLs specified in realtime also can not be used
; from manager.conf unless the storage driver is preloaded.
; Attempting to use a realtime stored named ACL before the
; driver is loaded will result in an invalid ACL which
; rejects all addresses.
;
; 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).
;
;
; Possible driver backends:
;
; "odbc" is shown in the examples below, but is not the only valid realtime
; engine. Here are several of the possible options:
; odbc ... res_config_odbc
; sqlite ... res_config_sqlite
; sqlite3 ... res_config_sqlite3
; pgsql ... res_config_pgsql
; curl ... res_config_curl
; ldap ... res_config_ldap
; mysql ... res_config_mysql (available via add-ons in menuselect)
;
; Note: The res_config_pgsql and res_config_sqlite backends configure the
; database used in their respective configuration files and ignore the
; database name configured in this file.
;
;iaxusers => odbc,asterisk
;iaxpeers => odbc,asterisk
;sippeers => odbc,asterisk
;sipregs => odbc,asterisk ; (avoid sipregs if possible, e.g. by using a view)
;ps_endpoints => odbc,asterisk
;ps_auths => odbc,asterisk
;ps_aors => odbc,asterisk
;ps_domain_aliases => odbc,asterisk
;ps_endpoint_id_ips => odbc,asterisk
;ps_outbound_publishes => odbc,asterisk
;ps_inbound_publications = odbc,asterisk
;ps_asterisk_publications = odbc,asterisk
;voicemail => odbc,asterisk
;extensions => odbc,asterisk
;meetme => mysql,general
;queues => odbc,asterisk
;queue_members => odbc,asterisk
;queue_rules => odbc,asterisk
;acls => 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.

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//
// Example AEL config file
//
//
// Static extension configuration file, used by
// the pbx_ael module. This is where you configure all your
// inbound and outbound calls in Asterisk.
//
// This configuration file is reloaded
// - With the "ael reload" command in the CLI
// - With the "reload" command (that reloads everything) in the CLI
// The "Globals" category contains global variables that can be referenced
// in the dialplan by using the GLOBAL dialplan function:
// ${GLOBAL(VARIABLE)}
// ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
// Unix/Linux environmental variables are reached with the ENV dialplan
// function: ${ENV(VARIABLE)}
//
// NOTE! NOTE! NOTE!
// Asterisk by default will load both extensions.conf and extensions.ael files.
// Upon loading these files the dialplans generated from both with be merged,
// so you must make sure that you don't have any overlapping contexts or global
// variables. If you do, then unexpected behavior may result when the data is
// merged.
// NOTE! NOTE! NOTE!
globals {
CONSOLE-AEL="Console/dsp"; // Console interface for demo
//CONSOLE-AEL=Zap/1;
//CONSOLE-AEL=Phone/phone0;
IAXINFO-AEL=guest; // IAXtel username/password
//IAXINFO-AEL="myuser:mypass";
OUTBOUND-TRUNK="Zap/g2"; // Trunk interface
//
// Note the 'g2' in the OUTBOUND-TRUNK variable above. It specifies which group (defined
// in chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in
// the specified group. The four possible options are:
//
// g: select the lowest-numbered non-busy DAHDI channel
// (aka. ascending sequential hunt group).
// G: select the highest-numbered non-busy DAHDI channel
// (aka. descending sequential hunt group).
// r: use a round-robin search, starting at the next highest channel than last
// time (aka. ascending rotary hunt group).
// R: use a round-robin search, starting at the next lowest channel than last
// time (aka. descending rotary hunt group).
//
OUTBOUND-TRUNKMSD=1; // MSD digits to strip (usually 1 or 0)
//OUTBOUND-TRUNK2=IAX2/user:pass@provider;
};
//
// Any category other than "General" and "Globals" represent
// extension contexts, which are collections of extensions.
//
// Extension names may be numbers, letters, or combinations
// thereof. If an extension name is prefixed by a '_'
// character, it is interpreted as a pattern rather than a
// literal. In patterns, some characters have special meanings:
//
// X - any digit from 0-9
// Z - any digit from 1-9
// N - any digit from 2-9
// [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
// . - wildcard, matches anything remaining (e.g. _9011. matches
// anything starting with 9011 excluding 9011 itself)
// ! - wildcard, causes the matching process to complete as soon as
// it can unambiguously determine that no other matches are possible
//
// For example the extension _NXXXXXX would match normal 7 digit dialings,
// while _1NXXNXXXXXX would represent an area code plus phone number
// preceded by a one.
//
// Each step of an extension is ordered by priority, which must
// always start with 1 to be considered a valid extension. The priority
// "next" or "n" means the previous priority plus one, regardless of whether
// the previous priority was associated with the current extension or not.
// The priority "same" or "s" means the same as the previously specified
// priority, again regardless of whether the previous entry was for the
// same extension. Priorities may be immediately followed by a plus sign
// and another integer to add that amount (most useful with 's' or 'n').
// Priorities may then also have an alias, or label, in
// parenthesis after their name which can be used in goto situations
//
// Contexts contain several lines, one for each step of each
// extension, which can take one of two forms as listed below,
// with the first form being preferred. One may include another
// context in the current one as well, optionally with a
// date and time. Included contexts are included in the order
// they are listed.
//
//context name {
// exten-name => {
// application(arg1,arg2,...);
//
// Timing list for includes is
//
// <time range>|<days of week>|<days of month>|<months>
//
// includes {
// daytime|9:00-17:00|mon-fri|*|*;
// };
//
// ignorepat can be used to instruct drivers to not cancel dialtone upon
// receipt of a particular pattern. The most commonly used example is
// of course '9' like this:
//
// ignorepat => 9;
//
// so that dialtone remains even after dialing a 9.
//};
//
// Sample entries for extensions.conf
//
//
context ael-dundi-e164-canonical {
//
// List canonical entries here
//
// 12564286000 => &ael-std-exten(6000,IAX2/foo);
// _125642860XX => Dial(IAX2/otherbox/${EXTEN:7});
};
context ael-dundi-e164-customers {
//
// If you are an ITSP or Reseller, list your customers here.
//
//_12564286000 => Dial(SIP/customer1);
//_12564286001 => Dial(IAX2/customer2);
};
context ael-dundi-e164-via-pstn {
//
// If you are freely delivering calls to the PSTN, list them here
//
//_1256428XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Expose all of 256-428
//_1256325XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Ditto for 256-325
};
context ael-dundi-e164-local {
//
// Context to put your dundi IAX2 or SIP user in for
// full access
//
includes {
ael-dundi-e164-canonical;
ael-dundi-e164-customers;
ael-dundi-e164-via-pstn;
};
};
context ael-dundi-e164-switch {
//
// Just a wrapper for the switch
//
switches {
DUNDi/e164;
};
};
context ael-dundi-e164-lookup {
//
// Locally to lookup, try looking for a local E.164 solution
// then try DUNDi if we don't have one.
//
includes {
ael-dundi-e164-local;
ael-dundi-e164-switch;
};
//
};
//
// DUNDi can also be implemented as a Macro instead of using
// the Local channel driver.
//
macro ael-dundi-e164(exten) {
//
// ARG1 is the extension to Dial
//
goto ${exten}|1;
return;
};
//
// Here are the entries you need to participate in the IAXTEL
// call routing system. Most IAXTEL numbers begin with 1-700, but
// there are exceptions. For more information, and to sign
// up, please go to www.gnophone.com or www.iaxtel.com
//
context ael-iaxtel700 {
_91700XXXXXXX => Dial(IAX2/${IAXINFO-AEL}@iaxtel.com/${EXTEN:1}@iaxtel);
};
//
// The SWITCH statement permits a server to share the dialplan with
// another server. Use with care: Reciprocal switch statements are not
// allowed (e.g. both A -> B and B -> A), and the switched server needs
// to be on-line or else dialing can be severly delayed.
//
context ael-iaxprovider {
switches {
// IAX2/user:[key]@myserver/mycontext;
};
};
context ael-trunkint {
//
// International long distance through trunk
//
includes {
ael-dundi-e164-lookup;
};
_9011. => {
&ael-dundi-e164(${EXTEN:4});
Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
};
};
context ael-trunkld {
//
// Long distance context accessed through trunk
//
includes {
ael-dundi-e164-lookup;
};
_91NXXNXXXXXX => {
&ael-dundi-e164(${EXTEN:1});
Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
};
};
context ael-trunklocal {
//
// Local seven-digit dialing accessed through trunk interface
//
_9NXXXXXX => {
Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
};
};
context ael-trunktollfree {
//
// Long distance context accessed through trunk interface
//
_91800NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
_91888NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
_91877NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
_91866NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
};
context ael-international {
//
// Master context for international long distance
//
ignorepat => 9;
includes {
ael-longdistance;
ael-trunkint;
};
};
context ael-longdistance {
//
// Master context for long distance
//
ignorepat => 9;
includes {
ael-local;
ael-trunkld;
};
};
context ael-local {
//
// Master context for local, toll-free, and iaxtel calls only
//
ignorepat => 9;
includes {
ael-default;
ael-trunklocal;
ael-iaxtel700;
ael-trunktollfree;
ael-iaxprovider;
};
};
//
// You can use an alternative switch type as well, to resolve
// extensions that are not known here, for example with remote
// IAX switching you transparently get access to the remote
// Asterisk PBX
//
// switch => IAX2/user:password@bigserver/local
//
// An "lswitch" is like a switch but is literal, in that
// variable substitution is not performed at load time
// but is passed to the switch directly (presumably to
// be substituted in the switch routine itself)
//
// lswitch => Loopback/12${EXTEN}@othercontext
//
// An "eswitch" is like a switch but the evaluation of
// variable substitution is performed at runtime before
// being passed to the switch routine.
//
// eswitch => IAX2/context@${CURSERVER}
macro ael-std-exten-ael( ext , dev ) {
Dial(${dev}/${ext},20);
switch(${DIALSTATUS}) {
case BUSY:
Voicemail(${ext},b);
break;
default:
Voicemail(${ext},u);
};
catch a {
VoiceMailMain(${ext});
return;
};
return;
};
context ael-demo {
s => {
Wait(1);
Answer();
Set(TIMEOUT(digit)=5);
Set(TIMEOUT(response)=10);
restart:
Background(demo-congrats);
instructions:
for (x=0; ${x} < 3; x=${x} + 1) {
Background(demo-instruct);
WaitExten();
};
};
2 => {
Background(demo-moreinfo);
goto s|instructions;
};
3 => {
Set(LANGUAGE()=fr);
goto s|restart;
};
1000 => {
goto ael-default|s|1;
};
500 => {
Playback(demo-abouttotry);
Dial(IAX2/guest@misery.digium.com/s@default);
Playback(demo-nogo);
goto s|instructions;
};
600 => {
Playback(demo-echotest);
Echo();
Playback(demo-echodone);
goto s|instructions;
};
_1234 => &ael-std-exten-ael(${EXTEN}, "IAX2");
8500 => {
VoicemailMain();
goto s|instructions;
};
# => {
Playback(demo-thanks);
Hangup();
};
t => goto #|1;
i => Playback(invalid);
};
//
// If you wish to use AEL for your default context, remove it
// from extensions.conf (or change its name or comment it out)
// and then uncomment the one here.
//
context ael-default {
// By default we include the demo. In a production system, you
// probably don't want to have the demo there.
includes {
ael-demo;
};
//
// Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
// Note that you must have a [sipprovider] section in sip.conf whereas
// the otherprovider.net example does not require such a peer definition
//
//_41X. => Dial(SIP/${EXTEN:2}@sipprovider,,r);
//_42X. => Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT);
// Real extensions would go here. Generally you want real extensions to be
// 4 or 5 digits long (although there is no such requirement) and start with a
// single digit that is fairly large (like 6 or 7) so that you have plenty of
// room to overlap extensions and menu options without conflict. You can alias
// them with names, too, and use global variables
// 6245 => {
// hint(SIP/Grandstream1&SIP/Xlite1,Joe Schmoe); // Channel hints for presence
// Dial(SIP/Grandstream1,20,rt); // permit transfer
// Dial(${HINT}/5245},20,rtT); // Use hint as listed
// switch(${DIALSTATUS}) {
// case BUSY:
// Voicemail(6245,b);
// return;
// default:
// Voicemail(6245,u);
// return;
// };
// };
// 6361 => Dial(IAX2/JaneDoe,,rm); // ring without time limit
// 6389 => Dial(MGCP/aaln/1@192.168.0.14);
// 6394 => Dial(Local/6275/n); // this will dial ${MARK}
// 6275 => &ael-stdexten(6275,${MARK}); // assuming ${MARK} is something like DAHDI/2
// mark => goto 6275|1; // alias mark to 6275
// 6536 => &ael-stdexten(6236,${WIL}); // Ditto for wil
// wil => goto 6236|1;
//
// Some other handy things are an extension for checking voicemail via
// voicemailmain
//
// 8500 => {
// VoicemailMain();
// Hangup();
// };
//
// Or a conference room (you'll need to edit meetme.conf to enable this room)
//
// 8600 => Meetme(1234);
//
// Or playing an announcement to the called party, as soon it answers
//
// 8700 => Dial(${MARK},30,A(/path/to/my/announcemsg))
//
// For more information on applications, just type "show applications" at your
// friendly Asterisk CLI prompt.
//
// 'show application <command>' will show details of how you
// use that particular application in this file, the dial plan.
//
}

862
extensions.conf Normal file
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@@ -0,0 +1,862 @@
; extensions.conf - the Asterisk dial plan
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "dialplan reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command "dialplan save" too
;
writeprotect=no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk's best guess. This is the default.
;
; If autofallthrough is not set, then if an extension runs out of
; things to do, Asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
;
;autofallthrough=no
;
;
;
; If extenpatternmatchnew is set (true, yes, etc), then a new algorithm that uses
; a Trie to find the best matching pattern is used. In dialplans
; with more than about 20-40 extensions in a single context, this
; new algorithm can provide a noticeable speedup.
; With 50 extensions, the speedup is 1.32x
; with 88 extensions, the speedup is 2.23x
; with 138 extensions, the speedup is 3.44x
; with 238 extensions, the speedup is 5.8x
; with 438 extensions, the speedup is 10.4x
; With 1000 extensions, the speedup is ~25x
; with 10,000 extensions, the speedup is 374x
; Basically, the new algorithm provides a flat response
; time, no matter the number of extensions.
;
; By default, the old pattern matcher is used.
;
; ****This is a new feature! *********************
; The new pattern matcher is for the brave, the bold, and
; the desperate. If you have large dialplans (more than about 50 extensions
; in a context), and/or high call volume, you might consider setting
; this value to "yes" !!
; Please, if you try this out, and are forced to return to the
; old pattern matcher, please report your reasons in a bug report
; on https://issues.asterisk.org. We have made good progress in providing
; something compatible with the old matcher; help us finish the job!
;
; This value can be switched at runtime using the cli command "dialplan set extenpatternmatchnew true"
; or "dialplan set extenpatternmatchnew false", so you can experiment to your hearts content.
;
;extenpatternmatchnew=no
;
; If clearglobalvars is set, global variables will be cleared
; and reparsed on a dialplan reload, or Asterisk reload.
;
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one of its included files, will remain set to the previous value.
;
; NOTE: A complication sets in, if you put your global variables into
; the AEL file, instead of the extensions.conf file. With clearglobalvars
; set, a "reload" will often leave the globals vars cleared, because it
; is not unusual to have extensions.conf (which will have no globals)
; load after the extensions.ael file (where the global vars are stored).
; So, with "reload" in this particular situation, first the AEL file will
; clear and then set all the global vars, then, later, when the extensions.conf
; file is loaded, the global vars are all cleared, and then not set, because
; they are not stored in the extensions.conf file.
;
clearglobalvars=no
;
; User context is where entries from users.conf are registered. The
; default value is 'default'
;
;userscontext=default
;
; You can include other config files, use the #include command
; (without the ';'). Note that this is different from the "include" command
; that includes contexts within other contexts. The #include command works
; in all asterisk configuration files.
;#include "filename.conf"
;#include <filename.conf>
;#include filename.conf
;
; You can execute a program or script that produces config files, and they
; will be inserted where you insert the #exec command. The #exec command
; works on all asterisk configuration files. However, you will need to
; activate them within asterisk.conf with the "execincludes" option. They
; are otherwise considered a security risk.
;#exec /opt/bin/build-extra-contexts.sh
;#exec /opt/bin/build-extra-contexts.sh --foo="bar"
;#exec </opt/bin/build-extra-contexts.sh --foo="bar">
;#exec "/opt/bin/build-extra-contexts.sh --foo=\"bar\""
;
; The "Globals" category contains global variables that can be referenced
; in the dialplan with the GLOBAL dialplan function:
; ${GLOBAL(VARIABLE)}
; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
; Unix/Linux environmental variables can be reached with the ENV dialplan
; function: ${ENV(VARIABLE)}
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=DAHDI/G2 ; Trunk interface
;
; Note the 'G2' in the TRUNK variable above. It specifies which group (defined
; in chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use
; in the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy DAHDI channel
; (aka. ascending sequential hunt group).
; G: select the highest-numbered non-busy DAHDI channel
; (aka. descending sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than last
; time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than last
; time (aka. descending rotary hunt group).
;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:pass@provider
;FREENUMDOMAIN=mydomain.com ; domain to send on outbound
; freenum calls (uses outbound-freenum
; context)
;
; WARNING WARNING WARNING WARNING
; If you load any other extension configuration engine, such as pbx_ael.so,
; your global variables may be overridden by that file. Please take care to
; use only one location to set global variables, and you will likely save
; yourself a ton of grief.
; WARNING WARNING WARNING WARNING
;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
; ! - wildcard, causes the matching process to complete as soon as
; it can unambiguously determine that no other matches are possible
;
; For example, the extension _NXXXXXX would match normal 7 digit dialings,
; while _1NXXNXXXXXX would represent an area code plus phone number
; preceded by a one.
;
; Each step of an extension is ordered by priority, which must always start
; with 1 to be considered a valid extension. The priority "next" or "n" means
; the previous priority plus one, regardless of whether the previous priority
; was associated with the current extension or not. The priority "same" or "s"
; means the same as the previously specified priority, again regardless of
; whether the previous entry was for the same extension. Priorities may be
; immediately followed by a plus sign and another integer to add that amount
; (most useful with 's' or 'n'). Priorities may then also have an alias, or
; label, in parentheses after their name which can be used in goto situations.
;
; Contexts contain several lines, one for each step of each extension. One may
; include another context in the current one as well, optionally with a date
; and time. Included contexts are included in the order they are listed.
; Switches may also be included within a context. The order of matching within
; a context is always exact extensions, pattern match extensions, includes, and
; switches. Includes are always processed depth-first. So for example, if you
; would like a switch "A" to match before context "B", simply put switch "A" in
; an included context "C", where "C" is included in your original context
; before "B".
;
;[context]
;
;autohints = yes
; If enabled for a context, a device state hint will be automatically created in
; the context with the name of the device and updated with device state changes.
;
;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
;
; Timing list for includes is
;
; <time range>,<days of week>,<days of month>,<months>[,<timezone>]
;
; Note that ranges may be specified to wrap around the ends. Also, minutes are
; fine-grained only down to the closest even minute.
;
;include => daytime,9:00-17:00,mon-fri,*,*
;include => weekend,*,sat-sun,*,*
;include => weeknights,17:02-8:58,mon-fri,*,*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon receipt
; of a particular pattern. The most commonly used example is of course '9'
; like this:
;
;ignorepat => 9
;
; so that dialtone remains even after dialing a 9. Please note that ignorepat
; only works with channels which receive dialtone from the PBX, such as DAHDI,
; Phone, and VPB. Other channels, such as SIP and MGCP, which generate their
; own dialtone and converse with the PBX only after a number is complete, are
; generally unaffected by ignorepat (unless DISA or another method is used to
; generate a dialtone after answering the channel).
;
;
; Sample entries for extensions.conf
;
;
[dundi-e164-canonical]
;include => stdexten
;
; List canonical entries here
;
;exten => 12564286000,1,Gosub(6000,stdexten(IAX2/foo))
;exten => 12564286000,n,Goto(default,s,1) ; exited Voicemail
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
[dundi-e164-customers]
;
; If you are an ITSP or Reseller, list your customers here.
;
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)
[dundi-e164-via-pstn]
;
; If you are freely delivering calls to the PSTN, list them here
;
;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325
[dundi-e164-local]
;
; Context to put your dundi IAX2 or SIP user in for
; full access
;
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn
[dundi-e164-switch]
;
; Just a wrapper for the switch
;
switch => DUNDi/e164
[dundi-e164-lookup]
;
; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don't have one.
;
include => dundi-e164-local
include => dundi-e164-switch
;
; DUNDi can also be implemented as a Macro instead of using
; the Local channel driver.
;
[macro-dundi-e164]
;
; ARG1 is the extension to Dial
;
; Extension "s" is not a wildcard extension that matches "anything".
; In macros, it is the start extension. In most other cases,
; you have to goto "s" to execute that extension.
;
; Note: In old versions of Asterisk the PBX in some cases defaulted to
; extension "s" when a given extension was wrong (like in AMI originate).
; This is no longer the case.
;
; For wildcard matches, see above - all pattern matches start with
; an underscore.
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup
;
; Here are the entries you need to participate in the IAXTEL
; call routing system. Most IAXTEL numbers begin with 1-700, but
; there are exceptions. For more information, and to sign
; up, please go to www.gnophone.com or www.iaxtel.com
;
[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
;
; The SWITCH statement permits a server to share the dialplan with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext
[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
;Include parkedcalls (or the context you define in features conf)
;to enable call parking.
include => parkedcalls
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
; Asterisk PBX
;
; switch => IAX2/user:password@bigserver/local
;
; An "lswitch" is like a switch but is literal, in that
; variable substitution is not performed at load time
; but is passed to the switch directly (presumably to
; be substituted in the switch routine itself)
;
; lswitch => Loopback/12${EXTEN}@othercontext
;
; An "eswitch" is like a switch but the evaluation of
; variable substitution is performed at runtime before
; being passed to the switch routine.
;
; eswitch => IAX2/context@${CURSERVER}
; The following two contexts are a template to enable the ability to dial
; ISN numbers. For more information about what an ISN number is, please see
; http://www.freenum.org.
;
; This is the dialing hook. use:
; include => outbound-freenum
[outbound-freenum]
; We'll add more digits as needed. The purpose is to dial things
; like extension numbers at domains (ITAD number) so we're matching
; on lengths of 1 through 6 prior to the separator (the asterisk [*])
;
exten => _X*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
[outbound-freenum2]
; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
;
exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)}) ; make sure the suffix is all digits as well
same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
same => n,Set(TIMEOUT(absolute)=10800)
same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ; perform our lookup with freenum.org
same => n,GotoIf($["${isnresult}" != ""]?from)
same => n,Set(DIALSTATUS=CONGESTION)
same => n,Goto(fn-CONGESTION,1)
same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial) ; check if we set the FREENUMDOMAIN global variable in [global]
same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ; if we did set it, then we'll use it for our outbound dialing domain
same => n(dial),Dial(SIP/${isnresult},40)
same => n,Goto(fn-${DIALSTATUS},1)
exten => fn-BUSY,1,Busy()
exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same => n,Congestion()
[macro-trunkdial]
;
; Standard trunk dial macro (hangs up on a dialstatus that should
; terminate call)
; ${ARG1} - What to dial
;
exten => s,1,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup()
exten => s-BUSY,1,Hangup()
exten => _s-.,1,NoOp
[stdexten]
;
; Standard extension subroutine:
; ${EXTEN} - Extension
; ${ARG1} - Device(s) to ring
; ${ARG2} - Optional context in Voicemail
;
; Note that the current version will drop through to the next priority in the
; case of their pressing '#'. This gives more flexibility in what do to next:
; you can prompt for a new extension, or drop the call, or send them to a
; general delivery mailbox, or...
;
; The use of the LOCAL() function is purely for convenience. Any variable
; initially declared as LOCAL() will disappear when the innermost Gosub context
; in which it was declared returns. Note also that you can declare a LOCAL()
; variable on top of an existing variable, and its value will revert to its
; previous value (before being declared as LOCAL()) upon Return.
;
exten => _X.,50000(stdexten),NoOp(Start stdexten)
exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
exten => _X.,n,Set(LOCAL(dev)=${ARG1})
exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
exten => _X.,n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})
exten => _X.,n,Dial(${dev},20) ; Ring the interface, 20 seconds maximum
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => stdexten-NOANSWER,1,VoiceMail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
exten => stdexten-BUSY,1,VoiceMail(${mbx},b) ; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,Return() ; If they press #, return to start
exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoiceMailMain(${mbx}) ; If they press *, send the user into VoicemailMain
exten => a,n,Return()
[stdPrivacyexten]
;
; Standard extension subroutine:
; ${ARG1} - Extension
; ${ARG2} - Device(s) to ring
; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
; ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
; ${ARG5} - Context in voicemail (if empty, then "default")
;
; See above note in stdexten about priority handling on exit.
;
exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
exten => _X.,n,Set(LOCAL(ext)=${ARG1})
exten => _X.,n,Set(LOCAL(dev)=${ARG2})
exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
exten => _X.,n,Set(LOCAL(cntx)=${ARG5})
exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
exten => _X.,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
; option (or use P for databased call _X.creening)
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => stdexten-NOANSWER,1,VoiceMail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
exten => stdexten-BUSY,1,VoiceMail(${mbx},b) ; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
exten => stdexten-BUSY,n,Return() ; If they press #, return to start
exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1) ; Callee chose to send this call to a polite "Don't call again" script.
exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1) ; Callee chose to send this call to a telemarketer torture script.
exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoiceMailMain(${mbx}) ; If they press *, send the user into VoicemailMain
exten => a,n,Return()
[macro-page]
;
; Paging macro:
;
; Check to see if SIP device is in use and DO NOT PAGE if they are
;
; ${ARG1} - Device to page
exten => s,1,ChanIsAvail(${ARG1},s) ; s is for ANY call
exten => s,n,GotoIf($[${AVAILSTATUS} = "1"]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA") ; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp() ; Add others here and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1})
exten => s,n(fail),Hangup()
[demo]
include => stdexten
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait(1) ; Wait a second, just for fun
exten => s,n,Answer() ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
exten => s,n,WaitExten() ; Wait for an extension to be dialed.
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)
exten => 3,1,Set(CHANNEL(language)=fr) ; Set language to french
exten => 3,n,Goto(s,restart) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
exten => 1234,n,Goto(default,s,1) ; exited Voicemail
exten => 1235,1,VoiceMail(1234,u) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,VoiceMail(1234,b) ; Unless busy
;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,Hangup() ; Hang them up.
;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry) ; Let them know what's going on
exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default) ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.
;
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo() ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over
;
; You can use the Macro Page to intercom a individual user
exten => 76245,1,Macro(page,SIP/Grandstream1)
; or if your peernames are the same as extensions
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
;
;
; System Wide Page at extension 7999
;
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)
; Give voicemail at extension 8500
;
exten => 8500,1,VoiceMailMain()
exten => 8500,n,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)
;
; The page context calls up the page macro that sets variables needed for auto-answer
; It is in is own context to make calling it from the Page() application as simple as
; Local/{peername}@page
;
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})
;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,n,Background(thanks) ; "Thanks for calling press 1 for sales, 2 for support, ..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts) ; "Thanks for calling the sales department. Press 1 for steve, 2 for..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)
[public]
;
; ATTENTION: If your Asterisk is connected to the internet and you do
; not have allowguest=no in sip.conf, everybody out there may use your
; public context without authentication. In that case you want to
; double check which services you offer to the world.
;
include => demo
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => demo
;
; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
; Note that you must have a [sipprovider] section in sip.conf
;
;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,,r)
; Real extensions would go here. Generally you want real extensions to be
; 4 or 5 digits long (although there is no such requirement) and start with a
; single digit that is fairly large (like 6 or 7) so that you have plenty of
; room to overlap extensions and menu options without conflict. You can alias
; them with names, too, and use global variables
;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed
;exten => 6245,n,VoiceMail(6245,u) ; Voicemail (unavailable)
;exten => 6245,s+1,Hangup ; s+1, same as n
;exten => 6245,dial+101,VoiceMail(6245,b) ; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}
;exten => 6275,1,Gosub(${EXTEN},stdexten(${MARK}))
; assuming ${MARK} is something like DAHDI/2
;exten => 6275,n,Goto(default,s,1) ; exited Voicemail
;exten => mark,1,Goto(6275,1) ; alias mark to 6275
;exten => 6536,1,Gosub(${EXTEN},stdexten(${WIL}))
; Ditto for wil
;exten => 6536,n,Goto(default,s,1) ; exited Voicemail
;exten => wil,1,Goto(6236,1)
;If you want to subscribe to the status of a parking space, this is
;how you do it. Subscribe to extension 6600 in sip, and you will see
;the status of the first parking lot with this extensions' help
;exten => 6600,hint,park:701@parkedcalls
;exten => 6600,1,noop
;
;To subscribe to the availability of a free member in the 'markq' queue.
;Note: '_avail' is added to the QueueName
;exten => 8501,hint,Queue:markq_avail
;exten => 8501,1,Queue(markq)
;
; You can also monitor the status of a queue by providing a hint for a
; particular queue name.
;exten => 8502,hint,Queue:markq
;exten => 8502,1,Queue(markq)
;
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;
;exten => 8500,1,VoicemailMain
;exten => 8500,n,Hangup
;
; Or a conference room (you'll need to edit meetme.conf to enable this room)
;
;exten => 8600,1,Meetme(1234)
;
; Or playing an announcement to the called party, as soon it answers
;
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
; example of a compartmentalized company called "acme"
;
; this is the context that your incoming IAX/SIP trunk dumps you in...
;[acme-incoming]
;exten => s,1,Wait(1)
;exten => s,n,Answer()
;exten => s,n(menu),Playback(acme/vm-brief-menu)
;exten => s,n(exten),Background(vm-enter-num-to-call)
;exten => s,n,WaitExten(5)
;exten => s,n(goodbye),Playback(vm-goodbye)
;exten => s,n(end),Hangup()
;
;include => acme-extens
;
;exten => i,1,Playback(vm-invalid)
;exten => i,n,Goto(s,exten) ; optionally, transfer to operator
;
;exten => t,1,Goto(s,goodbye)
;
; this is the context our internal SIP hardphones use (see sip.conf)
;
;[acme-internal]
;exten => s,1,Answer()
;exten => s,n(exten),Background(vm-enter-num-to-call)
;exten => s,n,WaitExten(5)
;exten => s,n(goodbye),Playback(vm-goodbye)
;exten => s,n(end),Hangup()
;
;include => trunkint
;include => trunkld
;include => trunklocal
;
;include => acme-extens
;
; you can test what your system sounds like to outside callers by dialing this
;exten => 777,1,DISA(no-password,acme-incoming)
;
; grouping of acme's extensions... never used directly, always included.
;
;[acme-extens]
;include => stdexten
;exten => 111,1,Gosub(111,stdexten(SIP/pete_1,acme))
;exten => 111,n,Goto(s,exten)
;
;exten => 112,1,Gosub(112,stdexten(SIP/nancy_1,acme))
;exten => 112,n,Goto(s,end)
;
; end of acme example
;
; Time context: you can patch this in via the following.
;
; [acme-internal]
; ...
; exten => 777,1,Gosub(time)
; exten => 777,n,Hangup()
;
; ...
; include => time
;
; Note: if you're geographically spread out, you can have SIP extensions
; specify their own local timezone in sip.conf as:
;
; [boi]
; type=friend
; context=acme-internal
; callerid="Boise Ofc. <2083451111>"
; ...
; ; use system-wide default timezone of MST7MDT
;
; [lws]
; type=friend
; context=acme-internal
; callerid="Lewiston Ofc. <2087431111>"
; ...
; setvar=timezone=PST8PDT
;
; "timezone" isn't a 'reserved' name in any way, and other places where
; the timezone is significant (e.g. calls to "SayUnixTime()", etc) will
; require modification as well. Note that voicemail.conf already has
; a mechanism for timezones.
;
[time]
exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
exten => _X.,n,Wait(0.25)
exten => _X.,n,Answer()
; the amount of delay is set for English; you may need to adjust this time
; for other languages if there's no pause before the synchronizing beep.
exten => _X.,n,Set(FUTURETIME=$[${EPOCH} + 12])
exten => _X.,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
exten => _X.,n,SayPhonetic(z)
; use the timezone associated with the extension (sip only), or system-wide
; default if one hasn't been set.
exten => _X.,n,SayUnixTime(${FUTURETIME},${timezone},HNS)
exten => _X.,n,Playback(spy-local)
exten => _X.,n,WaitUntil(${FUTURETIME})
exten => _X.,n,Playback(beep)
exten => _X.,n,Return()
;
; ANI context: use in the same way as "time" above
;
[ani]
exten => _X.,40000(ani),NoOp(ANI: ${EXTEN})
exten => _X.,n,Wait(0.25)
exten => _X.,n,Answer()
exten => _X.,n,Playback(vm-from)
exten => _X.,n,SayDigits(${CALLERID(ani)})
exten => _X.,n,Wait(1.25)
exten => _X.,n,SayDigits(${CALLERID(ani)}) ; playback again in case of missed digit
exten => _X.,n,Return()
; For more information on applications, just type "core show applications" at your
; friendly Asterisk CLI prompt.
;
; "core show application <command>" will show details of how you
; use that particular application in this file, the dial plan.
; "core show functions" will list all dialplan functions
; "core show function <COMMAND>" will show you more information about
; one function. Remember that function names are UPPER CASE.

156
extensions_minivm.conf Normal file
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; MINI-VOICEMAIL dialplan example
; ---------------------------------------------------------------------------------------
;
; This is an example on how to use the Mini-Voicemail system to build
; voicemail systems.
;
;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
; A macro to test the MINIVMACCOUNT dialplan function
; Currently, accountcode and pincode is not used in the application
; They where added to be used in dialplan scripting
;
;
[macro-minivmfunctest]
exten => s,1,set(account=${ARGV1})
exten => minivm,n,verbose(1,-------------------- Minivm Function test - Accoutn ${account} -------------)
exten => s,n,verbose(1,---- Has account: ${MINIVMACCOUNT(${account}:hasaccount)})
exten => s,n,verbose(1,---- Fullname: ${MINIVMACCOUNT(${account}:fullname)})
exten => s,n,verbose(1,---- Email: ${MINIVMACCOUNT(${account}:email)})
exten => s,n,verbose(1,---- Pager: ${MINIVMACCOUNT(${account}:pager)})
exten => s,n,verbose(1,---- E-mail template: ${MINIVMACCOUNT(${account}:etemplate)})
exten => s,n,verbose(1,---- Pager template: ${MINIVMACCOUNT(${account}:ptemplate)})
exten => s,n,verbose(1,---- Account code: ${MINIVMACCOUNT(${account}:accountcode)})
exten => s,n,verbose(1,---- Path: ${MINIVMACCOUNT(${account}:path)})
exten => s,n,verbose(1,---- Pincode: ${MINIVMACCOUNT(${account}:pincode)})
exten => s,n,verbose(1,---- Time zone: ${MINIVMACCOUNT(${account}:timezone)})
exten => s,n,verbose(1,---- Language: ${MINIVMACCOUNT(${account}:language)})
; This requires setvar=customerclass=gold in the account configuration
exten => s,n,verbose(1,---- Var:customerclass: ${MINIVMACCOUNT(${account}:customerclass)})
[minivm-scenario1]
; minivmtest tests the dialplan function MINIVMACCOUNT
; Check the output in the console with verbose set
exten => minivmtest,1,answer
exten => minivmtest,n,wait(0.5)
exten => minivmtest,n,set(ACCOUNT=do-not-spam-me@example.com)
exten => minivmtest,n,macro(minivmfunctest, ${ACCOUNT})
exten => minivmtest,n,playback(beep)
exten => minivmtest,n,hangup
;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
; "minivm" tests a full scenario
; Remember that users may hangup
; This works both for users with accounts in minivm.conf and by just giving an e-mail address
; without configuring an account
exten => minivm,1,answer
exten => minivm,n,wait(0.5) ; Wait for Voip channels to settle
exten => minivm,n,set(account=oej@example.com)
exten => minivm,n,noop(------------------------------------------- Minivm Greet -------------)
exten => minivm,n,minivmgreet(${account})
exten => minivm,n,verbose(1,-- MINIVM_GREET_STATUS = ${MINIVM_GREET_STATUS} )
exten => minivm,n,noop(------------------------------------------- Minivm Record -------------)
exten => minivm,n,minivmRecord(${account},b)
exten => minivm,n,goto(minivmcleanup,1)
; Cleanup after recording or hangup
exten => minivmcleanup,1,noop(------------------------------------------- Minivm Notify -------------)
;Increment voicemail counter with 1. The counter will be used in the e-mail message
;and in the filename
exten => minivmcleanup,n,gotoif($[${MINIVM_RECORD_STATUS} != SUCCESS]?minivmrecordfailure,1)
exten => minivmcleanup,n,set(MINIVMCOUNTER(${account}:voicemailcounter:inc)=1)
exten => minivmcleanup,n,set(MVM_COUNTER = ${MINIVMCOUNTER(${account}:voicemailcounter)})
exten => minivmcleanup,n,minivmNotify(${account})
exten => minivmcleanup,n,verbose(1,-- MINIVM_NOTIFY_STATUS = ${MINIVM_NOTIFY_STATUS} )
; Now, clean up after sending voicemail
exten => minivmcleanup,n,noop(------------------------------------------- Minivm Delete -------------)
exten => minivmcleanup,n,minivmdelete()
exten => minivmcleanup,n,verbose(1,-- MINIVM_DELETE_STATUS = ${MINIVM_DELETE_STATUS} )
;Recording failed
exten => minivmrecordfailure,1,playback(vm-sorry)
exten => minivmrecordfailure,n,wait(1)
exten => minivmrecordfailure,n,hangup
; If the user hangs up during the recording, we need to clean up
; And send notifications
exten => h,1,gotoif($[x${MINIVM_DELETE_STATUS} != x] ?h,stop)
exten => h,n,noop(------------------------------------------- HANGUP during voicemail recording -------------)
exten => h,n,goto(minivmcleanup,1)
exten => h,n(stop),noop(---Minivm DONE----)
;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
; Extension to record a greeting message
; Call this like:
; macro(recordgreetings,alice@atlanta.example.com)
;
[macro-recordgreetings]
exten => s,1,answer
exten => s,n,wait(0.5)
exten => s,n,set(account=${ARGV1])
; This file give extra options not available here, needs to be edited
; Change of password does not work
exten => s,n(menu),background(vm-options)
exten => 1,1,setvar(option=u)
exten => 1,n,macro(minivmrec,${account},${option})
exten => 1,n,goto(menu)
exten => 2,1,setvar(option=b)
exten => 2,n,macro(minivmrec,${account},${option})
exten => 2,n,goto(menu)
exten => 3,1,setvar(option=n)
exten => 3,n,macro(minivmrec,${account},${option})
exten => 3,n,goto(menu)
exten => 4,1,setvar(option=t)
exten => 4,n,macro(minivmrec,${account},${option})
exten => 4,n,goto(menu)
exten => *,1,playback(vm-thankyou)
exten => *,n,wait(1)
exten => *,n,hangup
exten => i,1,playback(invalid)
exten => i,n,goto(menu)
[macro-minivmrec]
exten => s,1,gotoif(${MINIVMACCOUNT(${account}:hasaccount)}?record)
; Account is not configured in minivm.conf or realtime
; Phony message, add something useful here
exten => s,n,playback(privacy-incorrect)
exten => s,n,macroreturn
exten => record,1,minivmappmess(${ARGV1},${ARGV2})
exten => record,n,noop(Recording status: ${MINIVM_APPMESS_STATUS})
exten => record,n,macroreturn
;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
; To set a counter and use a template for voicemail to users without acounts
; use something like this
;
; email address is in the "account" channel variable. Set from ast_db or a script
; based on called ID
exten => sendvoicemail,1,answer
exten => sendvoicemail,n,wait(0.5)
exten => sendvoicemail,n,set(domain=${CUT(account,@,2)})
exten => sendvoicemail,n,set(country=${CUT(domain,.,2)})
exten => sendvoicemail,n,minivmgreet(${account})
exten => sendvoicemail,n,minivmRecord(${account},b)
exten => sendvoicemail,n,goto(sendvmcleanup)
exten => sendvmcleanup,1,gotoif($[${MINIVM_RECORD_STATUS} != SUCCESS]?done)
; The counter is set in the domain directory, so we don't create one directory per user
; The counter has the email in the name of the counter, increase it
; Set the MVM_COUNTER variable that we use in the template
exten => sendvmcleanup,n,set(MINIVMCOUNTER(${account}:voicemailcounter:inc)=1)
exten => sendvmcleanup,n,set(MVM_COUNTER = ${MINIVMCOUNTER(${account}:voicemailcounter)})
; Increase a domain counter too, to see how many voicemails are sent to this domain
; This is just for statistics
exten => sendvmcleanup,n,set(MINIVMCOUNTER(${domain}:${domain}-all:inc) = 1)
; Send voicemail in e-mail with country-specific template
; The template need to be defined in minivm.conf
;
exten => sendvmcleanup,n,minivmNotify(${account}, ${country}_email)
exten => sendvmcleanup,n,minivmDelete()
exten => sendvmcleanup,n(done),wait(0.5)
exten => sendvmcleanup,n,hangup
exten => h,1,gotoif($[${MINIVM_RECORD_STATUS} = SUCCESS]?sendvmcleanup,1))

122
features.conf Normal file
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;
; Sample Call Features (transfer, monitor/mixmonitor, etc) configuration
;
; Note: From Asterisk 12 - All parking lot configuration is now done in res_parking.conf
[general]
;transferdigittimeout => 3 ; Number of seconds to wait between digits when transferring a call
; (default is 3 seconds)
;xfersound = beep ; to indicate an attended transfer is complete
;xferfailsound = beeperr ; to indicate a failed transfer
;pickupexten = *8 ; Configure the pickup extension. (default is *8)
;pickupsound = beep ; to indicate a successful pickup (default: no sound)
;pickupfailsound = beeperr ; to indicate that the pickup failed (default: no sound)
;featuredigittimeout = 1000 ; Max time (ms) between digits for
; feature activation (default is 1000 ms)
;recordingfailsound = beeperr ; indicates that a one-touch monitor or one-touch mixmonitor feature failed
; to be applied to the call. (default: no sound)
;atxfernoanswertimeout = 15 ; Timeout for answer on attended transfer default is 15 seconds.
;atxferdropcall = no ; If someone does an attended transfer, then hangs up before the transfer
; target answers, then by default, the system will try to call back the
; person that did the transfer. If this is set to "yes", the ringing
; transfer target is immediately transferred to the transferee.
;atxferloopdelay = 10 ; Number of seconds to sleep between retries (if atxferdropcall = no)
;atxfercallbackretries = 2 ; Number of times to attempt to send the call back to the transferer.
; By default, this is 2.
;transferdialattempts = 3 ; Number of times that a transferer may attempt to dial an extension before
; being kicked back to the original call.
;transferretrysound = "beep" ; Sound to play when a transferer fails to dial a valid extension.
;transferinvalidsound = "beeperr" ; Sound to play when a transferer fails to dial a valid extension and is out of retries.
;atxferabort = *1 ; cancel the attended transfer
;atxfercomplete = *2 ; complete the attended transfer, dropping out of the call
;atxferthreeway = *3 ; complete the attended transfer, but stay in the call. This will turn the call into a multi-party bridge
;atxferswap = *4 ; swap to the other party. Once an attended transfer has begun, this options may be used multiple times
; Note that the DTMF features listed below only work when two channels have answered and are bridged together.
; They can not be used while the remote party is ringing or in progress. If you require this feature you can use
; chan_local in combination with Answer to accomplish it.
[featuremap]
;blindxfer => #1 ; Blind transfer (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
;disconnect => *0 ; Disconnect (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
;automon => *1 ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
;atxfer => *2 ; Attended transfer -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
;parkcall => #72 ; Park call (one step parking) -- Make sure to set the K and/or k option in the Dial() app call!
;automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!
[applicationmap]
; Note that the DYNAMIC_FEATURES channel variable must be set to use the features
; defined here. The value of DYNAMIC_FEATURES should be the names of the features
; to allow the channel to use separated by '#'. For example:
;
; Set(__DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3)
;
; (Note: The two leading underscores allow these feature settings to be set
; on the outbound channels, as well. Otherwise, only the original channel
; will have access to these features.)
;
; The syntax for declaring a dynamic feature is any of the following:
;
;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_Class]]
;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,"<AppArguments>"[,MOH_Class]]
;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>([<AppArguments>])[,MOH_Class]
;
; FeatureName -> This is the name of the feature used when setting the
; DYNAMIC_FEATURES variable to enable usage of this feature.
; DTMF_sequence -> This is the key sequence used to activate this feature.
; ActivateOn -> This is the channel of the call that the application will be executed
; on. Valid values are "self" and "peer". "self" means run the
; application on the same channel that activated the feature. "peer"
; means run the application on the opposite channel from the one that
; has activated the feature.
; ActivatedBy -> ActivatedBy is no longer honored. The feature is activated by which
; channel DYNAMIC_FEATURES includes the feature is on. Use a pre-dial
; handler to set different values for DYNAMIC_FEATURES on the channels.
; Historic values are: "caller", "callee", and "both".
; Application -> This is the application to execute.
; AppArguments -> These are the arguments to be passed into the application. If you need
; commas in your arguments, you should use either the second or third
; syntax, above.
; MOH_Class -> This is the music on hold class to play while the idle
; channel waits for the feature to complete. If left blank,
; no music will be played.
;
;
; IMPORTANT NOTE: The applicationmap is not intended to be used for all Asterisk
; applications. When applications are used in extensions.conf, they are executed
; by the PBX core. In this case, these applications are executed outside of the
; PBX core, so it does *not* make sense to use any application which has any
; concept of dialplan flow. Examples of this would be things like Goto,
; Background, WaitExten, and many more. The exceptions to this are Gosub and
; Macro routines which must complete for the call to continue.
;
; Enabling these features means that the PBX needs to stay in the media flow and
; media will not be re-directed if DTMF is sent in the media stream.
;
; Example Usage:
;
;testfeature => #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play
; ;tt-monkeys to the opposite channel
;
; Set arbitrary channel variables, based upon CALLERID number (Note that the application
; argument contains commas)
;retrieveinfo => #8,peer,Set(ARRAY(CDR(mark),CDR(name))=${ODBC_FOO(${CALLERID(num)})})
;
;pauseMonitor => #1,self/callee,Pausemonitor ;Allow the callee to pause monitoring
; ;on their channel
;unpauseMonitor => #3,self/callee,UnPauseMonitor ;Allow the callee to unpause monitoring
; ;on their channel
; Dynamic Feature Groups:
; Dynamic feature groups are groupings of features defined in [applicationmap]
; that can have their own custom key mappings. To give a channel access to a dynamic
; feature group, add the group name to the value of the DYNAMIC_FEATURES variable.
;
; example:
; [myGroupName] ; defines the group named myGroupName
; testfeature => #9 ; associates testfeature with the group and the keycode '#9'.
; pauseMonitor => ; associates pauseMonitor with the group and uses the keycode specified
; ; in the [applicationmap].

34
festival.conf Normal file
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;
; Festival Configuration
;
[general]
;
; Host which runs the festival server (default : localhost);
;
;host=localhost
;
; Port on host where the festival server runs (default : 1314)
;
;port=1314
;
; Use cache (yes, no - defaults to no)
;
;usecache=yes
;
; 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=/var/lib/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
;
;

107
followme.conf Normal file
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; Find-Me / Follow-Me Configuration File
[general]
;
featuredigittimeout=>5000
; The number of ms to wait for a digit input for the callee on whether to take the call or
; not before we consider them "done" entering digits.
;
enable_callee_prompt=>true
; Enable prompting a callee to either accept or reject the forwarded call.
; If disabled, the optional prompting for caller name (option 'a') is
; likewise disabled. Enabled by default.
;
takecall=>1
; The global default keypress for the callee to take taking the current call. This can be
; a single digit or multiple digits. Default is "1".
; Note this string must not be a substring of declinecall.
;
declinecall=>2
; The global default keypress for the callee to decline taking the current call. This can
; be a single digit or multiple digits. Default is "2".
; Note this string must not be a substring of takecall.
;
call_from_prompt=>followme/call-from
; The global default for the 'Incoming call from' message.
;
norecording_prompt=>followme/no-recording
; The global default for the 'You have an incoming call' message when the caller elects
; not to leave their name or the option isn't set for them to do so.
;
options_prompt=>followme/options
; The global default for the 'Press 1 to accept this call or press 2 to decline it' message.
;
pls_hold_prompt=>followme/pls-hold-while-try
; The global default for 'Please hold while we try and connect your call' message.
;
status_prompt=>followme/status
; The global default for 'The party you're calling isn't at their desk' message.
;
sorry_prompt=>followme/sorry
; The global default for 'I'm sorry, but we were unable to locate your party' message.
;
connecting_prompt=>
; The global default sound file name for 'Please say hello to the caller' message.
; Setting to an empty string skips playing the prompt. The default is no prompt
; file name.
[default]
musicclass=>default
; The moh class that should be used for the caller while they are waiting to be connected.
context=>default
; The context to dial the numbers from
number=>01233456,25
; The a follow-me number to call. The format is:
; number=> <number to call[&2nd #[&...]]>[,<timeout value in seconds>[,<order in follow-me>]]
; You can specify as many of these numbers as you like. They will be dialed in the
; order that you specify them in the config file OR as specified with the order field
; on the number prompt. As you can see from the example, forked dialing of multiple
; numbers in the same step is supported with this application if you'd like to dial
; multiple numbers in the same followme step.
;
; The timeout value is the amount of time allowed between the time the dialing step
; starts and the callee answers. The callee then has until the timeout of the last
; step to make a choice on whether to take the call or not. That being the case,
; you may want to make the timeout on the last step longer to give enough time to
; make the choice to accept or not.
enable_callee_prompt=>true
; Enable prompting the callee to accept the forwarded call. The default
; is the global value.
takecall=>1
; The keypress for the callee to take taking the current call. This can be
; a single digit or multiple digits. Default is the global default.
; Note this string must not be a substring of declinecall.
;
declinecall=>2
; The keypress for the callee to decline taking the current call. This can
; be a single digit or multiple digits. Default is the global default.
; Note this string must not be a substring of takecall.
;
call_from_prompt=>followme/call-from
; The 'Incoming call from' message prompt. Default is the global default.
;
norecording_prompt=>followme/no-recording
; The 'You have an incoming call' message prompt when the caller elects
; not to leave their name or the option isn't set for them to do so. Default
; is the global default.
;
options_prompt=>followme/options
; The 'Press 1 to accept this call or press 2 to decline it' message prompt.
; Default is the global default.
;
pls_hold_prompt=>followme/pls-hold-while-try
; The 'Please hold while we try and connect your call' message prompt.
; Default is the global default.
;
status_prompt=>followme/status
; The 'The party you're calling isn't at their desk' message prompt.
; Default is the global default.
;
sorry_prompt=>followme/sorry
; The 'I'm sorry, but we were unable to locate your party' message prompt. Default
; is the global default.
;
connecting_prompt=>
; The sound file name for 'Please say hello to the caller' message. Default is the
; global default.
;

32
hep.conf Normal file
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;
; res_hep Module configuration for Asterisk
;
;
; Note that this configuration file is consumed by res_hep, which is responsible
; for the HEPv3 protocol manipulation and managing the connection to the Homer
; capture server. Additional modules provide specific messages to be sent to
; the Homer server:
; - res_hep_pjsip: Send SIP messages transmitted/received by the PJSIP stack
; - res_hep_rtcp: Send RTCP information (all channels)
;
; All settings are currently set in the general section.
[general]
enabled = no ; Enable/disable forwarding of packets to a
; HEP server. Default is "yes".
capture_address = 192.168.1.1:9061 ; The address of the HEP capture server.
capture_password = foo ; If specified, the authorization passsword
; for the HEP server. If not specified, no
; authorization password will be sent.
capture_id = 1234 ; A unique integer identifier for this
; server. This ID will be embedded sent
; with each packet from this server.
uuid_type = call-id ; Specify the preferred source for the Homer
; correlation UUID. Valid options are:
; - 'call-id' for the PJSIP or chan_sip SIP
; Call-ID
; - 'channel' for the Asterisk channel name
; Note: If 'call-id' is specified but the
; channel is not PJSIP or chan_sip then the
; Asterisk channel name will be used instead.

126
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;
; Asterisk Builtin mini-HTTP server
;
;
; Note about Asterisk documentation:
; If Asterisk was installed from a tarball, then the HTML documentation should
; be installed in the static-http/docs directory which is
; (/var/lib/asterisk/static-http/docs) on linux by default. If the Asterisk
; HTTP server is enabled in this file by setting the "enabled", "bindaddr",
; and "bindport" options, then you should be able to view the documentation
; remotely by browsing to:
; http://<server_ip>:<bindport>/static/docs/index.html
;
[general]
;
; The name of the server, advertised in both the Server field in HTTP
; response message headers, as well as the <address /> element in certain HTTP
; response message bodies. If not furnished here, "Asterisk/{version}" will be
; used as a default value for the Server header field and the <address />
; element. Setting this property to a blank value will result in the omission
; of the Server header field from HTTP response message headers and the
; <address /> element from HTTP response message bodies.
;
servername=Asterisk
;
; Whether HTTP/HTTPS interface is enabled or not. Default is no.
; This also affects manager/rawman/mxml access (see manager.conf)
;
;enabled=yes
;
; Address to bind to, both for HTTP and HTTPS. You MUST specify
; a bindaddr in order for the HTTP server to run. There is no
; default value.
;
bindaddr=127.0.0.1
;
; Port to bind to for HTTP sessions (default is 8088)
;
;bindport=8088
;
; Prefix allows you to specify a prefix for all requests
; to the server. The default is blank. If uncommented
; all requests must begin with /asterisk
;
;prefix=asterisk
;
; sessionlimit specifies the maximum number of httpsessions that will be
; allowed to exist at any given time. (default: 100)
;
;sessionlimit=100
;
; session_inactivity specifies the number of milliseconds to wait for
; more data over the HTTP connection before closing it.
;
; Default: 30000
;session_inactivity=30000
;
; session_keep_alive specifies the number of milliseconds to wait for
; the next HTTP request over a persistent connection.
;
; Set to 0 to disable persistent HTTP connections.
; Default: 15000
;session_keep_alive=15000
;
; Whether Asterisk should serve static content from static-http
; Default is no.
;
;enablestatic=yes
;
; Redirect one URI to another. This is how you would set a
; default page.
; Syntax: redirect=<from here> <to there>
; For example, if you are using the Asterisk-gui,
; it is convenient to enable the following redirect:
;
;redirect = / /static/config/index.html
;
; HTTPS support. In addition to enabled=yes, you need to
; explicitly enable tls, define the port to use,
; and have a certificate somewhere.
;tlsenable=yes ; enable tls - default no.
;tlsbindaddr=0.0.0.0:8089 ; address and port to bind to - default is bindaddr and port 8089.
;
;tlscertfile=</path/to/certificate.pem> ; path to the certificate file (*.pem) only.
;tlsprivatekey=</path/to/private.pem> ; path to private key file (*.pem) only.
; If no path is given for tlscertfile or tlsprivatekey, default is to look in current
; directory. If no tlsprivatekey is given, default is to search tlscertfile for private key.
;
; To produce a certificate you can e.g. use openssl. This places both the cert and
; private in same .pem file.
; openssl req -new -x509 -days 365 -nodes -out /tmp/foo.pem -keyout /tmp/foo.pem
;
; tlscipher= ; The list of allowed ciphers
; ; if none are specified the following cipher
; ; list will be used instead:
; ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES256-GCM-SHA384:
; ECDHE-ECDSA-AES256-GCM-SHA384:DHE-RSA-AES128-GCM-SHA256:DHE-DSS-AES128-GCM-SHA256:
; kEDH+AESGCM:ECDHE-RSA-AES128-SHA256:ECDHE-ECDSA-AES128-SHA256:ECDHE-RSA-AES128-SHA:
; ECDHE-ECDSA-AES128-SHA:ECDHE-RSA-AES256-SHA384:ECDHE-ECDSA-AES256-SHA384:
; ECDHE-RSA-AES256-SHA:ECDHE-ECDSA-AES256-SHA:DHE-RSA-AES128-SHA256:DHE-RSA-AES128-SHA:
; DHE-DSS-AES128-SHA256:DHE-RSA-AES256-SHA256:DHE-DSS-AES256-SHA:DHE-RSA-AES256-SHA:
; AES128-GCM-SHA256:AES256-GCM-SHA384:AES128-SHA256:AES256-SHA256:AES128-SHA:AES256-SHA:
; AES:CAMELLIA:DES-CBC3-SHA:!aNULL:!eNULL:!EXPORT:!DES:!RC4:!MD5:!PSK:!aECDH:
; !EDH-DSS-DES-CBC3-SHA:!EDH-RSA-DES-CBC3-SHA:!KRB5-DES-CBC3-SHA
;
; tlsdisablev1=yes ; Disable TLSv1 support - if not set this defaults to "yes"
; tlsdisablev11=yes ; Disable TLSv1.1 support - if not set this defaults to "no"
; tlsdisablev12=yes ; Disable TLSv1.2 support - if not set this defaults to "no"
;
; tlsservercipherorder=yes ; Use the server preference order instead of the client order
; ; Defaults to "yes"
;
; The post_mappings section maps URLs to real paths on the filesystem. If a
; POST is done from within an authenticated manager session to one of the
; configured POST mappings, then any files in the POST will be placed in the
; configured directory.
;
;[post_mappings]
;
; NOTE: You need a valid HTTP AMI mansession_id cookie with the manager
; config permission to POST files.
;
; In this example, if the prefix option is set to "asterisk", then using the
; POST URL: /asterisk/uploads will put files in /var/lib/asterisk/uploads/.
;uploads = /var/lib/asterisk/uploads/
;

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@@ -0,0 +1,736 @@
;
; indications.conf
;
; Configuration file for location specific tone indications
;
;
; NOTE:
; When adding countries to this file, please keep them in alphabetical
; order according to the 2-character country codes!
;
; The [general] category is for certain global variables.
; All other categories are interpreted as location specific indications
;
[general]
country=us ; default location
; [example]
; description = string
; The full name of your country, in English.
; ringcadence = num[,num]*
; List of durations the physical bell rings.
; dial = tonelist
; Set of tones to be played when one picks up the hook.
; busy = tonelist
; Set of tones played when the receiving end is busy.
; congestion = tonelist
; Set of tones played when there is some congestion (on the network?)
; callwaiting = tonelist
; Set of tones played when there is a call waiting in the background.
; dialrecall = tonelist
; Not well defined; many phone systems play a recall dial tone after hook
; flash.
; record = tonelist
; Set of tones played when call recording is in progress.
; info = tonelist
; Set of tones played with special information messages (e.g., "number is
; out of service")
; 'name' = tonelist
; Every other variable will be available as a shortcut for the "PlayList" command
; but will not be used automatically by Asterisk.
;
;
; The tonelist itself is defined by a comma-separated sequence of elements.
; Each element consist of a frequency (f) with an optional duration (in ms)
; attached to it (f/duration). The frequency component may be a mixture of two
; frequencies (f1+f2) or a frequency modulated by another frequency (f1*f2).
; The implicit modulation depth is fixed at 90%, though.
; If the list element starts with a !, that element is NOT repeated,
; therefore, only if all elements start with !, the tonelist is time-limited,
; all others will repeat indefinitely.
;
; concisely:
; element = [!]freq[+|*freq2][/duration]
; tonelist = element[,element]*
;
[at]
description = Austria
ringcadence = 1000,5000
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
dial = 420
busy = 420/400,0/400
ring = 420/1000,0/5000
congestion = 420/200,0/200
callwaiting = 420/40,0/1960
dialrecall = 420
; RECORDTONE - not specified
record = 1400/80,0/14920
info = 950/330,1450/330,1850/330,0/1000
stutter = 380+420
[au]
description = Australia
; Reference http://www.acif.org.au/__data/page/3303/S002_2001.pdf
; Normal Ring
ringcadence = 400,200,400,2000
; Distinctive Ring 1 - Forwarded Calls
; 400,400,200,200,400,1400
; Distinctive Ring 2 - Selective Ring 2 + Operator + Recall
; 400,400,200,2000
; Distinctive Ring 3 - Multiple Subscriber Number 1
; 200,200,400,2200
; Distinctive Ring 4 - Selective Ring 1 + Centrex
; 400,2600
; Distinctive Ring 5 - Selective Ring 3
; 400,400,200,400,200,1400
; Distinctive Ring 6 - Multiple Subscriber Number 2
; 200,400,200,200,400,1600
; Distinctive Ring 7 - Multiple Subscriber Number 3 + Data Privacy
; 200,400,200,400,200,1600
; Tones
dial = 413+438
busy = 425/375,0/375
ring = 413+438/400,0/200,413+438/400,0/2000
; XXX Congestion: Should reduce by 10 db every other cadence XXX
congestion = 425/375,0/375,420/375,0/375
callwaiting = 425/200,0/200,425/200,0/4400
dialrecall = 413+438
; Record tone used for Call Intrusion/Recording or Conference
record = !425/1000,!0/15000,425/360,0/15000
info = 425/2500,0/500
; Other Australian Tones
; The STD "pips" indicate the call is not an untimed local call
std = !525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100
; Facility confirmation tone (eg. Call Forward Activated)
facility = 425
; Message Waiting "stutter" dialtone
stutter = 413+438/100,0/40
; Ringtone for calls to Telstra mobiles
ringmobile = 400+450/400,0/200,400+450/400,0/2000
[bg]
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
description = Bulgaria
ringcadence = 1000,4000
;
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/250,0/250
callwaiting = 425/150,0/150,425/150,0/4000
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
record = 1400/425,0/15000
info = 950/330,1400/330,1800/330,0/1000
stutter = 425/1500,0/100
[br]
description = Brazil
ringcadence = 1000,4000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/4000
congestion = 425/250,0/250,425/750,0/250
callwaiting = 425/50,0/1000
; Dialrecall not used in Brazil standard (using UK standard)
dialrecall = 350+440
; Record tone is not used in Brazil, use busy tone
record = 425/250,0/250
; Info not used in Brazil standard (using UK standard)
info = 950/330,1400/330,1800/330
stutter = 350+440
[be]
description = Belgium
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadence = 1000,3000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/3000
congestion = 425/167,0/167
callwaiting = 1400/175,0/175,1400/175,0/3500
; DIALRECALL - not specified
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
; RECORDTONE - not specified
record = 1400/500,0/15000
info = 900/330,1400/330,1800/330,0/1000
stutter = 425/1000,0/250
[ch]
description = Switzerland
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadence = 1000,4000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/200,0/200,425/200,0/4000
; DIALRECALL - not specified
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
; RECORDTONE - not specified
record = 1400/80,0/15000
info = 950/330,1400/330,1800/330,0/1000
stutter = 425+340/1100,0/1100
[cl]
description = Chile
; According to specs from Telefonica CTC Chile
ringcadence = 1000,3000
dial = 400
busy = 400/500,0/500
ring = 400/1000,0/3000
congestion = 400/200,0/200
callwaiting = 400/250,0/8750
dialrecall = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
record = 1400/500,0/15000
info = 950/333,1400/333,1800/333,0/1000
stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
[cn]
description = China
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadence = 1000,4000
dial = 450
busy = 450/350,0/350
ring = 450/1000,0/4000
congestion = 450/700,0/700
callwaiting = 450/400,0/4000
dialrecall = 450
record = 950/400,0/10000
info = 450/100,0/100,450/100,0/100,450/100,0/100,450/400,0/400
; STUTTER - not specified
stutter = 450+425
[cz]
description = Czech Republic
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadence = 1000,4000
dial = 425/330,0/330,425/660,0/660
busy = 425/330,0/330
ring = 425/1000,0/4000
congestion = 425/165,0/165
callwaiting = 425/330,0/9000
; DIALRECALL - not specified
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425/330,0/330,425/660,0/660
; RECORDTONE - not specified
record = 1400/500,0/14000
info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
; STUTTER - not specified
stutter = 425/450,0/50
[de]
description = Germany
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
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 - not specified
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
; RECORDTONE - not specified
record = 1400/80,0/15000
info = 950/330,1400/330,1800/330,0/1000
stutter = 425+400
[dk]
description = Denmark
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadence = 1000,4000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = !425/200,!0/600,!425/200,!0/3000,!425/200,!0/200,!425/200,0
; DIALRECALL - not specified
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
; RECORDTONE - not specified
record = 1400/80,0/15000
info = 950/330,1400/330,1800/330,0/1000
; STUTTER - not specified
stutter = 425/450,0/50
[ee]
description = Estonia
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadence = 1000,4000
dial = 425
busy = 425/300,0/300
ring = 425/1000,0/4000
congestion = 425/200,0/200
; CALLWAIT not in accordance to ITU
callwaiting = 950/650,0/325,950/325,0/30,1400/1300,0/2600
; DIALRECALL - not specified
dialrecall = 425/650,0/25
; RECORDTONE - not specified
record = 1400/500,0/15000
; INFO not in accordance to ITU
info = 950/650,0/325,950/325,0/30,1400/1300,0/2600
; STUTTER not specified
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
[es]
description = Spain
ringcadence = 1500,3000
dial = 425
busy = 425/200,0/200
ring = 425/1500,0/3000
congestion = 425/200,0/200,425/200,0/200,425/200,0/600
callwaiting = 425/175,0/175,425/175,0/3500
dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425
record = 1400/500,0/15000
info = 950/330,0/1000
dialout = 500
; STUTTER not specified
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
[fi]
description = Finland
ringcadence = 1000,4000
dial = 425
busy = 425/300,0/300
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/150,0/150,425/150,0/8000
dialrecall = 425/650,0/25
record = 1400/500,0/15000
info = 950/650,0/325,950/325,0/30,1400/1300,0/2600
stutter = 425/650,0/25
[fr]
description = France
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadence = 1500,3500
; Dialtone can also be 440+330
dial = 440
busy = 440/500,0/500
ring = 440/1500,0/3500
; CONGESTION - not specified
congestion = 440/250,0/250
callwait = 440/300,0/10000
; DIALRECALL - not specified
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
; RECORDTONE - not specified
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330
stutter = !440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,440
[gr]
description = Greece
ringcadence = 1000,4000
dial = 425/200,0/300,425/700,0/800
busy = 425/300,0/300
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/150,0/150,425/150,0/8000
dialrecall = 425/650,0/25
record = 1400/400,0/15000
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
stutter = 425/650,0/25
[hu]
description = Hungary
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadence = 1250,3750
dial = 425
busy = 425/300,0/300
ring = 425/1250,0/3750
congestion = 425/300,0/300
callwaiting = 425/40,0/1960
dialrecall = 425+450
; RECORDTONE - not specified
record = 1400/400,0/15000
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
stutter = 350+375+400
[il]
description = Israel
ringcadence = 1000,3000
dial = 414
busy = 414/500,0/500
ring = 414/1000,0/3000
congestion = 414/250,0/250
callwaiting = 414/100,0/100,414/100,0/100,414/600,0/3000
dialrecall = !414/100,!0/100,!414/100,!0/100,!414/100,!0/100,414
record = 1400/500,0/15000
info = 1000/330,1400/330,1800/330,0/1000
stutter = !414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,414
[in]
description = India
ringcadence = 400,200,400,2000
dial = 400*25
busy = 400/750,0/750
ring = 400*25/400,0/200,400*25/400,0/2000
congestion = 400/250,0/250
callwaiting = 400/200,0/100,400/200,0/7500
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,!1800/330,0/1000
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,!0/100,400*25
[it]
description = Italy
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadence = 1000,4000
dial = 425/200,0/200,425/600,0/1000
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/400,0/100,425/250,0/100,425/150,0/14000
dialrecall = 470/400,425/400
record = 1400/400,0/15000
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
stutter = 470/400,425/400
[lt]
description = Lithuania
ringcadence = 1000,4000
dial = 425
busy = 425/350,0/350
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/150,0/150,425/150,0/4000
; DIALRECALL - not specified
dialrecall = 425/500,0/50
; RECORDTONE - not specified
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
; STUTTER - not specified
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
[jp]
description = Japan
ringcadence = 1000,2000
dial = 400
busy = 400/500,0/500
ring = 400+15/1000,0/2000
congestion = 400/500,0/500
callwaiting = 400+16/500,0/8000
dialrecall = !400/200,!0/200,!400/200,!0/200,!400/200,!0/200,400
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0
stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
[mx]
description = Mexico
ringcadence = 2000,4000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/4000
congestion = 425/250,0/250
callwaiting = 425/200,0/600,425/200,0/10000
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
record = 1400/500,0/15000
info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
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,!0/100,425
[my]
description = Malaysia
ringcadence = 2000,4000
dial = 425
busy = 425/500,0/500
ring = 425/400,0/200,425/400,0/2000
congestion = 425/500,0/500
; STUTTER - not specified
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
[nl]
description = Netherlands
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
ringcadence = 1000,4000
; Most of these 425's can also be 450's
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/250,0/250
callwaiting = 425/500,0/9500
; DIALRECALL - not specified
dialrecall = 425/500,0/50
; RECORDTONE - not specified
record = 1400/500,0/15000
info = 950/330,1400/330,1800/330,0/1000
stutter = 425/500,0/50
[no]
description = Norway
ringcadence = 1000,4000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/200,0/600,425/200,0/10000
dialrecall = 470/400,425/400
record = 1400/400,0/15000
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
stutter = 470/400,425/400
[nz]
description = New Zealand
; Reference = http://www.telepermit.co.nz/TNA102.pdf
ringcadence = 400,200,400,2000
dial = 400
busy = 400/500,0/500
ring = 400+450/400,0/200,400+450/400,0/2000
congestion = 400/250,0/250
callwaiting = !400/200,!0/3000,!400/200,!0/3000,!400/200,!0/3000,!400/200
dialrecall = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
record = 1400/425,0/15000
info = 400/750,0/100,400/750,0/100,400/750,0/100,400/750,0/400
stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
unobtainable = 400/75,0/100,400/75,0/100,400/75,0/100,400/75,0/400
[ph]
; reference http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
description = Philippines
ringcadence = 1000,4000
dial = 425
busy = 480+620/500,0/500
ring = 425+480/1000,0/4000
congestion = 480+620/250,0/250
callwaiting = 440/300,0/10000
; DIALRECALL - not specified
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
; RECORDTONE - not specified
record = 1400/500,0/15000
; INFO - not specified
info = !950/330,!1400/330,!1800/330,0
; STUTTER - not specified
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,!0/100,425
[pl]
description = Poland
ringcadence = 1000,4000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/500,0/500
callwaiting = 425/150,0/150,425/150,0/4000
; DIALRECALL - not specified
dialrecall = 425/500,0/50
; RECORDTONE - not specified
record = 1400/500,0/15000
; 950/1400/1800 3x0.33 on 1.0 off repeated 3 times
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000
; STUTTER - not specified
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
[pt]
description = Portugal
ringcadence = 1000,5000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/5000
congestion = 425/200,0/200
callwaiting = 440/300,0/10000
dialrecall = 425/1000,0/200
record = 1400/500,0/15000
info = 950/330,1400/330,1800/330,0/1000
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
[ru]
; References:
; http://www.minsvyaz.ru/site.shtml?id=1806
; http://www.aboutphone.info/lib/gost/45-223-2001.html
description = Russian Federation / ex Soviet Union
ringcadence = 1000,4000
dial = 425
busy = 425/350,0/350
ring = 425/1000,0/4000
congestion = 425/175,0/175
callwaiting = 425/200,0/5000
record = 1400/400,0/15000
info = 950/330,1400/330,1800/330,0/1000
dialrecall = 425/400,0/40
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
[se]
description = Sweden
ringcadence = 1000,5000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/5000
congestion = 425/250,0/750
callwaiting = 425/200,0/500,425/200,0/9100
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
record = 1400/500,0/15000
info = !950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,0
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
; stutter = 425/320,0/20 ; Real swedish standard, not used for now
[sg]
description = Singapore
; Singapore
; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf
; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz
ringcadence = 400,200,400,2000
dial = 425
ring = 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90%
busy = 425/750,0/750
congestion = 425/250,0/250
callwaiting = 425*24/300,0/200,425*24/300,0/3200
stutter = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425
info = 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference
dialrecall = 425*24/500,0/500,425/500,0/2500 ; unspecified in IDA reference, use repeating Holding Tone A,B
record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s
; additionally defined in reference
nutone = 425/2500,0/500
intrusion = 425/250,0/2000
warning = 425/624,0/4376 ; end of period tone, warning
acceptance = 425/125,0/125
holdinga = !425*24/500,!0/500 ; followed by holdingb
holdingb = !425/500,!0/2500
[th]
description = Thailand
ringcadence = 1000,4000
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
dial = 400*50
busy = 400/500,0/500
ring = 420/1000,0/5000
congestion = 400/300,0/300
callwaiting = 1000/400,10000/400,1000/400
; DIALRECALL - not specified - use special dial tone instead.
dialrecall = 400*50/400,0/100,400*50/400,0/100
; RECORDTONE - not specified
record = 1400/500,0/15000
; INFO - specified as an announcement - use special information tones instead
info = 950/330,1400/330,1800/330
; STUTTER - not specified
stutter = !400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,400
[uk]
description = United Kingdom
ringcadence = 400,200,400,2000
; These are the official tones taken from BT SIN350. The actual tones
; used by BT include some volume differences so sound slightly different
; from Asterisk-generated ones.
dial = 350+440
; Special dial is the intermittent dial tone heard when, for example,
; you have a divert active on the line
specialdial = 350+440/750,440/750
; Busy is also called "Engaged"
busy = 400/375,0/375
; "Congestion" is the Beep-bip engaged tone
congestion = 400/400,0/350,400/225,0/525
; "Special Congestion" is not used by BT very often if at all
specialcongestion = 400/200,1004/300
unobtainable = 400
ring = 400+450/400,0/200,400+450/400,0/2000
callwaiting = 400/100,0/4000
; BT seem to use "Special Call Waiting" rather than just "Call Waiting" tones
specialcallwaiting = 400/250,0/250,400/250,0/250,400/250,0/5000
; "Pips" used by BT on payphones. (Sounds wrong, but this is what BT claim it
; is and I've not used a payphone for years)
creditexpired = 400/125,0/125
; These two are used to confirm/reject service requests on exchanges that
; don't do voice announcements.
confirm = 1400
switching = 400/200,0/400,400/2000,0/400
; This is the three rising tones Doo-dah-dee "Special Information Tone",
; usually followed by the BT woman saying an appropriate message.
info = 950/330,0/15,1400/330,0/15,1800/330,0/1000
; Not listed in SIN350
record = 1400/500,0/60000
stutter = 350+440/750,440/750
[us]
description = United States / North America
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
[us-old]
description = United States Circa 1950/ North America
ringcadence = 2000,4000
dial = 600*120
busy = 500*100/500,0/500
ring = 420*40/2000,0/4000
congestion = 500*100/250,0/250
callwaiting = 440/300,0/10000
dialrecall = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0
stutter = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120
[tw]
description = Taiwan
; http://nemesis.lonestar.org/reference/telecom/signaling/dialtone.html
; http://nemesis.lonestar.org/reference/telecom/signaling/busy.html
; http://www.iproducts.com.tw/ee/kylink/06ky-1000a.htm
; http://www.pbx-manufacturer.com/ky120dx.htm
; http://www.nettwerked.net/tones.txt
; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/taiw_sup/taiw2.htm
;
; busy tone 480+620Hz 0.5 sec. on ,0.5 sec. off
; reorder tone 480+620Hz 0.25 sec. on,0.25 sec. off
; ringing tone 440+480Hz 1 sec. on ,2 sec. off
;
ringcadence = 1000,4000
dial = 350+440
busy = 480+620/500,0/500
ring = 440+480/1000,0/2000
congestion = 480+620/250,0/250
callwaiting = 350+440/250,0/250,350+440/250,0/3250
dialrecall = 300/1500,0/500
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
[ve]
; Tone definition source for ve found on
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
description = Venezuela / South America
ringcadence = 1000,4000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/250,0/250
callwaiting = 400+450/300,0/6000
dialrecall = 425
record = 1400/500,0/15000
info = !950/330,!1440/330,!1800/330,0/1000
; STUTTER - not specified
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
[za]
description = South Africa
; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/safr_sup/saf02.htm
; (definitions for other countries can also be found there)
; Note, though, that South Africa uses two switch types in their network --
; Alcatel switches -- mainly in the Western Cape, and Siemens elsewhere.
; The former use 383+417 in dial, ringback etc. The latter use 400*33
; I've provided both, uncomment the ones you prefer
ringcadence = 400,200,400,2000
; dial/ring/callwaiting for the Siemens switches:
dial = 400*33
ring = 400*33/400,0/200,400*33/400,0/2000
callwaiting = 400*33/250,0/250,400*33/250,0/250,400*33/250,0/250,400*33/250,0/250
; dial/ring/callwaiting for the Alcatel switches:
; dial = 383+417
; ring = 383+417/400,0/200,383+417/400,0/2000
; callwaiting = 383+417/250,0/250,383+417/250,0/250,383+417/250,0/250,383+417/250,0/250
congestion = 400/250,0/250
busy = 400/500,0/500
dialrecall = 350+440
; XXX Not sure about the RECORDTONE
record = 1400/500,0/10000
info = 950/330,1400/330,1800/330,0/330
stutter = !400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,400*33

166
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;
; Logging Configuration
;
; In this file, you configure logging to files or to
; the syslog system.
;
; "logger reload" at the CLI will reload configuration
; of the logging system.
[general]
;
; Customize the display of debug message time stamps
; this example is the ISO 8601 date format (yyyy-mm-dd HH:MM:SS)
;
; see strftime(3) Linux manual for format specifiers. Note that there is also
; a fractional second parameter which may be used in this field. Use %1q
; for tenths, %2q for hundredths, etc.
;
;dateformat=%F %T ; ISO 8601 date format
;dateformat=%F %T.%3q ; with milliseconds
;
;
; This makes Asterisk write callids to log messages
; (defaults to yes)
;use_callids = no
;
; This appends the hostname to the name of the log files.
;appendhostname = yes
;
; This determines whether or not we log queue events to a file
; (defaults to yes).
;queue_log = no
;
; Determines whether the queue_log always goes to a file, even
; when a realtime backend is present (defaults to no).
;queue_log_to_file = yes
;
; Set the queue_log filename
; (defaults to queue_log)
;queue_log_name = queue_log
;
; When using realtime for the queue log, use GMT for the timestamp
; instead of localtime. The default of this option is 'no'.
;queue_log_realtime_use_gmt = yes
;
; Log rotation strategy:
; none: Do not perform any logrotation at all. You should make
; very sure to set up some external logrotate mechanism
; as the asterisk logs can get very large, very quickly.
; sequential: Rename archived logs in order, such that the newest
; has the highest sequence number [default]. When
; exec_after_rotate is set, ${filename} will specify
; the new archived logfile.
; rotate: Rotate all the old files, such that the oldest has the
; highest sequence number [this is the expected behavior
; for Unix administrators]. When exec_after_rotate is
; set, ${filename} will specify the original root filename.
; timestamp: Rename the logfiles using a timestamp instead of a
; sequence number when "logger rotate" is executed.
; When exec_after_rotate is set, ${filename} will
; specify the new archived logfile.
;rotatestrategy = rotate
;
; Run a system command after rotating the files. This is mainly
; useful for rotatestrategy=rotate. The example allows the last
; two archive files to remain uncompressed, but after that point,
; they are compressed on disk.
;
; exec_after_rotate=gzip -9 ${filename}.2
;
;
; For each file, specify what to log.
;
; For console logging, you set options at start of
; Asterisk with -v for verbose and -d for debug
; See 'asterisk -h' for more information.
;
; Directory for log files is configures in asterisk.conf
; option astlogdir
;
; All log messages go to a queue serviced by a single thread
; which does all the IO. This setting controls how big that
; queue can get (and therefore how much memory is allocated)
; before new messages are discarded.
; The default is 1000
;logger_queue_limit = 250
;
;
[logfiles]
;
; Format is:
;
; logger_name => [formatter]levels
;
; The name of the logger dictates not only the name of the logging
; channel, but also its type. Valid types are:
; - 'console' - The root console of Asterisk
; - 'syslog' - Linux syslog, with facilities specified afterwards with
; a period delimiter, e.g., 'syslog.local0'
; - 'filename' - The name of the log file to create. This is the default
; for log channels.
;
; Filenames can either be relative to the standard Asterisk log directory
; (see 'astlogdir' in asterisk.conf), or absolute paths that begin with
; '/'.
;
; An optional formatter can be specified prior to the log levels sent
; to the log channel. The formatter is defined immediately preceeding the
; levels, and is enclosed in square brackets. Valid formatters are:
; - [default] - The default formatter, this outputs log messages using a
; human readable format.
; - [json] - Log the output in JSON. Note that JSON formatted log entries,
; if specified for a logger type of 'console', will be formatted
; per the 'default' formatter for log messages of type VERBOSE.
; This is due to the remote consoles intepreting verbosity
; outside of the logging subsystem.
;
; Log levels include the following, and are specified in a comma delineated
; list:
; debug
; notice
; warning
; error
; verbose(<level>)
; dtmf
; fax
; security
;
; Verbose takes an optional argument, in the form of an integer level.
; Verbose messages with higher levels will not be logged to the file. If
; the verbose level is not specified, it will log verbose messages following
; the current level of the root console.
;
; Special level name "*" means all levels, even dynamic levels registered
; by modules after the logger has been initialized (this means that loading
; and unloading modules that create/remove dynamic logger levels will result
; in these levels being included on filenames that have a level name of "*",
; without any need to perform a 'logger reload' or similar operation).
; Note that there is no value in specifying both "*" and specific level names
; for a filename; the "*" level means all levels. The only exception is if
; you need to specify a specific verbose level. e.g, "verbose(3),*".
;
; We highly recommend that you DO NOT turn on debug mode if you are simply
; running a production system. Debug mode turns on a LOT of extra messages,
; most of which you are unlikely to understand without an understanding of
; the underlying code. Do NOT report debug messages as code issues, unless
; you have a specific issue that you are attempting to debug. They are
; messages for just that -- debugging -- and do not rise to the level of
; something that merit your attention as an Asterisk administrator. Debug
; messages are also very verbose and can and do fill up logfiles quickly;
; this is another reason not to have debug mode on a production system unless
; you are in the process of debugging a specific issue.
;
;debug => debug
;security => security
console => notice,warning,error
;console => notice,warning,error,debug
messages => notice,warning,error
;full => notice,warning,error,debug,verbose,dtmf,fax
;
;full-json => [json]debug,verbose,notice,warning,error,dtmf,fax
;
;syslog keyword : This special keyword logs to syslog facility
;
;syslog.local0 => notice,warning,error
;

156
manager.conf Normal file
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;
; AMI - The Asterisk Manager Interface
;
; Third party application call management support and PBX event supervision
;
; Use the "manager show commands" at the CLI to list available manager commands
; and their authorization levels.
;
; "manager show command <command>" will show a help text.
;
; ---------------------------- SECURITY NOTE -------------------------------
; Note that you should not enable the AMI on a public IP address. If needed,
; block this TCP port with iptables (or another FW software) and reach it
; with IPsec, SSH, or SSL vpn tunnel. You can also make the manager
; interface available over http/https if Asterisk's http server is enabled in
; http.conf and if both "enabled" and "webenabled" are set to yes in
; this file. Both default to no. httptimeout provides the maximum
; timeout in seconds before a web based session is discarded. The
; default is 60 seconds.
;
[general]
enabled = no
;webenabled = yes
port = 5038
bindaddr = 0.0.0.0
; Parameters that control AMI over TLS. ("enabled" must be set too).
; You can open a connection to this socket with e.g.
;
; openssl s_client -connect my_host:5039
;
;tlsenable=no ; set to YES to enable it
;tlsbindaddr=0.0.0.0:5039 ; address and port to bind to, default to bindaddr and port 5039
;tlscertfile=/tmp/asterisk.pem ; path to the certificate.
;tlsprivatekey=/tmp/private.pem ; path to the private key, if no private given,
; if no tlsprivatekey is given, default is to search
; tlscertfile for private key.
;tlscipher=<cipher string> ; string specifying which SSL ciphers to use or not use
;
;allowmultiplelogin = yes ; IF set to no, rejects manager logins that are already in use.
; ; The default is yes.
;
;displayconnects = yes
;
; Add a Unix epoch timestamp to events (not action responses)
;
;timestampevents = yes
;brokeneventsaction = yes ; Restore previous behavior that caused the events
; action to not return a response in certain
; circumstances. Defaults to 'no'.
;
; Display certain channel variables every time a channel-oriented
; event is emitted:
;
;channelvars = var1,var2,var3
; debug = on ; enable some debugging info in AMI messages (default off).
; Also accessible through the "manager debug" CLI command.
; authtimeout specifies the maximum number of seconds a client has to
; authenticate. If the client does not authenticate beofre this timeout
; expires, the client will be disconnected. (default: 30 seconds)
;authtimeout = 30
; authlimit specifies the maximum number of unauthenticated sessions that will
; be allowed to connect at any given time.
;authlimit = 50
;httptimeout = 60
; a) httptimeout sets the Max-Age of the http cookie
; b) httptimeout is the amount of time the webserver waits
; on a action=waitevent request (actually its httptimeout-10)
; c) httptimeout is also the amount of time the webserver keeps
; a http session alive after completing a successful action
;[mark]
;secret = mysecret
;deny=0.0.0.0/0.0.0.0
;permit=209.16.236.73/255.255.255.0
;acl=named_acl_example ; use a named ACL from acl.conf
;
;
;setvar=PBXACCOUNT=edvina
; The setvar option defines channel variables that will be set when this account
; originates a call. You can define multiple setvar= commands for one manager
; user.
;
;eventfilter=Event: Newchannel
;eventfilter=Channel: (PJ)?SIP/(james|jim|john)-
;eventfilter=!Channel: DAHDI/
; The eventfilter option is used to whitelist or blacklist events per user.
; A filter consists of an (unanchored) regular expression that is run on the
; entire event data. If the first character of the filter is an exclamation
; mark (!), the filter is appended to the blacklist instead of the whitelist.
; After first checking the read access below, the regular expression filters
; are processed as follows:
; - If no filters are configured all events are reported as normal.
; - If there are white filters only: implied black all filter processed first,
; then white filters.
; - If there are black filters only: implied white all filter processed first,
; then black filters.
; - If there are both white and black filters: implied black all filter processed
; first, then white filters, and lastly black filters.
;
; If the device connected via this user accepts input slowly,
; the timeout for writes to it can be increased to keep it
; from being disconnected (value is in milliseconds)
;
; writetimeout = 100
;
;displayconnects = yes ; Display on CLI user login/logoff
;
; Authorization for various classes
;
; Read authorization permits you to receive asynchronous events, in general.
; Write authorization permits you to send commands and get back responses. The
; following classes exist:
;
; all - All event classes below (including any we may have missed).
; system - General information about the system and ability to run system
; management commands, such as Shutdown, Restart, and Reload. This
; class also includes dialplan manipulation actions such as
; DialplanExtensionAdd and DialplanExtensionRemove.
; call - Information about channels and ability to set information in a
; running channel.
; log - Logging information. Read-only. (Defined but not yet used.)
; verbose - Verbose information. Read-only. (Defined but not yet used.)
; agent - Information about queues and agents and ability to add queue
; members to a queue.
; user - Permission to send and receive UserEvent.
; config - Ability to read and write configuration files.
; command - Permission to run CLI commands. Write-only.
; dtmf - Receive DTMF events. Read-only.
; reporting - Ability to get information about the system.
; cdr - Output of cdr_manager, if loaded. Read-only.
; dialplan - Receive NewExten and VarSet events. Read-only.
; originate - Permission to originate new calls. Write-only.
; agi - Output AGI commands executed. Input AGI command to execute.
; cc - Call Completion events. Read-only.
; aoc - Permission to send Advice Of Charge messages and receive Advice
; - Of Charge events.
; test - Ability to read TestEvent notifications sent to the Asterisk Test
; Suite. Note that this is only enabled when the TEST_FRAMEWORK
; compiler flag is defined.
; security - Security Events. Read-only.
; message - Permissions to send out of call messages. Write-only
;
;read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan
;write = system,call,agent,user,config,command,reporting,originate,message

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minivm.conf Normal file
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;
; Mini-Voicemail Configuration
; for the MiniVM set of applications
;
; MiniVM consists of the following dialplan applications
; MinivmGreet Play personal prompts for busy/unavailable/temporary messages or default prompts
; MinivmRecord Record voice prompts to account directory or default directory
; MinivmNotify Notify via e-mail or pager - with or without attachment
; MinivmDelete Delete voice prompt (filename as argument or channel variable set by MinivmRecord)
;
; MiniVM works without accounts (just give e-mail address as argument) or with accounts in
; this configuration file or realtime. The idea is to build voicemail as building blocks so that
; a complete and adaptive voicemail system can be built in the dialplan
;
; ----------------------------- Variables to use in subject, from and message body ------------------
; Change the from, body and/or subject, variables:
; MVM_NAME, MVM_DUR, MVM_MSGNUM, VM_MAILBOX, MVM_CALLERID, MVM_CIDNUM,
; MVM_CIDNAME, MVM_DATE
;
; In addition to these, you can set the MVM_COUNTER channel variable in the
; dial plan and use that as a counter. It will also be used in the file name
; of the media file attached to the message
;
; Note: The emailbody config row can only be up to 512 characters due to a
; limitation in the Asterisk configuration subsystem.
; To create longer mails, use the templatefile option when creating the template
; ---------------------------------------------------------------------------------------------------
[general]
; Default format for storing and sending voicemail
; (only one format. Can also be set on a per-mailbox level)
format=wav49
;format=gsm
;
;Turn on logfile with the following syntax. One line per voicemail received
;with minivmRecord()
; Mailbox:domain:macrocontext:exten:priority:callerchan:callerid:origdate:origtime:duration:durationstatus:accountcode
;logfile=/var/log/asterisk/minivm.log
; Who the e-mail notification should appear to come from
serveremail=asterisk
;serveremail=asterisk@asterisk.example.com
; Minimum length of a voicemail message in seconds for the message to be kept
; The default is no minimum.
;minmessage=3
; How many seconds of silence before we end the recording
maxsilence=10
; Silence threshold (what we consider silence: the lower, the more sensitive)
silencethreshold=128
; How long greeting messages (busy/unavailable/temp/name) are allowed to be, in seconds
;maxgreet=120
; If you need to have an external program, i.e. /usr/bin/myapp called when a
; voicemail is received by the server. The arguments are
;
; <app> <username@domain> <callerid-name> <callerid-number>
;
;externnotify=/usr/bin/myapp
; The character set for voicemail messages can be specified here
;charset=ISO-8859-1
; Skip the "[PBX]:" string from the message title
;pbxskip=yes
; Change the From: string
; You can override the default program to send e-mail if you wish, too
; This is used both for e-mail and pager messages
;mailcmd=/usr/sbin/sendmail -t
;
; -------------Default e-mail message template (used if no templates are used) ------
;fromstring=The Asterisk PBX
;
;emailsubject=[PBX]: New message ${MVM_COUNER} in mailbox ${VM_MAILBOX}
; The following definition is very close to the default, but the default shows
; just the CIDNAME, if it is not null, otherwise just the CIDNUM, or "an unknown
; caller", if they are both null.
;emailbody=Dear ${MVM_NAME}:\n\n\tjust wanted to let you know you were just left a ${MVM_DUR} long message (number ${MVM_COUNTER})\nin mailbox ${MVM_MAILBOX} from ${MVM_CALLERID}, on ${MVM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n
;
; Set the date format on outgoing mails. Valid arguments can be found on the
; strftime(3) man page
;
; Default
emaildateformat=%A, %B %d, %Y at %r
; 24h date format
;emaildateformat=%A, %d %B %Y at %H:%M:%S
;
; -------------Default pager message template (used if no templates are used) ------
; You can also change the Pager From: string, the pager body and/or subject.
; The above defined variables also can be used here
;pagerfromstring=The Asterisk PBX
;pagersubject=New VM ${MVM_COUNTER}
;pagerbody=New ${MVM_DUR} long msg in box ${MVM_MAILBOX}\nfrom ${MVM_CALLERID}, on ${MVM_DATE}
;
;
; -------------Timezone definitions (used in voicemail accounts) -------------------
;
; Users may be located in different timezones, or may have different
; message announcements for their introductory message when they enter
; the voicemail system. Set the message and the timezone each user
; hears here. Set the user into one of these zones with the tz= attribute
; in the options field of the mailbox. Of course, language substitution
; still applies here so you may have several directory trees that have
; alternate language choices.
;
; Look in /usr/share/zoneinfo/ for names of timezones.
; Look at the manual page for strftime for a quick tutorial on how the
; variable substitution is done on the values below.
;
; Supported values:
; 'filename' filename of a soundfile (single ticks around the filename
; required)
; ${VAR} variable substitution
; A or a Day of week (Saturday, Sunday, ...)
; B or b or h Month name (January, February, ...)
; d or e numeric day of month (first, second, ..., thirty-first)
; Y Year
; I or l Hour, 12 hour clock
; H Hour, 24 hour clock (single digit hours preceded by "oh")
; k Hour, 24 hour clock (single digit hours NOT preceded by "oh")
; M Minute, with 00 pronounced as "o'clock"
; N Minute, with 00 pronounced as "hundred" (US military time)
; P or p AM or PM
; Q "today", "yesterday" or ABdY
; (*note: not standard strftime value)
; q "" (for today), "yesterday", weekday, or ABdY
; (*note: not standard strftime value)
; R 24 hour time, including minute
;
; The message here is not used in mini-voicemail, but stays for
; backwards compatibility
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
; ---------------------- Message body templates---------------------
; [template-name] ; "template-" is a verbatim marker
; fromaddress = Your Friendly Asterisk Server
; fromemail = asteriskvm@digium.com
; subject = <string>
; attachmedia = yes | no ; Add media file as attachment?
; dateformat = <formatstring> ; See above
; charset = <charset> ; Mime charset definition for e-mail messages
; locale = <locale> ; Locale for LC_TIME - to get weekdays in local language
; ; See your O/S documentation for proper settings for setlocale()
; templatefile = <filename> ; File name (relative to Asterisk configuration directory,
; or absolute
; messagebody = Format ; Message body definition with variables
;
[template-sv_SE_email]
messagebody=Hej ${MVM_NAME}:\n\n\tDu har f<>tt ett r<>stbrevl<76>de-meddelande fr<66>n ${MVM_CALLERID}.\nL<6E>ngd: ${MVM_DUR}\nMailbox ${MVM_MAILBOX}\nDatum: ${MVM_DATE}. \nMeddelandet bifogas det h<>r brevet. Om du inte kan l<>sa det, kontakta intern support. \nH<6E>lsningar\n\n\t\t\t\t--Asterisk\n
subject = Du har f<>tt r<>stmeddelande (se bilaga)
fromemail = swedish-voicemail-service@stockholm.example.com
fromaddress = Asterisk R<>stbrevl<76>da
charset=iso-8859-1
attachmedia=yes
dateformat=%A, %d %B %Y at %H:%M:%S
locale=sv_SE
[template-en_US_email]
messagebody=Dear ${MVM_NAME}:\n\n\tjust wanted to let you know you were just left a ${MVM_DUR} long message \nin mailbox ${MVM_MAILBOX} from ${MVM_CALLERID}, on ${MVM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n
subject = New voicemail
charset=ascii
attachmedia=yes
dateformat=%A, %B %d, %Y at %r
;[template-sv_SE_pager]
;templatefile = templates/pager_sv_se.txt
;subject = Du har f<>tt voicemail
;charset=iso-8859-1
;attachmedia=no
;locale=sv_SE
;[template-nb_NO_email]
;templatefile = templates/email_nb_NO.txt
;subject = Du har f<>tt voicemail
;charset=iso-8859-1
;locale=nb_NO
;[template-en_US_email_southern]
;templatefile = templates/email_en_US.txt
;subject = Y'all got voicemail, honey!
;charset=ascii
;[template-en_UK_email]
;templatefile = templates/email_en_us.txt
;subject = Dear old chap, you've got an electronic communique
;charset=ascii
; ---------------------- Mailbox accounts --------------------------
;Template for mailbox definition - all options
;
; [username@domain] ; Has to be unique within domain (MWM_USERNAME, MWM_DOMAIN)
; etemplate = sv_SE ; Email template from [templates]
; ptemplate = en_US ; Pager template from [templates]
; email = userpart@domain ; Extra e-mail address (overrides mailbox name)
; pager = pageremail@domain ; E-mail address for pager messages
; fullname = Mark Spencer ; Full name (MWM_NAME)
; options = ; E-mail options, se below
; accountcode = ; Account code (read in dialplan function MINIVMACCOUNT)
; pincode = ; Numeric pin code (read in dialplan function MINIVMACCOUNT)
; timezone=se ; Time zone
; serveremail = asterisk@digium.com ; Who to send email from (overrides template if set)
; externnotify = <application> ; External application for this account
; volgain = ; Volume gain setting (requires "sox")
; setvar=SERVICENAME=Voop.com Networks ; Extra variables to use in template
; Remember that you can use Asterisk Configuration Templates (ACT)
; [template@example.com](!) ; Declare template
; setvar=customerdomain=example.com
; setvar=customerclass=gold
; etemplate = sv_se_email
; serveremail = voicemail@example.com
; [user2@example.com](template@example.com) ; Declare user2 account using template
; fullname = Olle E. Johansson
; ; User inherits everything from template

44
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;
; Asterisk configuration file
;
; Module Loader configuration file
;
[modules]
autoload=yes
;
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger initialization) can be loaded
; using 'preload'. 'preload' forces a module and the modules it
; is known to depend upon to be loaded earlier than they normally get
; loaded.
;
; NOTE: There is no good reason left to use 'preload' anymore. It was
; historically required to preload realtime driver modules so you could
; map Asterisk core configuration files to Realtime storage.
; This is no longer needed.
;
;preload => your_special_module.so
;
; If you want Asterisk to fail if a module does not load, then use
; the "require" keyword. Asterisk will exit with a status code of 2
; if a required module does not load.
;
; require = chan_pjsip.so
;
; If you want you can combine with preload
; preload-require = your_special_module.so
;
;load => res_musiconhold.so
;
; Load one of: chan_oss, alsa, or console (portaudio).
; By default, load chan_oss only (automatically).
;
noload => chan_alsa.so
;noload => chan_oss.so
noload => chan_console.so
noload => res_hep.so
noload => res_hep_pjsip.so
noload => res_hep_rtcp.so
;

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musiconhold.conf Normal file
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;
; Music on Hold -- Sample Configuration
;
[general]
;cachertclasses=yes ; use 1 instance of moh class for all users who are using it,
; decrease consumable cpu cycles and memory
; disabled by default
;preferchannelclass=yes ; Prefer the musicclass as defined by CHANNEL(musicclass),
; over a class set by an application (e.g. a musicclass set on a queue).
; Defaults to yes.
; valid mode options:
; files -- read files from a directory in any Asterisk supported
; media format
; quietmp3 -- default
; mp3 -- loud
; mp3nb -- unbuffered
; quietmp3nb -- quiet unbuffered
; custom -- run a custom application (See examples below)
; =========
; File-based (native) music on hold
; =========
;
; This plays files directly from the specified directory, no external
; processes are required. Files are played in normal sorting order
; (same as a sorted directory listing), and no volume or other
; sound adjustments are available. If the file is available in
; the same format as the channel's codec, then it will be played
; without transcoding (same as Playback would do in the dialplan).
; Files can be present in as many formats as you wish, and the
; 'best' format will be chosen at playback time.
;
; The path specified can be either an absolute path (starts with '/'),
; or a relative path; relative paths are interpreted as being relative
; to the 'astdatalibdir' in asterisk.conf, which defaults to
; /var/lib/asterisk.
;
; NOTE:
; If you are not using "autoload" in modules.conf, then you
; must ensure that the format modules for any formats you wish
; to use are loaded _before_ res_musiconhold. If you do not do
; this, res_musiconhold will skip the files it is not able to
; understand when it loads.
;
[default]
mode=files
directory=moh
;
;[native-random]
;mode=files
;directory=moh
;digit=# ; If this option is set for a class, then when callers are
; ; listening to music on hold, they can press this digit, and
; ; they will switch to listening to this music class.
;announcement=queue-thankyou ;If this option is set for a class, then
; ; when callers get put on hold, the specified sound will be
; ; be played to them. Also, if using modes that Asterisk
; ; controls the playlist for (files, mp3, etc), the same
; ; sound will also be played between MOH songs.
;sort=random ; Sort the files in random order. Other sort options are:
; ; 'alpha' and 'randstart'. If 'alpha', the files are sorted
; ; in alphabetical order. If 'randstart', the files are sorted
; ; in alphabetical order as well, but the first file is chosen
; ; at random. If unspecified, the sort order is undefined.
;[native-alphabetical]
;mode=files
;directory=moh
;sort=alpha ; Sort the files in alphabetical order.
; =========
; Other (non-native) playback methods
; =========
;[manual]
;mode=custom
; Note that with mode=custom, a directory is not required, such as when reading
; from a stream.
;directory=/var/lib/asterisk/mohmp3
;application=/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s
;[ulawstream]
;mode=custom
;application=/usr/bin/streamplayer 192.168.100.52 888
;format=ulaw ; The 'format' option specifies the audio format that the
; ; 'application' will provide to Asterisk. In this example,
; ; streamplayer will output ulaw samples so we need to set the
; ; format to ulaw so that Asterisk knows how to interpret the
; ; incoming audio.
; mpg123 on Solaris does not always exit properly; madplay may be a better
; choice
;[solaris]
;mode=custom
;directory=/var/lib/asterisk/mohmp3
;application=/site/sw/bin/madplay -Q -o raw:- --mono -R 8000 -a -12
; By default, when res_musiconhold reloads or unloads, it sends a HUP signal
; to custom applications (and all descendants), waits 100ms, then sends a
; TERM signal, waits 100ms, then finally sends a KILL signal. An application
; which is interacting with an external device and/or spawns children of its
; own may not be able to exit cleanly in the default times, expecially if sent
; a KILL signal, or if it's children are getting signals directly from
; res_musiconhoild. To allow extra time, the 'kill_escalation_delay'
; class option can be used to set the number of milliseconds res_musiconhold
; waits before escalating kill signals, with the default being the current
; 100ms. To control to whom the signals are sent, the "kill_method"
; class option can be set to "process_group" (the default, existing behavior),
; which sends signals to the application and its descendants directly, or
; "process" which sends signals only to the application itself.
;[sox_from_device]
;mode=custom
;directory=/var/lib/asterisk/mohmp3
;application=/usr/bin/sox -q -t alsa -c 2 -r 48000 hw:1 -c 1 -r 8000 -t raw -s -
; Wait 500ms before escalating kill signals
;kill_escalation_delay=500
; Send signals to just the child process instead of all descendants
;kill_method=process

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[general]
; This section applies only to the default sip.conf/users.conf config provider
; embedded in res_phoneprov. Other providers may provide their own default settings.
; The default behavior of res_phoneprov will be to set the SERVER template variable to
; the IP address that the phone uses to contact the provisioning server and the
; SERVER_PORT variable to the bindport setting in sip.conf. Unless you have a very
; unusual setup, you should not need to set serveraddr, serveriface, or serverport.
;serveraddr=192.168.1.1 ; Override address to send to the phone to use as server address.
;serveriface=eth0 ; Same as above, except an ethernet interface.
; Useful for when the interface uses DHCP and the asterisk http
; server listens on a different IP than chan_sip.
;serverport=5060 ; Override port to send to the phone to use as server port.
default_profile=polycom ; The default profile to use if none specified in users.conf
; You can define profiles for different phones specifying what files to register
; with the provisioning server. You can define either static files, or dynamically
; generated files that can have dynamic names and point to templates that variables
; can be substituted into. You can also set arbitrary variables for the profiles
; templates to have access to. Profiles are shared across all config providers.
; Example:
;[example]
;mime_type => application/octet-stream
;static_file => example/firmware
;static_file => example/default.cfg,text/xml
;${TOUPPER(${MAC})}.cfg => templates/example-mac.cfg
;setvar => DB_CIDNAME=${ODBC_CID_NAME_LOOKUP(${USERNAME})}
; Dynamically generated files have a filename registered with variable substitution
; with variables obtained from various config providers. The default provider
; embedded in res_phoneprov reads users.conf. Other providers will have their own
; sources for the variables and may provide additional variables not listed here.
; Built in variables and the options in users.conf that they come from
; MAC (macaddress)
; USERNAME (username)
; DISPLAY_NAME (fullname)
; SECRET (secret)
; LABEL (label)
; CALLERID (cid_number)
; VOCIEMAIL_EXTEN (vmexten)
; EXTENSION_LENGTH (localextenlength)
; LINE
; LINEKEYS
; Built-in variables and the options in phoneprov.conf that they come from
; SERVER (server)
; SERVER_PORT (serverport)
; Built-in variables for managing timezones and daylight savings time.
; TZOFFSET
; DST_ENABLE
; DST_START_MONTH
; DST_START_MDAY
; DST_START_HOUR
; DST_END_MONTH
; DST_END_MDAY
; DST_END_HOUR
; TIMEZONE
[polycom]
staticdir => configs/ ; Sub directory of AST_DATA_DIR/phoneprov that static files reside
; in. This allows a request to /phoneprov/sip.cfg to pull the file
; from /phoneprov/configs/sip.cfg
mime_type => text/xml ; Default mime type to use if one isn't specified or the
; extension isn't recognized
static_file => bootrom.ld,application/octet-stream ; Static files the phone will download
static_file => bootrom.ver,plain/text ; static_file => filename,mime-type
static_file => sip.ld,application/octet-stream
static_file => sip.ver,plain/text
static_file => sip.cfg
static_file => custom.cfg
static_file => 2201-06642-001.bootrom.ld,application/octet-stream
static_file => 2201-06642-001.sip.ld,application/octet-stream
static_file => 2345-11000-001.bootrom.ld,application/octet-stream
static_file => 2345-11300-001.bootrom.ld,application/octet-stream
static_file => 2345-11300-010.bootrom.ld,application/octet-stream
static_file => 2345-11300-010.sip.ld,application/octet-stream
static_file => 2345-11402-001.bootrom.ld,application/octet-stream
static_file => 2345-11402-001.sip.ld,application/octet-stream
static_file => 2345-11500-001.bootrom.ld,application/octet-stream
static_file => 2345-11500-010.bootrom.ld,application/octet-stream
static_file => 2345-11500-020.bootrom.ld,application/octet-stream
static_file => 2345-11500-030.bootrom.ld,application/octet-stream
static_file => 2345-11500-030.sip.ld,application/octet-stream
static_file => 2345-11500-040.bootrom.ld,application/octet-stream
static_file => 2345-11500-040.sip.ld,application/octet-stream
static_file => 2345-11600-001.bootrom.ld,application/octet-stream
static_file => 2345-11600-001.sip.ld,application/octet-stream
static_file => 2345-11605-001.bootrom.ld,application/octet-stream
static_file => 2345-11605-001.sip.ld,application/octet-stream
static_file => 2345-12200-001.bootrom.ld,application/octet-stream
static_file => 2345-12200-001.sip.ld,application/octet-stream
static_file => 2345-12200-002.bootrom.ld,application/octet-stream
static_file => 2345-12200-002.sip.ld,application/octet-stream
static_file => 2345-12200-004.bootrom.ld,application/octet-stream
static_file => 2345-12200-004.sip.ld,application/octet-stream
static_file => 2345-12200-005.bootrom.ld,application/octet-stream
static_file => 2345-12200-005.sip.ld,application/octet-stream
static_file => 2345-12365-001.bootrom.ld,application/octet-stream
static_file => 2345-12365-001.sip.ld,application/octet-stream
static_file => 2345-12500-001.bootrom.ld,application/octet-stream
static_file => 2345-12500-001.sip.ld,application/octet-stream
static_file => 2345-12560-001.bootrom.ld,application/octet-stream
static_file => 2345-12560-001.sip.ld,application/octet-stream
static_file => 2345-12600-001.bootrom.ld,application/octet-stream
static_file => 2345-12600-001.sip.ld,application/octet-stream
static_file => 2345-12670-001.bootrom.ld,application/octet-stream
static_file => 2345-12670-001.sip.ld,application/octet-stream
static_file => 3111-15600-001.bootrom.ld,application/octet-stream
static_file => 3111-15600-001.sip.ld,application/octet-stream
static_file => 3111-40000-001.bootrom.ld,application/octet-stream
static_file => 3111-40000-001.sip.ld,application/octet-stream
static_file => SoundPointIPWelcome.wav,application/octet-stream
static_file => SoundPointIPLocalization/Japanese_Japan/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Norwegian_Norway/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Spanish_Spain/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Portuguese_Portugal/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/English_United_Kingdom/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/English_United_States/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Russian_Russia/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Italian_Italy/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Chinese_China/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Swedish_Sweden/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/English_Canada/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/German_Germany/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/French_France/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Danish_Denmark/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Dutch_Netherlands/SoundPointIP-dictionary.xml
static_file => SoundPointIPLocalization/Korean_Korea/SoundPointIP-dictionary.xml
${MAC}.cfg => 000000000000.cfg ; Dynamically generated files.
${MAC}-phone.cfg => 000000000000-phone.cfg ; (relative to AST_DATA_DIR/phoneprov)
config/${MAC} => polycom.xml ; Dynamic Filename => template file
${MAC}-directory.xml => 000000000000-directory.xml
setvar => CUSTOM_CONFIG=/var/lib/asterisk/phoneprov/configs/custom.cfg ; Custom variable
;snom 300, 320, 360, 370, 820, 821, 870 support
snom-${MAC}.xml => snom-mac.xml

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; Common pjproject options
;
;[startup]
; NOTES: The name of this section in the pjproject.conf configuration file must
; remain startup or the configuration will not be applied.
;
;cache_pools = yes ; Cache pjproject memory pools for performance
; Disable this option to help track down pool content
; mismanagement when using valgrind or MALLOC_DEBUG.
; The cache gets in the way of determining if the
; pool contents are used after being freed and who
; freed it.
; Default yes
;log_level=default ; Initial maximum pjproject logging level to log
; Valid values are: 0-6, and default
;
; Note: This option is needed very early in the startup
; process so it can only be read from config files because
; the modules for other methods have not been loaded yet.
;type= ; Must be of type startup (default: "")
;========================LOG_MAPPINGS SECTION OPTIONS===============================
;[log_mappings]
; SYNOPSIS: Provides pjproject to Asterisk log level mappings.
; NOTES: The name of this section in the pjproject.conf configuration file must
; remain log_mappings or the configuration will not be applied.
; The defaults mentioned below only apply if this file or the 'log_mappings'
; object can't be found. If the object is found, there are no defaults. If
; you don't specify an entry, nothing will be logged for that level.
;
; These logging level meanings are typically used by pjproject:
; - 0: fatal error
; - 1: error
; - 2: warning
; - 3: info
; - 4: debug
; - 5: trace
; - 6: more detailed trace
;
;asterisk_error = ; A comma separated list of pjproject log levels to map to
; Asterisk errors.
; (default: "0,1")
;asterisk_warning = ; A comma separated list of pjproject log levels to map to
; Asterisk warnings.
; (default: "2")
;asterisk_notice = ; A comma separated list of pjproject log levels to map to
; Asterisk notices.
; (default: "")
;asterisk_verbose = ; A comma separated list of pjproject log levels to map to
; Asterisk verbose.
; (default: "")
;asterisk_debug = ; A comma separated list of pjproject log levels to map to
; Asterisk debug
; (default: "3,4,5,6")
;type= ; Must be of type log_mappings (default: "")

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pjsip_notify.conf Normal file
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; 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
; Digium
[digium-check-cfg]
Event=>check-sync
; 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=>check-sync\;reboot=true
; Cisco
[cisco-check-cfg]
Event=>check-sync

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; PJSIP Wizard Configuration Samples and Quick Reference
;
; This file has several very basic configuration examples, to serve as a quick
; reference to jog your memory when you need to write up a new configuration.
; It is not intended to teach PJSIP configuration or serve as an exhaustive
; reference of options and potential scenarios.
;
; This file has two main sections.
; First, manually written examples to serve as a handy reference.
; Second, a list of all possible PJSIP config options by section. This is
; pulled from the XML config help. It only shows the synopsis for every item.
; If you want to see more detail please check the documentation sources
; mentioned at the top of this file.
; ============================================================================
; NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
;
; This file does not maintain the complete option documentation.
; ============================================================================
; Documentation
;
; The official documentation is at http://wiki.asterisk.org
; You can read the XML configuration help via Asterisk command line with
; "config show help res_pjsip_config_wizard", then you can drill down through
; the various sections and their options.
;
;============EXAMPLE WIZARD CONFIGURATION FOR A PHONE=======================
; This config would create an endpoint, aor with dynamic contact, inbound
; auth, a phoneprov object and a dialplan hint for extension 1000.
;[myphone]
;type = wizard
;accepts_auth = yes
;accepts_registrations = yes
;has_phoneprov = yes
;transport = ipv4
;has_hint = yes
;hint_exten = 1000
;inbound_auth/username = testname
;inbound_auth/password = test password
;endpoint/allow = ulaw
;endpoint/context = default
;phoneprov/MAC = 001122aa4455
;phoneprov/PROFILE = profile1
;============EXAMPLE WIZARD CONFIGURATION FOR AN ITSP TRUNK=================
; This ITSP has 2 servers available and requires registration.
; This config would create an endpoint, an aor with 2 static contacts, an
; outbound auth, an identify with 2 matches, and 2 registrations.
;[mytrunk]
;type = wizard
;sends_auth = yes
;sends_registrations = yes
;transport = ipv4
;remote_hosts = sip1.myitsp.com:5060,sip2.myitsp.com:5060
;outbound_auth/username = testname
;outbound_auth/password = test password
;endpoint/allow = ulaw
;endpoint/context = default
;========================WIZARD SECTION OPTIONS===============================
;[wizard]
; SYNOPSIS: Provides configuration wizard for common scenarios.
;sends_auth= ; Will create an outbound auth object for the endpoint and
; registration.
; If yes, outbound/username must be specified.
; (default = "no")
;accepts_auth= ; Will create an inbound auth object for the endpoint.
; If yes, inbound/username must be specified.
; (default = "no")
;sends_registrations= ; Will create an outbound registration object and an
; identify match for each host in remote_hosts (which
; must be specified).
; sends_auth must also be specified.
; (default: "no")
;sends_line_with_registrations= ; Setting this to true will cause the wizard to
; skip the creation of an identify object to match
; incoming requests to the endpoint and instead add the
; line and endpoint parameters to the outbound
; registration object.
; (default: "no")
;accepts_registrations= ; Will create an aor with dynamic contacts which will
; accept registrations.
; accepts_auth must also be specified.
; (default: "no")
;remote_hosts= ; A comma separated list of remote hosts in the form of
; <ipaddress | hostname>[:port] [,<ipaddress | hostname>[:port] ] ...
; If specified, a static contact for each host will be created
; in the aor. If accepts_registrations is no, an identify
; object is also created with a match line for each remote host.
; If an aor/contact or match/identify is explicitly supplied,
; remote_hosts will not be used to automatically create contacts
; or matches respectively.
; (default: "")
;outbound_proxy= ; Setting this is a shortcut for setting
; endpoint/outbound_proxy
; aor/outbound_proxy
; registration/outbound_proxy
;transport= ; The transport to use for the endpoint and registrations
; (default: the pjsip default)
;server_uri_pattern= ; The pattern used to construct the registration
; server_uri. The replaceable parameter ${REMOTE_HOST} isa
; available for use.
; (default: "sip:${REMOTE_HOST}")
;client_uri_pattern= ; The pattern used to construct the registration client_uri.
; The replaceable parameters ${REMOTE_HOST} and ${USERNAME}
; are available for use.
; (default: "sip:${USERNAME}@${REMOTE_HOST}")
;contact_pattern= ; The pattern used to construct the aor contact.
; The replaceable parameter ${REMOTE_HOST} is available
; for use.
; (default: "sip:${REMOTE_HOST}")
;has_phoneprov= ; Will create a phoneprov object.
; If yes, phoneprov/MAC must be specified.
; (default: "no")
;has_hint= ; Create hint and optionally a default application.
; (default: "no")
;hint_context ; Any hints created for this wizard will be placed in this
; context.
; (default: endpoint/context)
;hint_exten ; If specified, a PJSIP/<wizard_id> hint will be created
; for this extension in 'hint_context'.
; context.
; (default: none)
;hint_application ; If specified, an extension will be placed in 'hint_context'
; at priority 1 that calls this application. Could be any
; valid dialplan expression like
; "Gosub(stdexten,${EXTEN},1(${HINT}))"
; (default: "Dial(${HINT})")
;endpoint/<param> ; Any parameters to be passed directly to and validated
;aor/<param> ; by their respective objects.
;inbound_auth/<param>
;outbound_auth/<param>
;identify/<param>
;registration/<param>
;phoneprov/<param>
;type= ; Must be of type wizard (default: "")

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[general]
; Look for queue rules in the queue_rules database table through RealTime. Note
; that this option is not strictly "RealTime", in the sense that the queue
; rules are only loaded and parsed during module load/reload. Queue rules
; must have a unique rule name and support relative min/max penalties.
;
; realtime_rules = yes
;
; It is possible to change the value of the QUEUE_MAX_PENALTY, QUEUE_MIN_PENALTY and QUEUE_RAISE_PENALTY
; channel variables in mid-call by defining rules in the queue for when to do so. This can allow for
; a call to be opened to more members or potentially a different set of members.
; The advantage to changing members this way as opposed to inserting the caller into a
; different queue with more members or reinserting the caller into the same queue with a different
; QUEUE_MAX_PENALTY or QUEUE_MIN_PENALTY set is that the caller does not lose his place in the queue.
;
; QUEUE_MAX_PENALTY, QUEUE_MIN_PENALTY and QUEUE_RAISE_PENALTY only apply to a queue call, and are only
; modified by these rules if they are initially set in the dialplan.
;
; If QUEUE_MIN_PENALTY is set, agents with a lower penalty value will not be considered for the caller.
; If QUEUE_MAX_PENALTY is set, agents with a higher penalty value will not be considered for the caller.
; If QUEUE_RAISE_PENALTY is set, agents with a lower penalty will be treated as having a penalty = QUEUE_RAISE_PENALTY.
;
; QUEUE_RAISE_PENALTY example:
; - Agent 1 has penalty 1
; - Agent 2 has penalty 2
; - the queue rule is set to:
; penaltychange => 30,,,2
;
; Prior to the 30 second mark, Agent 1 will take priority over Agent 2 for call distribution.
; After 30 seconds, Agent 1's priority is bumped to 2 by the penaltychange, so both agents are treated equally.
;
; Note: There is a limitation to these rules; a caller will follow the penaltychange rules for
; the queue that were defined at the time the caller entered the queue. If an update to the rules is
; made during the caller's stay in the queue, these will not be reflected for that caller.
;
; The syntax for these rules is
; penaltychange => <number of seconds into the call>,<absolute or relative change to QUEUE_MAX_PENALTY>[,absolute or relative change to QUEUE_MIN_PENALTY][,absolute or relative change to QUEUE_RAISE_PENALTY]
;
; Example:
; [myrule]
; penaltychange => 30,+3 ; 30 seconds into the call increase the QUEUE_MAX_PENALTY by 3, no change to QUEUE_MIN_PENALTY
; penaltychange => 60,10,5 ; 60 seconds into the call increase the QUEUE_MAX_PENALTY to 10 and increase the QUEUE_MIN_PENALTY to 5
; penaltychange => 75,,7 ; 75 seconds into the call keep the QUEUE_MAX_PENALTY the same and increase the QUEUE_MIN_PENALTY to 7
; penaltychange => 90,,,20 ; 90 seconds into the call leave QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY untouched and set QUEUE_RAISE_PENALTY to 20

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[general]
;
; Global settings for call queues
;
; Persistent Members
; Store each dynamic member in each queue in the astdb so that
; when asterisk is restarted, each member will be automatically
; read into their recorded queues. Default is 'no'.
;
persistentmembers = yes
;
; AutoFill Behavior
; The old behavior of the queue (autofill=no) is to have a serial type behavior
; in that the queue will make all waiting callers wait in the queue
; even if there is more than one available member ready to take
; calls until the head caller is connected with the member they
; were trying to get to. The next waiting caller in line then
; becomes the head caller, and they are then connected with the
; next available member and all available members and waiting callers
; waits while this happens. The new behavior, enabled by setting
; autofill=yes makes sure that when the waiting callers are connecting
; with available members in a parallel fashion until there are
; no more available members or no more waiting callers. This is
; probably more along the lines of how a queue should work and
; in most cases, you will want to enable this behavior. If you
; do not specify or comment out this option, it will default to yes.
;
;autofill = no
;
; Monitor Type
; By setting monitor-type = MixMonitor, when specifying monitor-format
; to enable recording of queue member conversations, app_queue will
; now use the new MixMonitor application instead of Monitor so
; the concept of "joining/mixing" the in/out files now goes away
; when this is enabled. You can set the default type for all queues
; here, and then also change monitor-type for individual queues within
; queue by using the same configuration parameter within a queue
; configuration block. If you do not specify or comment out this option,
; it will default to the old 'Monitor' behavior to keep backward
; compatibility.
;
monitor-type = MixMonitor
;
; UpdateCDR behavior.
; This option is implemented to mimic chan_agents behavior of populating
; CDR dstchannel field of a call with an agent name, which you can set
; at the login time with AddQueueMember membername parameter.
;
; updatecdr = no
;
; Note that a timeout to fail out of a queue may be passed as part of
; an application call from extensions.conf:
; Queue(queuename,[options],[optionalurl],[announceoverride],[timeout])
; example: Queue(dave,t,,,45)
; shared_lastcall will make the lastcall and calls received be the same in
; members logged in more than one queue. This is useful to make the queue
; respect the wrapuptime of another queue for a shared member.
; The default value is no.
;
;shared_lastcall=no
;
; Negative_penalty_invalid will treat members with a negative penalty as logged off
;
;negative_penalty_invalid = no
;
; log_membername_as_agent will cause app_queue to log the membername rather than
; the interface for the ADDMEMBER and REMOVEMEMBER events when a state_interface
; is set. The default value (no) maintains backward compatibility.
;
;log_membername_as_agent = no
;
;[markq]
;
; A sample call queue
;
; Musicclass sets which music applies for this particular call queue.
; The only class which can override this one is if the MOH class is set
; directly on the channel using Set(CHANNEL(musicclass)=whatever) in the
; dialplan.
;
;musicclass = default
;
; An announcement may be specified which is played for the member as
; soon as they answer a call, typically to indicate to them which queue
; this call should be answered as, so that agents or members who are
; listening to more than one queue can differentiated how they should
; engage the customer
;
;announce = queue-markq
;
; A strategy may be specified. Valid strategies include:
;
; ringall - ring all available channels until one answers (default)
; leastrecent - ring interface which was least recently hung up by this queue
; fewestcalls - ring the one with fewest completed calls from this queue
; random - ring random interface
; rrmemory - round robin with memory, remember where we left off last ring pass
; rrordered - same as rrmemory, except the queue member order from config file
; is preserved
; linear - rings interfaces in the order specified in this configuration file.
; If you use dynamic members, the members will be rung in the order in
; which they were added
; wrandom - rings random interface, but uses the member's penalty as a weight
; when calculating their metric. So a member with penalty 0 will have
; a metric somewhere between 0 and 1000, and a member with penalty 1 will
; have a metric between 0 and 2000, and a member with penalty 2 will have
; a metric between 0 and 3000. Please note, if using this strategy, the member
; penalty is not the same as when using other queue strategies. It is ONLY used
; as a weight for calculating metric.
;
;strategy = ringall
;
; Second settings for service level (default 0)
; Used for service level statistics (calls answered within service level time frame)
; There are 2 Service Level Performance Calculations SL and SL2
; SL = (total calls answered within threshold / total calls answered) * 100
; SL2 = ((calls answered within threshold + calls abandoned within threshold) / (answered calls + abandoned calls)) * 100
;servicelevel = 60
;
; A context may be specified, in which if the user types a SINGLE
; digit extension while they are in the queue, they will be taken out
; of the queue and sent to that extension in this context.
;
;context = qoutcon
;
; A limit can be set to disregard penalty settings when the queue has
; too few members. No penalty will be weighed in if there are only X
; or fewer queue members. (default 0)
;
;penaltymemberslimit = 5
;
; ---------------------QUEUE TIMING OPTIONS------------------------------------
; A Queue has two different "timeout" values associated with it. One is the
; timeout parameter configured in queues.conf. This timeout specifies the
; amount of time to try ringing a member's phone before considering the
; member to be unavailable. The other timeout value is the second argument
; to the Queue() application. This timeout represents the absolute amount
; of time to allow a caller to stay in the queue before the caller is
; removed from the queue. In certain situations, these two timeout values
; may clash. For instance, if the timeout in queues.conf is set to 5 seconds,
; the retry value in queues.conf is set to 4, and the second argument to Queue()
; is 10, then the following may occur:
;
; A caller places a call to a queue.
; The queue selects a member and attempts to ring that member.
; The member's phone is rung for 5 seconds and he does not answer.
; The retry time of 4 seconds occurs.
; The queue selects a second member to call.
;
; How long does that second member's phone ring? Does it ring for 5 seconds
; since the timeout set in app_queue is 5 seconds? Does it ring for 1 second since
; the caller has been in the queue for 9 seconds and is supposed to be removed after
; being in the queue for 10 seconds? This is configurable with the timeoutpriority
; option. By setting the timeoutpriority to "conf" then you are saying that you would
; rather use the time specified in the configuration file even if it means having the
; caller stay in the queue longer than the time specified in the application argument.
; For the scenario described above, timeoutpriority=conf would result in the second
; member's phone ringing for 5 seconds. By specifying "app" as the value for
; timeoutpriority, you are saying that the timeout specified as the argument to the
; Queue application is more important. In the scenario above, timeoutpriority=app
; would result in the second member's phone ringing for 1 second.
;
; There are a few exceptions to the priority rules. For instance, if timeoutpriority=appp
; and the configuration file timeout is set to 0, but the application argument timeout is
; non-zero, then the timeoutpriority is ignored and the application argument is used as
; the timeout. Furthermore, if no application argument timeout is specified, then the
; timeoutpriority option is ignored and the configuration file timeout is always used
; when calling queue members.
;
; In timeoutpriority=conf mode however timeout specified in config file will take higher
; priority than timeout in application arguments, so if config file has timeout 0, each
; queue member will be called indefineately and application timeout will be checked only
; after this call attempt. This is useful for having queue members with custom timeouts
; specified within Dial application of Local channel, and allows handling NO ANSWER which
; would otherwise be interrupted by queue destroying child channel on timeout.
;
; The default value for timeoutpriority is "app" since this was how previous versions of
; Asterisk behaved.
;
;timeout = 15
;retry = 5
;timeoutpriority = app|conf
;
; ----------------------END QUEUE TIMING OPTIONS---------------------------------
; Weight of queue - when compared to other queues, higher weights get
; first shot at available channels when the same channel is included in
; more than one queue.
;
;weight=0
;
; After a successful call, how long to wait before sending a potentially
; free member another call (default is 0, or no delay)
;
;wrapuptime=15
;
; Autofill will follow queue strategy but push multiple calls through
; at same time until there are no more waiting callers or no more
; available members. The per-queue setting of autofill allows you
; to override the default setting on an individual queue level.
;
;autofill=yes
;
; Autopause will pause a queue member if they fail to answer a call
; no: Member will not be paused
; yes: Member will be paused only in the queue where the timeout took place
; all: Memeber will be paused in all queues he/she is a member
;autopause=yes
;
; Autopausedelay delay autopause for autopausedelay seconds from the
; last call if a member has not taken a call the delay has no effect.
;autopausedelay=60
;
; Autopausebusy controls whether or not a queue member is set as paused
; automatically upon the member device reporting busy. The autopausedelay
; option applies. Defaults to 'no'.
;autopausebusy=no
;
; Autopauseunavail controls whether or not a queue member is set as paused
; automatically upon the member device reporting congestion. The autopausedely
; option applies. Defaults to 'no'.
;autopauseunavail=no
;
; Maximum number of people waiting in the queue (0 for unlimited)
;
;maxlen = 0
;
; Note: for below queue channel options (setinterfacevar, setqueueentryvar,
; setqueuevar) if the caller channel is a local channel and optimizations
; is enabled then after optimization has occurred only the queue member
; channel will contain the variables.
;
; If set to yes, just prior to the caller being bridged with a queue member
; the following variables will be set on the caller and queue member channels:
; MEMBERINTERFACE is the interface name (eg. Agent/1234)
; MEMBERNAME is the member name (eg. Joe Soap)
; MEMBERCALLS is the number of calls that interface has taken,
; MEMBERLASTCALL is the last time the member took a call.
; MEMBERPENALTY is the penalty of the member
; MEMBERDYNAMIC indicates if a member is dynamic or not
; MEMBERREALTIME indicates if a member is realtime or not
;
;setinterfacevar=no
;
; If set to yes, just prior to the caller being bridged with a queue member
; the following variables will be set on the caller and queue member channels:
; QEHOLDTIME callers hold time
; QEORIGINALPOS original position of the caller in the queue
;
;setqueueentryvar=no
;
; If set to yes, the following variables will be set
; just prior to the caller being bridged with a queue member (set on the
; caller and queue member channels) and just prior to the caller
; leaving the queue
; QUEUENAME name of the queue
; QUEUEMAX maxmimum number of calls allowed
; QUEUESTRATEGY the strategy of the queue;
; QUEUECALLS number of calls currently in the queue
; QUEUEHOLDTIME current average hold time
; QUEUECOMPLETED number of completed calls for the queue
; QUEUEABANDONED number of abandoned calls
; QUEUESRVLEVEL queue service level
; QUEUESRVLEVELPERF current service level performance
;
;setqueuevar=no
; if set, run this macro when connected to the queue member
; you can override this macro by setting the macro option on
; the queue application
;
;membermacro=macro_name[,arg1[,...][,argN]]
; if set, run this gosub when connected to the queue member
; you can override this gosub by setting the gosub option on
; the queue application
;
;membergosub=gosub_context_name[,arg1[,...][,argN]]
; How often to announce queue position and/or estimated
; holdtime to caller (0=off)
; Note that this value is ignored if the caller's queue
; position has changed (see min-announce-frequency)
;
;announce-frequency = 90
;
; The absolute minimum time between the start of each
; queue position and/or estimated holdtime announcement
; This is useful for avoiding constant announcements
; when the caller's queue position is changing frequently
; (see announce-frequency)
;
;min-announce-frequency = 15
;
; How often to make any periodic announcement (see periodic-announce)
;
;periodic-announce-frequency=60
;
; Should the periodic announcements be played in a random order? Default is no.
;
;random-periodic-announce=no
;
; If set to yes, the periodic announcement frequency will be timed from the end
; of each announcement rather than from the start of each announcement. This
; defaults to off.
;
;relative-periodic-announce=yes
;
; Should we include estimated hold time in position announcements?
; Either yes, no, or only once.
; Hold time will be announced as the estimated time.
;
;announce-holdtime = yes|no|once
;
; Queue position announce?
; Valid values are "yes," "no," "limit," or "more." If set to "no," then the caller's position will
; never be announced. If "yes," then the caller's position in the queue will be announced
; to the caller. If set to "more," then if the number of callers is more than the number
; specified by the announce-position-limit option, then the caller will hear that there
; are more than that many callers waiting (i.e. if a caller number 6 is in a queue with the
; announce-position-limit set to 5, then that caller will hear that there are more than 5
; callers waiting). If set to "limit," then only callers within the limit specified by announce-position-limit
; will have their position announced.
;
;announce-position = yes
;
; If enabled, play announcements to the first user waiting in the Queue. This may mean
; that announcements are played when an agent attempts to connect to the waiting user,
; which may delay the time before the agent and the user can communicate. Disabled by
; default.
;
; announce-to-first-user = no
;
; If you have specified "limit" or "more" for the announce-position option, then the following
; value is what is used to determine what announcement to play to waiting callers. If you have
; set the announce-position option to anything else, then this will have no bearing on queue operation
;
;announce-position-limit = 5
;
; What's the rounding time for the seconds?
; If this is non-zero, then we announce the seconds as well as the minutes
; rounded to this value.
; Valid values are 0, 5, 10, 15, 20, and 30.
;
; announce-round-seconds = 10
;
; Only announce the caller's position if it has improved since the last announcement.
; The default value is no.
;
; announce-position-only-up = yes
;
; Use these sound files in making position/holdtime announcements. The
; defaults are as listed below -- change only if you need to.
;
; Keep in mind that you may also prevent a sound from being played if you
; explicitly set a sound to be an empty string. For example, if you want to
; prevent the queue from playing queue-thankyou, you may set the sound using
; the following line:
;
; queue-thankyou=
;
; ("You are now first in line.")
;queue-youarenext = queue-youarenext
; ("There are")
;queue-thereare = queue-thereare
; ("calls waiting.")
;queue-callswaiting = queue-callswaiting
; ("The current est. holdtime is")
;queue-holdtime = queue-holdtime
; ("minute.")
;queue-minute = queue-minute
; ("minutes.")
;queue-minutes = queue-minutes
; ("seconds.")
;queue-seconds = queue-seconds
; ("Thank you for your patience.")
;queue-thankyou = queue-thankyou
; ("Hold time")
;queue-reporthold = queue-reporthold
; ("All reps busy / wait for next")
;periodic-announce = queue-periodic-announce
;
; A set of periodic announcements can be defined by separating
; periodic announcements to reproduce by commas. For example:
;periodic-announce = queue-periodic-announce,your-call-is-important,please-wait
;
; The announcements will be played in the order in which they are defined. After
; playing the last announcement, the announcements begin again from the beginning.
;
; Calls may be recorded using Asterisk's monitor/MixMonitor resource
; This can be enabled from within the Queue application, starting recording
; when the call is actually picked up; thus, only successful calls are
; recorded, and you are not recording while people are listening to MOH.
; To enable monitoring, simply specify "monitor-format"; it will be disabled
; otherwise.
;
; You can specify the monitor filename with by calling
; Set(MONITOR_FILENAME=foo)
; Otherwise it will use MONITOR_FILENAME=${UNIQUEID}
;
; Pick any one valid extension for monitor format recording. If you leave
; monitor-format commented out, it will not record calls.
;
; monitor-format = gsm|wav|wav49
;
; Monitor Type
; By setting monitor-type = MixMonitor, when specifying monitor-format
; to enable recording of queue member conversations, app_queue will
; now use the new MixMonitor application instead of Monitor so
; the concept of "joining/mixing" the in/out files now goes away
; when this is enabled. If you do not specify or comment out this option,
; it will default to the old 'Monitor' behavior to keep backward
; compatibility.
;
; monitor-type = MixMonitor
;
; ----------------------- TYPE MIXMONITOR OPTIONS -----------------------------
;
;
; You can specify the options supplied to MixMonitor by calling (from the dialplan)
; Set(MONITOR_OPTIONS=av(<x>)V(<x>)W(<x>))
; The 'b' option for MixMonitor (only save audio to the file while bridged) is
; implied.
;
; You can specify a post recording command to be executed after the end of
; recording by calling (from the dialplan)
;
; Set(MONITOR_EXEC=mv /var/spool/asterisk/monitor/^{MONITOR_FILENAME} /tmp/^{MONITOR_FILENAME})
;
; or
;
; Set(MONITOR_EXEC=mv /var/spool/asterisk/monitor/^{MIXMONITOR_FILENAME} /tmp/^{MIXMONITOR_FILENAME})
;
; If you choose to use the latter, you will not be able to switch the monitor-type back to Monitor
; without changing this in the dialplan.
;
;
; The command specified within the contents of MONITOR_EXEC will be executed when
; the recording is over. Any strings matching ^{X} will be unescaped to ${X} and
; all variables will be evaluated just prior to recording being started.
;
; The contents of MONITOR_FILENAME will also be unescaped from ^{X} to ${X} and
; all variables will be evaluated just prior to recording being started.
;
; ---------------------- Queue Empty Options ----------------------------------
;
; Asterisk has provided the "joinempty" and "leavewhenempty" options for a while
; with tenuous definitions of what they actually mean. The "joinempty" option controls
; whether a caller may join a queue depending on several factors of member availability.
; Similarly, then leavewhenempty option controls whether a caller may remain in a queue
; he has already joined. Both options take a comma-separated list of factors which
; contribute towards whether a caller may join/remain in the queue. The list of
; factors which contribute to these option is as follows:
;
; paused: a member is not considered available if he is paused
; penalty: a member is not considered available if his penalty is less than QUEUE_MAX_PENALTY
; inuse: a member is not considered available if he is currently on a call
; ringing: a member is not considered available if his phone is currently ringing
; unavailable: This applies mainly to Agent channels. If the agent is a member of the queue
; but has not logged in, then do not consider the member to be available
; invalid: Do not consider a member to be available if he has an "invalid" device state.
; This generally is caused by an error condition in the member's channel driver.
; unknown: Do not consider a member to be available if we are unable to determine the member's
; current device state.
; wrapup: A member is not considered available if he is currently in his wrapuptime after
; taking a call.
;
; For the "joinempty" option, when a caller attempts to enter a queue, the members of that
; queue are examined. If all members are deemed to be unavailable due to any of the conditions
; listed for the "joinempty" option, then the caller will be unable to enter the queue. For the
; "leavewhenempty" option, the state of the members of the queue are checked periodically during
; the caller's stay in the queue. If all of the members are unavailable due to any of the above
; conditions, then the caller will be removed from the queue.
;
; Some examples:
;
;joinempty = paused,inuse,invalid
;
; A caller will not be able to enter a queue if at least one member cannot be found
; who is not paused, on the phone, or who has an invalid device state.
;
;leavewhenempty = inuse,ringing
;
; A caller will be removed from the queue if at least one member cannot be found
; who is not on the phone, or whose phone is not ringing.
;
; For the sake of backwards-compatibility, the joinempty and leavewhenempty
; options also accept the strings "yes" "no" "strict" and "loose". The following
; serves as a translation for these values:
;
; yes - (empty) for joinempty; penalty,paused,invalid for leavewhenempty
; no - penalty,paused,invalid for joinempty; (empty) for leavewhenempty
; strict - penalty,paused,invalid,unavailable
; loose - penalty,invalid
;
; If you wish to report the caller's hold time to the member before they are
; connected to the caller, set this to yes.
;
; reportholdtime = no
;
; If you want the queue to avoid sending calls to members whose devices are
; known to be 'in use' (via the channel driver supporting that device state)
; uncomment this option. This can be controlled on a per member basis by
; setting 'ringinuse' on that member. This can be done in the member definition,
; in the 'ringinuse' field on a realtime member, via the QUEUE_MEMBER dialplan
; function, or with CLI/AMI. By default, the per member value will be the same
; as the queue's ringinuse value if it isn't set on the member deliberately.
; (Note: only the SIP channel driver currently is able to report 'in use'.)
; ringinuse = no
;
; If you wish to have a delay before the member is connected to the caller (or
; before the member hears any announcement messages), set this to the number of
; seconds to delay.
;
; memberdelay = 0
;
; If timeoutrestart is set to yes, then the timeout for an agent to answer is
; reset if a BUSY or CONGESTION is received. This can be useful if agents
; are able to cancel a call with reject or similar.
;
; timeoutrestart = no
;
; If you wish to implement a rule defined in queuerules.conf (see
; configs/queuerules.conf.sample from the asterisk source directory for
; more information about penalty rules) by default, you may specify this
; by setting defaultrule to the rule's name
;
; defaultrule = myrule
;
; Each member of this call queue is listed on a separate line in
; the form technology/dialstring. "member" means a normal member of a
; queue. An optional penalty may be specified after a comma, such that
; entries with higher penalties are considered last. An optional member
; name may also be specified after a second comma, which is used in log
; messages as a "friendly name". Multiple interfaces may share a single
; member name. An optional state interface may be specified after a third
; comma. This interface will be the one for which app_queue receives device
; state notifications, even though the first interface specified is the one
; that is actually called.
;
; A hint can also be used in place of the state interface using the format
; hint:<extension>@<context>. If no context is specified then 'default' will
; be used.
;
; It is important to ensure that channel drivers used for members are loaded
; before app_queue.so itself or they may be marked invalid until reload. This
; can be accomplished by explicitly listing them in modules.conf before
; app_queue.so. Additionally, if you use Local channels as queue members, you
; must also preload pbx_config.so and chan_local.so (or pbx_ael.so, pbx_lua.so,
; or pbx_realtime.so, depending on how your dialplan is configured).
;
; syntax: member => interface,[,penalty][,membername][,state_interface][,ringinuse][,wrapuptime]
;
;member => DAHDI/1
;member => DAHDI/2,10
;member => DAHDI/3,10,Bob Johnson
;member => Local/1001@agents,0,May Flowers,Agent:1001
;member => Local/1002@agents,0,John Doe,Agent:1002
;member => Local/1000@default,0,John Smith,SIP/1000
;member => Local/2000@default,0,Lorem Ipsum,SIP/2000,no

35
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; Define a realtime database name to use in extconfig.conf
;
;[asterisk]
;dbfile => /var/lib/asterisk/realtime.sqlite3
;
; debug - Turn on debugging information
;debug=yes
;
; requirements - At startup, each realtime family will make requirements
; on the backend. There are several strategies for handling requirements:
; warn - Warn if the required column does not exist.
; createclose - Create columns as close to the requirements as possible.
; createchar - Create char columns only
;
;requirements=warn
;
; batch - SQLite 3 write performance can be greatly improved by wrapping
; multiple writes in transactions. This option specifies the duration in
; milliseconds of auto-generated transactions. Any changes made during an
; unfinished transaction will be immediately available to the same database
; connection, but any external connections could see a delay up to the value
; of this setting. It is also possible that if asterisk crashes, any changes
; made during this time could be lost. Due to the nearly 100x performance
; benefit, the default is 100 ms. Set to 0 to disable batching.
;
; PLEASE NOTE: If you need to write to the database from another connection
; you will need to set batch=0 as the transactions will cause the database
; to lock for writing.
;
;batch=1000
;
; busy timeout - timeout in miliseconds to stop waiting for locked database.
; More info at https://www.sqlite.org/c3ref/busy_timeout.html
;
;busy_timeout=1000

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; Generic Fax Application 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: 4800
;minrate=4800
; 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
; T.38 Negotiation Timeout in milliseconds
; Default: 5000
t38timeout=5000

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;
; Configuration file for res_config_ldap
;
; Realtime configuration
; ----------------------
; In order to use this module, you start
; in extconfig.conf with a configuration like this:
;
; sippeers = ldap,"dc=myDomain,dc=myDomainExt",sip
; extensions = ldap,"dc=myDomain,dc=myDomainExt",extensions
; sip.conf = ldap,"dc=myDomain,dc=myDomainExt",config
;
; In the case of LDAP the last keyword in each line above specifies
; a section in this file.
;
; LDAP schema and ldif files can be located in contrib/scripts.
; TLS support
; -----------
; Note that you can configure an ldaps: url here to get TLS support.
; Detailed configuration of certificates and supported CAs is done in your
; ldap.conf file for OpenLDAP clients on your system.
; This requires that you have OpenLDAP libraries compiled with TLS support
; *********************************************************************************
; NOTE: res_ldap.conf should be chmod 600 because it contains the plain-text LDAP
; password to an account with WRITE access to the asterisk configuration.
; *********************************************************************************
[_general]
;
; Specify one of either host and port OR url. URL is preferred, as you can
; use more options.
;host=192.168.1.1 ; LDAP host
;port=389
;url=ldap://ldap3.mydomain.com:3890
;protocol=3 ; Version of the LDAP protocol to use; default is 3.
;basedn=dc=example,dc=tld ; Base DN
;user=cn=asterisk,dc=example,dc=tld ; Bind DN
;pass=MyPassword ; Bind password
; Configuration Table
[config]
;
; additionalFilter - This specifies an additional set of criteria to be used
; when querying the LDAP server.
;
additionalFilter=(objectClass=AstConfig)
;
; Attributes mapping (asterisk variable name = ldap attribute name)
; When Asterisk requests the variable by the name of the value on the left,
; this module will look up the attribute listed on the right.
;
filename = AstConfigFilename
category = AstConfigCategory
variable_name = AstConfigVariableName
variable_value = AstConfigVariableValue
cat_metric = AstConfigCategoryMetric
commented = AstConfigCommented
;
; Extensions Table
;
[extensions]
context = AstExtensionContext
exten = AstExtensionExten
priority = AstExtensionPriority
app = AstExtensionApplication
appdata = AstExtensionApplicationData
additionalFilter=(objectClass=AstExtension)
;
; Sip Users Table
;
[sip]
name = cn ; We use the "cn" as the default value for name on the line above
; because objectClass=AsteriskSIPUser does not include a uid as an allowed field
; If your entry combines other objectClasses and uid is available, you may
; prefer to change the line to be name = uid, especially if your LDAP entries
; contain spaces in the cn field.
; You may also find it appropriate to use something completely different.
; This is possible by changing the line above to name = AstAccountName (or whatever you
; prefer).
;
amaflags = AstAccountAMAFlags
callgroup = AstAccountCallGroup
callerid = AstAccountCallerID
directmedia = AstAccountDirectMedia
context = AstAccountContext
dtmfmode = AstAccountDTMFMode
fromuser = AstAccountFromUser
fromdomain = AstAccountFromDomain
fullcontact = AstAccountFullContact
fullcontact = gecos
host = AstAccountHost
insecure = AstAccountInsecure
mailbox = AstAccountMailbox
md5secret = AstAccountRealmedPassword ; Must be an MD5 hash. Field value can start with
; {md5} but it is not required.
; Generate the password via the md5sum command, e.g.
; echo "my_password" | md5sum
nat = AstAccountNAT
deny = AstAccountDeny
permit = AstAccountPermit
pickupgroup = AstAccountPickupGroup
port = AstAccountPort
qualify = AstAccountQualify
restrictcid = AstAccountRestrictCID
rtptimeout = AstAccountRTPTimeout
rtpholdtimeout = AstAccountRTPHoldTimeout
type = AstAccountType
disallow = AstAccountDisallowedCodec
allow = AstAccountAllowedCodec
MusicOnHold = AstAccountMusicOnHold
regseconds = AstAccountExpirationTimestamp
regcontext = AstAccountRegistrationContext
regexten = AstAccountRegistrationExten
CanCallForward = AstAccountCanCallForward
ipaddr = AstAccountIPAddress
defaultuser = AstAccountDefaultUser
regserver = AstAccountRegistrationServer
lastms = AstAccountLastQualifyMilliseconds
supportpath = AstAccountPathSupport
additionalFilter=(objectClass=AsteriskSIPUser)
;
; IAX Users Table
;
[iax]
amaflags = AstAccountAMAFlags
callerid = AstAccountCallerID
context = AstAccountContext
fullcontact = AstAccountFullContact
fullcontact = gecos
host = AstAccountHost
mailbox = AstAccountMailbox
md5secret = AstAccountRealmedPassword ; Must be an MD5 hash. Field value can start with
; {md5} but it is not required.
; Generate the password via the md5sum command, e.g.
; echo "my_password" | md5sum
deny = AstAccountDeny
permit = AstAccountPermit
port = AstAccountPort
qualify = AstAccountQualify
type = AstAccountType
disallow = AstAccountDisallowedCodec
allow = AstAccountAllowedCodec
regseconds = AstAccountExpirationTimestamp
regcontext = AstAccountRegistrationContext
regexten = AstAccountRegistrationExten
notransfer = AstAccountNoTransfer
lastms = AstAccountLastQualifyMilliseconds
additionalFilter=(objectClass=AstAccountIAX)
;
; A Test Family
;
[testfamily]
MyUSERID = uid
additionalFilter=(objectClass=*)
[accounts]
amaflags = AstAccountAMAFlags
callgroup = AstAccountCallGroup
callerid = AstAccountCallerID
directmedia = AstAccountDirectMedia
context = AstAccountContext
dtmfmode = AstAccountDTMFMode
fromuser = AstAccountFromUser
fromdomain = AstAccountFromDomain
fullcontact = AstAccountFullContact
fullcontact = gecos
host = AstAccountHost
insecure = AstAccountInsecure
mailbox = AstAccountMailbox
md5secret = AstAccountRealmedPassword ; Must be an MD5 hash. Field value can start with
; {md5} but it is not required.
; Generate the password via the md5sum command, e.g.
; echo "my_password" | md5sum
nat = AstAccountNAT
deny = AstAccountDeny
permit = AstAccountPermit
pickupgroup = AstAccountPickupGroup
port = AstAccountPort
qualify = AstAccountQualify
restrictcid = AstAccountRestrictCID
rtptimeout = AstAccountRTPTimeout
rtpholdtimeout = AstAccountRTPHoldTimeout
type = AstAccountType
disallow = AstAccountDisallowedCodec
allow = AstAccountAllowedCodec
MusicOnHold = AstAccountMusicOnHold
regseconds = AstAccountExpirationTimestamp
regcontext = AstAccountRegistrationContext
regexten = AstAccountRegistrationExten
CanCallForward = AstAccountCanCallForward
additionalFilter=(objectClass=AstAccount)

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[general]
;parkeddynamic = yes ; Enables dynamically created parkinglots. (default is no)
; If the option is enabled then the following
; variables can be used to dynamically create
; new parking lots.
;
; The PARKINGDYNAMIC variable specifies the
; parking lot to use as a template to create
; a dynamic parking lot. It is an error to
; specify a non-existent parking lot for the
; template. If not set then the default
; parking lot is used as the template.
;
; The PARKINGDYNCONTEXT variable specifies
; the dialplan context to use for the newly
; created dynamic parking lot. If not set
; then the context from the parking lot
; template is used. The context is created
; if it does not already exist and the new
; parking lot needs to create extensions.
;
; The PARKINGDYNEXTEN variable specifies the
; parkext to use for the newly created dynamic
; parking lot. If not set then the parkext
; is used from the parking lot template. If
; the template does not specify a parkext
; then no extensions are created for the
; newly created parking lot. The dynamic
; parking lot cannot be created if it needs
; to create extensions that overlap existing
; parking lot extensions. The only exception
; to this is for the parkext extension and
; only if neither of the overlaping parking
; lot's parkext is exclusive.
;
; The PARKINGDYNPOS variable specifies the
; parking positions to use for the newly
; created dynamic parking lot. If not set
; then the parkpos from the parking lot
; template is used.
; A parking lot named 'default' will automatically be used when no other
; named parking lot is indicated for use by the park application or a
; channel's parkinglot function and PARKINGLOT channel variable. This parking
; lot is guaranteed to exist and will be created even if default is left out of
; the configuration file.
[default] ; Default Parking Lot
parkext => 700 ; What extension to dial to park. (optional; if
; specified, extensions will be created for parkext and
; the whole range of parkpos)
;
; Note: Generated parking extensions cannot overlap.
; The only exception is if neither overlapping parkext
; is exclusive.
;parkext_exclusive=yes ; Specify that the parkext created for this parking lot
; will only access this parking lot. (default is no)
parkpos => 701-720 ; What range of parking spaces to use - must be numeric
; Creates these spaces as extensions if parkext is set.
; Since this value is interpreted numerically, leading 0's
; will be ignored (so expect 00700-00720 to map to 700-720)
context => parkedcalls ; Which context parked calls and the default park
;parkinghints = no ; Add hints priorities automatically for parkpos
; extensions if parkext is set
;parkingtime => 45 ; Number of seconds a call can be parked before returning
;comebacktoorigin = yes ; Setting this option configures the behavior of call parking when the
; parked call times out (See the parkingtime option). The default value is 'yes'.
;
; 'yes' - When the parked call times out, attempt to send the call back to the peer
; that parked this call. This is done by saving off the name of the channel
; that parked the call. The call will return to the context 'park-dial' and
; an extension created based on the name of the channel that originally parked
; the call. This extension will be created automatically to do a Dial() to the
; device that originally parked the call for comebacktodialtime seconds. If the
; call is not answered, the call will proceed to the next priority (usually none
; unless you deliberately set up a catch-all second priority in the park-call
; context) in the dialplan for extension matching the peer name (same as how
; peer names are flattened into extensions when comebacktoorigin is 'no').
;
; 'no' - This option is useful for performing custom dialplan functionality prior to
; sending the call back to the extension that initially parked the call, or to
; an entirely different destination.
;
; When the parked call times out, send it back to the dialplan. The location
; will be defined by the comebackcontext option. The extension will be built from
; the saved channel name that parked the call. For example, if a SIP peer named
; '0004F2040001' parked this call, the extension will be 'SIP_0004F2040001'.
; (Note that an underscore is used here because the '/' character has a special
; meaning in extension names for CallerID matching.) If this extension does not
; exist, the call will be sent to the 's' extension, instead. Finally, if the 's'
; extension of 'parkedcallstimeout' does not exist, the call will fall back to the
; 's' extension of the 'default' context.
;
; Additionally, in this example an extension of 'SIP_0004F2040001' will be
; created in the 'park-dial' context. This extension will be set up to do a
; Dial() to 'SIP/0004F2040001'.
;
; During the timeout procedure, the following variables are set
; PARKING_SPACE - extension that the call was parked in prior to timing out
; PARKEDLOT - name of the lot that the call was parked in prior to timing out
; PARKER - dial string to call the device that parked the call
;comebackdialtime = 30 ; When a parked call times out, this is the number of seconds to dial the device that
; originally parked the call. It is also available as a channel variable COMEBACKDIALTIME
; after a parked call has timed out.
; The default value is 30 seconds.
;comebackcontext = parkedcallstimeout
; The context a timed out call will return to if comebcktoorigin=no.
; The default value is 'parkedcallstimeout'.
;courtesytone = beep ; Sound file to play to when someone picks up a parked call
; and also when the Touch Monitor is activated/deactivated.
; Default is no tone.
;parkedplay = caller ; Who to play courtesytone to when picking up a parked call.
; One of: parked, caller, both (default is caller)
;parkedcalltransfers = caller ; Enables or disables DTMF based transfers when picking up a parked call.
; one of: callee, caller, both, no (default is no)
;parkedcallreparking = caller ; Enables or disables DTMF based parking when picking up a parked call.
; one of: callee, caller, both, no (default is no)
;parkedcallhangup = caller ; Enables or disables DTMF based hangups when picking up a parked call.
; one of: callee, caller, both, no (default is no)
;findslot => next ; Sets the method for selecting parking spaces when a call is parked
; 'next' - use the next parking space from the most recently used one.
; 'first' - use the lowest numbered parking space available
;parkedmusicclass = default ; This is the MOH class to use for the parked channel
; as long as the class is not set on the channel directly
; using Set(CHANNEL(musicclass)=whatever) in the dialplan
;*** Define another parking lot
;
; The parkinglot used can be set with the CHANNEL(parkinglot) dialplan function or by
; setting the 'parkinglot' configuration for a channel in its configuration file.
;
; Parking lots can now be any named configuration category aside from
; 'general' which is reserved for general options. They no longer need to be
; prefixed with 'parkinglot_'
;
;[edvina]
;context => edvina_park
;parkpos => 800-850
;findslot => next
;comebacktoorigin = no
;comebackdialtime = 90
;comebackcontext = edvinapark-timeout
;parkedmusicclass = edvina
;
; Since edvina doesn't define parkext, extensions won't automatically be
; created for parking to it or for retrieving calls from it. These can be
; created manually in the dial plan by using the Park and ParkedCall
; applications.

32
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;; Sample res_pktccops.conf
;
;[general]
;gateinfoperiod => 60 ; default 60s
;gatetimeout = 150 ; default 150
;t1 => 250 ; default 250s
;t7 => 200 ; default 200s
;t8 => 300 ; default 300s
;keepalive => 60 ; default 60s
;
;[teszt]
;host => 192.168.0.24
;pool => 10.0.1.0 10.0.1.255
;pool => 10.0.3.0 10.0.3.255
;pool => 10.0.7.0 10.0.8.255
;pool => 10.0.10.0 10.0.11.255
;
;[general]
;gateinfoperiod => 60 ; default 60s
;gatetimeout = 150 ; default 150
;t1 => 250 ; default 250s
;t7 => 200 ; default 200s
;t8 => 300 ; default 300s
;keepalive => 60 ; default 60s
;
;[test]
;host => 192.168.0.24
;pool => 10.0.1.0 10.0.1.255
;pool => 10.0.3.0 10.0.3.255
;pool => 10.0.7.0 10.0.8.255
;pool => 10.0.10.0 10.0.11.255
;

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res_snmp.conf Normal file
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;
; Configuration file for res_snmp
; --------------------------------
;
; Res_snmp can run as a subagent or standalone SNMP agent. The standalone snmp
; agent is based on net-snmp and will read a configuration file called
; asterisk.conf in the net-snmp configuration file path, starting with
; /etc/snmp on many systems.
;
; If you use the subagent model, you need to enable agentx in snmpd.conf
; Note that you can only run one Asterisk on the system in this case.
;
; See https://wiki.asterisk.org/wiki/display/AST/Simple+Network+Management+Protocol+(SNMP)+Support
; to get more information about
; snmp support in Asterisk
[general]
; We run as a subagent per default -- to run as a full agent
; we must run as root (to be able to bind to port 161)
;subagent = yes
; SNMP must be explicitly enabled to be active
;enabled = yes

27
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;
; Configuration file for the res_stun_monitor module
;
; The res_stun_monitor module sends STUN requests to a configured STUN server
; periodically. If the monitor detects a change in the external IP address or port
; provided by the STUN server an event is sent out internally within Asterisk
; to alert all listeners to that event of the change.
; The current default listeners for the network change event include chan_sip
; and chan_iax. Both of these channel drivers by default react to this event
; by renewing all outbound registrations. This allows the endpoints Asterisk
; is registering with to become aware of the address change and know the new
; location.
;
[general]
;
; ---- STUN Server configuration ---
; Setting the 'stunaddr' option to a valid address enables the STUN monitor.
;
;stunaddr = mystunserver.com ; Address of the STUN server to query.
; Valid form:
; [(hostname | IP-address) [':' port]]
; The port defaults to the standard STUN port (3478).
; Set to an empty value to disable STUN monitoring.
; Default is disabled.
;stunrefresh = 30 ; Number of seconds between STUN refreshes.
; Default is 30.

23
resolver_unbound.conf Normal file
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; Unbound DNS Resolver Configuration
;
; This file serves as a reference for the configurable options within the
; unbound DNS resolver.
[general]
;hosts = /etc/hosts ; Full path to a hosts file which contains a mapping of
; ; hostnames to addresses. If "system" is specified then
; ; the system specific hosts file will be used. (default: system)
;resolv = /etc/resolv.conf ; Full path to a resolv.conf which contains the nameservers
; ; to use for resolution. If "system" is specified then the
; ; system specific resolv.conf file will be used. (default: system)
;nameserver = 127.0.0.1 ; An explicit nameserver to use for queries. If this option
; ; is specified multiple times the first configured one will
; ; be treated as the primary with each subsequent one being
; ; a backup. If the resolv options is also specified the
; ; nameservers from it will be tried after all nameserver
; ; options.
;debug = 99 ; The debug level to run the unbound resolver at. While
; ; there is no explicit range the higher the number the more
; ; debug is output.
;ta_file = /etc/asterisk/dnssec_keys ; Full path to a trusted anchors key file. These keys are
; ; used to verify DNSSEC signed results.

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;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000
;
; Whether to enable or disable UDP checksums on RTP traffic
;
;rtpchecksums=no
;
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of a second)
;
;dtmftimeout=3000
; rtcpinterval = 5000 ; Milliseconds between rtcp reports
;(min 500, max 60000, default 5000)
;
; Enable strict RTP protection. This will drop RTP packets that do not come
; from the recoginized source of the RTP stream. Strict RTP qualifies RTP
; packet stream sources before accepting them upon initial connection and
; when the connection is renegotiated (e.g., transfers and direct media).
; Initial connection and renegotiation starts a learning mode to qualify
; stream source addresses. Once Asterisk has recognized a stream it will
; allow other streams to qualify and replace the current stream for 5
; seconds after starting learning mode. Once learning mode completes the
; current stream is locked in and cannot change until the next
; renegotiation.
; Valid options are "no" to disable strictrtp, "yes" to enable strictrtp,
; and "seqno", which does the same thing as strictrtp=yes, but only checks
; to make sure the sequence number is correct rather than checking the time
; interval as well.
; This option is enabled by default.
; strictrtp=yes
;
; Number of packets containing consecutive sequence values needed
; to change the RTP source socket address. This option only comes
; into play while using strictrtp=yes. Consider changing this value
; if rtp packets are dropped from one or both ends after a call is
; connected. This option is set to 4 by default.
; probation=8
;
; Whether to enable or disable ICE support. This option is enabled by default.
; icesupport=false
;
; Hostname or address for the STUN server used when determining the external
; IP address and port an RTP session can be reached at. The port number is
; optional. If omitted the default value of 3478 will be used. This option is
; disabled by default.
;
; e.g. stundaddr=mystun.server.com:3478
;
; stunaddr=
;
; Some multihomed servers have IP interfaces that cannot reach the STUN
; server specified by stunaddr. Blacklist those interface subnets from
; trying to send a STUN packet to find the external IP address.
; Attempting to send the STUN packet needlessly delays processing incoming
; and outgoing SIP INVITEs because we will wait for a response that can
; never come until we give up on the response.
; * Multiple subnets may be listed.
; * Blacklisting applies to IPv4 only. STUN isn't needed for IPv6.
; * Blacklisting applies when binding RTP to specific IP addresses and not
; the wildcard 0.0.0.0 address. e.g., A PJSIP endpoint binding RTP to a
; specific address using the bind_rtp_to_media_address and media_address
; options. Or the PJSIP endpoint specifies an explicit transport that binds
; to a specific IP address.
;
; e.g. stun_blacklist = 192.168.1.0/255.255.255.0
; stun_blacklist = 10.32.77.0/255.255.255.0
;
; stun_blacklist =
;
; Hostname or address for the TURN server to be used as a relay. The port
; number is optional. If omitted the default value of 3478 will be used.
; This option is disabled by default.
;
; e.g. turnaddr=myturn.server.com:34780
;
; turnaddr=
;
; Username used to authenticate with TURN relay server.
; turnusername=
;
; Password used to authenticate with TURN relay server.
; turnpassword=
;
; Subnets to exclude from ICE host, srflx and relay discovery. This is useful
; to optimize the ICE process where a system has multiple host address ranges
; and/or physical interfaces and certain of them are not expected to be used
; for RTP. For example, VPNs and local interconnections may not be suitable or
; necessary for ICE. Multiple subnets may be listed. If left unconfigured,
; all discovered host addresses are used.
;
; e.g. ice_blacklist = 192.168.1.0/255.255.255.0
; ice_blacklist = 10.32.77.0/255.255.255.0
;
; ice_blacklist =
;
[ice_host_candidates]
;
; When Asterisk is behind a static one-to-one NAT and ICE is in use, ICE will
; expose the server's internal IP address as one of the host candidates.
; Although using STUN (see the 'stunaddr' configuration option) will provide a
; publicly accessible IP, the internal IP will still be sent to the remote
; peer. To help hide the topology of your internal network, you can override
; the host candidates that Asterisk will send to the remote peer.
;
; IMPORTANT: Only use this functionality when your Asterisk server is behind a
; one-to-one NAT and you know what you're doing. If you do define anything
; here, you almost certainly will NOT want to specify 'stunaddr' or 'turnaddr'
; above.
;
; The format for these overrides is:
;
; <local address> => <advertised address>
;
; The following will replace 192.168.1.10 with 1.2.3.4 during ICE
; negotiation:
;
;192.168.1.10 => 1.2.3.4
;
; You can define an override for more than 1 interface if you have a multihomed
; server. Any local interface that is not matched will be passed through
; unaltered. Both IPv4 and IPv6 addresses are supported.

1619
sip.conf Normal file
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57
sip_notify.conf Normal file
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; 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
; Digium
[digium-check-cfg]
Event=>check-sync
; 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=>check-sync\;reboot=true
; Cisco
[cisco-check-cfg]
Event=>check-sync

75
smdi.conf Normal file
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; Asterisk SMDI configuration
[interfaces]
; Specify serial ports to listen for SMDI messages on below. These will be
; referenced later in chan_dahdi.conf. If you do not specify any interfaces
; then SMDI will be disabled. Interfaces can have several different attributes
; associated with them.
; Set the number of stop bits to use per character here. The default is no,
; in which case one stop bit will be used.
;twostopbits = no
; Character size or bit length is the size of each character sent across the
; link. Character size can be 7 or 8. The default is 7.
;charsize = 7
; If you need parity checking enabled you can turn it on here. Acceptable
; values are even, odd, and none. The default is even.
;paritybit = even
; The baudrate to use for this port. Acceptable values are 1200, 2400, 4800,
; and 9600. The default is 9600.
;baudrate = 1200
; Often the numbering scheme for a set of mailboxes or extensions will not be 7
; or 10 digits (as SMDI requires). Use the msdstrip option to strip unused
; digits from the start of numbers.
;msdstrip = 0
; Occasionally Asterisk and the SMDI switch may become out of sync. If this
; happens, Asterisk will appear one or several calls behind as it processes
; voicemail requests. To prevent this from happening, adjust the msgexpirytime.
; This will make Asterisk discard old SMDI messages that have not yet been
; processed. The default expiry time is 30000 milliseconds.
;msgexpirytime = 30000
;smdiport => /dev/ttyS0
[mailboxes]
; This section configures parameters related to MWI handling for the SMDI link.
; This option configures the polling interval used to check to see if the
; mailboxes have any new messages. This option is specified in seconds.
; The default value is 10 seconds.
;
;pollinginterval=10
; Every other entry in this section of the configuration file is interpreted as
; a mapping between the mailbox ID on the SMDI link, and the local Asterisk
; mailbox name. In many cases, they are the same thing, but they still must be
; listed here so that this module knows which mailboxes it needs to pay
; attention to.
;
; Syntax:
; <SMDI mailbox ID>=<Asterisk Mailbox Name>[@Asterisk Voicemail Context]
;
; If no Asterisk voicemail context is specified, "default" will be assumed.
;
; Before specifying mailboxes, you must specify an SMDI interface. All mailbox
; definitions that follow will correspond to that SMDI interface. If you specify
; another interface, then all definitions following that will correspond to the
; new interface.
;
;smdiport=/dev/ttyS0
;2565551234=1234@vmcontext1
;2565555678=5678@vmcontext2
;smdiport=/dev/ttyS1
;2565559999=9999

78
sorcery.conf Normal file
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; Sample configuration file for Sorcery Data Access Layer
;
; Wizards
;
; Wizards are the persistence mechanism for objects. They are loaded as Asterisk modules and register
; themselves with the sorcery core. All implementation specific details of how objects are persisted is isolated
; within wizards.
;
;
; Caching
;
; A wizard can optionally be marked as an object cache by adding "/cache" to the object type within the mapping.
; If an object is returned from a non-object cache it is immediately given to the cache to be created. Multiple
; object caches can be configured for a single object type.
;
;
; Object Type Mappings
;
; To allow configuration of where and how an object is persisted object mappings can be defined within this file
; on a per-module basis. The mapping consists of the object type, options, wizard name, and wizard configuration
; data. This has the following format:
;
; object type [/options] = wizard name, wizard configuration data
;
; For example to configure an in-memory wizard for the 'bob' object type:
;
; bob = memory
;
; Or to configure the object type 'joe' from a configuration file:
;
; joe = config,joe.conf
;
; Note that an object type can have multiple mappings defined. Each mapping will be consulted in the order in which
; it appears within the configuration file. This means that if you are configuring a wizard as a cache it should
; appear as the first mapping so the cache is consulted before all other mappings.
;
;
; The following object mappings are used by the unit test to test certain functionality of sorcery.
;
[test_sorcery_section]
test=memory
[test_sorcery_cache]
test/cache=test
test=memory
;
; The following object mapping is the default mapping of external MWI mailbox
; objects to give persistence to the message counts.
;
;[res_mwi_external]
;mailboxes=astdb,mwi_external
;
; The following object mappings set PJSIP objects to use realtime database mappings from extconfig
; with the table names used when automatically generating configuration from the alembic script.
;
;[res_pjsip]
;endpoint=realtime,ps_endpoints
;auth=realtime,ps_auths
;aor=realtime,ps_aors
;domain_alias=realtime,ps_domain_aliases
;[res_pjsip_endpoint_identifier_ip]
;identify=realtime,ps_endpoint_id_ips
;[res_pjsip_outbound_publish]
;outbound-publish=realtime,ps_outbound_publishes
;[res_pjsip_pubsub]
;inbound-publication=realtime,ps_inbound_publications
;[res_pjsip_publish_asterisk]
;asterisk-publication=realtime,ps_asterisk_publications

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stasis.conf Normal file
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[threadpool]
;initial_size = 5 ; Initial size of the threadpool.
; ; 0 means the threadpool has no threads initially
; ; until a task needs a thread.
;idle_timeout_sec = 20 ; Number of seconds a thread should be idle before
; ; dying. 0 means threads never time out.
;max_size = 50 ; Maximum number of threads in the Stasis threadpool.
; ; 0 means no limit to the number of threads in the
; ; threadpool.
[declined_message_types]
; This config section contains the names of message types that should be prevented
; from being created. By default, all message types are allowed to be created.
;
; Using this functionality requires knowledge of the names of internal stasis
; message types which is generally the same as the name of the accessor function.
;
; Use of this functionality may break more complex functionality in Asterisk
; such as CEL, CDR, transfers, etc. and will likely cause related messages in ARI
; and AMI to go missing.
; decline=stasis_app_recording_snapshot_type
; decline=stasis_app_playback_snapshot_type
; decline=stasis_test_message_type
; decline=confbridge_start_type
; decline=confbridge_end_type
; decline=confbridge_join_type
; decline=confbridge_leave_type
; decline=confbridge_start_record_type
; decline=confbridge_stop_record_type
; decline=confbridge_mute_type
; decline=confbridge_unmute_type
; decline=confbridge_talking_type
; decline=cel_generic_type
; decline=ast_bridge_snapshot_type
; decline=ast_bridge_merge_message_type
; decline=ast_channel_entered_bridge_type
; decline=ast_channel_left_bridge_type
; decline=ast_blind_transfer_type
; decline=ast_attended_transfer_type
; decline=ast_endpoint_snapshot_type
; decline=ast_endpoint_state_type
; decline=ast_device_state_message_type
; decline=ast_test_suite_message_type
; decline=ast_mwi_state_type
; decline=ast_mwi_vm_app_type
; decline=ast_format_register_type
; decline=ast_format_unregister_type
; decline=ast_manager_get_generic_type
; decline=ast_parked_call_type
; decline=ast_channel_snapshot_type
; decline=ast_channel_dial_type
; decline=ast_channel_varset_type
; decline=ast_channel_hangup_request_type
; decline=ast_channel_dtmf_begin_type
; decline=ast_channel_dtmf_end_type
; decline=ast_channel_hold_type
; decline=ast_channel_unhold_type
; decline=ast_channel_chanspy_start_type
; decline=ast_channel_chanspy_stop_type
; decline=ast_channel_fax_type
; decline=ast_channel_hangup_handler_type
; decline=ast_channel_moh_start_type
; decline=ast_channel_moh_stop_type
; decline=ast_channel_monitor_start_type
; decline=ast_channel_monitor_stop_type
; decline=ast_channel_agent_login_type
; decline=ast_channel_agent_logoff_type
; decline=ast_channel_talking_start
; decline=ast_channel_talking_stop
; decline=ast_security_event_type
; decline=ast_named_acl_change_type
; decline=ast_local_bridge_type
; decline=ast_local_optimization_begin_type
; decline=ast_local_optimization_end_type
; decline=stasis_subscription_change_type
; decline=ast_multi_user_event_type
; decline=stasis_cache_clear_type
; decline=stasis_cache_update_type
; decline=ast_network_change_type
; decline=ast_system_registry_type
; decline=ast_cc_available_type
; decline=ast_cc_offertimerstart_type
; decline=ast_cc_requested_type
; decline=ast_cc_requestacknowledged_type
; decline=ast_cc_callerstopmonitoring_type
; decline=ast_cc_callerstartmonitoring_type
; decline=ast_cc_callerrecalling_type
; decline=ast_cc_recallcomplete_type
; decline=ast_cc_failure_type
; decline=ast_cc_monitorfailed_type
; decline=ast_presence_state_message_type
; decline=ast_rtp_rtcp_sent_type
; decline=ast_rtp_rtcp_received_type
; decline=ast_call_pickup_type
; decline=aoc_s_type
; decline=aoc_d_type
; decline=aoc_e_type
; decline=dahdichannel_type
; decline=mcid_type
; decline=session_timeout_type
; decline=cdr_read_message_type
; decline=cdr_write_message_type
; decline=cdr_prop_write_message_type
; decline=corosync_ping_message_type
; decline=agi_exec_start_type
; decline=agi_exec_end_type
; decline=agi_async_start_type
; decline=agi_async_exec_type
; decline=agi_async_end_type
; decline=queue_caller_join_type
; decline=queue_caller_leave_type
; decline=queue_caller_abandon_type
; decline=queue_member_status_type
; decline=queue_member_added_type
; decline=queue_member_removed_type
; decline=queue_member_pause_type
; decline=queue_member_penalty_type
; decline=queue_member_ringinuse_type
; decline=queue_agent_called_type
; decline=queue_agent_connect_type
; decline=queue_agent_complete_type
; decline=queue_agent_dump_type
; decline=queue_agent_ringnoanswer_type
; decline=meetme_join_type
; decline=meetme_leave_type
; decline=meetme_end_type
; decline=meetme_mute_type
; decline=meetme_talking_type
; decline=meetme_talk_request_type
; decline=appcdr_message_type
; decline=forkcdr_message_type
; decline=cdr_sync_message_type

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statsd.conf Normal file
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[general]
;enabled = yes ; When set to yes, statsd support is enabled
;server = 127.0.0.1 ; server[:port] of statsd server to use.
; If not specified, the port is 8125
;prefix = ; Prefix to prepend to all metrics
;add_newline = no ; Append a newline to every event. This is
; useful if you want to run a fake statsd
; server using netcat (nc -lu 8125)

83
telcordia-1.adsi Normal file
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;
; Asterisk default ADSI script
;
;
; Begin with the preamble requirements
;
DESCRIPTION "Telcordia Demo" ; Name of vendor
VERSION 0x02 ; Version of stuff
;SECURITY "_AST" ; Security code
SECURITY 0x0000 ; Security code
FDN 0x0000000f ; Descriptor number
;
; Predefined strings
;
DISPLAY "talkingto" IS "Talking To" "$Call1p" WRAP
DISPLAY "titles" IS "20th Century IQ Svc"
DISPLAY "newcall" IS "New Call From" "$Call1p" WRAP
DISPLAY "ringing" IS "Ringing"
;
; Begin state definitions
;
STATE "callup" ; Call is currently up
STATE "inactive" ; No active call
;
; Begin soft key definitions
;
KEY "CB_OH" IS "Block" OR "Call Block"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "*60"
SUBSCRIPT "offHook"
ENDKEY
KEY "CB" IS "Block" OR "Call Block"
SENDDTMF "*60"
ENDKEY
;
; Begin main subroutine
;
SUB "main" IS
IFEVENT NEARANSWER THEN
CLEAR
SHOWDISPLAY "talkingto" AT 1
GOTO "stableCall"
ENDIF
IFEVENT OFFHOOK THEN
CLEAR
SHOWDISPLAY "titles" AT 1
SHOWKEYS "CB"
GOTO "offHook"
ENDIF
IFEVENT IDLE THEN
CLEAR
SHOWDISPLAY "titles" AT 1
SHOWKEYS "CB_OH"
ENDIF
IFEVENT CALLERID THEN
CLEAR
SHOWDISPLAY "newcall" AT 1
ENDIF
ENDSUB
SUB "offHook" IS
IFEVENT FARRING THEN
CLEAR
SHOWDISPLAY "ringing" AT 1
ENDIF
IFEVENT FARANSWER THEN
CLEAR
SHOWDISPLAY "talkingto" AT 1
GOTO "stableCall"
ENDIF
ENDSUB
SUB "stableCall" IS
ENDSUB

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udptl.conf Normal file
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;
; UDPTL Configuration (UDPTL is one of the transports for T.38)
;
[general]
;
; UDPTL start and UDPTL end configure start and end addresses
;
udptlstart=4000
udptlend=4999
;
; Whether to enable or disable UDP checksums on UDPTL traffic
;
;udptlchecksums=no
;
; The number of error correction entries in a UDPTL packet
;
udptlfecentries = 3
;
; The span over which parity is calculated for FEC in a UDPTL packet
;
udptlfecspan = 3
;
; Some VoIP providers will only accept an offer with an even-numbered
; UDPTL port. Set this option so that Asterisk will only attempt to use
; even-numbered ports when negotiating T.38. Default is no.
use_even_ports = no

113
users.conf Normal file
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;
; User configuration
;
; Creating entries in users.conf is a "shorthand" for creating individual
; entries in each configuration file. Using users.conf is not intended to
; provide you with as much flexibility as using the separate configuration
; files (e.g. sip.conf, iax.conf, etc) but is intended to accelerate the
; simple task of adding users. Note that creating individual items (e.g.
; custom SIP peers, IAX friends, etc.) will allow you to override specific
; parameters within this file. Parameter names here are the same as they
; appear in the other configuration files. There is no way to change the
; value of a parameter here for just one subsystem.
;
[general]
;
; Full name of a user
;
fullname = New User
;
; Starting point of allocation of extensions
;
userbase = 6000
;
; Create voicemail mailbox and use use macro-stdexten
;
hasvoicemail = yes
;
; Set voicemail mailbox 6000 password to 1234
;
vmsecret = 1234
;
; Create SIP Peer
;
hassip = yes
;
; Create IAX friend
;
hasiax = yes
;
; Create H.323 friend
;
;hash323 = yes
;
; Create manager entry
;
hasmanager = no
;
; Set permissions for manager entry (see manager.conf.sample for documentation)
; (defaults to *all* permissions)
;managerread = system,call,log,verbose,command,agent,user,config
;managerwrite = system,call,log,verbose,command,agent,user,config
;
;
; MAC Address for res_phoneprov
;
;macaddress = 112233445566
;
; Auto provision the phone with res_phoneprov
;
;autoprov = yes
;
; Line Keys for hardphone
;
;LINEKEYS = 1
;
; Line number for hardphone
;
;linenumber = 1
;
; Local Caller ID number used with res_phoneprov and Asterisk GUI
;
;cid_number = 6000
;
; Remaining options are not specific to users.conf entries but are general.
;
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1
;nat = no
;[6000]
;fullname = Joe User
;description = Courtesy Phone In Lobby ; Used to provide a description of the
; peer in console output
;email = joe@foo.bar
;secret = 1234
;dahdichan = 1
;hasvoicemail = yes
;vmsecret = 1234
;hassip = yes
;hasiax = no
;hash323 = no
;hasmanager = no
;callwaiting = no
;context = international
;
; Some administrators choose alphanumeric extensions, but still want their
; users to be reachable by traditional numeric extensions, specified by the
; alternateexts entry.
;
;alternateexts = 7057,3249
;macaddress = 112233445566
;autoprov = yes
;LINEKEYS = 1
;linenumber = 1
;cid_number = 6000

64
xmpp.conf Normal file
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[general]
;debug=yes ; Enable debugging (disabled by default).
;autoprune=yes ; Auto remove users from buddy list. Depending on your
; setup (ie, using your personal Gtalk account for a test)
; you might lose your contacts list. Default is 'no'.
;autoregister=yes ; Auto register users from buddy list.
;collection_nodes=yes ; Enable support for XEP-0248 for use with
; distributed device state. Default is 'no'.
;pubsub_autocreate=yes ; Whether or not the PubSub server supports/is using
; auto-create for nodes. If it is, we have to
; explicitly pre-create nodes before publishing them.
; Default is 'no'.
;auth_policy=accept ; Auto accept users' subscription requests (default).
; Set to deny for auto denial.
;[asterisk]
;type=client ; Client or Component connection
;serverhost=astjab.org ; Route to server for example, talk.google.com
;pubsub_node=pubsub.astjab.org ; Node to use for publishing events via PubSub
;username=asterisk@astjab.org/asterisk ; Username with optional resource.
;secret=blah ; Password
;refresh_token=TOKEN_VALUE ; Refresh token issued by Google OAuth 2.0 protocol.
; `secret` must NOT be set if you use OAuth.
; See https://developers.google.com/identity/protocols/OAuth2WebServer
; for more details.
; For test reasons you can obtain one on the page
; https://developers.google.com/oauthplayground/
; 1. Click on Settings icon, check "Use your own OAuth credentials"
; and enter your Client ID and Client Secret (see below).
; 2. Input the scope https://www.googleapis.com/auth/googletalk
; and push "Authorize APIs" button.
; 3. Approve permissions.
; 4. On section "Step 2" push "Exchange authorization code for tokens"
; and get your Refresh token.
;oauth_clientid=OAUTH_CLIENT_ID_VALUE ; The application's client id to authorize using Google OAuth 2.0 protocol.
;oauth_secret=OAUTH_SECRET_VALUE ; The application's client secret to authorize using Google OAuth 2.0 protocol.
; 1. Create new Project on the page:
; https://console.cloud.google.com/apis/credentials/oauthclient
; 2. Create new Application ID on the same page with type Web-application.
; In section "Allowed URI redirections" put the path to the corresponding
; script on your site or https://developers.google.com/oauthplayground
; if you would like to obtain refresh_token from users by hand
; (for example, for test reasons).
; 3. Client ID and Client Secret will be shown and available on the same page.
;priority=1 ; Resource priority
;port=5222 ; Port to use defaults to 5222
;usetls=yes ; Use tls or not
;usesasl=yes ; Use sasl or not
;buddy=mogorman@astjab.org ; Manual addition of buddy to list.
; For distributed events, these buddies are
; automatically added in the whitelist as
; 'owners' of the node(s).
;distribute_events=yes ; Whether or not to distribute events using
; this connection. Default is 'no'.
;status=available ; One of: chat, available, away, xaway, or dnd
;statusmessage="I am available" ; Have custom status message for Asterisk
;timeout=5 ; Timeout (in seconds) on the message stack, defaults to 5.
; Messages stored longer than this value will be deleted by Asterisk.
; This option applies to incoming messages only, which are intended to
; be processed by the JABBER_RECEIVE dialplan function.
;sendtodialplan=yes ; Send incoming messages into the dialplan. Off by default.
;context=messages ; Dialplan context to send incoming messages to. If not set,
; "default" will be used.
;forceoldssl=no ; Force the use of old-style SSL.
;keepalive=