Remove asterisk configs (better one soon)

This commit is contained in:
mpmc 2023-06-23 15:34:46 +01:00
parent aeba465e76
commit f09865b0e5
16 changed files with 0 additions and 1568 deletions

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#!/bin/bash
# Run this script with "(sudo) bash <filename> <args>".
# Exit on error.
#set -e
# Debug
set -eux
apt install acl
chmod o=-rwx,g=rwx+s -v /var/log/asterisk/ /var/spool/asterisk/ /var/lib/asterisk
chown root:asterisk -v /var/log/asterisk/ /var/spool/asterisk/ /var/lib/asterisk
chown asterisk:asterisk -vR /var/log/asterisk/* /var/spool/asterisk/* /var/lib/asterisk/*
setfacl -d -m g::rwx /var/log/asterisk/ /var/spool/asterisk/ /var/lib/asterisk

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;
; /etc/asterisk/asterisk.conf
[directories](!)
astcachedir => /var/cache/asterisk
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /run/asterisk
astlogdir => /var/log/asterisk
astsbindir => /usr/sbin
[options]
;verbose = 3
;debug = 3
;trace = 0 ; Set the trace level.
;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.
runuser = asterisk
rungroup = asterisk
;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_GB ; 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 and configuration file access from
; external sources (AMI, etc.) These functions
; (such as SHELL) are considered dangerous
; because they can allow privilege escalation.
; Configuration files are considered dangerous
; if they exist outside of the Asterisk
; configuration directory.
; 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.
;hide_messaging_ami_events = no; This option, if enabled, will
; suppress all of the Message/ast_msg_queue channel's
; housekeeping AMI and ARI channel events. This can
; reduce the load on the manager and ARI applications
; when the Digium Phone Module for Asterisk is in use.
; Changing the following lines may compromise your security.
;[files]
;astctlpermissions = 0660
;astctlowner = root
;astctlgroup = apache
;astctl = asterisk.ctl

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# /etc/systemd/system/asterisk.service
# "Borrowed" from https://g1fef.co.uk/asterisk-systemd-startup-script/
[Unit]
Description=Asterisk PBX and telephony daemon.
Wants=network.target
After=network.target
[Service]
Type=simple
User=asterisk
Group=asterisk
RuntimeDirectory=asterisk
Environment=HOME=/var/lib/asterisk
WorkingDirectory=/var/lib/asterisk
ExecStart=/usr/sbin/asterisk -f -C /etc/asterisk/asterisk.conf
ExecStop=/usr/sbin/asterisk -rx 'core stop now'
ExecReload=/usr/sbin/asterisk -rx 'core reload'
# safe_asterisk emulation
Restart=always
RestartSec=10
#Nice=0
#UMask=0002
LimitCORE=infinity
#LimitNOFILE=
# Prevent duplication of logs with color codes to /var/log/messages
#StandardOutput=null
PrivateTmp=true
[Install]
WantedBy=multi-user.target

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; /etc/asterisk/extensions.conf
; Remember, context names for each SIP account are specified in pjsip_wizard.conf.
; First, some safeguards against abuse of the built-in contexts.
[default]
exten => _X.,1,Hangup(3)
[globals]
; trunk caller IDs.
TRUNK_ONE_CID = +442012345678
TRUNK_TWO_CID = +442087456210
; Default queue
QUEUE_ONE = queue-one
[subscribe]
exten => _X.,hint,PJSIP/${EXTEN}
[from-trunk-one]
exten => _X.,1,Queue(${QUEUE_ONE},nr,,,50)
; same => n,Answer(500)
; same => n,Voicemail(${VOICEMAIL_BOX},su)
same => n,Hangup()
[outbound-one]
exten => _X.,1,Verbose(1, "OUTBOUND TRUNK ONE ${TRUNK_ONE_CID}")
same => n,Set(CALLERID(all)=${TRUNK_ONE_CID})
same => n,Dial(PJSIP/${EXTEN}@trunk-one)
same => n,GotoIf($["${DIALSTATUS}"="CONGESTION"]?outbound-two,${EXTEN},1)
same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?outbound-two,${EXTEN},1)
same => n,Hangup()
[outbound-two]
exten => _X.,1,Verbose(1, "OUTBOUND TRUNK TWO ${TRUNK_TWO_CID}")
same => n,Set(CALLERID(all)=${TRUNK_TWO_CID})
same => n,Dial(PJSIP/${EXTEN}@trunk-two)
same => n,GotoIf($["${DIALSTATUS}"="CONGESTION"]?default,${EXTEN},1)
same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?default,${EXTEN},1)
same => n,Hangup()
[app-lastcallreturn]
exten => 1471,1,Wait(1)
same => n,Answer(1)
same => n,Playback(last-num-to-call)
same => n,Set(number=${DB(lastcaller/${CALLERID(num)})})
same => n,GotoIf($["${number}" = ""]?s-nonum,1)
same => n,SayDigits(${number})
same => n,Wait(2)
same => n,SayDigits(${number})
same => n,Wait(1)
same => n,Playback(to-call-num-press)
same => n,Playback(digits/1)
exten => 1,1,Goto(outbound-one,${number},1)
exten => s-nonum,1,Playback(unidentified-no-callback)
same => n,Hangup()
[from-ext]
exten => 123,1,goto(default,${EXTEN},1)
same => n,Hangup()
exten => 1471,1,Goto(app-lastcallreturn,${EXTEN},1)
same => n,Hangup()
exten => _X.,1,Dial(PJSIP/outbound-one/${EXTEN})
same => n,GotoIf($["${DIALSTATUS}"="CONGESTION"]?outbound-two,${EXTEN},1)
same => n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?outbound-two,${EXTEN},1)
same => n,Hangup()
[to-ext]
exten => _X.,1,Set(DB(lastcaller/${ARG1})=${CALLERID(num)})
same => n,Dial(PJSIP/${EXTEN})
same => n,Hangup()

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;
; /etc/asterisk/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=uk ; 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
[id]
description = Indonesia
; 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 = 425/250,0/250
callwaiting = 425/150,0/150,425/150,0/10000
info = !950/330,!1400/330,!1800/330,0/1000
[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

View File

@ -1,200 +0,0 @@
;
; /etc/asterisk/logger.conf
;
; 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 = no
;
; 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
;~
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
;~
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
;
; Any custom logging levels you may want to use, which can then
; be sent to logging channels. The maximum number of custom
; levels is 16, but not all of these may be available if modules
; in Asterisk define their own.
;custom_levels = foobar,important,compliance
;
[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.
; - [plain] - The plain formatter, this outputs log messages using a
; human readable format with the addition of function name
; and line number. No color escape codes are ever printed
; nor are verbose messages treated specially.
; - [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 interpreting verbosity
; outside of the logging subsystem.
;
; Log levels include the following, and are specified in a comma delineated
; list:
; debug
; trace
; notice
; warning
; error
; verbose(<level>)
; dtmf
; fax
; security
; <customlevel>
;
; Verbose takes an optional argument, in the form of an integer level. The
; verbose level can be set per logfile. 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.
;
; Debug has multiple levels like verbose. However, it is a system wide setting
; and cannot be specified per logfile. You specify the debug level elsewhere
; such as the CLI 'core set debug 3', starting Asterisk with '-ddd', or in
; asterisk.conf 'debug=3'.
;
; 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. Both
; debug and trace messages are also very verbose and can and do fill up
; logfiles quickly. This is another reason not to have debug or trace
; modes on a production system unless you are in the process of debugging
; a specific issue.
;
;debug.log => error,warning,notice,verbose,debug
;trace.log => trace
;security.log => security
console => notice,warning,error
;console => notice,warning,error,debug
messages.log => [default]notice,warning,error,verbose
;full.log => notice,warning,error,debug,verbose,dtmf,fax
;~
;full.log => [default]notice,warning,error,debug,verbose,dtmf,fax
;
;full-json.log => [json]debug,verbose,notice,warning,error,dtmf,fax
;
;syslog keyword : This special keyword logs to syslog facility
;
;syslog.local0 => notice,warning,error
;
; A log level defined in 'custom_levels' above
;important.log = important

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@ -1,57 +0,0 @@
;
; /etc/asterisk/modules.conf
;
; 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
;~
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: console (portaudio).
; By default, load chan_console only (automatically).
;
;noload = chan_console.so
;
; Do not load res_hep and kin unless you are using HEP monitoring
; <http://sipcapture.org> in your network.
;
noload = res_hep.so
noload = res_hep_pjsip.so
noload = res_hep_rtcp.so
;
; Load one of the voicemail modules as they are mutually exclusive.
; By default, load app_voicemail only (automatically).
;
;noload = app_voicemail.so
noload = app_voicemail_imap.so
noload = app_voicemail_odbc.so

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@ -1,27 +0,0 @@
;
; /etc/asterisk/pjsip.conf
;
; Modifed and "borrowed" (read: stolen) from https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
; This template contains default settings for all transports.
[transport-defaults](!)
type = transport
bind = 0.0.0.0:15000
; For communication to any addresses within local_nets, Asterisk will not apply
; NAT-related workarounds.
local_net = 127.0.0.0/8
local_net = 10.0.0.0/8
local_net = 172.16.0.0/12
local_net = 192.168.0.0/16
; If you have a public static IP for your Asterisk server, set it here.
;external_media_address = xxx.xxx.xxx.xxx
;external_signaling_address = xxx.xxx.xxx.xxx
; The following UDP and TCP transports will inherit from the defaults.
[transport-udp](transport-defaults)
protocol = udp
;[transport-tcp](transport-defaults)
;protocol = tcp

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@ -1,56 +0,0 @@
;
; /etc/asterisk/pjsip_wizard.conf
;
; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
;
; This template contains default settings for all SIP trunks.
[trunk-defaults](!)
type = wizard
; Send media to the address and port on the incoming packet, regardless of what
; the SIP headers say (NAT workaround).
endpoint/rtp_symmetric = yes
; Rewrite the SIP contact to the address and port of the request (NAT workaround).
endpoint/rewrite_contact = yes
; Send the Remote-Party-ID SIP header. Some providers need this.
endpoint/send_rpid = yes
; We use alaw mainly in Europe.
endpoint/allow = !all,alaw
; Call encryption.
endpoint/media_encryption = no
; Load trunks.
#tryinclude pjsip_wizard_trunk_one.conf
;
;
; This template contains default settings for all local extensions.
[extension-defaults](!)
type = wizard
; Require clients to register.
accepts_registrations = yes
; Require clients to authenticate.
accepts_auth = yes
; When simultaneous logins from the same account exceed max_contacts, disconnect
; the oldest session.
aor/remove_existing = yes
; allow just alaw
endpoint/allow = !all,alaw
; Context name for BLF/presence subscriptions. This can be any string of your
; choosing.
endpoint/subscribe_context = subscribe
; Load user extensions.
#tryinclude pjsip_wizard_users.conf

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@ -1,24 +0,0 @@
;
; /etc/asterisk/pjsip_wizard_trunk_one.conf
;
; For a local system to register as a trunk.
;
[trunk-one](trunk-defaults)
; Accept inbound auth for this trunk.
accepts_auth = yes
; UDP transport please.
transport = transport-udp
; What's the username?
inbound_auth/username = trunk-one
; The Password?
inbound_auth/password = BedknobsArentSupposedToWorkThatWayUNLESSYOUREBRAVE
; Choose a context name for incoming calls from this account. You'll use this
; name in your dialplan.
endpoint/context = from-trunk-one
; Only one registration please.
aor/max_contacts = 1

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@ -1,34 +0,0 @@
;
; /etc/asterisk/pjsip_wizard_trunk_two.conf
;
; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
;
[trunk-two](trunk-defaults)
; Require SIP authentication.
sends_auth = yes
; Require SIP registration.
sends_registrations = yes
; If registration fails, keep trying until x tries.
registration/max_retries = 1000
; Don't assume an authentication failure is permanent.
registration/auth_rejection_permanent = no
; Perform a connectivity check every 30 seconds.
aor/qualify_frequency = 30
; UDP transport please.
transport = transport-udp
; Hostname and port for your SIP provider.
remote_hosts = voiceless.aa.net.uk:5060
; Choose a context name for incoming calls from this account. You'll use this
; name in your dialplan.
endpoint/context = from-trunk-two
; Your SIP provider will give you these credentials.
outbound_auth/username = +442012345678
outbound_auth/password = NoIDontHaveAnyMONEY

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@ -1,37 +0,0 @@
; /etc/asterisk/pjsip_wizard_users.conf
;
; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
;
[1000](extension-defaults)
; Dialplan context name for calls originating from this account.
endpoint/context = from-ext
; Voicemail address.
endpoint/mailboxes = 1000@default
; Internal Caller ID string for this device.
endpoint/callerid = 1000 <1000>
; Username for SIP account. By convention, this should be the extension number.
inbound_auth/username = 1000
; Password for SIP account (you can choose whatever password you like).
inbound_auth/password = IfYouTolerateThisThenYourChildrenWillBeNext!
; Maximum number of simultaneous logins for this account.
aor/max_contacts = 1
; Check connectivity every 30 seconds.
aor/qualify_frequency = 30
; Set connectivity check timeout to 3 seconds.
aor/qualify_timeout = 3.0
; IMPORTANT! This setting determines whether the audio stream will be proxied
; through the Asterisk server.
;
; If this device is directly reachable by the internet (either by a publicly
; routable IP, or static port mappings on your router), choose YES.
;
; Otherwise, if this device is hidden behind NAT, choose NO.
endpoint/direct_media = no

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@ -1,49 +0,0 @@
; /etc/asterisk/queues.conf
;
; Global queue settings go in this section.
;
; Modifed and "borrowed" (read: stolen) https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
;
[general]
; Persist dynamic member lists in the astdb.
persistentmembers = yes
; Some options for more intuitive queue behavior.
autofill = yes
monitor-type = MixMonitor
shared_lastcall = yes
log_membername_as_agent = yes
;;;;;;;;;;;;;;;;;;;
; Queue Definitions
;;;;;;;;;;;;;;;;;;;
; The first queue is for incoming calls to our lines.
[queue-one]
; For each incoming call, ring all members of the queue.
strategy = ringall
; Max number of seconds a caller waits in the queue.
timeout = 90
; Don't announce estimated hold time, etc.
announce-frequency = 0
announce-holdtime = no
announce-position = no
periodic-announce-frequency = 0
; Allow ringing even when no queue members are present.
joinempty = yes
leavewhenempty = no
; Ring member phones even when they are on another call.
ringinuse = yes
; Queue Members
;
; Each member is specified with the following format:
; member => INTERFACE,PENALTY,FRIENDLY_NAME,PRESENCE_INTERFACE
;
; The "penalty" value is not interesting for our use case.
; With PJSIP, the BLF/Presence interface is identical to the standard interface name.
member => to-ext/1000,0,1000,to-ext/1000

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@ -1,7 +0,0 @@
;
; /etc/asterisk/rtp.conf
;
[general]
stunaddr = stun.aa.net.uk:3478
stunrefresh = 30

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@ -1,8 +0,0 @@
;
; /etc/asterisk/rtp.conf
;
; Modifed and "borrowed" (read: stolen) from https://www.sacredheartsc.com/blog/building-a-personal-voip-system/
;
[general]
rtpstart=15010
rtpend=15999

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@ -1,47 +0,0 @@
[general]
format = wav49|gsm|wav
[zonemessages]
; 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
;
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'
european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM
uk=Europe/London|'vm-received' a d b 'digits/at' HM
[default]
; Note: The rest of the system must reference mailboxes defined here as mailbox@default.
1000 => 1000,Default Mailbox,,,,Tz=uk