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Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.

Browse Source 
Let's try that again, this time removing trailing whitespace and not leading
whitespace.  I can't believe no one noticed.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
certified/1.8.6
Sean Bright 18 years ago
parent a7d813cae7
commit f22962a0c1
57 changed files with 1730 additions and 1730 deletions
Show Stats Download Patch File Download Diff File

14
configs/agents.conf.sample
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@ -32,14 +32,14 @@ persistentagents=yes
; Define autologoffunavail to have agents automatically logged
; out when the extension that they are at returns a CHANUNAVAIL
; status when a call is attempted to be sent there.
; Default is "no".
; Default is "no".
;
;autologoffunavail=yes
;
; Define ackcall to require a DTMF acknowledgement when
; an agent logs in using agentcallbacklogin. Default is "no".
; Can also be set to "always", which will also require AgentLogin
; agents to acknowledge calls. Use the acceptdtmf option to
; agents to acknowledge calls. Use the acceptdtmf option to
; configure what DTMF key press should be used to acknowledge the
; call. The default is '#'.
;
@ -70,14 +70,14 @@ persistentagents=yes
;
;goodbye => goodbye_file
;
; Define updatecdr. This is whether or not to change the source
; channel in the CDR record for this call to agent/agent_id so
; Define updatecdr. This is whether or not to change the source
; channel in the CDR record for this call to agent/agent_id so
; that we know which agent generates the call
;
;updatecdr=no
;
; Group memberships for agents (may change in mid-file)
;
;
;group=3
;group=1,2
;group=
@ -85,7 +85,7 @@ persistentagents=yes
; --------------------------------------------------
; This section is devoted to recording agent's calls
; The keywords are global to the chan_agent channel driver
;
;
; Enable recording calls addressed to agents. It's turned off by default.
;recordagentcalls=yes
;
@ -100,7 +100,7 @@ persistentagents=yes
; /var/spool/asterisk/monitor
;savecallsin=/var/calls
;
; An optional custom beep sound file to play to always-connected agents.
; An optional custom beep sound file to play to always-connected agents.
;custom_beep=beep
;
; --------------------------------------------------

4
configs/ais.conf.sample
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@ -1,5 +1,5 @@
;
; Sample configuration file for res_ais
; Sample configuration file for res_ais
; * SAForum AIS (Application Interface Specification)
;
; More information on the AIS specification is available from the SAForum.
@ -76,7 +76,7 @@
;
; This example would be used for a node that has phones directly registered
; to it, but does not have direct access to voicemail. So, this node wants
; to be informed about MWI state changes on other voicemail server nodes, but
; to be informed about MWI state changes on other voicemail server nodes, but
; is not capable of publishing any state changes.
;
; [mwi]

8
configs/alarmreceiver.conf.sample
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@ -7,7 +7,7 @@
[general]
;
;
; Specify a timestamp format for the metadata section of the event files
; Default is %a %b %d, %Y @ %H:%M:%S %Z
@ -32,7 +32,7 @@ timestampformat = %a %b %d, %Y @ %H:%M:%S %Z
eventspooldir = /tmp
;
;
; The alarmreceiver app can either log the events one-at-a-time to individual
; files in the spool directory, or it can store them until the caller
; disconnects and write them all to one file.
@ -46,7 +46,7 @@ logindividualevents = no
; The timeout for receiving the first DTMF digit is adjustable from 1000 msec.
; to 10000 msec. The default is 2000 msec. Note: if you wish to test the
; receiver by entering digits manually, set this to a reasonable time out
; like 10000 milliseconds.
; like 10000 milliseconds.
fdtimeout = 2000
@ -54,7 +54,7 @@ fdtimeout = 2000
; The timeout for receiving subsequent DTMF digits is adjustable from
; 110 msec. to 4000 msec. The default is 200 msec. Note: if you wish to test
; the receiver by entering digits manually, set this to a reasonable time out
; like 4000 milliseconds.
; like 4000 milliseconds.
;
sdtimeout = 200

22
configs/alsa.conf.sample
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@ -39,23 +39,23 @@ extension=s
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; ALSA channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The ALSA channel can't accept jitter,
; thus an enabled jitterbuffer on the receive ALSA side will always
; be used if the sending side can create jitter.
; ALSA channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The ALSA channel can't accept jitter,
; thus an enabled jitterbuffer on the receive ALSA side will always
; be used if the sending side can create jitter.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

10
configs/amd.conf.sample
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@ -4,15 +4,15 @@
[general]
initial_silence = 2500 ; Maximum silence duration before the greeting.
; If exceeded then MACHINE.
; If exceeded then MACHINE.
greeting = 1500 ; Maximum length of a greeting. If exceeded then MACHINE.
after_greeting_silence = 800 ; Silence after detecting a greeting.
; If exceeded then HUMAN
; If exceeded then HUMAN
total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide
; on a HUMAN or MACHINE
; on a HUMAN or MACHINE
min_word_length = 100 ; Minimum duration of Voice to considered as a word
between_words_silence = 50 ; Minimum duration of silence after a word to consider
; the audio what follows as a new word
; the audio what follows as a new word
maximum_number_of_words = 3 ; Maximum number of words in the greeting.
; If exceeded then MACHINE
; If exceeded then MACHINE
silence_threshold = 256

180
configs/asterisk.adsi
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@ -35,39 +35,39 @@ DISPLAY "empty" IS "asdf"
; Begin soft key definitions
;
KEY "callfwd" IS "CallFwd" OR "Call Forward"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "*60"
GOTO "offHook"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "*60"
GOTO "offHook"
ENDKEY
KEY "vmail_OH" IS "VMail" OR "Voicemail"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "8500"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "8500"
ENDKEY
KEY "vmail" IS "VMail" OR "Voicemail"
SENDDTMF "8500"
SENDDTMF "8500"
ENDKEY
KEY "backspace" IS "BackSpc" OR "Backspace"
BACKSPACE
BACKSPACE
ENDKEY
KEY "cwdisable" IS "CWDsble" OR "Disable Call Wait"
SENDDTMF "*70"
SETFLAG "nocallwaiting"
SHOWDISPLAY "cwdisabled" AT 4
TIMERCLEAR
TIMERSTART 1
SENDDTMF "*70"
SETFLAG "nocallwaiting"
SHOWDISPLAY "cwdisabled" AT 4
TIMERCLEAR
TIMERSTART 1
ENDKEY
KEY "cidblock" IS "CIDBlk" OR "Block Callerid"
SENDDTMF "*67"
SETFLAG "nocallwaiting"
SENDDTMF "*67"
SETFLAG "nocallwaiting"
ENDKEY
;
@ -75,85 +75,85 @@ ENDKEY
;
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
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
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
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
IFEVENT REORDER THEN
SHOWDISPLAY "callended" AT 2
ENDIF
ENDSUB

22
configs/cdr.conf.sample
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@ -14,12 +14,12 @@
;enable=yes
; Define whether or not to log unanswered calls. Setting this to "yes" will
; report every attempt to ring a phone in dialing attempts, when it was not
; report every attempt to ring a phone in dialing attempts, when it was not
; answered. For example, if you try to dial 3 extensions, and this option is "yes",
; you will get 3 CDR's, one for each phone that was rung. Default is "no". Some
; find this information horribly useless. Others find it very valuable. Note, in "yes"
; mode, you will see one CDR, with one of the call targets on one side, and the originating
; channel on the other, and then one CDR for each channel attempted. This may seem
; channel on the other, and then one CDR for each channel attempted. This may seem
; redundant, but cannot be helped.
;unanswered = no
@ -67,7 +67,7 @@
; Normally, the 'billsec' field logged to the backends (text files or databases)
; is simply the end time (hangup time) minus the answer time in seconds. Internally,
; asterisk stores the time in terms of microseconds and seconds. By setting
; asterisk stores the time in terms of microseconds and seconds. By setting
; initiatedseconds to 'yes', you can force asterisk to report any seconds
; that were initiated (a sort of round up method). Technically, this is
; when the microsecond part of the end time is greater than the microsecond
@ -78,19 +78,19 @@
;
; CHOOSING A CDR "BACKEND" (what kind of output to generate)
;
; To choose a backend, you have to make sure either the right category is
; defined in this file, or that the appropriate config file exists, and has the
; To choose a backend, you have to make sure either the right category is
; defined in this file, or that the appropriate config file exists, and has the
; proper definitions in it. If there are any problems, usually, the entry will
; silently ignored, and you get no output.
;
; Also, please note that you can generate CDR records in as many formats as you
;
; Also, please note that you can generate CDR records in as many formats as you
; wish. If you configure 5 different CDR formats, then each event will be logged
; in 5 different places! In the example config files, all formats are commented
; out except for the cdr-csv format.
;
; Here are all the possible back ends:
;
; csv, custom, manager, odbc, pgsql, radius, sqlite, tds
; csv, custom, manager, odbc, pgsql, radius, sqlite, tds
; (also, mysql is available via the asterisk-addons, due to licensing
; requirements)
; (please note, also, that other backends can be created, by creating
@ -104,7 +104,7 @@
; backend is marked with XXX, you know that the "configure" command could not find
; the required libraries for that option.
;
; To get CDRs to be logged to the plain-jane /var/log/asterisk/cdr-csv/Master.csv
; To get CDRs to be logged to the plain-jane /var/log/asterisk/cdr-csv/Master.csv
; file, define the [csv] category in this file. No database necessary. The example
; config files are set up to provide this kind of output by default.
;
@ -126,7 +126,7 @@
; shows that the modules are available, and the cdr_pgsql.conf file exists, and
; has a [global] section with the proper variables defined.
;
; For logging to radius databases, make sure all the proper libs are installed, that
; For logging to radius databases, make sure all the proper libs are installed, that
; "make menuselect" shows that the modules are available, and the [radius]
; category is defined in this file, and in that section, make sure the 'radiuscfg'
; variable is properly pointing to an existing radiusclient.conf file.
@ -135,7 +135,7 @@
; which is usually /var/log/asterisk. Of course, the proper libraries should be available
; during the 'configure' operation.
;
; For tds logging, make sure the proper libraries are available during the 'configure'
; For tds logging, make sure the proper libraries are available during the 'configure'
; phase, and that cdr_tds.conf exists and is properly set up with a [global] category.
;
; Also, remember, that if you wish to log CDR info to a database, you will have to define

130
configs/chan_dahdi.conf.sample
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@ -6,7 +6,7 @@
; will reload the configuration file, but not all configuration options
; are re-configured during a reload (signalling, as well as PRI and
; SS7-related settings cannot be changed on a reload).
;
;
; This file documents many configuration variables. Normally unless you know
; what a variable means or that it should be changed, there's no reason to
; un-comment those lines.
@ -21,11 +21,11 @@
;
; Trunk groups are used for NFAS or GR-303 connections.
;
; Group: Defines a trunk group.
; Group: Defines a trunk group.
; trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]
;
; trunkgroup is the numerical trunk group to create
; dchannel is the DAHDI channel which will have the
; dchannel is the DAHDI channel which will have the
; d-channel for the trunk.
; backup1 is an optional list of backup d-channels.
;
@ -85,7 +85,7 @@
; example, if you set 'national', you will be unable to dial local or
; international numbers.
;
; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's
; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's
; numbering plan). In North America, the typical use is sending the 10 digit
; callerID number and setting the prilocaldialplan to 'national' (the default).
; Only VERY rarely will you need to change this.
@ -98,12 +98,12 @@
; national: National ISDN
; international: International ISDN
; dynamic: Dynamically selects the appropriate dialplan
; redundant: Same as dynamic, except that the underlying number is not
; redundant: Same as dynamic, except that the underlying number is not
; changed (not common)
;
;pridialplan=unknown
;prilocaldialplan=national
;
;
; pridialplan may be also set at dialtime, by prefixing the dialled number with
; one of the following letters:
; U - Unknown
@ -133,27 +133,27 @@
;
; PRI caller ID prefixes based on the given TON/NPI (dialplan)
; This is especially needed for EuroISDN E1-PRIs
;
;
; None of the prefix settings can be changed on reload.
;
; sample 1 for Germany
; sample 1 for Germany
;internationalprefix = 00
;nationalprefix = 0
;localprefix = 0711
;privateprefix = 07115678
;unknownprefix =
;unknownprefix =
;
; sample 2 for Germany
; sample 2 for Germany
;internationalprefix = +
;nationalprefix = +49
;localprefix = +49711
;privateprefix = +497115678
;unknownprefix =
;unknownprefix =
;
; PRI resetinterval: sets the time in seconds between restart of unused
; B channels; defaults to 'never'.
;
;resetinterval = 3600
;resetinterval = 3600
;
; Overlap dialing mode (sending overlap digits)
; Cannot be changed on a reload.
@ -168,7 +168,7 @@
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to work
; with all telcos.
;
;
; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT
; inband: Signal Busy/Congestion using in-band tones (default)
;
@ -206,7 +206,7 @@
; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
; T308: Wait for RELEASE acknowledge (default 4000 ms)
; T309: Maintain active calls on Layer 2 disconnection (default -1,
; T309: Maintain active calls on Layer 2 disconnection (default -1,
; Asterisk clears calls)
; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s
; May vary in other ISDN standards (Q.931 1993 : 90000 ms)
@ -284,11 +284,11 @@
; (see below). The 'signalling' format specified will be the inbound signalling
; format. If you only specify 'signalling', then it will be the format for
; both inbound and outbound.
;
; outsignalling can only be one of:
;
; outsignalling can only be one of:
; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd,
; featdmf, featdmf_ta, e911, fgccama, fgccamamf
;
;
; outsignalling cannot be changed on a reload.
;
;signalling=featdmf
@ -318,9 +318,9 @@
; None of them will update on a reload.
;
; How long generated tones (DTMF and MF) will be played on the channel
; (in milliseconds).
; (in milliseconds).
;
; This is a global, rather than a per-channel setting. It will not be
; This is a global, rather than a per-channel setting. It will not be
; updated on a reload.
;
;toneduration=100
@ -354,7 +354,7 @@ usecallerid=yes
; What signals the start of caller ID
; ring = a ring signals the start (default)
; polarity = polarity reversal signals the start
; polarity_IN = polarity reversal signals the start, for India,
; polarity_IN = polarity reversal signals the start, for India,
; for dtmf dialtone detection; using DTMF.
; (see doc/India-CID.txt)
;
@ -381,7 +381,7 @@ usecallerid=yes
; fsk,rpas - the FXO line is monitored for MWI FSK spills preceded
; by a ring pulse alert signal.
; neon - The fxo line is monitored for the presence of NEON pulses
; indicating MWI.
; indicating MWI.
; When detected, an internal Asterisk MWI event is generated so that any other
; part of Asterisk that cares about MWI state changes is notified, just as if
; the state change came from app_voicemail.
@ -432,7 +432,7 @@ usecallingpres=yes
;
; Some countries (UK) have ring tones with different ring tones (ring-ring),
; which means the caller ID needs to be set later on, and not just after
; the first ring, as per the default (1).
; the first ring, as per the default (1).
;
;sendcalleridafter = 2
;
@ -472,10 +472,10 @@ cancallforward=yes
;
callreturn=yes
;
; Stutter dialtone support: If a mailbox is specified without a voicemail
; context, then when voicemail is received in a mailbox in the default
; Stutter dialtone support: If a mailbox is specified without a voicemail
; context, then when voicemail is received in a mailbox in the default
; voicemail context in voicemail.conf, taking the phone off hook will cause a
; stutter dialtone instead of a normal one.
; stutter dialtone instead of a normal one.
;
; If a mailbox is specified *with* a voicemail context, the same will result
; if voicemail received in mailbox in the specified voicemail context.
@ -486,9 +486,9 @@ callreturn=yes
;
; for any other voicemail context, the following will produce the stutter tone:
;
;mailbox=1234@context
;mailbox=1234@context
;
; Enable echo cancellation
; Enable echo cancellation
; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
; actually set the number of taps of cancellation.
;
@ -552,7 +552,7 @@ echocancelwhenbridged=yes
;
; There are several independent gain settings:
; rxgain: gain for the rx (receive - into Asterisk) channel. Default: 0.0
; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel.
; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel.
; Default: 0.0
; cid_rxgain: set the gain just for the caller ID sounds Asterisk
; emits. Default: 5.0 .
@ -581,9 +581,9 @@ pickupgroup=1
; Channel variable to be set for all calls from this channel
;setvar=CHANNEL=42
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended transfer.
; cause the given audio file to
; be played upon completion of
; an attended transfer.
;
; Specify whether the channel should be answered immediately or if the simple
@ -600,10 +600,10 @@ pickupgroup=1
;
; caller ID can be set to "asreceived" or a specific number if you want to
; override it. Note that "asreceived" only applies to trunk interfaces.
; fullname sets just the
; fullname sets just the
;
; fullname: sets just the name part.
; cid_number: sets just the number part:
; cid_number: sets just the number part:
;
;callerid = 123456
;
@ -642,7 +642,7 @@ pickupgroup=1
;smdiport=/dev/ttyS0
;
; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D
; etc, it can be useful to perform busy detection either in an effort to
; etc, it can be useful to perform busy detection either in an effort to
; detect hangup or for detecting busies. This enables listening for
; the beep-beep busy pattern.
;
@ -685,8 +685,8 @@ pickupgroup=1
;
;hanguponpolarityswitch=yes
;
; polarityonanswerdelay: minimal time period (ms) between the answer
; polarity switch and hangup polarity switch.
; polarityonanswerdelay: minimal time period (ms) between the answer
; polarity switch and hangup polarity switch.
; (default: 600ms)
;
; On trunk interfaces (FXS) it can be useful to attempt to follow the progress
@ -699,7 +699,7 @@ pickupgroup=1
; with "progzone".
;
; progzone also affects the pattern used for buzydetect (unless
; busypattern is set explicitly). The possible values are:
; busypattern is set explicitly). The possible values are:
; us (default)
; ca (alias for 'us')
; cr (Costa Rica)
@ -741,7 +741,7 @@ pickupgroup=1
;faxdetect=no
;
; When 'faxdetect' is used, one could use 'faxbuffers' to configure the DAHDI
; transmit buffer policy. The default is *OFF*. When this configuration
; transmit buffer policy. The default is *OFF*. When this configuration
; option is used, the faxbuffer policy will be used for the life of the call
; after a fax tone is detected. The faxbuffer policy is reverted after the
; call is torn down. The sample below will result in 6 buffers and a full
@ -792,23 +792,23 @@ pickupgroup=1
;
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The DAHDI channel can't accept jitter,
; thus an enabled jitterbuffer on the receive DAHDI side will always
; be used if the sending side can create jitter.
; DAHDI channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The DAHDI channel can't accept jitter,
; thus an enabled jitterbuffer on the receive DAHDI side will always
; be used if the sending side can create jitter.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a DAHDI
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
@ -834,7 +834,7 @@ pickupgroup=1
; parameters that were specified above its declaration.
;
; For GR-303, CRV's are created like channels except they must start with the
; trunk group followed by a colon, e.g.:
; trunk group followed by a colon, e.g.:
;
; crv => 1:1
; crv => 2:1-2,5-8
@ -908,15 +908,15 @@ pickupgroup=1
; A range of -1 will force it to always match.
; Anything lower than -1 would presumably cause it to never match.
;
;dring1=95,0,0
;dring1context=internal1
;dring1=95,0,0
;dring1context=internal1
;dring1range=10
;dring2=325,95,0
;dring2context=internal2
;dring2=325,95,0
;dring2context=internal2
;dring2range=10
; If no pattern is matched here is where we go.
;context=default
;channel => 1
;channel => 1
; ---------------- Options for use with signalling=ss7 -----------------
; None of them can be changed by a reload.
@ -945,12 +945,12 @@ pickupgroup=1
;
;ss7_calling_nai=dynamic
;
;
; sample 1 for Germany
;
; sample 1 for Germany
;ss7_internationalprefix = 00
;ss7_nationalprefix = 0
;ss7_subscriberprefix =
;ss7_unknownprefix =
;ss7_subscriberprefix =
;ss7_unknownprefix =
;
; This option is used to disable automatic sending of ACM when the call is started
@ -1056,7 +1056,7 @@ pickupgroup=1
; 'stack' is for very verbose output of the channel and context call stack, only useful
; if you are debugging a crash or want to learn how the library works. The stack logging
; will be only enabled if the openr2 library was compiled with -DOR2_TRACE_STACKS
; You can mix up values, like: loglevel=error,debug,mf to log just error, debug and
; You can mix up values, like: loglevel=error,debug,mf to log just error, debug and
; multi frequency messages
; 'all' is a special value to log all the activity
; 'nothing' is a clean-up value, in case you want to not log any activity for
@ -1110,20 +1110,20 @@ pickupgroup=1
; You most likely dont need this feature. Default is yes.
; When this is set to yes, all calls that are offered (incoming calls) which
; DNIS is valid (exists in extensions.conf) and pass collect call validation
; DNIS is valid (exists in extensions.conf) and pass collect call validation
; will be accepted with a Group B tone (either call with charge or not, depending on mfcr2_charge_calls)
; with this set to 'no' then the call will NOT be accepted on offered, and the call will start its
; execution in extensions.conf without being accepted until the channel is answered (either with Answer() or
; any other application resulting in the channel being answered).
; any other application resulting in the channel being answered).
; This can be set to 'no' if your telco or PBX needs the hangup cause to be set accurately
; when this option is set to no you must explicitly accept the call with DAHDIAcceptR2Call
; or implicitly through the Answer() application.
; or implicitly through the Answer() application.
; mfcr2_accept_on_offer=yes
; WARNING: advanced users only! I really mean it
; this parameter is commented by default because
; YOU DON'T NEED IT UNLESS YOU REALLY GROK MFC/R2
; READ COMMENTS on doc/r2proto.conf in openr2 package
; READ COMMENTS on doc/r2proto.conf in openr2 package
; for more info
; mfcr2_advanced_protocol_file=/path/to/r2proto.conf
@ -1171,7 +1171,7 @@ pickupgroup=1
; chan_dahdi.conf and [general] in users.conf - one section's configuration
; does not affect another one's.
;
; Instead of letting common configuration values "slide through" you can
; Instead of letting common configuration values "slide through" you can
; use configuration templates to easily keep the common part in one
; place and override where needed.
;

10
configs/cli_aliases.conf.sample
Unescape View File

@ -13,8 +13,8 @@ template = friendly ; By default, include friendly aliases
;template = asterisk12 ; Asterisk 1.2 style syntax
;template = asterisk14 ; Asterisk 1.4 style syntax
;template = individual_custom ; see [individual_custom] example below which
; includes a list of aliases from an external
; file
; includes a list of aliases from an external
; file
; Because the Asterisk CLI syntax follows a "module verb argument" syntax,
@ -70,7 +70,7 @@ pri intense debug span=pri set debug 2 span
; 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
; [general] section
[asterisk12_tpl](!)
show channeltypes=core show channeltypes
@ -92,7 +92,7 @@ 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 hints=core show hints
show globals=core show globals
show function=core show function
show application=core show application
@ -102,7 +102,7 @@ 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
show codec=core show codec
moh classes show=moh show classes
moh files show=moh show files
agi no debug=agi debug off

2
configs/cli_permissions.conf.sample
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@ -23,7 +23,7 @@
[general]
default_perm=permit ; To leave asterisk working as normal
; we should set this parameter to 'permit'
; we should set this parameter to 'permit'
;
; Follows the per-users permissions configs.
;

36
configs/console.conf.sample
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@ -5,7 +5,7 @@
[general]
; Set this option to "yes" to enable automatically answering calls on the
; console. This is very useful if the console is used as an intercom.
; console. This is very useful if the console is used as an intercom.
; The default value is "no".
;
;autoanswer = no
@ -21,7 +21,7 @@
;extension = s
; Set the default CallerID for created channels.
;
;
;callerid = MyName Here <(256) 428-6000>
; Set the default language for created channels.
@ -34,7 +34,7 @@
; The default is "no".
;
;overridecontext = no ; if 'no', the last @ will start the context
; if 'yes' the whole string is an extension.
; if 'yes' the whole string is an extension.
; Default Music on Hold class to use when this channel is placed on hold in
@ -46,23 +46,23 @@
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; Console channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The Console channel can't accept jitter,
; thus an enabled jitterbuffer on the receive Console side will always
; be used if the sending side can create jitter.
; Console channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The Console channel can't accept jitter,
; thus an enabled jitterbuffer on the receive Console side will always
; be used if the sending side can create jitter.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a Console
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
@ -76,8 +76,8 @@
[default]
input_device = default ; When configuring an input device and output device,
output_device = default ; use the name that you see when you run the "console
; list available" CLI command. If you say "default", the
; system default input and output devices will be used.
; list available" CLI command. If you say "default", the
; system default input and output devices will be used.
autoanswer = no
context = default
extension = s
@ -86,5 +86,5 @@ language = en
overridecontext = no
mohinterpret = default
active = yes ; This option should only be set for one console.
; It means that it is the active console to be
; used from the Asterisk CLI.
; It means that it is the active console to be
; used from the Asterisk CLI.

4
configs/dnsmgr.conf.sample
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@ -1,5 +1,5 @@
[general]
;enable=yes ; enable creation of managed DNS lookups
; default is 'no'
; default is 'no'
;refreshinterval=1200 ; refresh managed DNS lookups every <n> seconds
; default is 300 (5 minutes)
; default is 300 (5 minutes)

26
configs/dundi.conf.sample
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@ -1,6 +1,6 @@
;
; DUNDi configuration file
;
;
; For more information about DUNDi, see http://www.dundi.com
;
;
@ -50,9 +50,9 @@
ttl=32
;
; If we don't get ACK to our DPDISCOVER within 2000ms, and autokill is set
; to yes, then we cancel the whole thing (that's enough time for one
; to yes, then we cancel the whole thing (that's enough time for one
; retransmission only). This is used to keep things from stalling for a long
; time for a host that is not available, but would be ill advised for bad
; time for a host that is not available, but would be ill advised for bad
; connections. In addition to 'yes' or 'no' you can also specify a number
; of milliseconds. See 'qualify' for individual peers to turn on for just
; a specific peer.
@ -60,7 +60,7 @@ ttl=32
autokill=yes
;
; pbx_dundi creates a rotating key called "secret", under the family
; 'secretpath'. The default family is dundi (resulting in
; 'secretpath'. The default family is dundi (resulting in
; the key being held at dundi/secret).
;
;secretpath=dundi
@ -78,8 +78,8 @@ autokill=yes
;
; The "mappings" section maps DUNDi contexts
; to contexts on the local asterisk system. Remember
; that numbers that are made available under the e164
; DUNDi context are regulated by the DUNDi General Peering
; that numbers that are made available under the e164
; DUNDi context are regulated by the DUNDi General Peering
; Agreement (GPA) if you are a member of the DUNDi E.164
; Peering System.
;
@ -108,14 +108,14 @@ autokill=yes
;
; Further options may include:
;
; nounsolicited: No unsolicited calls of any type permitted via this
; nounsolicited: No unsolicited calls of any type permitted via this
; route
; nocomunsolicit: No commercial unsolicited calls permitted via
; nocomunsolicit: No commercial unsolicited calls permitted via
; this route
; residential: This number is known to be a residence
; commercial: This number is known to be a business
; mobile: This number is known to be a mobile phone
; nocomunsolicit: No commercial unsolicited calls permitted via
; nocomunsolicit: No commercial unsolicited calls permitted via
; this route
; nopartial: Do not search for partial matches
;
@ -163,7 +163,7 @@ autokill=yes
;
; host - What their host is
;
; order - What search order to use. May be 'primary', 'secondary',
; order - What search order to use. May be 'primary', 'secondary',
; 'tertiary' or 'quartiary'. In large systems, it is beneficial
; to only query one up-stream host in order to maximize caching
; value. Adding one with primary and one with secondary gives you
@ -187,7 +187,7 @@ autokill=yes
; the local system. Set "all" to deny this host to
; lookup all contexts.
;
; model - inbound, outbound, or symmetric for whether we receive
; model - inbound, outbound, or symmetric for whether we receive
; requests only, transmit requests only, or do both.
;
; precache - Utilize/Permit precaching with this peer (to pre
@ -241,7 +241,7 @@ autokill=yes
;inkey = littleguy
;outkey = ourkey
;include = e164 ; In this case used only for precaching
;permit = e164
;permit = e164
;qualify = yes
;
@ -254,7 +254,7 @@ autokill=yes
;register = yes
;inkey = dhcp34
;permit = all ; In this case used only for precaching
;include = all
;include = all
;qualify = yes
;outkey=foo

2
configs/extconfig.conf.sample
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@ -7,7 +7,7 @@
;
[settings]
;
; Static configuration files:
; Static configuration files:
;
; file.conf => driver,database[,table]
;

420
configs/extensions.ael.sample
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@ -3,49 +3,49 @@
//
//
// 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
// 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)}
// ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
// Unix/Linux environmental variables are 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 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
};
//
// Any category other than "General" and "Globals" represent
// extension contexts, which are collections of extensions.
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 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
};
//
// 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 '_'
@ -56,12 +56,12 @@ TRUNKMSD=1; // MSD digits to strip (usually 1 or 0)
// 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
// . - 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,
// 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.
//
@ -72,8 +72,8 @@ TRUNKMSD=1; // MSD digits to strip (usually 1 or 0)
// 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
// 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
@ -87,11 +87,11 @@ TRUNKMSD=1; // MSD digits to strip (usually 1 or 0)
// exten-name => {
// application(arg1,arg2,...);
//
// Timing list for includes is
// Timing list for includes is
//
// <time range>|<days of week>|<days of month>|<months>
//
// includes {
// includes {
// daytime|9:00-17:00|mon-fri|*|*;
// };
//
@ -110,73 +110,73 @@ TRUNKMSD=1; // MSD digits to strip (usually 1 or 0)
//
//
context ael-dundi-e164-canonical {
//
// List canonical entries here
//
// 12564286000 => &ael-std-exten(6000,IAX2/foo);
// _125642860XX => Dial(IAX2/otherbox/${EXTEN:7});
//
// List canonical entries here
//
// 12564286000 => &ael-std-exten(6000,IAX2/foo);
// _125642860XX => Dial(IAX2/otherbox/${EXTEN:7});
};
context ael-dundi-e164-customers {
//
// If you are an ITSP or Reseller, list your customers here.
//
//_12564286000 => Dial(SIP/customer1);
//_12564286001 => Dial(IAX2/customer2);
//
// If you are an ITSP or Reseller, list your customers here.
//
//_12564286000 => Dial(SIP/customer1);
//_12564286001 => Dial(IAX2/customer2);
};
context ael-dundi-e164-via-pstn {
//
// If you are freely delivering calls to the PSTN, list them here
//
//_1256428XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Expose all of 256-428
//_1256325XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Ditto for 256-325
//
// If you are freely delivering calls to the PSTN, list them here
//
//_1256428XXXX => Dial(DAHDI/G2/${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 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
//
//
// Just a wrapper for the switch
//
switches {
DUNDi/e164;
};
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;
};
//
//
// 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.
// 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;
goto ${exten}|1;
return;
};
//
@ -186,7 +186,7 @@ return;
// up, please go to www.gnophone.com or www.iaxtel.com
//
context ael-iaxtel700 {
_91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel);
_91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel);
};
//
@ -196,99 +196,99 @@ _91700XXXXXXX => Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel);
// to be on-line or else dialing can be severly delayed.
//
context ael-iaxprovider {
switches {
// IAX2/user:[key]@myserver/mycontext;
};
switches {
// IAX2/user:[key]@myserver/mycontext;
};
};
context ael-trunkint {
//
// International long distance through trunk
//
includes {
ael-dundi-e164-lookup;
};
_9011. => {
&ael-dundi-e164(${EXTEN:4});
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
//
// International long distance through trunk
//
includes {
ael-dundi-e164-lookup;
};
_9011. => {
&ael-dundi-e164(${EXTEN:4});
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
};
context ael-trunkld {
//
// Long distance context accessed through trunk
//
includes {
ael-dundi-e164-lookup;
};
_91NXXNXXXXXX => {
&ael-dundi-e164(${EXTEN:1});
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
//
// Long distance context accessed through trunk
//
includes {
ael-dundi-e164-lookup;
};
_91NXXNXXXXXX => {
&ael-dundi-e164(${EXTEN:1});
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
};
context ael-trunklocal {
//
// Local seven-digit dialing accessed through trunk interface
//
_9NXXXXXX => {
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
//
// Local seven-digit dialing accessed through trunk interface
//
_9NXXXXXX => {
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
};
context ael-trunktollfree {
//
// Long distance context accessed through trunk interface
//
//
// Long distance context accessed through trunk interface
//
_91800NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91888NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91877NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91866NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91800NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91888NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91877NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
_91866NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
};
context ael-international {
//
// Master context for international long distance
//
ignorepat => 9;
includes {
ael-longdistance;
ael-trunkint;
};
//
// Master context for international long distance
//
ignorepat => 9;
includes {
ael-longdistance;
ael-trunkint;
};
};
context ael-longdistance {
//
// Master context for long distance
//
ignorepat => 9;
includes {
ael-local;
ael-trunkld;
};
//
// Master context for long distance
//
ignorepat => 9;
includes {
ael-local;
ael-trunkld;
};
};
context ael-local {
//
// Master context for local, toll-free, and iaxtel calls only
//
ignorepat => 9;
includes {
ael-default;
ael-trunklocal;
ael-iaxtel700;
ael-trunktollfree;
ael-iaxprovider;
};
//
// Master context for local, toll-free, and iaxtel calls only
//
ignorepat => 9;
includes {
ael-default;
ael-trunklocal;
ael-iaxtel700;
ael-trunktollfree;
ael-iaxprovider;
};
};
//
// You can use an alternative switch type as well, to resolve
// extensions that are not known here, for example with remote
// extensions that are not known here, for example with remote
// IAX switching you transparently get access to the remote
// Asterisk PBX
//
//
// switch => IAX2/user:password@bigserver/local
//
// An "lswitch" is like a switch but is literal, in that
@ -306,69 +306,69 @@ ael-iaxprovider;
macro ael-std-exten-ael( ext , dev ) {
Dial(${dev}/${ext},20);
switch(${DIALSTATUS}) {
case BUSY:
Voicemail(${ext},b);
break;
default:
Voicemail(${ext},u);
};
catch a {
VoiceMailMain(${ext});
return;
};
return;
Dial(${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);
s => {
Wait(1);
Answer();
Set(TIMEOUT(digit)=5);
Set(TIMEOUT(response)=10);
restart:
Background(demo-congrats);
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);
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);
};
@ -380,12 +380,12 @@ i => Playback(invalid);
context ael-default {
// By default we include the demo. In a production system, you
// By default we include the demo. In a production system, you
// probably don't want to have the demo there.
includes {
ael-demo;
};
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
@ -443,6 +443,6 @@ ael-demo;
// friendly Asterisk CLI prompt.
//
// 'show application <command>' will show details of how you
// use that particular application in this file, the dial plan.
// use that particular application in this file, the dial plan.
//
}

74
configs/extensions.conf.sample
Unescape View File

@ -1,21 +1,21 @@
; 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
; 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.
; 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.
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
@ -30,8 +30,8 @@ writeprotect=no
; 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
; 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
@ -41,7 +41,7 @@ writeprotect=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.
; 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
@ -49,15 +49,15 @@ writeprotect=no
; 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
; Basically, the new algorithm provides a flat response
; time, no matter the number of extensions.
;
; By default, the old pattern matcher is used.
; 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 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
; 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
@ -69,7 +69,7 @@ writeprotect=no
;
;extenpatternmatchnew=no
;
; If clearglobalvars is set, global variables will be cleared
; 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
@ -108,7 +108,7 @@ clearglobalvars=no
;#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
; 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.
@ -153,8 +153,8 @@ TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
; yourself a ton of grief.
; WARNING WARNING WARNING WARNING
;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
; 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 '_'
@ -165,12 +165,12 @@ TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
; 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
; . - 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,
; 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.
;
@ -197,7 +197,7 @@ TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;[context]
;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
;
; Timing list for includes is
; Timing list for includes is
;
; <time range>,<days of week>,<days of month>,<months>[,<timezone>]
;
@ -246,7 +246,7 @@ TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;
; If you are freely delivering calls to the PSTN, list them here
;
;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325
[dundi-e164-local]
@ -272,15 +272,15 @@ switch => DUNDi/e164
include => dundi-e164-local
include => dundi-e164-switch
;
; DUNDi can also be implemented as a Macro instead of using
; the Local channel driver.
; 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,
; In macros, it is the start extension. In most other cases,
; you have to goto "s" to execute that extension.
;
; For wildcard matches, see above - all pattern matches start with
@ -367,10 +367,10 @@ include => iaxprovider
include => parkedcalls
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; 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
@ -388,7 +388,7 @@ include => parkedcalls
[macro-trunkdial]
;
; Standard trunk dial macro (hangs up on a dialstatus that should
; Standard trunk dial macro (hangs up on a dialstatus that should
; terminate call)
; ${ARG1} - What to dial
;
@ -430,7 +430,7 @@ exten => stdexten-NOANSWER,n,NoOp(Finish stdexten NOANSWER)
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
exten => stdexten-BUSY,1,Voicemail(${mbx},b)
; If busy, send to voicemail w/ busy announce
; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,NoOp(Finish stdexten BUSY)
exten => stdexten-BUSY,n,Return() ; If they press #, return to start
@ -458,8 +458,8 @@ 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(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
; option (or use P for databased call _X.creening)
exten => _X.,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
; option (or use P for databased call _X.creening)
exten => _X.,n,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
@ -520,8 +520,8 @@ 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 => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,n,Gosub(stdexten(1234,${GLOBAL(CONSOLE)}))
exten => 1234,n,Goto(default,s,1) ; exited Voicemail
@ -583,7 +583,7 @@ exten => 8500,n,Goto(s,6)
;
; The page context calls up the page macro that sets variables needed for auto-answer
; It is in is own context to make calling it from the Page() application as simple as
; It is in is own context to make calling it from the Page() application as simple as
; Local/{peername}@page
;
[page]
@ -610,7 +610,7 @@ exten => _X.,1,Macro(page,SIP/${EXTEN})
[default]
;
; By default we include the demo. In a production system, you
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => demo
@ -640,11 +640,11 @@ include => demo
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}
;exten => 6275,1,Gosub(stdexten(6275,${MARK}))
; assuming ${MARK} is something like DAHDI/2
; assuming ${MARK} is something like DAHDI/2
;exten => 6275,n,Goto(default,s,1) ; exited Voicemail
;exten => mark,1,Goto(6275,1) ; alias mark to 6275
;exten => 6536,1,Gosub(stdexten(6236,${WIL}))
; Ditto for wil
; Ditto for wil
;exten => 6536,n,Goto(default,s,1) ; exited Voicemail
;exten => wil,1,Goto(6236,1)
@ -723,7 +723,7 @@ include => demo
; friendly Asterisk CLI prompt.
;
; "core show application <command>" will show details of how you
; use that particular application in this file, the dial plan.
; 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.

174
configs/extensions.lua.sample
Unescape View File

@ -22,20 +22,20 @@ TRUNKMSD = 1
-- 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
-- . - 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.
--
--
-- If your extension has special characters in it such as '.' and '!' you must
-- explicitly make it a string in the tabale definition:
--
@ -44,7 +44,7 @@ TRUNKMSD = 1
--
-- There are no priorities. All extensions to asterisk appear to have a single
-- priority as if they consist of a single priority.
--
--
-- Each context is defined as a table in the extensions table. The
-- context names should be strings.
--
@ -52,7 +52,7 @@ TRUNKMSD = 1
-- extension. This extension should be set to a table containing a list
-- of context names. Do not put references to tables in the includes
-- table.
--
--
-- include = {"a", "b", "c"};
--
-- Channel variables can be accessed thorugh the global 'channel' table.
@ -79,7 +79,7 @@ TRUNKMSD = 1
-- Also notice the absence of the following constructs from the examples above:
-- channel.func_name(1,2,3) = "value" -- this will NOT work
-- value = channel.func_name(1,2,3) -- this will NOT work as expected
--
--
--
-- Dialplan applications can be accessed through the global 'app' table.
--
@ -97,103 +97,103 @@ TRUNKMSD = 1
--
function outgoing_local(c, e)
app.dial("DAHDI/1/" .. e, "", "")
app.dial("DAHDI/1/" .. e, "", "")
end
function demo_instruct()
app.background("demo-instruct")
app.waitexten()
app.background("demo-instruct")
app.waitexten()
end
function demo_congrats()
app.background("demo-congrats")
demo_instruct()
app.background("demo-congrats")
demo_instruct()
end
-- Answer the chanel and play the demo sound files
function demo_start(context, exten)
app.wait(1)
app.answer()
app.wait(1)
app.answer()
channel.TIMEOUT("digit"):set(5)
channel.TIMEOUT("response"):set(10)
-- app.set("TIMEOUT(digit)=5")
-- app.set("TIMEOUT(response)=10")
channel.TIMEOUT("digit"):set(5)
channel.TIMEOUT("response"):set(10)
-- app.set("TIMEOUT(digit)=5")
-- app.set("TIMEOUT(response)=10")
demo_congrats(context, exten)
demo_congrats(context, exten)
end
function demo_hangup()
app.playback("demo-thanks")
app.hangup()
app.playback("demo-thanks")
app.hangup()
end
extensions = {
demo = {
s = demo_start;
["2"] = function()
app.background("demo-moreinfo")
demo_instruct()
end;
["3"] = function ()
channel.LANGUAGE():set("fr") -- set the language to french
demo_congrats()
end;
["1000"] = function()
app.goto("default", "s", 1)
end;
["1234"] = function()
app.playback("transfer", "skip")
-- do a dial here
end;
["1235"] = function()
app.voicemail("1234", "u")
end;
["1236"] = function()
app.dial("Console/dsp")
app.voicemail(1234, "b")
end;
["#"] = demo_hangup;
t = demo_hangup;
i = function()
app.playback("invalid")
demo_instruct()
end;
["500"] = function()
app.playback("demo-abouttotry")
app.dial("IAX2/guest@misery.digium.com/s@default")
app.playback("demo-nogo")
demo_instruct()
end;
["600"] = function()
app.playback("demo-echotest")
app.echo()
app.playback("demo-echodone")
demo_instruct()
end;
["8500"] = function()
app.voicemailmain()
demo_instruct()
end;
};
default = {
-- by default, do the demo
include = {"demo"};
};
["local"] = {
["_NXXXXXX"] = outgoing_local;
};
demo = {
s = demo_start;
["2"] = function()
app.background("demo-moreinfo")
demo_instruct()
end;
["3"] = function ()
channel.LANGUAGE():set("fr") -- set the language to french
demo_congrats()
end;
["1000"] = function()
app.goto("default", "s", 1)
end;
["1234"] = function()
app.playback("transfer", "skip")
-- do a dial here
end;
["1235"] = function()
app.voicemail("1234", "u")
end;
["1236"] = function()
app.dial("Console/dsp")
app.voicemail(1234, "b")
end;
["#"] = demo_hangup;
t = demo_hangup;
i = function()
app.playback("invalid")
demo_instruct()
end;
["500"] = function()
app.playback("demo-abouttotry")
app.dial("IAX2/guest@misery.digium.com/s@default")
app.playback("demo-nogo")
demo_instruct()
end;
["600"] = function()
app.playback("demo-echotest")
app.echo()
app.playback("demo-echodone")
demo_instruct()
end;
["8500"] = function()
app.voicemailmain()
demo_instruct()
end;
};
default = {
-- by default, do the demo
include = {"demo"};
};
["local"] = {
["_NXXXXXX"] = outgoing_local;
};
}

4
configs/extensions_minivm.conf.sample
Unescape View File

@ -1,4 +1,4 @@
; MINI-VOICEMAIL dialplan example
; MINI-VOICEMAIL dialplan example
; ---------------------------------------------------------------------------------------
; ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
;
@ -10,7 +10,7 @@
; 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})

50
configs/features.conf.sample
Unescape View File

@ -5,52 +5,52 @@
[general]
parkext => 700 ; What extension to dial to park (all parking lots)
parkpos => 701-720 ; What extensions to park calls on. (defafult parking lot)
; These needs to be numeric, as Asterisk starts from the start position
; and increments with one for the next parked call.
; These needs to be numeric, as Asterisk starts from the start position
; and increments with one for the next parked call.
context => parkedcalls ; Which context parked calls are in (default parking lot)
;parkinghints = no ; Add hints priorities automatically for parking slots (default is no).
;parkingtime => 45 ; Number of seconds a call can be parked for
; (default is 45 seconds)
;parkingtime => 45 ; Number of seconds a call can be parked for
; (default is 45 seconds)
;comebacktoorigin = yes ; Whether to return to the original calling extension upon parking
; timeout or to send the call to context 'parkedcallstimeout' at
; extension 's', priority '1' (default is yes).
;courtesytone = beep ; Sound file to play to the parked caller
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
; timeout or to send the call to context 'parkedcallstimeout' at
; extension 's', priority '1' (default is yes).
;courtesytone = beep ; Sound file to play to the parked caller
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
;parkedplay = caller ; Who to play the courtesy tone to when picking up a parked call
; one of: parked, caller, both (default is caller)
; 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)
; 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)
; 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)
; one of: callee, caller, both, no (default is no)
;parkedcallrecording = caller ; Enables or disables DTMF based one-touch recording when picking up a parked call.
; one of: callee, caller, both, no (default is no)
; one of: callee, caller, both, no (default is no)
;adsipark = yes ; if you want ADSI parking announcements
;findslot => next ; Continue to the 'next' free parking space.
; Defaults to 'first' available
;findslot => next ; Continue to the 'next' free parking space.
; Defaults to 'first' 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
; as long as the class is not set on the channel directly
; using Set(CHANNEL(musicclass)=whatever) in the dialplan
;transferdigittimeout => 3 ; Number of seconds to wait between digits when transferring a call
; (default is 3 seconds)
; (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)
;featuredigittimeout = 1000 ; Max time (ms) between digits for
; feature activation (default is 1000 ms)
;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 transferred
; caller is connected, then by default, the system will try to call back the
; person that did the transfer. If this is set to "yes", the callback will
; not be attempted and the transfer will just fail.
; caller is connected, then by default, the system will try to call back the
; person that did the transfer. If this is set to "yes", the callback will
; not be attempted and the transfer will just fail.
;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.
; By default, this is 2.
; 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

12
configs/festival.conf.sample
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@ -15,9 +15,9 @@
;
;usecache=yes
;
; If usecache=yes, a directory to store waveform cache files.
; 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).
; 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/
;
@ -25,10 +25,10 @@
;
; 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.
;
; %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
;
;

8
configs/followme.conf.sample
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@ -29,7 +29,7 @@ pls_hold_prompt=>followme/pls-hold-while-try
status_prompt=>followme/status
; The global default for 'The party you're calling isn't at their desk' message.
;
sorry_prompt=>followme/sorry
sorry_prompt=>followme/sorry
; The global default for 'I'm sorry, but we were unable to locate your party' message.
;
;
@ -41,9 +41,9 @@ context=>default
number=>01233456,25
; The a follow-me number to call. The format is:
; number=> <number to call[&2nd #[&3rd #]]> [, <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
; 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
; 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.
; It's also important to note that the timeout value is not the same
@ -79,7 +79,7 @@ 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
sorry_prompt=>followme/sorry
; The 'I'm sorry, but we were unable to locate your party' message prompt. Default
; is the global default.

8
configs/func_odbc.conf.sample
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@ -76,10 +76,10 @@ readsql=${ARG1}
; ODBC_ANTIGF - A blacklist.
[ANTIGF]
dsn=mysql1,mysql2 ; Use mysql1 as the primary handle, but fall back to mysql2
; if mysql1 is down. Supports up to 5 comma-separated
; DSNs. "dsn" may also be specified as "readhandle" and
; "writehandle", if it is important to separate reads and
; writes to different databases.
; if mysql1 is down. Supports up to 5 comma-separated
; DSNs. "dsn" may also be specified as "readhandle" and
; "writehandle", if it is important to separate reads and
; writes to different databases.
readsql=SELECT COUNT(*) FROM exgirlfriends WHERE callerid='${SQL_ESC(${ARG1})}'
syntax=<callerid>
synopsis=Check if a specified callerid is contained in the ex-gf database

12
configs/gtalk.conf.sample
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@ -2,19 +2,19 @@
;context=default ;;Context to dump call into
;bindaddr=0.0.0.0 ;;Address to bind to
;allowguest=yes ;;Allow calls from people not in
;;list of peers
;;list of peers
;
;[guest] ;;special account for options on guest account
;disallow=all
;disallow=all
;allow=ulaw
;context=guest
;
;[ogorman]
;username=ogorman@gmail.com ;;username of the peer your
;;calling or accepting calls from
;username=ogorman@gmail.com ;;username of the peer your
;;calling or accepting calls from
;disallow=all
;allow=ulaw
;context=default
;context=default
;connection=asterisk ;;client or component in jabber.conf
;;for the call to leave on.
;;for the call to leave on.
;

48
configs/h323.conf.sample
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@ -44,7 +44,7 @@ port = 1720
; or
;dtmfmode=cisco:121
;
; Set the gatekeeper
; Set the gatekeeper
; DISCOVER - Find the Gk address using multicast
; DISABLE - Disable the use of a GK
; <IP address> or <Host name> - The acutal IP address or hostname of your GK
@ -70,9 +70,9 @@ port = 1720
;
;UserByAlias=no
;
; Default context gets used in siutations where you are using
; the GK routed model or no type=user was found. This gives you
; the ability to either play an invalid message or to simply not
; Default context gets used in siutations where you are using
; the GK routed model or no type=user was found. This gives you
; the ability to either play an invalid message or to simply not
; use user authentication at all.
;
;context=default
@ -122,27 +122,27 @@ port = 1720
;
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; H323 channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The H323 channel can accept jitter,
; thus an enabled jitterbuffer on the receive H323 side will only
; be used if the sending side can create jitter and jbforce is
; also set to yes.
; H323 channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The H323 channel can accept jitter,
; thus an enabled jitterbuffer on the receive H323 side will only
; be used if the sending side can create jitter and jbforce is
; also set to yes.
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a H323
; channel. Defaults to "no".
; channel. Defaults to "no".
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a H323
; channel. Two implementations are currenlty available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; channel. Two implementations are currenlty available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
@ -153,7 +153,7 @@ port = 1720
; and Gatekeeper, if there is one.
;
; Example: if someone calls time@your.asterisk.box.com
; Asterisk will send the call to the extension 'time'
; Asterisk will send the call to the extension 'time'
; in the context default
;
; [default]
@ -161,13 +161,13 @@ port = 1720
; exten => time,2,Playback,current-time
;
; Keyword's 'prefix' and 'e164' are only make sense when
; used with a gatekeeper. You can specify either a prefix
; used with a gatekeeper. You can specify either a prefix
; or E.164 this endpoint is responsible for terminating.
;
;
; Example: The H.323 alias 'det-gw' will tell the gatekeeper
; to route any call with the prefix 1248 to this alias. Keyword
; e164 is used when you want to specifiy a full telephone
; number. So a call to the number 18102341212 would be
; number. So a call to the number 18102341212 would be
; routed to the H.323 alias 'time'.
;
;[time]
@ -182,10 +182,10 @@ port = 1720
;
;
; Inbound H.323 calls from BillyBob would land in the incoming
; context with a maximum of 4 concurrent incoming calls
;
; context with a maximum of 4 concurrent incoming calls
;
;
; Note: If keyword 'incominglimit' are omitted Asterisk will not
; Note: If keyword 'incominglimit' are omitted Asterisk will not
; enforce any maximum number of concurrent calls.
;
;[BillyBob]

2
configs/http.conf.sample
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@ -38,7 +38,7 @@ bindaddr=127.0.0.1
;enablestatic=yes
;
; Redirect one URI to another. This is how you would set a
; default page.
; 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:

86
configs/iax.conf.sample
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@ -12,11 +12,11 @@
[general]
;bindport=4569 ; bindport and bindaddr may be specified
; ; NOTE: bindport must be specified BEFORE
; bindaddr or may be specified on a specific
; bindaddr if followed by colon and port
; (e.g. bindaddr=192.168.0.1:4569)
; bindaddr or may be specified on a specific
; bindaddr if followed by colon and port
; (e.g. bindaddr=192.168.0.1:4569)
;bindaddr=192.168.0.1 ; more than once to bind to multiple
; ; addresses, but the first will be the
; ; addresses, but the first will be the
; ; default
;
; Set iaxcompat to yes if you plan to use layered switches or
@ -36,7 +36,7 @@
;
; For increased security against brute force password attacks
; enable "delayreject" which will delay the sending of authentication
; reject for REGREQ or AUTHREP if there is a password.
; reject for REGREQ or AUTHREP if there is a password.
;
;delayreject=yes
;
@ -60,7 +60,7 @@
;
;accountcode=lss0101
;
; You may specify a global default language for users.
; You may specify a global default language for users.
; Can be specified also on a per-user basis
; If omitted, will fallback to english
;
@ -111,7 +111,7 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
;
; forcejitterbuffer=yes|no: in the ideal world, when we bridge VoIP channels
; we don't want to do jitterbuffering on the switch, since the endpoints
; can each handle this. However, some endpoints may have poor jitterbuffers
; can each handle this. However, some endpoints may have poor jitterbuffers
; themselves, so this option will force * to always jitterbuffer, even in this
; case.
;
@ -166,7 +166,7 @@ forcejitterbuffer=no
;
; With a large amount of traffic on IAX2 trunks, there is a risk of bad voice quality due to
; the fact that the IAX2 trunking scheme depends on the Linux system to handle fragmentation of
; UDP packets. This may not be very efficient.
; UDP packets. This may not be very efficient.
; This setting sets the maximum transmission unit for IAX2 UDP trunking.
; default is 1240 bytes. Zero disables this functionality and let's the O/S handle fragmentation.
;
@ -177,7 +177,7 @@ forcejitterbuffer=no
; encryption = yes
;
; Force encryption insures no connection is established unless both sides support
; encryption. By turning this option on, encryption is automatically turned on as well.
; encryption. By turning this option on, encryption is automatically turned on as well.
;
; forceencryption = yes
@ -211,7 +211,7 @@ forcejitterbuffer=no
; Sample Registration for iaxtel
;
; Visit http://www.iaxtel.com to register with iaxtel. Replace "user"
; and "pass" with your username and password for iaxtel. Incoming
; and "pass" with your username and password for iaxtel. Incoming
; calls arrive at the "s" extension of "default" context.
;
;register => user:pass@iaxtel.com
@ -228,7 +228,7 @@ forcejitterbuffer=no
;register => FWDNumber:passwd@iax.fwdnet.net
;
;
; You can disable authentication debugging to reduce the amount of
; You can disable authentication debugging to reduce the amount of
; debugging traffic.
;
;authdebug=no
@ -256,7 +256,7 @@ forcejitterbuffer=no
autokill=yes
;
; codecpriority controls the codec negotiation of an inbound IAX call.
; This option is inherited to all user entities. It can also be defined
; This option is inherited to all user entities. It can also be defined
; in each user entity separately which will override the setting in general.
;
; The valid values are:
@ -284,29 +284,29 @@ autokill=yes
;allowfwdownload=yes
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
; just like friends added from the config file only on a
; as-needed basis? (yes|no)
; just like friends added from the config file only on a
; as-needed basis? (yes|no)
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
; If set to yes, when a IAX2 peer registers successfully,
; the ip address, the origination port, the registration period,
; and the username of the peer will be set to database via realtime.
; If not present, defaults to 'yes'.
; If set to yes, when a IAX2 peer registers successfully,
; the ip address, the origination port, the registration period,
; and the username of the peer will be set to database via realtime.
; If not present, defaults to 'yes'.
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
; as if it had just registered? (yes|no|<seconds>)
; If set to yes, when the registration expires, the friend will
; vanish from the configuration until requested again.
; If set to an integer, friends expire within this number of
; seconds instead of the registration interval.
; as if it had just registered? (yes|no|<seconds>)
; If set to yes, when the registration expires, the friend will
; vanish from the configuration until requested again.
; If set to an integer, friends expire within this number of
; seconds instead of the registration interval.
;rtignoreregexpire=yes ; When reading a peer from Realtime, if the peer's registration
; has expired based on its registration interval, used the stored
; address information regardless. (yes|no)
; has expired based on its registration interval, used the stored
; address information regardless. (yes|no)
;parkinglot=edvina ; Default parkinglot for IAX peers and users
; This can also be configured per device
; Parkinglots are defined in features.conf
; This can also be configured per device
; Parkinglots are defined in features.conf
; Guest sections for unauthenticated connection attempts. Just specify an
; empty secret, or provide no secret section.
@ -357,7 +357,7 @@ inkeys=freeworlddialup
; across the net. "md5" uses a challenge/response md5 sum arrangement, but
; still requires both ends have plain text access to the secret. "rsa" allows
; unidirectional secret knowledge through public/private keys. If "rsa"
; authentication is used, "inkeys" is a list of acceptable public keys on the
; authentication is used, "inkeys" is a list of acceptable public keys on the
; local system that can be used to authenticate the remote peer, separated by
; the ":" character. "outkey" is a single, private key to use to authenticate
; to the other side. Public keys are named /var/lib/asterisk/keys/<name>.pub
@ -377,13 +377,13 @@ inkeys=freeworlddialup
;auth=md5,plaintext,rsa
;secret=markpasswd
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended transfer.
; cause the given audio file to
; be played upon completion of
; an attended transfer.
;dbsecret=mysecrets/place ; Secrets can be stored in astdb, too
;transfer=no ; Disable IAX native transfer
;transfer=mediaonly ; When doing IAX native transfers, transfer
; only media stream
;transfer=mediaonly ; When doing IAX native transfers, transfer
; only media stream
;jitterbuffer=yes ; Override global setting an enable jitter buffer
; ; for this user
;maxauthreq=10 ; Set maximum number of outstanding AUTHREQs waiting for replies. Any further authentication attempts will be blocked
@ -395,10 +395,10 @@ inkeys=freeworlddialup
;language=en ; Use english as default language
;encryption=yes ; Enable IAX2 encryption. The default is no.
;keyrotate=off ; This is a compatibility option for older versions of
; ; IAX2 that do not support key rotation with encryption.
; ; This option will disable the IAX_COMMAND_RTENC message.
; ; IAX2 that do not support key rotation with encryption.
; ; This option will disable the IAX_COMMAND_RTENC message.
; ; default is on.
; ;
; ;
;
; Peers may also be specified, with a secret and
; a remote hostname.
@ -414,20 +414,20 @@ host=216.207.245.47
;mask=255.255.255.255
;qualify=yes ; Make sure this peer is alive
;qualifysmoothing = yes ; use an average of the last two PONG
; results to reduce falsely detected LAGGED hosts
; Default: Off
; results to reduce falsely detected LAGGED hosts
; Default: Off
;qualifyfreqok = 60000 ; how frequently to ping the peer when
; everything seems to be ok, in milliseconds
; everything seems to be ok, in milliseconds
;qualifyfreqnotok = 10000 ; how frequently to ping the peer when it's
; either LAGGED or UNAVAILABLE, in milliseconds
; either LAGGED or UNAVAILABLE, in milliseconds
;jitterbuffer=no ; Turn off jitter buffer for this peer
;
;encryption=yes ; Enable IAX2 encryption. The default is no.
;keyrotate=off ; This is a compatibility option for older versions of
; ; IAX2 that do not support key rotation with encryption.
; ; This option will disable the IAX_COMMAND_RTENC message.
; ; IAX2 that do not support key rotation with encryption.
; ; This option will disable the IAX_COMMAND_RTENC message.
; ; default is on.
; ;
; ;
; Peers can remotely register as well, so that they can be mobile. Default
; IP's can also optionally be given but are not required. Caller*ID can be
; suggested to the other side as well if it is for example a phone instead of

2
configs/iaxprov.conf.sample
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@ -7,7 +7,7 @@
; Templates provide a group of settings from which provisioning takes place.
; A template may be based upon any template that has been specified before
; it. If the template that an entry is based on is not specified then it is
; presumed to be 'default' (unless it is the first of course).
; presumed to be 'default' (unless it is the first of course).
;
; Templates which begin with 'si-' are used for provisioning units with
; specific service identifiers. For example the entry "si-000364000126"

6
configs/indications.conf.sample
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@ -516,7 +516,7 @@ callwaiting = 425/150,0/150,425/150,0/4000
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
; 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
@ -567,7 +567,7 @@ stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/1
[sg]
description = Singapore
; Singapore
; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf
; 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
@ -691,7 +691,7 @@ 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
; 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

10
configs/jabber.conf.sample
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@ -1,14 +1,14 @@
[general]
;debug=yes ;;Turn on debugging 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'.
;;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.
;[asterisk] ;;label
;type=client ;;Client or Component connection
;serverhost=astjab.org ;;Route to server for example,
;; talk.google.com
;; talk.google.com
;username=asterisk@astjab.org/asterisk ;;Username with optional resource.
;secret=blah ;;Password
;priority=1 ;;Resource priority
@ -17,7 +17,7 @@
;usesasl=yes ;;Use sasl or not
;buddy=mogorman@astjab.org ;;Manual addition of buddy to list.
;status=available ;;One of: chat, available, away,
;; xaway, or dnd
;; xaway, or dnd
;statusmessage="I am available" ;;Have custom status message for
;;Asterisk.
;;Asterisk.
;timeout=100 ;;Timeout on the message stack.

12
configs/jingle.conf.sample
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@ -2,19 +2,19 @@
;context=default ;;Context to dump call into
;bindaddr=0.0.0.0 ;;Address to bind to
;allowguest=yes ;;Allow calls from people not in
;;list of peers
;;list of peers
;
;[guest] ;;special account for options on guest account
;disallow=all
;disallow=all
;allow=ulaw
;context=guest
;
;[ogorman]
;username=ogorman@gmail.com ;;username of the peer your
;;calling or accepting calls from
;username=ogorman@gmail.com ;;username of the peer your
;;calling or accepting calls from
;disallow=all
;allow=ulaw
;context=default
;context=default
;connection=asterisk ;;client or component in jabber.conf
;;for the call to leave on.
;;for the call to leave on.
;

4
configs/logger.conf.sample
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@ -15,7 +15,7 @@
; 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
;
@ -90,7 +90,7 @@ console => notice,warning,error
messages => notice,warning,error
;full => notice,warning,error,debug,verbose
;syslog keyword : This special keyword logs to syslog facility
;syslog keyword : This special keyword logs to syslog facility
;
;syslog.local0 => notice,warning,error
;

20
configs/manager.conf.sample
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@ -1,6 +1,6 @@
;
; AMI - The Asterisk Manager Interface
;
;
; Third party application call management support and PBX event supervision
;
; This configuration file is read every time someone logs in
@ -13,11 +13,11 @@
; ---------------------------- 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
; 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
; 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]
@ -27,9 +27,9 @@ port = 5038
;httptimeout = 60
; a) httptimeout sets the Max-Age of the http cookie
; b) httptimeout is the amount of time the webserver waits
; 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
; c) httptimeout is also the amount of time the webserver keeps
; a http session alive after completing a successful action
bindaddr = 0.0.0.0
@ -44,8 +44,8 @@ bindaddr = 0.0.0.0
;tlsbindaddr=0.0.0.0 ; address to bind to, default to bindaddr
;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.
; 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.
@ -58,7 +58,7 @@ bindaddr = 0.0.0.0
;timestampevents = yes
; debug = on ; enable some debugging info in AMI messages (default off).
; Also accessible through the "manager debug" CLI command.
; Also accessible through the "manager debug" CLI command.
;[mark]
;secret = mysecret
;deny=0.0.0.0/0.0.0.0
@ -72,7 +72,7 @@ bindaddr = 0.0.0.0
;
;displayconnects = yes ; Display on CLI user login/logoff
;
; Authorization for various classes
; 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

18
configs/meetme.conf.sample
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@ -5,13 +5,13 @@
[general]
;audiobuffers=32 ; The number of 20ms audio buffers to be used
; when feeding audio frames from non-DAHDI channels
; into the conference; larger numbers will allow
; for the conference to 'de-jitter' audio that arrives
; at different timing than the conference's timing
; source, but can also allow for latency in hearing
; the audio from the speaker. Minimum value is 2,
; maximum value is 32.
; when feeding audio frames from non-DAHDI channels
; into the conference; larger numbers will allow
; for the conference to 'de-jitter' audio that arrives
; at different timing than the conference's timing
; source, but can also allow for latency in hearing
; the audio from the speaker. Minimum value is 2,
; maximum value is 32.
;
; Conferences may be scheduled from realtime?
;schedule=yes
@ -34,12 +34,12 @@
;
[rooms]
;
; Usage is conf => confno[,pin][,adminpin]
; Usage is conf => confno[,pin][,adminpin]
;
; Note that once a participant has called the conference, a change to the pin
; number done in this file will not take effect until there are no more users
; in the conference and it goes away. When it is created again, it will have
; the new pin number.
;
;conf => 1234
;conf => 1234
;conf => 2345,9938

64
configs/mgcp.conf.sample
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@ -13,27 +13,27 @@
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; MGCP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The MGCP channel can accept jitter,
; thus an enabled jitterbuffer on the receive MGCP side will only
; be used if the sending side can create jitter and jbforce is
; also set to yes.
; MGCP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The MGCP channel can accept jitter,
; thus an enabled jitterbuffer on the receive MGCP side will only
; be used if the sending side can create jitter and jbforce is
; also set to yes.
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a MGCP
; channel. Defaults to "no".
; channel. Defaults to "no".
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a MGCP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
@ -47,27 +47,27 @@
;; The MGCP channel supports the following service codes:
;; # - Transfer
;; *67 - Calling Number Delivery Blocking
;; *70 - Cancel Call Waiting
;; *72 - Call Forwarding Activation
;; *73 - Call Forwarding Deactivation
;; *78 - Do Not Disturb Activation
;; *79 - Do Not Disturb Deactivation
;; *67 - Calling Number Delivery Blocking
;; *70 - Cancel Call Waiting
;; *72 - Call Forwarding Activation
;; *73 - Call Forwarding Deactivation
;; *78 - Do Not Disturb Activation
;; *79 - Do Not Disturb Deactivation
;; *8 - Call pick-up
;
; known to work with Swissvoice IP10s
;[192.168.1.20]
;context=local
;host=192.168.1.20
;callerid = "John Doe" <123>
; known to work with Swissvoice IP10s
;[192.168.1.20]
;context=local
;host=192.168.1.20
;callerid = "John Doe" <123>
;callgroup=0 ; in the range from 0 to 63
;pickupgroup=0 ; in the range from 0 to 63
;nat=no
;threewaycalling=yes
;nat=no
;threewaycalling=yes
;transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer
;callwaiting=yes ; this might be a cause of trouble for ip10s
;cancallforward=yes
;line => aaln/1
;cancallforward=yes
;line => aaln/1
;
;[dph100]
@ -79,7 +79,7 @@
;context=local
;host=dynamic
;dtmfmode=none ; DTMF Mode can be 'none', 'rfc2833', or 'inband' or
; 'hybrid' which starts in none and moves to inband. Default is none.
; 'hybrid' which starts in none and moves to inband. Default is none.
;slowsequence=yes ; The DPH100M does not follow MGCP standards for sequencing
;line => aaln/1
@ -87,11 +87,11 @@
;[192.168.1.20]
;accountcode = 1000 ; record this in cdr as account identification for billing
;amaflags = billing ; record this in cdr as flagged for 'billing',
; 'documentation', or 'omit'
; 'documentation', or 'omit'
;context = local
;host = 192.168.1.20
;wcardep = aaln/* ; enables wildcard endpoint and sets it to 'aaln/*'
; another common format is '*'
;wcardep = aaln/* ; enables wildcard endpoint and sets it to 'aaln/*'
; another common format is '*'
;callerid = "Duane Cox" <123> ; now lets setup line 1 using per endpoint configuration...
;callwaiting = no
;callreturn = yes

76
configs/minivm.conf.sample
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@ -16,7 +16,7 @@
; 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
@ -89,43 +89,43 @@ emaildateformat=%A, %B %d, %Y at %r
;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:
; 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"
; ${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
; P or p AM or PM
; Q "today", "yesterday" or ABdY
; (*note: not standard strftime value)
; (*note: not standard strftime value)
; q "" (for today), "yesterday", weekday, or ABdY
; (*note: not standard strftime value)
; R 24 hour time, including minute
;
; (*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
; backwards compatibility
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
@ -141,27 +141,27 @@ military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
; 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
; 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
; or absolute
; messagebody = Format ; Message body definition with variables
;
[template-sv_SE_email]
[template-sv_SE_email]
messagebody=Hej ${MVM_NAME}:\n\n\tDu har fått ett röstbrevlåde-meddelande från ${MVM_CALLERID}.\nLängd: ${MVM_DUR}\nMailbox ${MVM_MAILBOX}\nDatum: ${MVM_DATE}. \nMeddelandet bifogas det här brevet. Om du inte kan läsa det, kontakta intern support. \nHälsningar\n\n\t\t\t\t--Asterisk\n
subject = Du har fått röstmeddelande (se bilaga)
fromemail = swedish-voicemail-service@stockholm.example.com
fromaddress = Asterisk Röstbrevlåda
charset=iso-8859-1
attachmedia=yes
attachmedia=yes
dateformat=%A, %d %B %Y at %H:%M:%S
locale=sv_SE
[template-en_US_email]
[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
attachmedia=yes
dateformat=%A, %B %d, %Y at %r
;[template-sv_SE_pager]
@ -180,12 +180,12 @@ dateformat=%A, %B %d, %Y at %r
;[template-en_US_email_southern]
;templatefile = templates/email_en_US.txt
;subject = Y'all got voicemail, honey!
;charset=ascii
;charset=ascii
;[template-en_UK_email]
;templatefile = templates/email_en_us.txt
;subject = Dear old chap, you've got an electronic communique
;charset=ascii
;charset=ascii
;----------------------- Mailbox accounts --------------------------
;Template for mailbox definition - all options

24
configs/misdn.conf.sample
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@ -111,26 +111,26 @@ crypt_keys=test,muh
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
; channel. Defaults to "no".
; channel. Defaults to "no".
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

4
configs/modules.conf.sample
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@ -21,7 +21,7 @@ autoload=yes
; Uncomment the following if you wish to use the Speech Recognition API
;preload => res_speech.so
;
; If you want, load the GTK console right away.
; If you want, load the GTK console right away.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
@ -38,6 +38,6 @@ noload => chan_console.so
;
; Only load one timing interface. If DAHDI is available, use that as it will
; provide the best results.
;
;
;noload => res_timing_dahdi.so
;noload => res_timing_pthread.so

8
configs/musiconhold.conf.sample
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@ -3,14 +3,14 @@
;
[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
; decrease consumable cpu cycles and memory
; disabled by default
; valid mode options:
; files -- read files from a directory in any Asterisk supported
; files -- read files from a directory in any Asterisk supported
; media format
; quietmp3 -- default
; quietmp3 -- default
; mp3 -- loud
; mp3nb -- unbuffered
; quietmp3nb -- quiet unbuffered

56
configs/osp.conf.sample
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@ -1,17 +1,17 @@
;
; Open Settlement Protocol Sample Configuration File
;
; This file contains configuration of OSP server providers that are used by the
; Asterisk OSP module. The section "general" is reserved for global options.
; All other sections describe specific OSP Providers. The provider "default"
; is used when no provider is otherwise specified.
; This file contains configuration of OSP server providers that are used by the
; Asterisk OSP module. The section "general" is reserved for global options.
; All other sections describe specific OSP Providers. The provider "default"
; is used when no provider is otherwise specified.
;
; The "servicepoint" and "source" parameters must be configured. For most
; The "servicepoint" and "source" parameters must be configured. For most
; implementations the other parameters in this file can be left unchanged.
;
[general]
;
; Enable cryptographic acceleration hardware.
; Enable cryptographic acceleration hardware.
; The default value is no.
;
;accelerate=no
@ -23,9 +23,9 @@
;
;securityfeatures=no
;
; Defines the status of tokens that Asterisk will validate.
; 0 - signed tokens only
; 1 - unsigned tokens only
; Defines the status of tokens that Asterisk will validate.
; 0 - signed tokens only
; 1 - unsigned tokens only
; 2 - both signed and unsigned
; The default value is 0, i.e. the Asterisk will only validate signed tokens.
; If securityfeatures are disabled, Asterisk cannot validate signed tokens.
@ -45,37 +45,37 @@
;source=domain name or [IP address in brackets]
;
; Define path and file name of crypto files.
; The default path for crypto file is /var/lib/asterisk/keys. If no path is
; The default path for crypto file is /var/lib/asterisk/keys. If no path is
; defined, crypto files will in /var/lib/asterisk/keys directory.
;
; Specify the private key file name.
; If this parameter is unspecified or not present, the default name will be the
; osp.conf section name followed by "-privatekey.pem" (for example:
; Specify the private key file name.
; If this parameter is unspecified or not present, the default name will be the
; osp.conf section name followed by "-privatekey.pem" (for example:
; default-privatekey.pem)
; If securityfeatures are disabled, this parameter is ignored.
;
;privatekey=pkey.pem
;
; Specify the local certificate file.
; If this parameter is unspecified or not present, the default name will be the
; osp.conf section name followed by "- localcert.pem " (for example:
; default-localcert.pem)
; Specify the local certificate file.
; If this parameter is unspecified or not present, the default name will be the
; osp.conf section name followed by "- localcert.pem " (for example:
; default-localcert.pem)
; If securityfeatures are disabled, this parameter is ignored.
;
;localcert=localcert.pem
;
; Specify one or more Certificate Authority key file names. If none are listed,
; a single Certificate Authority key file name is added with the default name of
; the osp.conf section name followed by "-cacert_0.pem " (for example:
; Specify one or more Certificate Authority key file names. If none are listed,
; a single Certificate Authority key file name is added with the default name of
; the osp.conf section name followed by "-cacert_0.pem " (for example:
; default-cacert_0.pem)
; If securityfeatures are disabled, this parameter is ignored.
;
;cacert=cacert_0.pem
;
; Configure parameters for OSP communication between Asterisk OSP client and OSP
; servers.
; Configure parameters for OSP communication between Asterisk OSP client and OSP
; servers.
;
; maxconnections: Max number of simultaneous connections to the provider OSP
; maxconnections: Max number of simultaneous connections to the provider OSP
; server (default=20)
; retrydelay: Extra delay between retries (default=0)
; retrylimit: Max number of retries before giving up (default=2)
@ -86,18 +86,18 @@
;retrylimit=2
;timeout=500
;
; Set the authentication policy.
; Set the authentication policy.
; 0 - NO - Accept all calls.
; 1 - YES - Accept calls with valid token or no token. Block calls with
; invalid token.
; 2 - EXCLUSIVE - Accept calls with valid token. Block calls with invalid token
; 1 - YES - Accept calls with valid token or no token. Block calls with
; invalid token.
; 2 - EXCLUSIVE - Accept calls with valid token. Block calls with invalid token
; or no token.
; Default is 1,
; If securityfeatures are disabled, Asterisk cannot validate signed tokens.
;
;authpolicy=1
;
; Set the default destination protocol. The OSP module supports SIP, H323, and
; Set the default destination protocol. The OSP module supports SIP, H323, and
; IAX protocols. The default protocol is set to SIP.
;
;defaultprotocol=SIP

212
configs/oss.conf.sample
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@ -3,75 +3,75 @@
;
[general]
; General config options, with default values shown.
; You should use one section per device, with [general] being used
; for the first device and also as a template for other devices.
;
; All but 'debug' can go also in the device-specific sections.
;
; debug = 0x0 ; misc debug flags, default is 0
; Set the device to use for I/O
; device = /dev/dsp
; Optional mixer command to run upon startup (e.g. to set
; volume levels, mutes, etc.
; mixer =
; Software mic volume booster (or attenuator), useful for sound
; cards or microphones with poor sensitivity. The volume level
; is in dB, ranging from -20.0 to +20.0
; boost = n ; mic volume boost in dB
; Set the callerid for outgoing calls
; callerid = John Doe <555-1234>
; autoanswer = no ; no autoanswer on call
; autohangup = yes ; hangup when other party closes
; extension = s ; default extension to call
; context = default ; default context for outgoing calls
; language = "" ; default language
; If you set overridecontext to 'yes', then the whole dial string
; will be interpreted as an extension, which is extremely useful
; to dial SIP, IAX and other extensions which use the '@' character.
; The default is 'no' just for backward compatibility, but the
; suggestion is to change it.
; overridecontext = no ; if 'no', the last @ will start the context
; if 'yes' the whole string is an extension.
; low level device parameters in case you have problems with the
; device driver on your operating system. You should not touch these
; unless you know what you are doing.
; queuesize = 10 ; frames in device driver
; frags = 8 ; argument to SETFRAGMENT
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; OSS channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The OSS channel can't accept jitter,
; thus an enabled jitterbuffer on the receive OSS side will always
; be used if the sending side can create jitter.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
; General config options, with default values shown.
; You should use one section per device, with [general] being used
; for the first device and also as a template for other devices.
;
; All but 'debug' can go also in the device-specific sections.
;
; debug = 0x0 ; misc debug flags, default is 0
; Set the device to use for I/O
; device = /dev/dsp
; Optional mixer command to run upon startup (e.g. to set
; volume levels, mutes, etc.
; mixer =
; Software mic volume booster (or attenuator), useful for sound
; cards or microphones with poor sensitivity. The volume level
; is in dB, ranging from -20.0 to +20.0
; boost = n ; mic volume boost in dB
; Set the callerid for outgoing calls
; callerid = John Doe <555-1234>
; autoanswer = no ; no autoanswer on call
; autohangup = yes ; hangup when other party closes
; extension = s ; default extension to call
; context = default ; default context for outgoing calls
; language = "" ; default language
; If you set overridecontext to 'yes', then the whole dial string
; will be interpreted as an extension, which is extremely useful
; to dial SIP, IAX and other extensions which use the '@' character.
; The default is 'no' just for backward compatibility, but the
; suggestion is to change it.
; overridecontext = no ; if 'no', the last @ will start the context
; if 'yes' the whole string is an extension.
; low level device parameters in case you have problems with the
; device driver on your operating system. You should not touch these
; unless you know what you are doing.
; queuesize = 10 ; frames in device driver
; frags = 8 ; argument to SETFRAGMENT
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; OSS channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The OSS channel can't accept jitter,
; thus an enabled jitterbuffer on the receive OSS side will always
; be used if the sending side can create jitter.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
; below is an entry for a second console channel
; [card1]
; device = /dev/dsp1 ; alternate device
; device = /dev/dsp1 ; alternate device
; Below are the settings to support video. You can include them
; in your general configuration as [general](+,video)
@ -79,26 +79,26 @@
; Section names used here are only examples.
[my_video](!) ; you can just include in your config
videodevice = /dev/video0 ; uses your V4L webcam as video source
videodevice = X11 ; X11 grabber. Dragging on the local display moves the origin.
videocodec = h263 ; also h261, h263p, h264, mpeg4, ...
; video_size is the geometry used by the encoder.
; Depending on the codec your choice is restricted.
video_size = 352x288 ; the format WIDTHxHEIGHT is also ok
video_size = cif ; sqcif, qcif, cif, qvga, vga, ...
; You can also set the geometry used for the camera, local display and remote display.
; The local window is on the right, the remote window is on the left.
; Right clicking with the mouse on a video window increases the size,
; center-clicking reduces the size.
camera_size = cif
remote_size = cif
local_size = qcif
bitrate = 60000 ; rate told to ffmpeg.
fps = 5 ; frames per second from the source.
; qmin = 3 ; quantizer value passed to the encoder.
videodevice = /dev/video0 ; uses your V4L webcam as video source
videodevice = X11 ; X11 grabber. Dragging on the local display moves the origin.
videocodec = h263 ; also h261, h263p, h264, mpeg4, ...
; video_size is the geometry used by the encoder.
; Depending on the codec your choice is restricted.
video_size = 352x288 ; the format WIDTHxHEIGHT is also ok
video_size = cif ; sqcif, qcif, cif, qvga, vga, ...
; You can also set the geometry used for the camera, local display and remote display.
; The local window is on the right, the remote window is on the left.
; Right clicking with the mouse on a video window increases the size,
; center-clicking reduces the size.
camera_size = cif
remote_size = cif
local_size = qcif
bitrate = 60000 ; rate told to ffmpeg.
fps = 5 ; frames per second from the source.
; qmin = 3 ; quantizer value passed to the encoder.
; The keypad is made of an image (in any format supported by SDL_image)
; and some configuration entries indicating the location and function of buttons.
@ -115,30 +115,30 @@ fps = 5 ; frames per second from the source.
; diameter of the ellipse.
;
[my_skin](!)
keypad = /tmp/keypad.jpg
region = 1 rect 19 18 67 18 28
region = 2 rect 84 18 133 18 28
region = 3 rect 152 18 201 18 28
region = 4 rect 19 60 67 60 28
region = 5 rect 84 60 133 60 28
region = 6 rect 152 60 201 60 28
region = 7 rect 19 103 67 103 28
region = 8 rect 84 103 133 103 28
region = 9 rect 152 103 201 103 28
region = * rect 19 146 67 146 28
region = 0 rect 84 146 133 146 28
region = # rect 152 146 201 146 28
region = pickup rect 229 15 267 15 40
region = hangup rect 230 66 270 64 40
region = mute circle 232 141 264 141 33
region = sendvideo circle 235 185 266 185 33
region = autoanswer rect 228 212 275 212 50
keypad = /tmp/keypad.jpg
region = 1 rect 19 18 67 18 28
region = 2 rect 84 18 133 18 28
region = 3 rect 152 18 201 18 28
region = 4 rect 19 60 67 60 28
region = 5 rect 84 60 133 60 28
region = 6 rect 152 60 201 60 28
region = 7 rect 19 103 67 103 28
region = 8 rect 84 103 133 103 28
region = 9 rect 152 103 201 103 28
region = * rect 19 146 67 146 28
region = 0 rect 84 146 133 146 28
region = # rect 152 146 201 146 28
region = pickup rect 229 15 267 15 40
region = hangup rect 230 66 270 64 40
region = mute circle 232 141 264 141 33
region = sendvideo circle 235 185 266 185 33
region = autoanswer rect 228 212 275 212 50
; another skin with entries for the keypad and a small font
; to write to the message boards in the skin.
[skin2](!)
keypad = /tmp/kpad2.jpg
keypad_font = /tmp/font.png
keypad = /tmp/kpad2.jpg
keypad_font = /tmp/font.png
; to add video support, uncomment this and remember to install
; the keypad and keypad_font files to the right place

4
configs/phone.conf.sample
Unescape View File

@ -6,8 +6,8 @@
[interfaces]
;
; Select a mode, either the phone jack provides dialtone, reads digits,
; then starts PBX with the given extension (dialtone mode), or
; immediately provides the PBX without reading any digits or providing
; then starts PBX with the given extension (dialtone mode), or
; immediately provides the PBX without reading any digits or providing
; any dialtone (this is the immediate mode, the default). Also, you
; can set the mode to "fxo" if you have a linejack to make it operate
; properly. If you are using a Sigma Designs board you may set this to

62
configs/phoneprov.conf.sample
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@ -6,15 +6,15 @@
;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.
; 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
; can be substituted into. You can also set arbitrary variables for the profiles
; templates to have access to. Example:
;[example]
@ -43,48 +43,48 @@ default_profile=polycom ; The default profile to use if none specified in users.
[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
; 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
; 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 => 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 => 2345-11402-001.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 => 2345-12500-001.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 => 2345-12500-001.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
@ -112,6 +112,6 @@ 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
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

14
configs/queuerules.conf.sample
Unescape View File

@ -1,12 +1,12 @@
; It is possible to change the value of the QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY
; It is possible to change the value of the QUEUE_MAX_PENALTY and QUEUE_MIN_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.
; 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.
;
; 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
; 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 the caller's stay in the queue, these will not be reflected for that caller.
;
; The syntax for these rules is

94
configs/queues.conf.sample
Unescape View File

@ -13,12 +13,12 @@ persistentmembers = yes
; Keep queue statistics during a reload. Default is 'no'
;
keepstats = no
;
;
; AutoFill Behavior
; The old/current behavior of the queue has a serial type behavior
; The old/current behavior of the queue has 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
; 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
@ -26,8 +26,8 @@ keepstats = no
; 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
; 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 no
; to keep backward compatibility with the old behavior.
;
@ -36,22 +36,22 @@ autofill = yes
; 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
; 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
; 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.
; compatibility.
;
monitor-type = MixMonitor
;
; UpdateCDR behavior.
; 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.
;
; CDR dstchannel field of a call with an agent name, which you can set
; at the login time with AddQueueMember membername parameter.
;
; updatecdr = no
;
@ -134,7 +134,7 @@ shared_lastcall=no
; 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
@ -143,8 +143,8 @@ shared_lastcall=no
; 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
; 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.
;
@ -152,7 +152,7 @@ shared_lastcall=no
; 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
; 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
@ -170,8 +170,8 @@ shared_lastcall=no
;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
; 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
@ -196,21 +196,21 @@ shared_lastcall=no
;
;maxlen = 0
;
; If set to yes, just prior to the caller being bridged with a queue member
; If set to yes, just prior to the caller being bridged with a queue member
; the following variables will be set
; 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
; 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
; If set to yes, just prior to the caller being bridged with a queue member
; the following variables will be set:
; QEHOLDTIME callers hold time
; QEHOLDTIME callers hold time
; QEORIGINALPOS original position of the caller in the queue
;
;setqueueentryvar=no
@ -220,7 +220,7 @@ shared_lastcall=no
; 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;
; 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
@ -231,17 +231,17 @@ shared_lastcall=no
;setqueuevar=no
;
; if set, run this macro when connected to the queue member
; you can override this macro by setting the macro option on
; you can override this macro by setting the macro option on
; the queue application
;
; membermacro=somemacro
; How often to announce queue position and/or estimated
; 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
;announce-frequency = 90
;
; The absolute minimum time between the start of each
; queue position and/or estimated holdtime announcement
@ -300,26 +300,26 @@ shared_lastcall=no
;
; queue-thankyou=
;
; ("You are now first in line.")
;queue-youarenext = queue-youarenext
; ("There are")
; ("You are now first in line.")
;queue-youarenext = queue-youarenext
; ("There are")
;queue-thereare = queue-thereare
; ("calls waiting.")
; ("calls waiting.")
;queue-callswaiting = queue-callswaiting
; ("The current est. holdtime is")
; ("The current est. holdtime is")
;queue-holdtime = queue-holdtime
; ("minutes.")
; ("minutes.")
;queue-minutes = queue-minutes
; ("seconds.")
; ("seconds.")
;queue-seconds = queue-seconds
; ("Thank you for your patience.")
; ("Thank you for your patience.")
;queue-thankyou = queue-thankyou
; ("Hold time")
; ("Hold time")
;queue-reporthold = queue-reporthold
; ("All reps busy / wait for next")
; ("All reps busy / wait for next")
;periodic-announce = queue-periodic-announce
;
; A set of periodic announcements can be defined by separating
; 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
;
@ -358,7 +358,7 @@ shared_lastcall=no
;
; You can specify the options supplied to MixMonitor by calling
; Set(MONITOR_OPTIONS=av(<x>)V(<x>)W(<x>))
; The 'b' option for MixMonitor (only save audio to the file while bridged) is
; 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
@ -379,9 +379,9 @@ shared_lastcall=no
; 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
; 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
@ -394,14 +394,14 @@ shared_lastcall=no
; 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
@ -411,7 +411,7 @@ shared_lastcall=no
;
;leavewhenempty = inuse,ringing
;
; A caller will be removed from the queue if at least one member cannot be found
; 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
@ -461,7 +461,7 @@ shared_lastcall=no
;
; timeoutrestart = no
;
; If you wish to implement a rule defined in queuerules.conf (see
; 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
@ -501,5 +501,5 @@ shared_lastcall=no
;
;member => Agent/@1 ; Any agent in group 1
;member => Agent/:1,1 ; Any agent in group 1, wait for first
; available, but consider with penalty
; available, but consider with penalty

12
configs/res_odbc.conf.sample
Unescape View File

@ -1,4 +1,4 @@
;;; odbc setup file
;;; odbc setup file
; ENV is a global set of environmental variables that will get set.
; Note that all environmental variables can be seen by all connections,
@ -49,11 +49,11 @@ pre-connect => yes
sanitysql => select count(*) from systables
; forcecommit => no ; Default to committing uncommitted transactions?
; isolation => read_committed ; Isolation level; supported levels are:
; read_uncommitted, read_committed, repeatable_read,
; serializable. Note that not all databases support
; all isolation levels (e.g. Postgres only supports
; repeatable_read and serializable). See database
; documentation for further information.
; read_uncommitted, read_committed, repeatable_read,
; serializable. Note that not all databases support
; all isolation levels (e.g. Postgres only supports
; repeatable_read and serializable). See database
; documentation for further information.
;
; Many databases have a default of '\' to escape special characters. MS SQL
; Server does not.

2
configs/res_snmp.conf.sample
Unescape View File

@ -15,7 +15,7 @@
[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)
; 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

32
configs/rpt.conf.sample
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@ -28,13 +28,13 @@
;funcchar = * ; function lead-in character (defaults to '*')
;endchar = # ; command mode end character (defaults to '#')
;;nobusyout=yes ; (optional) Do not busy-out reverse-patch when
; normal patch in use
; normal patch in use
;hangtime=1000 ; squelch tail hang time (in ms) (optional)
;totime=100000 ; transmit time-out time (in ms) (optional)
;idtime=30000 ; id interval time (in ms) (optional)
;politeid=30000 ; time in milliseconds before ID timer
; expires to try and ID in the tail.
; (optional, default is 30000).
; expires to try and ID in the tail.
; (optional, default is 30000).
;idtalkover=|iwb6nil/rpt ; Talkover ID (optional) default is none
;unlinkedct=ct2 ; unlinked courtesy tone (optional) default is none
@ -69,13 +69,13 @@
;funcchar = * ; function lead-in character (defaults to '*')
;endchar = # ; command mode end character (defaults to '#')
;;nobusyout=yes ; (optional) Do not busy-out reverse-patch when
; normal patch in use
; normal patch in use
;hangtime=1000 ; squelch tail hang time (in ms) (optional)
;totime=100000 ; transmit time-out time (in ms) (optional)
;idtime=30000 ; id interval time (in ms) (optional)
;politeid=30000 ; time in milliseconds before ID timer
; expires to try and ID in the tail.
; (optional, default is 30000).
; expires to try and ID in the tail.
; (optional, default is 30000).
;idtalkover=|iwb6nil/rpt ; Talkover ID (optional) default is none
;unlinkedct=ct2 ; unlinked courtesy tone (optional) default is none
@ -86,9 +86,9 @@
; specify the rxchannel and the txchannel will be assumed from the rxchannel
;txchannel = DAHDI/6 ; Tx audio/signalling channel
;functions = functions-remote
;remote = ft897 ; Set remote=y for dumb remote or
; remote=ft897 for Yaesu FT-897 or
; remote=rbi for Doug Hall RBI1
;remote = ft897 ; Set remote=y for dumb remote or
; remote=ft897 for Yaesu FT-897 or
; remote=rbi for Doug Hall RBI1
;iobase = 0x378 ; Specify IO port for parallel port (optional)
;[functions-repeater]
@ -106,7 +106,7 @@
;6=autopatchup ; Autopatch up
;0=autopatchdn ; Autopatch down
;90=cop,1 ; System warm boot
;90=cop,1 ; System warm boot
;91=cop,2 ; System enable
;92=cop,3 ; System disable
@ -135,7 +135,7 @@
; Single frequencies are created by setting freq1 or freq2 to zero.
;
; |m - Morse escape sequence
;
;
; Sends Morse code at the telemetry amplitude and telemetry frequency as defined in the
; [morse] section.
;
@ -150,15 +150,15 @@
;ct1=|t(350,0,100,2048)(500,0,100,2048)(660,0,100,2048)
;ct2=|t(660,880,150,2048)
;ct3=|t(440,0,150,2048)
;ct2=|t(660,880,150,2048)
;ct3=|t(440,0,150,2048)
;ct4=|t(550,0,150,2048)
;ct5=|t(660,0,150,2048)
;ct6=|t(880,0,150,2048)
;ct7=|t(660,440,150,2048)
;ct8=|t(700,1100,150,2048)
;remotetx=|t(2000,0,75,2048)(0,0,75,0)(1600,0,75,2048);
;remotemon=|t(1600,0,75,2048)
;remotetx=|t(2000,0,75,2048)(0,0,75,0)(1600,0,75,2048);
;remotemon=|t(1600,0,75,2048)
;cmdmode=|t(900,903,200,2048)
;functcomplete=|t(1000,0,100,2048)(0,0,100,0)(1000,0,100,2048)
@ -168,7 +168,7 @@
;speed=20 ; Approximate speed in WPM
;frequency=800 ; Morse Telemetry Frequency
;amplitude=4096 ; Morse Telemetry Amplitude
;idfrequency=330 ; Morse ID Frequency
;idfrequency=330 ; Morse ID Frequency
;idamplitude=2048 ; Morse ID Amplitude
;[nodes]

4
configs/rtp.conf.sample
Unescape View File

@ -18,8 +18,8 @@ rtpend=20000
; allowed to continue (in 'samples', 1/8000 of a second)
;
;dtmftimeout=3000
; rtcpinterval = 5000 ; Milliseconds between rtcp reports
;(min 500, max 60000, default 5000)
; 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 source of the RTP stream. This option is

230
configs/say.conf.sample
Unescape View File

@ -1,11 +1,11 @@
;
;
; language configuration
;
[general]
mode=old ; method for playing numbers and dates
; old - using asterisk core function
; new - using this configuration file
; old - using asterisk core function
; new - using this configuration file
; The new language routines produce strings of the form
; prefix:[format:]data
@ -66,7 +66,7 @@ mode=old ; method for playing numbers and dates
; date:M:200604172030.00-4-102
; date:p:200604172030.00-4-102
;
;
;
; Remember, normally X Z N are special, and the search is
; case insensitive, so you must use [X] [N] [Z] .. if you
; want exact match.
@ -75,126 +75,126 @@ mode=old ; method for playing numbers and dates
; language-independent
[digit-base](!) ; base rule for digit strings
; XXX incomplete yet
_digit:[0-9] => digits/${SAY}
_digit:[-] => letters/dash
_digit:[*] => letters/star
_digit:[@] => letters/at
_digit:[0-9]. => digit:${SAY:0:1}, digit:${SAY:1}
; XXX incomplete yet
_digit:[0-9] => digits/${SAY}
_digit:[-] => letters/dash
_digit:[*] => letters/star
_digit:[@] => letters/at
_digit:[0-9]. => digit:${SAY:0:1}, digit:${SAY:1}
[date-base](!) ; base rules for dates and times
; the 'SAY' variable contains YYYYMMDDHHmm.ss-dow-doy
; these rule map the strftime attributes.
_date:Y:. => num:${SAY:0:4} ; year, 19xx
_date:[Bb]:. => digits/mon-$[${SAY:4:2}-1] ; month name, 0..11
_date:[Aa]:. => digits/day-${SAY:16:1} ; day of week
_date:[de]:. => num:${SAY:6:2} ; day of month
_date:[hH]:. => num:${SAY:8:2} ; hour
_date:[I]:. => num:$[${SAY:8:2} % 12] ; hour 0-12
_date:[M]:. => num:${SAY:10:2} ; minute
; XXX too bad the '?' function does not remove the quotes
; _date:[pP]:. => digits/$[ ${SAY:10:2} > 12 ? "p-m" :: "a-m"] ; am pm
_date:[pP]:. => digits/p-m ; am pm
_date:[S]:. => num:${SAY:13:2} ; seconds
; the 'SAY' variable contains YYYYMMDDHHmm.ss-dow-doy
; these rule map the strftime attributes.
_date:Y:. => num:${SAY:0:4} ; year, 19xx
_date:[Bb]:. => digits/mon-$[${SAY:4:2}-1] ; month name, 0..11
_date:[Aa]:. => digits/day-${SAY:16:1} ; day of week
_date:[de]:. => num:${SAY:6:2} ; day of month
_date:[hH]:. => num:${SAY:8:2} ; hour
_date:[I]:. => num:$[${SAY:8:2} % 12] ; hour 0-12
_date:[M]:. => num:${SAY:10:2} ; minute
; XXX too bad the '?' function does not remove the quotes
; _date:[pP]:. => digits/$[ ${SAY:10:2} > 12 ? "p-m" :: "a-m"] ; am pm
_date:[pP]:. => digits/p-m ; am pm
_date:[S]:. => num:${SAY:13:2} ; seconds
[en-base](!)
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
; enumeration
_e[n]um:X => digits/h-${SAY}
_e[n]um:1X => digits/h-${SAY}
_e[n]um:[2-9]0 => digits/h-${SAY}
_e[n]um:[2-9][1-9] => num:${SAY:0:1}0, digits/h-${SAY:1}
_e[n]um:[1-9]XX => num:${SAY:0:1}, digits/hundred, enum:${SAY:1}
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
; enumeration
_e[n]um:X => digits/h-${SAY}
_e[n]um:1X => digits/h-${SAY}
_e[n]um:[2-9]0 => digits/h-${SAY}
_e[n]um:[2-9][1-9] => num:${SAY:0:1}0, digits/h-${SAY:1}
_e[n]um:[1-9]XX => num:${SAY:0:1}, digits/hundred, enum:${SAY:1}
[it](digit-base,date-base)
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:1XX => digits/hundred, num:${SAY:1}
_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXX => digits/thousand, num:${SAY:1}
_[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousands, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousands, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousands, num:${SAY:3}
_[n]um:1XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:[2-9]XXXXXX => num:${SAY:0:1}, digits/millions, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:1XX => digits/hundred, num:${SAY:1}
_[n]um:[2-9]XX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXX => digits/thousand, num:${SAY:1}
_[n]um:[2-9]XXX => num:${SAY:0:1}, digits/thousands, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousands, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousands, num:${SAY:3}
_[n]um:1XXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:[2-9]XXXXXX => num:${SAY:0:1}, digits/millions, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
[en](en-base,date-base,digit-base)
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
[de](date-base,digit-base)
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:1}-and, digits/${SAY:0:1}0
_[n]um:1XX => digits/ein, digits/hundred, num:${SAY:1}
_[n]um:[2-9]XX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXX => digits/ein, digits/thousand, num:${SAY:1}
_[n]um:[2-9]XXX => digits/${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:X00XXX => digits/${SAY:0:1}, digits/hundred, digits/thousand, num:${SAY:3}
_[n]um:XXXXXX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXXXXX => digits/eine, digits/million, num:${SAY:1}
_[n]um:[2-9]XXXXXX => digits/${SAY:0:1}, digits/millions, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1X => digits/${SAY}
_[n]um:[2-9]0 => digits/${SAY}
_[n]um:[2-9][1-9] => digits/${SAY:1}-and, digits/${SAY:0:1}0
_[n]um:1XX => digits/ein, digits/hundred, num:${SAY:1}
_[n]um:[2-9]XX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXX => digits/ein, digits/thousand, num:${SAY:1}
_[n]um:[2-9]XXX => digits/${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:X00XXX => digits/${SAY:0:1}, digits/hundred, digits/thousand, num:${SAY:3}
_[n]um:XXXXXX => digits/${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:1XXXXXX => digits/eine, digits/million, num:${SAY:1}
_[n]um:[2-9]XXXXXX => digits/${SAY:0:1}, digits/millions, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/millions, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/millions, num:${SAY:3}
_datetime::. => date:AdBY 'digits/at' IMp:${SAY}
_date::. => date:AdBY:${SAY}
_time::. => date:IMp:${SAY}
[hu](digit-base,date-base)
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1[1-9] => digits/10en, digits/${SAY:1}
_[n]um:2[1-9] => digits/20on, digits/${SAY:1}
_[n]um:[1-9]0 => digits/${SAY}
_[n]um:[3-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
_datetime::. => date:YBdA k 'ora' M 'perc':${SAY}
_date::. => date:YBdA:${SAY}
_time::. => date:k 'ora' M 'perc':${SAY}
_[n]um:0. => num:${SAY:1}
_[n]um:X => digits/${SAY}
_[n]um:1[1-9] => digits/10en, digits/${SAY:1}
_[n]um:2[1-9] => digits/20on, digits/${SAY:1}
_[n]um:[1-9]0 => digits/${SAY}
_[n]um:[3-9][1-9] => digits/${SAY:0:1}0, num:${SAY:1}
_[n]um:XXX => num:${SAY:0:1}, digits/hundred, num:${SAY:1}
_[n]um:XXXX => num:${SAY:0:1}, digits/thousand, num:${SAY:1}
_[n]um:XXXXX => num:${SAY:0:2}, digits/thousand, num:${SAY:2}
_[n]um:XXXXXX => num:${SAY:0:3}, digits/thousand, num:${SAY:3}
_[n]um:XXXXXXX => num:${SAY:0:1}, digits/million, num:${SAY:1}
_[n]um:XXXXXXXX => num:${SAY:0:2}, digits/million, num:${SAY:2}
_[n]um:XXXXXXXXX => num:${SAY:0:3}, digits/million, num:${SAY:3}
_[n]um:XXXXXXXXXX => num:${SAY:0:1}, digits/billion, num:${SAY:1}
_[n]um:XXXXXXXXXXX => num:${SAY:0:2}, digits/billion, num:${SAY:2}
_[n]um:XXXXXXXXXXXX => num:${SAY:0:3}, digits/billion, num:${SAY:3}
_datetime::. => date:YBdA k 'ora' M 'perc':${SAY}
_date::. => date:YBdA:${SAY}
_time::. => date:k 'ora' M 'perc':${SAY}

642
configs/sip.conf.sample
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File diff suppressed because it is too large Load Diff

60
configs/skinny.conf.sample
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@ -5,22 +5,22 @@
bindaddr=0.0.0.0 ; Address to bind to
bindport=2000 ; Port to bind to, default tcp/2000
dateformat=M-D-Y ; M,D,Y in any order (6 chars max)
; "A" may also be used, but it must be at the end.
; Use M for month, D for day, Y for year, A for 12-hour time.
; "A" may also be used, but it must be at the end.
; Use M for month, D for day, Y for year, A for 12-hour time.
keepalive=120
;vmexten=8500 ; Systemwide voicemailmain pilot number
; It must be in the same context as the calling
; device/line
; It must be in the same context as the calling
; device/line
; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given line which registers or unregisters with
; us and have a "regexten=" configuration item.
; Multiple contexts may be specified by separating them with '&'. The
; us and have a "regexten=" configuration item.
; Multiple contexts may be specified by separating them with '&'. The
; actual extension is the 'regexten' parameter of the registering line or its
; name if 'regexten' is not provided. If more than one context is provided,
; the context must be specified within regexten by appending the desired
; context after '@'. More than one regexten may be supplied if they are
; context after '@'. More than one regexten may be supplied if they are
; separated by '&'. Patterns may be used in regexten.
;
;regcontext=skinnyregistrations
@ -38,27 +38,27 @@ keepalive=120
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
;jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; skinny channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The skinny channel can accept
; jitter, thus a jitterbuffer on the receive skinny side will be
; used only if it is forced and enabled.
; skinny channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The skinny channel can accept
; jitter, thus a jitterbuffer on the receive skinny side will be
; used only if it is forced and enabled.
;jbforce = no ; Forces the use of a jitterbuffer on the receive side of a skinny
; channel. Defaults to "no".
; channel. Defaults to "no".
;jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
;jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
;jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a
; skinny channel. Two implementations are currently available
; - "fixed" (with size always equals to jbmaxsize)
; - "adaptive" (with variable size, actually the new jb of IAX2).
; Defaults to fixed.
; skinny channel. Two implementations are currently available
; - "fixed" (with size always equals to jbmaxsize)
; - "adaptive" (with variable size, actually the new jb of IAX2).
; Defaults to fixed.
;jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
@ -93,8 +93,8 @@ keepalive=120
;vmexten=8500 ; Device level voicemailmain pilot number
;regexten=100
;context=inbound
;linelabel="Support Line" ; Displays next to the line
; button on 7940's and 7960s
;linelabel="Support Line" ; Displays next to the line
; button on 7940's and 7960s
;[110]
;callerid="John Chambers" <408-526-4000>
;context=did
@ -110,21 +110,21 @@ keepalive=120
;callerid="George W. Bush" <202-456-1414>
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended transfer.
; cause the given audio file to
; be played upon completion of
; an attended transfer.
;mailbox=500
;callwaiting=yes
;transfer=yes
;threewaycalling=yes
;context=default
;mohinterpret=default ; This option specifies a default music on hold class to
; use when put on hold if the channel's moh class was not
; explicitly set with Set(CHANNEL(musicclass)=whatever) and
; the peer channel did not suggest a class to use.
; use when put on hold if the channel's moh class was not
; explicitly set with Set(CHANNEL(musicclass)=whatever) and
; the peer channel did not suggest a class to use.
;mohsuggest=default ; This option specifies which music on hold class to suggest to the peer channel
; when this channel places the peer on hold. It may be specified globally or on
; a per-user or per-peer basis.
; when this channel places the peer on hold. It may be specified globally or on
; a per-user or per-peer basis.
[devices]

110
configs/sla.conf.sample
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@ -8,10 +8,10 @@
[general]
;attemptcallerid=no ; Attempt CallerID handling. The default value for this
; is "no" because CallerID handling with an SLA setup is
; known to not work properly in some situations. However,
; feel free to enable it if you would like. If you do, and
; you find problems, please do not report them.
; is "no" because CallerID handling with an SLA setup is
; known to not work properly in some situations. However,
; feel free to enable it if you would like. If you do, and
; you find problems, please do not report them.
; -------------------------------------
@ -22,30 +22,30 @@
;type=trunk ; This line is what marks this entry as a trunk.
;device=DAHDI/3 ; Map this trunk declaration to a specific device.
; NOTE: You can not just put any type of channel here.
; DAHDI channels can be directly used. IP trunks
; require some indirect configuration which is
; described in doc/asterisk.pdf.
; NOTE: You can not just put any type of channel here.
; DAHDI channels can be directly used. IP trunks
; require some indirect configuration which is
; described in doc/asterisk.pdf.
;autocontext=line1 ; This supports automatic generation of the dialplan entries
; if the autocontext option is used. Each trunk should have
; a unique context name. Then, in chan_dahdi.conf, this device
; should be configured to have incoming calls go to this context.
;autocontext=line1 ; This supports automatic generation of the dialplan entries
; if the autocontext option is used. Each trunk should have
; a unique context name. Then, in chan_dahdi.conf, this device
; should be configured to have incoming calls go to this context.
;ringtimeout=30 ; Set how long to allow this trunk to ring on an inbound call before hanging
; it up as an unanswered call. The value is in seconds.
;ringtimeout=30 ; Set how long to allow this trunk to ring on an inbound call before hanging
; it up as an unanswered call. The value is in seconds.
;barge=no ; If this option is set to "no", then no station will be
; allowed to join a call that is in progress. The default
; value is "yes".
; allowed to join a call that is in progress. The default
; value is "yes".
;hold=private ; This option configure hold permissions for this trunk.
; "open" - This means that any station can put this trunk
; on hold, and any station can retrieve it from
; hold. This is the default.
; "private" - This means that once a station puts the
; trunk on hold, no other station will be
; allowed to retrieve the call from hold.
; "open" - This means that any station can put this trunk
; on hold, and any station can retrieve it from
; hold. This is the default.
; "private" - This means that once a station puts the
; trunk on hold, no other station will be
; allowed to retrieve the call from hold.
;[line2]
;type=trunk
@ -60,9 +60,9 @@
;[line4]
;type=trunk
;device=Local/disa@line4_outbound ; A Local channel in combination with the Disa
; application can be used to support IP trunks.
; See doc/asterisk.pdf on more information on how
; IP trunks work.
; application can be used to support IP trunks.
; See doc/asterisk.pdf on more information on how
; IP trunks work.
;autocontext=line4
; --------------------------------------
@ -75,55 +75,55 @@
;device=SIP/station1 ; Each station must be mapped to a device.
;autocontext=sla_stations ; This supports automatic generation of the dialplan entries if
; the autocontext option is used. All stations can use the same
; context without conflict. The device for this station should
; have its context configured to the same one listed here.
;autocontext=sla_stations ; This supports automatic generation of the dialplan entries if
; the autocontext option is used. All stations can use the same
; context without conflict. The device for this station should
; have its context configured to the same one listed here.
;ringtimeout=10 ; Set a timeout for how long to allow the station to ring for an
; incoming call, in seconds.
;ringtimeout=10 ; Set a timeout for how long to allow the station to ring for an
; incoming call, in seconds.
;ringdelay=10 ; Set a time for how long to wait before beginning to ring this station
; once there is an incoming call, in seconds.
; once there is an incoming call, in seconds.
;hold=private ; This option configure hold permissions for this station. Note
; that if private hold is set in the trunk entry, that will override
; anything here. However, if a trunk has open hold access, but this
; station is set to private hold, then the private hold will be in
; effect.
; "open" - This means that once this station puts a call
; on hold, any other station is allowed to retrieve
; it. This is the default.
; "private" - This means that once this station puts a
; call on hold, no other station will be
; allowed to retrieve the call from hold.
; that if private hold is set in the trunk entry, that will override
; anything here. However, if a trunk has open hold access, but this
; station is set to private hold, then the private hold will be in
; effect.
; "open" - This means that once this station puts a call
; on hold, any other station is allowed to retrieve
; it. This is the default.
; "private" - This means that once this station puts a
; call on hold, no other station will be
; allowed to retrieve the call from hold.
;trunk=line1 ; Individually list all of the trunks that will appear on this station. This
; order is significant. It should be the same order as they appear on the
; phone. The order here defines the order of preference that the trunks will
; be used.
; order is significant. It should be the same order as they appear on the
; phone. The order here defines the order of preference that the trunks will
; be used.
;trunk=line2
;trunk=line3,ringdelay=5 ; A ring delay for the station can also be specified for a specific trunk.
; If a ring delay is specified both for the whole station and for a specific
; trunk on a station, the setting for the specific trunk will take priority.
; This value is in seconds.
; If a ring delay is specified both for the whole station and for a specific
; trunk on a station, the setting for the specific trunk will take priority.
; This value is in seconds.
;trunk=line4,ringtimeout=5 ; A ring timeout for the station can also be specified for a specific trunk.
; If a ring timeout is specified both for the whole station and for a specific
; trunk on a station, the setting for the specific trunk will take priority.
; This value is in seconds.
; If a ring timeout is specified both for the whole station and for a specific
; trunk on a station, the setting for the specific trunk will take priority.
; This value is in seconds.
;[station](!) ; When there are a lot of stations that are configured the same way,
; it is convenient to use a configuration template like this so that
; the common settings stay in one place.
; it is convenient to use a configuration template like this so that
; the common settings stay in one place.
;type=station
;autocontext=sla_stations
;trunk=line1
;trunk=line2
;trunk=line2
;trunk=line3
;trunk=line4
;trunk=line4
;[station2](station) ; Define a station that uses the configuration from the template "station".
;device=SIP/station2

70
configs/telcordia-1.adsi
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@ -28,15 +28,15 @@ STATE "inactive" ; No active call
; Begin soft key definitions
;
KEY "CB_OH" IS "Block" OR "Call Block"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "*60"
SUBSCRIPT "offHook"
OFFHOOK
VOICEMODE
WAITDIALTONE
SENDDTMF "*60"
SUBSCRIPT "offHook"
ENDKEY
KEY "CB" IS "Block" OR "Call Block"
SENDDTMF "*60"
SENDDTMF "*60"
ENDKEY
;
@ -44,38 +44,38 @@ ENDKEY
;
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
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
IFEVENT FARRING THEN
CLEAR
SHOWDISPLAY "ringing" AT 1
ENDIF
IFEVENT FARANSWER THEN
CLEAR
SHOWDISPLAY "talkingto" AT 1
GOTO "stableCall"
ENDIF
ENDSUB
SUB "stableCall" IS

32
configs/unistim.conf.sample
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@ -14,29 +14,29 @@ port=5000 ; UDP port
;keepalive=120 ; in seconds, default = 120
;public_ip= ; if asterisk is behind a nat, specify your public IP
;autoprovisioning=no ; Allow undeclared phones to register an extension. See README for important
; informations. no (default), yes, tn.
; informations. no (default), yes, tn.
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.
; SIP channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
; channel. Defaults to "no".
; channel. Defaults to "no".
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; channel. Two implementations are currently available - "fixed"
; (with size always equals to jbmaxsize) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
@ -63,13 +63,13 @@ port=5000 ; UDP port
;mailbox=1234 ; Specify the mailbox number. Used by Message Waiting Indication
;linelabel="Support" ; Softkey label for the next line=> entry, 9 char max.
;extension=none ; Add an extension into the dialplan. Only valid in context specified previously.
; none=don't add (default), ask=prompt user, line=use the line number
; none=don't add (default), ask=prompt user, line=use the line number
;line => 100 ; Only one line by device is currently supported.
; Beware ! only bookmark and softkey entries are allowed after line=>
; Beware ! only bookmark and softkey entries are allowed after line=>
;bookmark=Hans C.@123 ; Use a softkey to dial 123. Name : 9 char max
;bookmark=Mailbox@011@54 ; 54 shows a mailbox icon. See #define FAV_ICON_ for other values (32 to 63)
;bookmark=Test@*@USTM/violet ; Display an icon if violet is connected (dynamic), only for unistim device
;bookmark=4@Pager@54321@51 ; Display a pager icon and dial 54321 when softkey 4 is pressed
;bookmark=4@Pager@54321@51 ; Display a pager icon and dial 54321 when softkey 4 is pressed
;[violet]
;device=006038abcdef

22
configs/usbradio.conf.sample
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@ -30,23 +30,23 @@
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
; USBRADIO channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The USBRADIO channel can't accept jitter,
; thus an enabled jitterbuffer on the receive USBRADIO side will always
; be used if the sending side can create jitter.
; USBRADIO channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The USBRADIO channel can't accept jitter,
; thus an enabled jitterbuffer on the receive USBRADIO side will always
; be used if the sending side can create jitter.
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; and programs. Defaults to 1000.
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an USBRADIO
; channel. Two implementations are currenlty available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; channel. Two implementations are currenlty available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

4
configs/users.conf.sample
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@ -2,11 +2,11 @@
; 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
; 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
; 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.

182
configs/voicemail.conf.sample
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@ -65,7 +65,7 @@ maxlogins=3
;userscontext=default
;
; If you need to have an external program, i.e. /usr/bin/myapp
; called when a voicemail is left, delivered, or your voicemailbox
; called when a voicemail is left, delivered, or your voicemailbox
; is checked, uncomment this.
;externnotify=/usr/bin/myapp
@ -93,7 +93,7 @@ maxlogins=3
;directoryintro=dir-intro
; The character set for voicemail messages can be specified here
;charset=ISO-8859-1
; The ADSI feature descriptor number to download to
; The ADSI feature descriptor number to download to
;adsifdn=0000000F
; The ADSI security lock code
;adsisec=9BDBF7AC
@ -156,58 +156,58 @@ emaildateformat=%A, %B %d, %Y at %r
; ; enables polling mailboxes for changes. Normally, it will
; ; expect that changes are only made when someone called in
; ; to one of the voicemail applications.
; ; Examples of situations that would require this option are
; ; web interfaces to voicemail or an email client in the case
; ; Examples of situations that would require this option are
; ; web interfaces to voicemail or an email client in the case
; ; of using IMAP storage.
;
;pollfreq=30 ; If the "pollmailboxes" option is enabled, this option
; ; sets the polling frequency. The default is once every
; ; 30 seconds.
; If using IMAP storage, specify whether voicemail greetings should be stored
; If using IMAP storage, specify whether voicemail greetings should be stored
; via IMAP. If no, then greetings are stored as if IMAP storage were not enabled
;imapgreetings=no
; If imapgreetings=yes, then specify which folder to store your greetings in. If
; you do not specify a folder, then INBOX will be used
;greetingsfolder=INBOX
; Some IMAP server implementations store folders under INBOX instead of
; Some IMAP server implementations store folders under INBOX instead of
; using a top level folder (ex. INBOX/Friends). In this case, user
; imapparentfolder to set the parent folder. For example, Cyrus IMAP does
; NOT use INBOX as the parent. Default is to have no parent folder set.
;imapparentfolder=INBOX
;
; 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:
;
; 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"
; ${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
; P or p AM or PM
; Q "today", "yesterday" or ABdY
; (*note: not standard strftime value)
; (*note: not standard strftime value)
; q "" (for today), "yesterday", weekday, or ABdY
; (*note: not standard strftime value)
; R 24 hour time, including minute
;
;
; (*note: not standard strftime value)
; R 24 hour time, including minute
;
;
;
; Each mailbox is listed in the form <mailbox>=<password>,<name>,<email>,<pager_email>,<options>
; if the e-mail is specified, a message will be sent when a message is
@ -218,88 +218,88 @@ emaildateformat=%A, %B %d, %Y at %r
; Advanced options example is extension 4069
; NOTE: All options can be expressed globally in the general section, and
; overridden in the per-mailbox settings, unless listed otherwise.
;
;
; tz=central ; Timezone from zonemessages below. Irrelevant if envelope=no.
; attach=yes ; Attach the voicemail to the notification email *NOT* the pager email
; attachfmt=wav49 ; Which format to attach to the email. Normally this is the
; first format specified in the format parameter above, but this
; option lets you customize the format sent to particular mailboxes.
; Useful if Windows users want wav49, but Linux users want gsm.
; [per-mailbox only]
; saycid=yes ; Say the caller id information before the message. If not described,
; or set to no, it will be in the envelope
; cidinternalcontexts=intern ; Internal Context for Name Playback instead of
; extension digits when saying caller id.
; first format specified in the format parameter above, but this
; option lets you customize the format sent to particular mailboxes.
; Useful if Windows users want wav49, but Linux users want gsm.
; [per-mailbox only]
; saycid=yes ; Say the caller id information before the message. If not described,
; or set to no, it will be in the envelope
; cidinternalcontexts=intern ; Internal Context for Name Playback instead of
; extension digits when saying caller id.
; sayduration=no ; Turn on/off the duration information before the message. [ON by default]
; saydurationm=2 ; Specify the minimum duration to say. Default is 2 minutes
; dialout=fromvm ; Context to dial out from [option 4 from mailbox's advanced menu].
; If not specified, option 4 will not be listed and dialing out
; from within VoiceMailMain() will not be permitted.
sendvoicemail=yes ; Allow the user to compose and send a voicemail while inside
; VoiceMailMain() [option 5 from mailbox's advanced menu].
; If set to 'no', option 5 will not be listed.
; dialout=fromvm ; Context to dial out from [option 4 from mailbox's advanced menu].
; If not specified, option 4 will not be listed and dialing out
; from within VoiceMailMain() will not be permitted.
sendvoicemail=yes ; Allow the user to compose and send a voicemail while inside
; VoiceMailMain() [option 5 from mailbox's advanced menu].
; If set to 'no', option 5 will not be listed.
; searchcontexts=yes ; Current default behavior is to search only the default context
; if one is not specified. The older behavior was to search all contexts.
; This option restores the old behavior [DEFAULT=no]
; Note: If you have this option enabled, then you will be required to have
; unique mailbox names across all contexts. Otherwise, an ambiguity is created
; since it is impossible to know which mailbox to retrieve when one is requested.
; callback=fromvm ; Context to call back from
; if not listed, calling the sender back will not be permitted
; if one is not specified. The older behavior was to search all contexts.
; This option restores the old behavior [DEFAULT=no]
; Note: If you have this option enabled, then you will be required to have
; unique mailbox names across all contexts. Otherwise, an ambiguity is created
; since it is impossible to know which mailbox to retrieve when one is requested.
; callback=fromvm ; Context to call back from
; if not listed, calling the sender back will not be permitted
; exitcontext=fromvm ; Context to go to on user exit such as * or 0
; The default is the current context.
; The default is the current context.
; review=yes ; Allow sender to review/rerecord their message before saving it [OFF by default
; operator=yes ; Allow sender to hit 0 before/after/during leaving a voicemail to
; reach an operator. This option REQUIRES an 'o' extension in the
; same context (or in exitcontext, if set), as that is where the
; 0 key will send you. [OFF by default]
; envelope=no ; Turn on/off envelope playback before message playback. [ON by default]
; This does NOT affect option 3,3 from the advanced options menu
; reach an operator. This option REQUIRES an 'o' extension in the
; same context (or in exitcontext, if set), as that is where the
; 0 key will send you. [OFF by default]
; envelope=no ; Turn on/off envelope playback before message playback. [ON by default]
; This does NOT affect option 3,3 from the advanced options menu
; delete=yes ; After notification, the voicemail is deleted from the server. [per-mailbox only]
; This is intended for use with users who wish to receive their
; voicemail ONLY by email. Note: "deletevoicemail" is provided as an
; equivalent option for Realtime configuration.
; This is intended for use with users who wish to receive their
; voicemail ONLY by email. Note: "deletevoicemail" is provided as an
; equivalent option for Realtime configuration.
; volgain=0.0 ; Emails bearing the voicemail may arrive in a volume too
; quiet to be heard. This parameter allows you to specify how
; much gain to add to the message when sending a voicemail.
; NOTE: sox must be installed for this option to work.
; quiet to be heard. This parameter allows you to specify how
; much gain to add to the message when sending a voicemail.
; NOTE: sox must be installed for this option to work.
; nextaftercmd=yes ; Skips to the next message after hitting 7 or 9 to delete/save current message.
; [global option only at this time]
; [global option only at this time]
; forcename=yes ; Forces a new user to record their name. A new user is
; determined by the password being the same as
; the mailbox number. The default is "no".
; determined by the password being the same as
; the mailbox number. The default is "no".
; forcegreetings=no ; This is the same as forcename, except for recording
; greetings. The default is "no".
; greetings. The default is "no".
; hidefromdir=yes ; Hide this mailbox from the directory produced by app_directory
; The default is "no".
; The default is "no".
; tempgreetwarn=yes ; Remind the user that their temporary greeting is set
;messagewrap=no ; Enable next/last message to wrap around to
; first (from last) and last (from first) message
; The default is "no".
; first (from last) and last (from first) message
; The default is "no".
; minpassword=0 ; Enforce minimum password length
; vm-password=custom_sound
; Customize which sound file is used instead of the default
; prompt that says: "password"
; Customize which sound file is used instead of the default
; prompt that says: "password"
; vm-newpassword=custom_sound
; Customize which sound file is used instead of the default
; prompt that says: "Please enter your new password followed by
; the pound key."
; Customize which sound file is used instead of the default
; prompt that says: "Please enter your new password followed by
; the pound key."
; vm-passchanged=custom_sound
; Customize which sound file is used instead of the default
; prompt that says: "Your password has been changed."
; Customize which sound file is used instead of the default
; prompt that says: "Your password has been changed."
; vm-reenterpassword=custom_sound
; Customize which sound file is used instead of the default
; prompt that says: "Please re-enter your password followed by
; the pound key"
; Customize which sound file is used instead of the default
; prompt that says: "Please re-enter your password followed by
; the pound key"
; vm-mismatch=custom_sound
; Customize which sound file is used instead of the default
; prompt that says: "The passwords you entered and re-entered
; did not match. Please try again."
; Customize which sound file is used instead of the default
; prompt that says: "The passwords you entered and re-entered
; did not match. Please try again."
; vm-invalid-password=custom_sound
; Customize which sound file is used instead of the default
; prompt that says: ...
; Customize which sound file is used instead of the default
; prompt that says: ...
; listen-control-forward-key=# ; Customize the key that fast-forwards message playback
; listen-control-reverse-key=* ; Customize the key that rewinds message playback
; listen-control-pause-key=0 ; Customize the key that pauses/unpauses message playback

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