diff --git a/configs/agents.conf.sample b/configs/agents.conf.sample
index a006d275af12537f85f4f9159d75d33e5c991d04..3ac31332210ccca4131baa10dbbc5cc86c7efbee 100644
--- a/configs/agents.conf.sample
+++ b/configs/agents.conf.sample
@@ -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
;
; --------------------------------------------------
diff --git a/configs/ais.conf.sample b/configs/ais.conf.sample
index f0bccc639cb3488db07374583fdc8c227eda138c..a4428891f96142e0d53c0948fd8dddda6c813cd8 100644
--- a/configs/ais.conf.sample
+++ b/configs/ais.conf.sample
@@ -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]
diff --git a/configs/alarmreceiver.conf.sample b/configs/alarmreceiver.conf.sample
index 0ad23f8fc4d76f6936eef9c82a62566707132958..796470181e9a4b546135fcb3fc89b39a50505c06 100644
--- a/configs/alarmreceiver.conf.sample
+++ b/configs/alarmreceiver.conf.sample
@@ -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
diff --git a/configs/alsa.conf.sample b/configs/alsa.conf.sample
index 33c5a3fa8e493d0796cd546d61fecc3d1e0f70d6..f550306185ef3f9a30a030ec2a2598b649343159 100644
--- a/configs/alsa.conf.sample
+++ b/configs/alsa.conf.sample
@@ -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".
;-----------------------------------------------------------------------------------
diff --git a/configs/amd.conf.sample b/configs/amd.conf.sample
index e25c18e18280269b74ffcdaea8b47d1973317d56..ce4808a0ca3f73d3f38f42024213aa61d4c08249 100644
--- a/configs/amd.conf.sample
+++ b/configs/amd.conf.sample
@@ -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
diff --git a/configs/asterisk.adsi b/configs/asterisk.adsi
index 396de2c75085bc8e1e3d34fadcc2e6fa9343e856..a58952589613955531fc4dd15a2c209c1e2e8191 100644
--- a/configs/asterisk.adsi
+++ b/configs/asterisk.adsi
@@ -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
diff --git a/configs/cdr.conf.sample b/configs/cdr.conf.sample
index 195f88f32a4d906a97c0795cfd62f589d4cfa2b3..0c0413163b304e5668fdae6db42f3f918fb4b1b0 100644
--- a/configs/cdr.conf.sample
+++ b/configs/cdr.conf.sample
@@ -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
diff --git a/configs/chan_dahdi.conf.sample b/configs/chan_dahdi.conf.sample
index 76771fb3ed58228b5649e989587824d580c567fc..6d9847d2a0f0c6e9197a13cd269605bc4467ff7c 100644
--- a/configs/chan_dahdi.conf.sample
+++ b/configs/chan_dahdi.conf.sample
@@ -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.
;
diff --git a/configs/cli_aliases.conf.sample b/configs/cli_aliases.conf.sample
index cc1e2e6d31dda2986298de68bc8d25d960daf4d8..1d9cd9107d7948c6a0751942728c2d1a8d6bc1e1 100644
--- a/configs/cli_aliases.conf.sample
+++ b/configs/cli_aliases.conf.sample
@@ -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
diff --git a/configs/cli_permissions.conf.sample b/configs/cli_permissions.conf.sample
index 7cbad88f340c6dacd92e4ea228809864417b6606..4a6973f507db85ed7225a8e8f9e88cdc69b52353 100644
--- a/configs/cli_permissions.conf.sample
+++ b/configs/cli_permissions.conf.sample
@@ -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.
;
diff --git a/configs/console.conf.sample b/configs/console.conf.sample
index d7e586a6bd135a867a692199a6c7dd0ca81d9753..9bd502696aca321ccb60dc3d8f67d00aa8ae5741 100644
--- a/configs/console.conf.sample
+++ b/configs/console.conf.sample
@@ -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.
diff --git a/configs/dnsmgr.conf.sample b/configs/dnsmgr.conf.sample
index a2939dc10a665980e27c268e0938742731daa7f3..e34dbcf0a8e91fb67dd11abf42048c0d9cd0f149 100644
--- a/configs/dnsmgr.conf.sample
+++ b/configs/dnsmgr.conf.sample
@@ -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)
\ No newline at end of file
+ ; default is 300 (5 minutes)
\ No newline at end of file
diff --git a/configs/dundi.conf.sample b/configs/dundi.conf.sample
index 3eb1bd3208a1827664865fe9d2269dfd73853c89..1b6a174c047afeec76f82c26c403a5cc862b4979 100644
--- a/configs/dundi.conf.sample
+++ b/configs/dundi.conf.sample
@@ -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
diff --git a/configs/extconfig.conf.sample b/configs/extconfig.conf.sample
index 2f1554f6371c17d7e4d915a30cc401a85e99a16f..542bedb52030398b0ed4feb000aa288c1aa05ec1 100644
--- a/configs/extconfig.conf.sample
+++ b/configs/extconfig.conf.sample
@@ -7,7 +7,7 @@
;
[settings]
;
-; Static configuration files:
+; Static configuration files:
;
; file.conf => driver,database[,table]
;
diff --git a/configs/extensions.ael.sample b/configs/extensions.ael.sample
index c7720290afe495a484ebf09163f3a27a19e19bb0..69f441d1e66b0826fa8089090db34ed5bad53228 100644
--- a/configs/extensions.ael.sample
+++ b/configs/extensions.ael.sample
@@ -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.
//
}
diff --git a/configs/extensions.conf.sample b/configs/extensions.conf.sample
index 230576d45e238e0640f279af92cf2dd3b127f035..db93f12e3b04e89a8e813388e2083a8cf0534136 100644
--- a/configs/extensions.conf.sample
+++ b/configs/extensions.conf.sample
@@ -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.
diff --git a/configs/extensions.lua.sample b/configs/extensions.lua.sample
index 0bbb3aef116dc1e5b749f6be6270e805ff06f375..df32ec705953f11b8a7726df902ad8e355d798c1 100644
--- a/configs/extensions.lua.sample
+++ b/configs/extensions.lua.sample
@@ -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;
+ };
}
diff --git a/configs/extensions_minivm.conf.sample b/configs/extensions_minivm.conf.sample
index 75f87c16558d665303f7296f7b05352c8f9d81da..2f9d246370dad28623e1e203aedd6fd6653c6c9f 100644
--- a/configs/extensions_minivm.conf.sample
+++ b/configs/extensions_minivm.conf.sample
@@ -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})
diff --git a/configs/features.conf.sample b/configs/features.conf.sample
index 83aa696439a92126d6cb975efc53571e6175af01..23732d8245ce0a2fa3ef118a2df6a31bb18245c1 100644
--- a/configs/features.conf.sample
+++ b/configs/features.conf.sample
@@ -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
diff --git a/configs/festival.conf.sample b/configs/festival.conf.sample
index 774f1a16c7c967ff3d19eef94838525d9858a5e6..e9182171998e3800a8864f618a4654bd93d63056 100644
--- a/configs/festival.conf.sample
+++ b/configs/festival.conf.sample
@@ -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
;
;
diff --git a/configs/followme.conf.sample b/configs/followme.conf.sample
index a8a9955bbb9d5652a92908d5b1451ce004f8a43d..b11836a5cc36366352e04441cb01c94decf63113 100644
--- a/configs/followme.conf.sample
+++ b/configs/followme.conf.sample
@@ -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.
diff --git a/configs/func_odbc.conf.sample b/configs/func_odbc.conf.sample
index 2b67e53969c9daa768b683433a40606bf1f0d0e9..1bc11be2e6d6de397ecbb93c05321839f968561b 100644
--- a/configs/func_odbc.conf.sample
+++ b/configs/func_odbc.conf.sample
@@ -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
diff --git a/configs/gtalk.conf.sample b/configs/gtalk.conf.sample
index f3dd3f830a4199b90f2b1bad8a1f85befb6ad185..8873d0678f91fe71d736e84051b5d29af45bc321 100644
--- a/configs/gtalk.conf.sample
+++ b/configs/gtalk.conf.sample
@@ -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.
;
diff --git a/configs/h323.conf.sample b/configs/h323.conf.sample
index c2e5db3282326d07992a69348b9c55ee8a548c93..bdeb6b320121d0982d4229d760d779ab4c304d21 100644
--- a/configs/h323.conf.sample
+++ b/configs/h323.conf.sample
@@ -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]
diff --git a/configs/http.conf.sample b/configs/http.conf.sample
index a47a2d65346eca68ef6ad91809f91da8eb3d457e..017263b38922f7e4cd69dd79c96a70047ebead8a 100644
--- a/configs/http.conf.sample
+++ b/configs/http.conf.sample
@@ -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:
diff --git a/configs/iax.conf.sample b/configs/iax.conf.sample
index 259fe626b59423c91a37be0a47958ae5186c6fe4..562a4e6c861d6267c68cedfde19f35f3901de53c 100644
--- a/configs/iax.conf.sample
+++ b/configs/iax.conf.sample
@@ -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
diff --git a/configs/iaxprov.conf.sample b/configs/iaxprov.conf.sample
index 06891d78520dc9564ff4848344e2bf693d4d0edc..d3789dcdecebe61e0dca391d22ebb6025549ce4c 100644
--- a/configs/iaxprov.conf.sample
+++ b/configs/iaxprov.conf.sample
@@ -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"
diff --git a/configs/indications.conf.sample b/configs/indications.conf.sample
index 239dcd11ce0bc6d727e5ff29d08792e97c2c4717..c7ab52ea0c35232a7e32e0afb07f12176691cf74 100644
--- a/configs/indications.conf.sample
+++ b/configs/indications.conf.sample
@@ -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
diff --git a/configs/jabber.conf.sample b/configs/jabber.conf.sample
index 2990d8e919d5fa6d22d4b1d70e542fef5438c65c..6cfb755bde795f60f09dd261334eabd25ff19586 100644
--- a/configs/jabber.conf.sample
+++ b/configs/jabber.conf.sample
@@ -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.
diff --git a/configs/jingle.conf.sample b/configs/jingle.conf.sample
index f3dd3f830a4199b90f2b1bad8a1f85befb6ad185..8873d0678f91fe71d736e84051b5d29af45bc321 100644
--- a/configs/jingle.conf.sample
+++ b/configs/jingle.conf.sample
@@ -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.
;
diff --git a/configs/logger.conf.sample b/configs/logger.conf.sample
index fc64078daadc1f8b864323b89021d483572a40a8..c9e9890a7a740731210cd9ceb5260d54bec46bf0 100644
--- a/configs/logger.conf.sample
+++ b/configs/logger.conf.sample
@@ -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
;
diff --git a/configs/manager.conf.sample b/configs/manager.conf.sample
index 855b9e6bc87709d13da581358e3a75aa504e0503..28a8154019c4c6a93a7cf273dde238c82b2c6406 100644
--- a/configs/manager.conf.sample
+++ b/configs/manager.conf.sample
@@ -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
diff --git a/configs/meetme.conf.sample b/configs/meetme.conf.sample
index 05bcb893f017b815b59b6fec41cc8095e6bf43d7..c40c4606ea945cd3aeab340b58982fbaa65da1cd 100644
--- a/configs/meetme.conf.sample
+++ b/configs/meetme.conf.sample
@@ -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
diff --git a/configs/mgcp.conf.sample b/configs/mgcp.conf.sample
index 01c8fe77ca97dcbd914968d59bbb42cf1a807358..116b66cd036eb606117fb334901b603c2cb32a4e 100644
--- a/configs/mgcp.conf.sample
+++ b/configs/mgcp.conf.sample
@@ -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
diff --git a/configs/minivm.conf.sample b/configs/minivm.conf.sample
index 0e29dd96deca928f487a402b4b80da59d60e5a1c..55a39c869f0ebccf854656a5d83468ad9e0ede58 100644
--- a/configs/minivm.conf.sample
+++ b/configs/minivm.conf.sample
@@ -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
diff --git a/configs/misdn.conf.sample b/configs/misdn.conf.sample
index 08fb288f305614af25a1b5d91be6ef5cff04f756..f4ca486e9b28806b5a8f2aed6784516af481dd72 100644
--- a/configs/misdn.conf.sample
+++ b/configs/misdn.conf.sample
@@ -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".
;-----------------------------------------------------------------------------------
diff --git a/configs/modules.conf.sample b/configs/modules.conf.sample
index bdcc266cd470982d44909fb29c74ebdaa5d72b38..fdcc70a63ab44af8d05e781b4a34858b23ef9bde 100644
--- a/configs/modules.conf.sample
+++ b/configs/modules.conf.sample
@@ -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
diff --git a/configs/musiconhold.conf.sample b/configs/musiconhold.conf.sample
index 39df862bf245733feb4a75b94428885a2d21a371..714e2640305addc320ca692685af45d8e1b295c0 100644
--- a/configs/musiconhold.conf.sample
+++ b/configs/musiconhold.conf.sample
@@ -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
diff --git a/configs/osp.conf.sample b/configs/osp.conf.sample
index 5eccf85d51fe040b6c5695e0271085a025f1922e..08445bb951d39b4b9332f85c5a250e52e954358f 100644
--- a/configs/osp.conf.sample
+++ b/configs/osp.conf.sample
@@ -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
diff --git a/configs/oss.conf.sample b/configs/oss.conf.sample
index f0ed94ea693221f438a55d2ebb21e8e7173be67e..d29d3ac52aef324268a6914c686faa02fc6d8e80 100644
--- a/configs/oss.conf.sample
+++ b/configs/oss.conf.sample
@@ -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
diff --git a/configs/phone.conf.sample b/configs/phone.conf.sample
index 17204501ebe920b7bab119a9c4cb9ed07e8764c3..3d4a7c2dd99acde6a581191bb4acbd2fb6a9e837 100644
--- a/configs/phone.conf.sample
+++ b/configs/phone.conf.sample
@@ -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
diff --git a/configs/phoneprov.conf.sample b/configs/phoneprov.conf.sample
index f3df08c4995521ec44e7ceef832afa85aabafda7..c819d6d634d4a813a2951659e157049061990f0c 100644
--- a/configs/phoneprov.conf.sample
+++ b/configs/phoneprov.conf.sample
@@ -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
diff --git a/configs/queuerules.conf.sample b/configs/queuerules.conf.sample
index 5ab794be7c7753d5a5987b7b140f9ead3c80e6ec..ccabe2cfa415f82603f0a9c8939f7e3d2acb9516 100644
--- a/configs/queuerules.conf.sample
+++ b/configs/queuerules.conf.sample
@@ -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
diff --git a/configs/queues.conf.sample b/configs/queues.conf.sample
index fb45a2e9e085146ea09fcdd4377ba57f0a6961c8..db3ee35155c77068a8df2b23130b1733103122e4 100644
--- a/configs/queues.conf.sample
+++ b/configs/queues.conf.sample
@@ -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
diff --git a/configs/res_odbc.conf.sample b/configs/res_odbc.conf.sample
index 85bd8f45afcfea9793f20975918b6ade0a7a8c73..7e9405dde42bc3bf7b3e39143e429c2fe8a4276b 100644
--- a/configs/res_odbc.conf.sample
+++ b/configs/res_odbc.conf.sample
@@ -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.
diff --git a/configs/res_snmp.conf.sample b/configs/res_snmp.conf.sample
index fb7bfd7508735de20dce3569a2c5c1ca79917604..0aa042b0879d0128be00732e3e358ce8c438b332 100644
--- a/configs/res_snmp.conf.sample
+++ b/configs/res_snmp.conf.sample
@@ -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
diff --git a/configs/rpt.conf.sample b/configs/rpt.conf.sample
index 871793d657d5ee836e3afe897a19c83d17c41c6a..f1c86a11b2c79e943d5bad30a7f803cd2a4a5ce8 100644
--- a/configs/rpt.conf.sample
+++ b/configs/rpt.conf.sample
@@ -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]
diff --git a/configs/rtp.conf.sample b/configs/rtp.conf.sample
index 615b6fe46273a85b4d0a9a9c911d4e8be8616ee0..224dc2abe2ea7dd5299a22ac39e009d2b18e105c 100644
--- a/configs/rtp.conf.sample
+++ b/configs/rtp.conf.sample
@@ -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
diff --git a/configs/say.conf.sample b/configs/say.conf.sample
index f592b780af8ecb5562fcaead1dcbf2382dba986b..16311487544cb254f8360dc45fe3641a4f3f81ae 100644
--- a/configs/say.conf.sample
+++ b/configs/say.conf.sample
@@ -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}
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 862b482d424d189cf30fd1c105446a54a20602ef..20467f1aa1952ccc96da2f0712484403b54dd9a5 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -3,7 +3,7 @@
;
; SIP dial strings
;-----------------------------------------------------------
-; In the dialplan (extensions.conf) you can use several
+; In the dialplan (extensions.conf) you can use several
; syntaxes for dialing SIP devices.
; SIP/devicename
; SIP/username@domain (SIP uri)
@@ -17,11 +17,11 @@
; username@domain
; Call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
-;
+;
; devicename/extension
; If you define a SIP proxy as a peer below, you may call
-; SIP/proxyhostname/user or SIP/user@proxyhostname
-; where the proxyhostname is defined in a section below
+; SIP/proxyhostname/user or SIP/user@proxyhostname
+; where the proxyhostname is defined in a section below
; This syntax also works with ATA's with FXO ports
;
; SIP/username[:password[:md5secret[:authname]]]@host[:port]
@@ -54,7 +54,7 @@
; When naming devices, make sure you understand how Asterisk matches calls
; that come in.
; 1. Asterisk checks the SIP From: address username and matches against
-; names of devices with type=user
+; names of devices with type=user
; The name is the text between square brackets [name]
; 2. Asterisk checks the From: addres and matches the list of devices
; with a type=peer
@@ -64,14 +64,14 @@
; Don't mix extensions with the names of the devices. Devices need a unique
; name. The device name is *not* used as phone numbers. Phone numbers are
; anything you declare as an extension in the dialplan (extensions.conf).
-;
+;
; When setting up trunks, make sure there's no risk that any From: username
-; (caller ID) will match any of your device names, because then Asterisk
+; (caller ID) will match any of your device names, because then Asterisk
; might match the wrong device.
;
; Note: The parameter "username" is not the username and in most cases is
; not needed at all. Check below. In later releases, it's renamed
-; to "defaultuser" which is a better name, since it is used in
+; to "defaultuser" which is a better name, since it is used in
; combination with the "defaultip" setting.
;-----------------------------------------------------------------------------
@@ -81,25 +81,25 @@
; You are encouraged to use the dialplan groupcount functionality
; to enforce call limits instead of using this channel-specific method.
;
-; You can still set limits per device in sip.conf or in a database by using
+; You can still set limits per device in sip.conf or in a database by using
; "setvar" to set variables that can be used in the dialplan for various limits.
[general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
;match_auth_username=yes ; if available, match user entry using the
-; 'username' field from the authentication line
-; instead of the From: field.
+ ; 'username' field from the authentication line
+ ; instead of the From: field.
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
-; Default is enabled
+ ; Default is enabled
;realm=mydomain.tld ; Realm for digest authentication
-; defaults to "asterisk". If you set a system name in
-; asterisk.conf, it defaults to that system name
-; Realms MUST be globally unique according to RFC 3261
-; Set this to your host name or domain name
+ ; defaults to "asterisk". If you set a system name in
+ ; asterisk.conf, it defaults to that system name
+ ; Realms MUST be globally unique according to RFC 3261
+ ; Set this to your host name or domain name
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
-; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
+ ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
;
; Note that the TCP and TLS support for chan_sip is currently considered
@@ -109,20 +109,20 @@ udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0
;
tcpenable=no ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
-; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
+ ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
-; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
-; Remember that the IP address must match the common name (hostname) in the
-; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
+ ; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
+ ; Remember that the IP address must match the common name (hostname) in the
+ ; certificate, so you don't want to bind a TLS socket to multiple IP addresses.
;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem only) to use for TLS connections
-; default is to look for "asterisk.pem" in current directory
+ ; default is to look for "asterisk.pem" in current directory
;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem only) for TLS connections.
-; If no tlsprivatekey is specified, tlscertfile is searched for
-; for both public and private key.
+ ; If no tlsprivatekey is specified, tlscertfile is searched for
+ ; for both public and private key.
;tlscafile=</path/to/certificate>
; If the server your connecting to uses a self signed certificate
@@ -130,12 +130,12 @@ tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0
; verify the authenticity of their certificate.
;tlscadir=</path/to/ca/dir>
-; A directory full of CA certificates. The files must be named with
-; the CA subject name hash value.
-; (see man SSL_CTX_load_verify_locations for more info)
+; A directory full of CA certificates. The files must be named with
+; the CA subject name hash value.
+; (see man SSL_CTX_load_verify_locations for more info)
;tlsdontverifyserver=[yes|no]
-; If set to yes, don't verify the servers certificate when acting as
+; If set to yes, don't verify the servers certificate when acting as
; a client. If you don't have the server's CA certificate you can
; set this and it will connect without requiring tlscafile to be set.
; Default is no.
@@ -146,20 +146,20 @@ tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0
; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
;
;tlsclientmethod=tlsv1 ; values include tlsv1, sslv3, sslv2.
-; Specify protocol for outbound client connections.
-; If left unspecified, the default is sslv2.
+ ; Specify protocol for outbound client connections.
+ ; If left unspecified, the default is sslv2.
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
-; Note: Asterisk only uses the first host
-; in SRV records
-; Disabling DNS SRV lookups disables the
-; ability to place SIP calls based on domain
-; names to some other SIP users on the Internet
+ ; Note: Asterisk only uses the first host
+ ; in SRV records
+ ; Disabling DNS SRV lookups disables the
+ ; ability to place SIP calls based on domain
+ ; names to some other SIP users on the Internet
-;pedantic=yes ; Enable checking of tags in headers,
-; international character conversions in URIs
-; and multiline formatted headers for strict
-; SIP compatibility (defaults to "no")
+;pedantic=yes ; Enable checking of tags in headers,
+ ; international character conversions in URIs
+ ; and multiline formatted headers for strict
+ ; SIP compatibility (defaults to "no")
; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
;tos_sip=cs3 ; Sets TOS for SIP packets.
@@ -173,32 +173,32 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;cos_text=3 ; Sets 802.1p priority for RTP text packets.
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
-; and subscriptions (seconds)
+ ; and subscriptions (seconds)
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
-;qualifyfreq=60 ; Qualification: How often to check for the
-; host to be up in seconds
-; Set to low value if you use low timeout for
-; NAT of UDP sessions
+;qualifyfreq=60 ; Qualification: How often to check for the
+ ; host to be up in seconds
+ ; Set to low value if you use low timeout for
+ ; NAT of UDP sessions
;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
-; fully. Enable this option to not get error messages
-; when sending MWI to phones with this bug.
+ ; fully. Enable this option to not get error messages
+ ; when sending MWI to phones with this bug.
;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
-; the From: header as the "name" portion. Also fill the
-; "user" portion of the URI in the From: header with this
-; value if no fromuser is set
-; Default: empty
-;vmexten=voicemail ; dialplan extension to reach mailbox sets the
-; Message-Account in the MWI notify message
-; defaults to "asterisk"
+ ; the From: header as the "name" portion. Also fill the
+ ; "user" portion of the URI in the From: header with this
+ ; value if no fromuser is set
+ ; Default: empty
+;vmexten=voicemail ; dialplan extension to reach mailbox sets the
+ ; Message-Account in the MWI notify message
+ ; defaults to "asterisk"
;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
-; rather than advertising all joint codec capabilities. This
-; limits the other side's codec choice to exactly what we prefer.
+ ; rather than advertising all joint codec capabilities. This
+ ; limits the other side's codec choice to exactly what we prefer.
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
@@ -220,135 +220,135 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;mohsuggest=default
;
;parkinglot=plaza ; Sets the default parking lot for call parking
-; This may also be set for individual users/peers
-; Parkinglots are configured in features.conf
+ ; This may also be set for individual users/peers
+ ; Parkinglots are configured in features.conf
;language=en ; Default language setting for all users/peers
-; This may also be set for individual users/peers
+ ; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
;sendrpid = rpid ; Use the "Remote-Party-ID" header
-; to send the identity of the remote party
-; This is identical to sendrpid=yes
+ ; to send the identity of the remote party
+ ; This is identical to sendrpid=yes
;sendrpid = pai ; Use the "P-Asserted-Identity" header
-; to send the identity of the remote party
+ ; to send the identity of the remote party
;rpid_update = no ; In certain cases, the only method by which a connected line
-; change may be immediately transmitted is with a SIP UPDATE request.
-; If communicating with another Asterisk server, and you wish to be able
-; transmit such UPDATE messages to it, then you must enable this option.
-; Otherwise, we will have to wait until we can send a reinvite to
-; transmit the information.
+ ; change may be immediately transmitted is with a SIP UPDATE request.
+ ; If communicating with another Asterisk server, and you wish to be able
+ ; transmit such UPDATE messages to it, then you must enable this option.
+ ; Otherwise, we will have to wait until we can send a reinvite to
+ ; transmit the information.
;progressinband=never ; If we should generate in-band ringing always
-; use 'never' to never use in-band signalling, even in cases
-; where some buggy devices might not render it
-; Valid values: yes, no, never Default: never
+ ; use 'never' to never use in-band signalling, even in cases
+ ; where some buggy devices might not render it
+ ; Valid values: yes, no, never Default: never
;useragent=Asterisk PBX ; Allows you to change the user agent string
-; The default user agent string also contains the Asterisk
-; version. If you don't want to expose this, change the
-; useragent string.
+ ; The default user agent string also contains the Asterisk
+ ; version. If you don't want to expose this, change the
+ ; useragent string.
;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
-; Like the useragent parameter, the default user agent string
-; also contains the Asterisk version.
+ ; Like the useragent parameter, the default user agent string
+ ; also contains the Asterisk version.
;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
-; This field MUST NOT contain spaces
+ ; This field MUST NOT contain spaces
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
-; Note that promiscredir when redirects are made to the
-; local system will cause loops since Asterisk is incapable
-; of performing a "hairpin" call.
+ ; Note that promiscredir when redirects are made to the
+ ; local system will cause loops since Asterisk is incapable
+ ; of performing a "hairpin" call.
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
-; a valid phone number
+ ; a valid phone number
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
-; Other options:
-; info : SIP INFO messages (application/dtmf-relay)
-; shortinfo : SIP INFO messages (application/dtmf)
-; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
-; auto : Use rfc2833 if offered, inband otherwise
+ ; Other options:
+ ; info : SIP INFO messages (application/dtmf-relay)
+ ; shortinfo : SIP INFO messages (application/dtmf)
+ ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
+ ; auto : Use rfc2833 if offered, inband otherwise
;compactheaders = yes ; send compact sip headers.
;
;videosupport=yes ; Turn on support for SIP video. You need to turn this
-; on in this section to get any video support at all.
-; You can turn it off on a per peer basis if the general
-; video support is enabled, but you can't enable it for
-; one peer only without enabling in the general section.
-; If you set videosupport to "always", then RTP ports will
-; always be set up for video, even on clients that don't
-; support it. This assists callfile-derived calls and
-; certain transferred calls to use always use video when
-; available. [yes|NO|always]
+ ; on in this section to get any video support at all.
+ ; You can turn it off on a per peer basis if the general
+ ; video support is enabled, but you can't enable it for
+ ; one peer only without enabling in the general section.
+ ; If you set videosupport to "always", then RTP ports will
+ ; always be set up for video, even on clients that don't
+ ; support it. This assists callfile-derived calls and
+ ; certain transferred calls to use always use video when
+ ; available. [yes|NO|always]
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
-; Videosupport and maxcallbitrate is settable
-; for peers and users as well
-;callevents=no ; generate manager events when sip ua
-; performs events (e.g. hold)
+ ; Videosupport and maxcallbitrate is settable
+ ; for peers and users as well
+;callevents=no ; generate manager events when sip ua
+ ; performs events (e.g. hold)
;authfailureevents=no ; generate manager "peerstatus" events when peer can't
-; authenticate with Asterisk. Peerstatus will be "rejected".
+ ; authenticate with Asterisk. Peerstatus will be "rejected".
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
-; for any reason, always reject with an identical response
-; equivalent to valid username and invalid password/hash
-; instead of letting the requester know whether there was
-; a matching user or peer for their request. This reduces
-; the ability of an attacker to scan for valid SIP usernames.
+ ; for any reason, always reject with an identical response
+ ; equivalent to valid username and invalid password/hash
+ ; instead of letting the requester know whether there was
+ ; a matching user or peer for their request. This reduces
+ ; the ability of an attacker to scan for valid SIP usernames.
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
-; order instead of RFC3551 packing order (this is required
-; for Sipura and Grandstream ATAs, among others). This is
-; contrary to the RFC3551 specification, the peer _should_
-; be negotiating AAL2-G726-32 instead :-(
+ ; order instead of RFC3551 packing order (this is required
+ ; for Sipura and Grandstream ATAs, among others). This is
+ ; contrary to the RFC3551 specification, the peer _should_
+ ; be negotiating AAL2-G726-32 instead :-(
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
-;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
+;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls
; ; (could also be tcp,udp) - defining transports on the proxy line only
; ; applies for the global proxy, otherwise use the transport= option
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
-; your localnet setting. Unless you have some sort of strange network
-; setup you will not need to enable this.
+ ; your localnet setting. Unless you have some sort of strange network
+ ; setup you will not need to enable this.
;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
-; as any IP address used for staticly defined
-; hosts. This helps avoid the configuration
-; error of allowing your users to register at
-; the same address as a SIP provider.
+ ; as any IP address used for staticly defined
+ ; hosts. This helps avoid the configuration
+ ; error of allowing your users to register at
+ ; the same address as a SIP provider.
;contactdeny=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
-; register their phones.
+ ; register their phones.
;engine=asterisk ; RTP engine to use when communicating with the device
;
; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given peer who 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 peer 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=sipregistrations
;regextenonqualify=yes ; Default "no"
-; If you have qualify on and the peer becomes unreachable
-; this setting will enforce inactivation of the regexten
-; extension for the peer
+ ; If you have qualify on and the peer becomes unreachable
+ ; this setting will enforce inactivation of the regexten
+ ; extension for the peer
;
;--------------------------- SIP timers ----------------------------------------------------
-; These timers are used primarily in INVITE transactions.
+; These timers are used primarily in INVITE transactions.
; The default for Timer T1 is 500 ms or the measured run-trip time between
; Asterisk and the device if you have qualify=yes for the device.
;
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
-; Defaults to 100 ms
+ ; Defaults to 100 ms
;timert1=500 ; Default T1 timer
-; Defaults to 500 ms or the measured round-trip
-; time to a peer (qualify=yes).
+ ; Defaults to 500 ms or the measured round-trip
+ ; time to a peer (qualify=yes).
;timerb=32000 ; Call setup timer. If a provisional response is not received
-; in this amount of time, the call will autocongest
-; Defaults to 64*timert1
+ ; in this amount of time, the call will autocongest
+ ; Defaults to 64*timert1
;--------------------------- RTP timers ----------------------------------------------------
; These timers are currently used for both audio and video streams. The RTP timeouts
@@ -356,15 +356,15 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; The settings are settable in the global section as well as per device
;
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
-; on the audio channel
-; when we're not on hold. This is to be able to hangup
-; a call in the case of a phone disappearing from the net,
-; like a powerloss or grandma tripping over a cable.
+ ; on the audio channel
+ ; when we're not on hold. This is to be able to hangup
+ ; a call in the case of a phone disappearing from the net,
+ ; like a powerloss or grandma tripping over a cable.
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
-; on the audio channel
-; when we're on hold (must be > rtptimeout)
+ ; on the audio channel
+ ; when we're on hold (must be > rtptimeout)
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
-; (default is off - zero)
+ ; (default is off - zero)
;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
@@ -403,22 +403,22 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging by default, from
-; the moment the channel loads this configuration
-;recordhistory=yes ; Record SIP history by default
-; (see sip history / sip no history)
+ ; the moment the channel loads this configuration
+;recordhistory=yes ; Record SIP history by default
+ ; (see sip history / sip no history)
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
-; SIP history is output to the DEBUG logging channel
+ ; SIP history is output to the DEBUG logging channel
;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
; You can subscribe to the status of extensions with a "hint" priority
; (See extensions.conf.sample for examples)
-; chan_sip support two major formats for notifications: dialog-info and SIMPLE
+; chan_sip support two major formats for notifications: dialog-info and SIMPLE
;
; You will get more detailed reports (busy etc) if you have a call counter enabled
-; for a device.
+; for a device.
;
-; If you set the busylevel, we will indicate busy when we have a number of calls that
+; If you set the busylevel, we will indicate busy when we have a number of calls that
; matches the busylevel treshold.
;
; For queues, you will need this level of detail in status reporting, regardless
@@ -430,38 +430,38 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
-; Useful to limit subscriptions to local extensions
-; Settable per peer/user also
+ ; Useful to limit subscriptions to local extensions
+ ; Settable per peer/user also
;notifyringing = no ; Control whether subscriptions already INUSE get sent
-; RINGING when another call is sent (default: yes)
+ ; RINGING when another call is sent (default: yes)
;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
-; Turning on notifyringing and notifyhold will add a lot
-; more database transactions if you are using realtime.
+ ; Turning on notifyringing and notifyhold will add a lot
+ ; more database transactions if you are using realtime.
;notifycid = yes ; Control whether caller ID information is sent along with
-; dialog-info+xml notifications (supported by snom phones).
-; Note that this feature will only work properly when the
-; incoming call is using the same extension and context that
-; is being used as the hint for the called extension. This means
-; that it won't work when using subscribecontext for your sip
-; user or peer (if subscribecontext is different than context).
-; This is also limited to a single caller, meaning that if an
-; extension is ringing because multiple calls are incoming,
-; only one will be used as the source of caller ID. Specify
-; 'ignore-context' to ignore the called context when looking
-; for the caller's channel. The default value is 'no.' Setting
-; notifycid to 'ignore-context' also causes call-pickups attempted
-; via SNOM's NOTIFY mechanism to set the context for the call pickup
-; to PICKUPMARK.
+ ; dialog-info+xml notifications (supported by snom phones).
+ ; Note that this feature will only work properly when the
+ ; incoming call is using the same extension and context that
+ ; is being used as the hint for the called extension. This means
+ ; that it won't work when using subscribecontext for your sip
+ ; user or peer (if subscribecontext is different than context).
+ ; This is also limited to a single caller, meaning that if an
+ ; extension is ringing because multiple calls are incoming,
+ ; only one will be used as the source of caller ID. Specify
+ ; 'ignore-context' to ignore the called context when looking
+ ; for the caller's channel. The default value is 'no.' Setting
+ ; notifycid to 'ignore-context' also causes call-pickups attempted
+ ; via SNOM's NOTIFY mechanism to set the context for the call pickup
+ ; to PICKUPMARK.
;callcounter = yes ; Enable call counters on devices. This can be set per
-; device too.
+ ; device too.
;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
;
; This setting is available in the [general] section as well as in device configurations.
; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
-; both parties have T38 support enabled in their Asterisk configuration
+; both parties have T38 support enabled in their Asterisk configuration
; This has to be enabled in the general section for all devices to work. You can then
-; disable it on a per device basis.
+; disable it on a per device basis.
;
; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
;
@@ -469,21 +469,21 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
; Fax Detect will cause the SIP channel to jump to the 'fax' extension (if it exists)
; after T.38 is successfully negotiated.
-;
-; faxdetect = yes ; Default false
+;
+; faxdetect = yes ; Default false
;
;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register => [transport://]user[:secret[:authuser]]@domain[:port][/extension][~expiry]
;
-;
;
-; domain is either
+;
+; domain is either
; - domain in DNS
; - host name in DNS
; - the name of a peer defined below or in realtime
-; The domain is where you register your username, so your SIP uri you are registering to
+; The domain is where you register your username, so your SIP uri you are registering to
; is username@domain
;
; If no extension is given, the 's' extension is used. The extension needs to
@@ -514,7 +514,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
; Examples:
;
-;register => 1234:password@mysipprovider.com
+;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
@@ -536,9 +536,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
-; 0 = continue forever, hammering the other server
-; until it accepts the registration
-; Default is 0 tries, continue forever
+ ; 0 = continue forever, hammering the other server
+ ; until it accepts the registration
+ ; Default is 0 tries, continue forever
;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
; by other phones.
@@ -635,7 +635,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; nat = no ; default. Use NAT mode only according to RFC3581 (;rport)
; nat = yes ; Always ignore info and assume NAT
; nat = never ; Never attempt NAT mode or RFC3581 support
-; nat = route ; route = Assume NAT, don't send rport
+; nat = route ; route = Assume NAT, don't send rport
; ; (work around more UNIDEN bugs)
;----------------------------------- MEDIA HANDLING --------------------------------
@@ -645,43 +645,43 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
;
;canreinvite=yes ; Asterisk by default tries to redirect the
-; RTP media stream (audio) to go directly from
-; the caller to the callee. Some devices do not
-; support this (especially if one of them is behind a NAT).
-; The default setting is YES. If you have all clients
-; behind a NAT, or for some other reason wants Asterisk to
-; stay in the audio path, you may want to turn this off.
-
-; This setting also affect direct RTP
-; at call setup (a new feature in 1.4 - setting up the
-; call directly between the endpoints instead of sending
-; a re-INVITE).
+ ; RTP media stream (audio) to go directly from
+ ; the caller to the callee. Some devices do not
+ ; support this (especially if one of them is behind a NAT).
+ ; The default setting is YES. If you have all clients
+ ; behind a NAT, or for some other reason wants Asterisk to
+ ; stay in the audio path, you may want to turn this off.
+
+ ; This setting also affect direct RTP
+ ; at call setup (a new feature in 1.4 - setting up the
+ ; call directly between the endpoints instead of sending
+ ; a re-INVITE).
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
-; the call directly with media peer-2-peer without re-invites.
-; Will not work for video and cases where the callee sends
-; RTP payloads and fmtp headers in the 200 OK that does not match the
-; callers INVITE. This will also fail if canreinvite is enabled when
-; the device is actually behind NAT.
+ ; the call directly with media peer-2-peer without re-invites.
+ ; Will not work for video and cases where the callee sends
+ ; RTP payloads and fmtp headers in the 200 OK that does not match the
+ ; callers INVITE. This will also fail if canreinvite is enabled when
+ ; the device is actually behind NAT.
;canreinvite=nonat ; An additional option is to allow media path redirection
-; (reinvite) but only when the peer where the media is being
-; sent is known to not be behind a NAT (as the RTP core can
-; determine it based on the apparent IP address the media
-; arrives from).
+ ; (reinvite) but only when the peer where the media is being
+ ; sent is known to not be behind a NAT (as the RTP core can
+ ; determine it based on the apparent IP address the media
+ ; arrives from).
;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
-; instead of INVITE. This can be combined with 'nonat', as
-; 'canreinvite=update,nonat'. It implies 'yes'.
+ ; instead of INVITE. This can be combined with 'nonat', as
+ ; 'canreinvite=update,nonat'. It implies 'yes'.
;ignoresdpversion=yes ; By default, Asterisk will honor the session version
-; number in SDP packets and will only modify the SDP
-; session if the version number changes. This option will
-; force asterisk to ignore the SDP session version number
-; and treat all SDP data as new data. This is required
-; for devices that send us non standard SDP packets
-; (observed with Microsoft OCS). By default this option is
-; off.
+ ; number in SDP packets and will only modify the SDP
+ ; session if the version number changes. This option will
+ ; force asterisk to ignore the SDP session version number
+ ; and treat all SDP data as new data. This is required
+ ; for devices that send us non standard SDP packets
+ ; (observed with Microsoft OCS). By default this option is
+ ; off.
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
@@ -689,38 +689,38 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; source code.
;
;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)
;rtsavesysname=yes ; Save systemname in realtime database at registration
-; Default= no
+ ; Default= no
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
-; If set to yes, when a SIP UA registers successfully, the ip address,
-; the origination port, the registration period, and the username of
-; the UA will be set to database via realtime.
-; If not present, defaults to 'yes'. Note: realtime peers will
-; probably not function across reloads in the way that you expect, if
-; you turn this option off.
+ ; If set to yes, when a SIP UA registers successfully, the ip address,
+ ; the origination port, the registration period, and the username of
+ ; the UA will be set to database via realtime.
+ ; If not present, defaults to 'yes'. Note: realtime peers will
+ ; probably not function across reloads in the way that you expect, if
+ ; you turn this option off.
;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.
;ignoreregexpire=yes ; Enabling this setting has two functions:
-;
-; For non-realtime peers, when their registration expires, the
-; information will _not_ be removed from memory or the Asterisk database
-; if you attempt to place a call to the peer, the existing information
-; will be used in spite of it having expired
-;
-; For realtime peers, when the peer is retrieved from realtime storage,
-; the registration information will be used regardless of whether
-; it has expired or not; if it expires while the realtime peer
-; is still in memory (due to caching or other reasons), the
-; information will not be removed from realtime storage
+ ;
+ ; For non-realtime peers, when their registration expires, the
+ ; information will _not_ be removed from memory or the Asterisk database
+ ; if you attempt to place a call to the peer, the existing information
+ ; will be used in spite of it having expired
+ ;
+ ; For realtime peers, when the peer is retrieved from realtime storage,
+ ; the registration information will be used regardless of whether
+ ; it has expired or not; if it expires while the realtime peer
+ ; is still in memory (due to caching or other reasons), the
+ ; information will not be removed from realtime storage
;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
@@ -744,45 +744,45 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; allowexternaldomains=no
;domain=mydomain.tld,mydomain-incoming
-; Add domain and configure incoming context
-; for external calls to this domain
+ ; Add domain and configure incoming context
+ ; for external calls to this domain
;domain=1.2.3.4 ; Add IP address as local domain
-; You can have several "domain" settings
+ ; You can have several "domain" settings
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
-; Default is yes
+ ; Default is yes
;autodomain=yes ; Turn this on to have Asterisk add local host
-; name and local IP to domain list.
+ ; name and local IP to domain list.
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
-; non-peers, use your primary domain "identity"
-; for From: headers instead of just your IP
-; address. This is to be polite and
-; it may be a mandatory requirement for some
-; destinations which do not have a prior
-; account relationship with your server.
+ ; non-peers, use your primary domain "identity"
+ ; for From: headers instead of just your IP
+ ; address. This is to be polite and
+ ; it may be a mandatory requirement for some
+ ; destinations which do not have a prior
+ ; account relationship with your server.
;------------------------------ 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".
;-----------------------------------------------------------------------------------
@@ -793,20 +793,20 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; any credentials in peer/register definition if realm is matched.
;
; This way, Asterisk can authenticate for outbound calls to other
-; realms. We match realm on the proxy challenge and pick an set of
+; realms. We match realm on the proxy challenge and pick an set of
; credentials from this list
; Syntax:
; auth = <user>:<secret>@<realm>
; auth = <user>#<md5secret>@<realm>
; Example:
;auth=mark:topsecret@digium.com
-;
-; You may also add auth= statements to [peer] definitions
+;
+; You may also add auth= statements to [peer] definitions
; Peer auth= override all other authentication settings if we match on realm
;------------------------------------------------------------------------------
; DEVICE CONFIGURATION
-;
+;
; The SIP channel has two types of devices, the friend and the peer.
; * The type=friend is a device type that accepts both incoming and outbound calls,
; where Asterisk match on the From: username on incoming calls.
@@ -817,16 +817,16 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; trunks.
;
; For device names, we recommend using only a-z, numerics (0-9) and underscore
-;
+;
; For local phones, type=friend works most of the time
;
-; If you have one-way audio, you probably have NAT problems.
+; If you have one-way audio, you probably have NAT problems.
; If Asterisk is on a public IP, and the phone is inside of a NAT device
; you will need to configure nat option for those phones.
; Also, turn on qualify=yes to keep the nat session open
-;
-; Configuration options available
-; --------------------
+;
+; Configuration options available
+; --------------------
; context
; callingpres
; permit
@@ -895,7 +895,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
-; We match on IP address of the proxy for incoming calls
+; We match on IP address of the proxy for incoming calls
; since we can not match on username (caller id)
;type=peer
;context=from-fwd
@@ -906,7 +906,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;remotesecret=guessit ; Our password to their service
;defaultuser=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
-;fromdomain=provider.sip.domain
+;fromdomain=provider.sip.domain
;host=box.provider.com
;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
; ; accept both tcp and udp. The default transport type is only used for
@@ -919,7 +919,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;busylevel=2 ; Signal busy at 2 or more calls
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
;port=80 ; The port number we want to connect to on the remote side
-; Also used as "defaultport" in combination with "defaultip" settings
+ ; Also used as "defaultport" in combination with "defaultip" settings
;--- sample definition for a provider
;[provider1]
@@ -940,30 +940,30 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; the templates uncommented as they will not harm:
[basic-options](!) ; a template
-dtmfmode=rfc2833
-context=from-office
-type=friend
+ dtmfmode=rfc2833
+ context=from-office
+ type=friend
[natted-phone](!,basic-options) ; another template inheriting basic-options
-nat=yes
-canreinvite=no
-host=dynamic
+ nat=yes
+ canreinvite=no
+ host=dynamic
[public-phone](!,basic-options) ; another template inheriting basic-options
-nat=no
-canreinvite=yes
+ nat=no
+ canreinvite=yes
[my-codecs](!) ; a template for my preferred codecs
-disallow=all
-allow=ilbc
-allow=g729
-allow=gsm
-allow=g723
-allow=ulaw
+ disallow=all
+ allow=ilbc
+ allow=g729
+ allow=gsm
+ allow=g723
+ allow=ulaw
[ulaw-phone](!) ; and another one for ulaw-only
-disallow=all
-allow=ulaw
+ disallow=all
+ allow=ulaw
; and finally instantiate a few phones
;
@@ -979,34 +979,34 @@ allow=ulaw
; Standard configurations not using templates look like this:
;
;[grandstream1]
-;type=friend
+;type=friend
;context=from-sip ; Where to start in the dialplan when this phone calls
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
-; on incoming calls to Asterisk
+ ; on incoming calls to Asterisk
;host=192.168.0.23 ; we have a static but private IP address
-; No registration allowed
+ ; No registration allowed
;nat=no ; there is not NAT between phone and Asterisk
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
-; from the phone to asterisk (deprecated)
-; 1 for the explicit peer, 1 for the explicit user,
-; remember that a friend equals 1 peer and 1 user in
-; memory
-; There is no combined call counter for a "friend"
-; so there's currently no way in sip.conf to limit
-; to one inbound or outbound call per phone. Use
-; the group counters in the dial plan for that.
-;
+ ; from the phone to asterisk (deprecated)
+ ; 1 for the explicit peer, 1 for the explicit user,
+ ; remember that a friend equals 1 peer and 1 user in
+ ; memory
+ ; There is no combined call counter for a "friend"
+ ; so there's currently no way in sip.conf to limit
+ ; to one inbound or outbound call per phone. Use
+ ; the group counters in the dial plan for that.
+ ;
;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
;disallow=all ; need to disallow=all before we can use allow=
;allow=ulaw ; Note: In user sections the order of codecs
-; listed with allow= does NOT matter!
+ ; listed with allow= does NOT matter!
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
;allow=g729 ; Pass-thru only unless g729 license obtained
;callingpres=allowed_passed_screen ; Set caller ID presentation
-; See README.callingpres for more information
+ ; See README.callingpres for more information
;[xlite1]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
@@ -1029,16 +1029,16 @@ allow=ulaw
;context=from-sip ; Context for incoming calls from this user
;secret=blah
;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
-;language=de ; Use German prompts for this user
+;language=de ; Use German prompts for this user
;host=dynamic ; This peer register with us
;dtmfmode=inband ; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59 ; IP used until peer registers
;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
-;subscribemwi=yes ; Only send notifications if this phone
-; subscribes for mailbox notification
-;vmexten=voicemail ; dialplan extension to reach mailbox
-; sets the Message-Account in the MWI notify message
-; defaults to global vmexten which defaults to "asterisk"
+;subscribemwi=yes ; Only send notifications if this phone
+ ; subscribes for mailbox notification
+;vmexten=voicemail ; dialplan extension to reach mailbox
+ ; sets the Message-Account in the MWI notify message
+ ; defaults to global vmexten which defaults to "asterisk"
;disallow=all
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
@@ -1051,7 +1051,7 @@ allow=ulaw
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
;defaultuser=polly ; Username to use in INVITE until peer registers
;defaultip=192.168.40.123
-; Normally you do NOT need to set this parameter
+ ; Normally you do NOT need to set this parameter
;disallow=all
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no ; Polycom phones don't work properly with "never"
@@ -1061,17 +1061,17 @@ allow=ulaw
;type=friend
;secret=blah
;host=dynamic
-;insecure=port ; Allow matching of peer by IP address without
-; matching port number
+;insecure=port ; Allow matching of peer by IP address without
+ ; matching port number
;insecure=invite ; Do not require authentication of incoming INVITEs
;insecure=port,invite ; (both)
;qualify=1000 ; Consider it down if it's 1 second to reply
-; Helps with NAT session
-; qualify=yes uses default value
-;qualifyfreq=60 ; Qualification: How often to check for the
-; host to be up in seconds
-; Set to low value if you use low timeout for
-; NAT of UDP sessions
+ ; Helps with NAT session
+ ; qualify=yes uses default value
+;qualifyfreq=60 ; Qualification: How often to check for the
+ ; host to be up in seconds
+ ; Set to low value if you use low timeout for
+ ; NAT of UDP sessions
;
; Call group and Pickup group should be in the range from 0 to 63
;
@@ -1086,30 +1086,30 @@ allow=ulaw
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
;nat=yes ; This phone may be natted
-; Send SIP and RTP to the IP address that packet is
-; received from instead of trusting SIP headers
+ ; Send SIP and RTP to the IP address that packet is
+ ; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;canreinvite=no ; Asterisk by default tries to redirect the
-; RTP media stream (audio) to go directly from
-; the caller to the callee. Some devices do not
-; support this (especially if one of them is
-; behind a NAT).
+ ; RTP media stream (audio) to go directly from
+ ; the caller to the callee. Some devices do not
+ ; support this (especially if one of them is
+ ; behind a NAT).
;defaultip=192.168.0.4 ; IP address to use until registration
;defaultuser=goran ; Username to use when calling this device before registration
-; Normally you do NOT need to set this parameter
+ ; Normally you do NOT need to set this parameter
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from or to 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.
;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
-; You must have this turned on or DTMF reception will work improperly.
+ ; You must have this turned on or DTMF reception will work improperly.
;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
-; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
-; external IP address of the remote device. If port forwarding is done at the client side
-; then UDPTL will flow to the remote device.
+ ; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
+ ; external IP address of the remote device. If port forwarding is done at the client side
+ ; then UDPTL will flow to the remote device.
diff --git a/configs/skinny.conf.sample b/configs/skinny.conf.sample
index f5613ac2139172c7bf45e10de4dfc0de28d7b2d0..701723923d3493f9d8bf3740d12e54f316cdb557 100644
--- a/configs/skinny.conf.sample
+++ b/configs/skinny.conf.sample
@@ -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]
diff --git a/configs/sla.conf.sample b/configs/sla.conf.sample
index 9fdb3f3366325abadc08aa8d495f608e6a215c7e..c2015b622765345bbf6c8d801632822c577c621a 100644
--- a/configs/sla.conf.sample
+++ b/configs/sla.conf.sample
@@ -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
diff --git a/configs/telcordia-1.adsi b/configs/telcordia-1.adsi
index 96eb1db21b3138a5af7c84cb859a743b4ff009f6..1486aa95e6a126bc196e5376c359414a700fccb9 100644
--- a/configs/telcordia-1.adsi
+++ b/configs/telcordia-1.adsi
@@ -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
diff --git a/configs/unistim.conf.sample b/configs/unistim.conf.sample
index 2b61a86464f61812adcddad844be71ead16a6e4c..39cb99875e9423048d5cc3ba916fa2f1e5e450ef 100644
--- a/configs/unistim.conf.sample
+++ b/configs/unistim.conf.sample
@@ -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
diff --git a/configs/usbradio.conf.sample b/configs/usbradio.conf.sample
index 2b62ea809abb55b48539b0ba5f4dd716fce65ede..5ba9815ca1c0576ab30f07dd8f09954941948242 100644
--- a/configs/usbradio.conf.sample
+++ b/configs/usbradio.conf.sample
@@ -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".
;-----------------------------------------------------------------------------------
diff --git a/configs/users.conf.sample b/configs/users.conf.sample
index 171b891c1538375219d25b949ef285e2af500230..9258cd3d6d18faf59dc7332bbabd278c52d03761 100644
--- a/configs/users.conf.sample
+++ b/configs/users.conf.sample
@@ -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.
diff --git a/configs/voicemail.conf.sample b/configs/voicemail.conf.sample
index 5d639760876b5c7958aa9bd6e07ced143e64b642..25336425da2337f3c357dcfeae3e1c1e4f0d3a8e 100644
--- a/configs/voicemail.conf.sample
+++ b/configs/voicemail.conf.sample
@@ -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