Modified: trunk/configs/agents.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/agents.conf.sample?view=diff&rev=197535&r1=197534&r2=197535
==============================================================================
--- trunk/configs/agents.conf.sample (original)
+++ trunk/configs/agents.conf.sample Thu May 28 09:39:21 2009
@@ -32,14 +32,14 @@
; 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 @@
;
;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 @@
; --------------------------------------------------
; 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 @@
; /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
;
; --------------------------------------------------
Modified: trunk/configs/ais.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/ais.conf.sample?view=diff&rev=197535&r1=197534&r2=197535
==============================================================================
--- trunk/configs/ais.conf.sample (original)
+++ trunk/configs/ais.conf.sample Thu May 28 09:39:21 2009
@@ -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]
-;
+;
; 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 @@
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 @@
; 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 @@
; 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.
;
;------------------------------ 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".
;-----------------------------------------------------------------------------------
[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
; 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
Modified: trunk/configs/chan_dahdi.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/chan_dahdi.conf.sample?view=diff&rev=197535&r1=197534&r2=197535
==============================================================================
--- trunk/configs/chan_dahdi.conf.sample (original)
+++ trunk/configs/chan_dahdi.conf.sample Thu May 28 09:39:21 2009
@@ -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 =
-;
-; sample 2 for Germany
+;unknownprefix =
+;
+; 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).
-;
-; This is a global, rather than a per-channel setting. It will not be
+; (in milliseconds).
+;
+; This is a global, rather than a per-channel setting. It will not be
; updated on a reload.
;
;toneduration=100
@@ -354,7 +354,7 @@
; 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 @@
; 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 @@
;
; 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 @@
;
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 @@
;
; for any other voicemail context, the following will produce the stutter tone:
;
-;mailbox=1234@context
-;
-; Enable echo cancellation
+;mailbox=1234@context
+;
+; 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 @@
;
; 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 @@
; 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 @@
;
; 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 @@
;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 @@
;
;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 @@
; 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 @@
;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 @@
;
;------------------------------ 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 @@
; 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 @@
; 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 @@
;
;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 @@
; '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 @@
; 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 @@
; 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.
;
Modified: trunk/configs/cli_aliases.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/cli_aliases.conf.sample?view=diff&rev=197535&r1=197534&r2=197535
==============================================================================
--- trunk/configs/cli_aliases.conf.sample (original)
+++ trunk/configs/cli_aliases.conf.sample Thu May 28 09:39:21 2009
@@ -13,8 +13,8 @@
;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 @@
; 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 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 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
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.
;
; 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 @@
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