[svn-commits] seanbright: trunk r197535 - /trunk/configs/

SVN commits to the Digium repositories svn-commits at lists.digium.com
Thu May 28 09:39:27 CDT 2009


Author: seanbright
Date: Thu May 28 09:39:21 2009
New Revision: 197535

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=197535
Log:
Let's try that again, this time removing trailing whitespace and not leading
whitespace.  I can't believe no one noticed.

Modified:
    trunk/configs/agents.conf.sample
    trunk/configs/ais.conf.sample
    trunk/configs/alarmreceiver.conf.sample
    trunk/configs/alsa.conf.sample
    trunk/configs/amd.conf.sample
    trunk/configs/asterisk.adsi
    trunk/configs/cdr.conf.sample
    trunk/configs/chan_dahdi.conf.sample
    trunk/configs/cli_aliases.conf.sample
    trunk/configs/cli_permissions.conf.sample
    trunk/configs/console.conf.sample
    trunk/configs/dnsmgr.conf.sample
    trunk/configs/dundi.conf.sample
    trunk/configs/extconfig.conf.sample
    trunk/configs/extensions.ael.sample
    trunk/configs/extensions.conf.sample
    trunk/configs/extensions.lua.sample
    trunk/configs/extensions_minivm.conf.sample
    trunk/configs/features.conf.sample
    trunk/configs/festival.conf.sample
    trunk/configs/followme.conf.sample
    trunk/configs/func_odbc.conf.sample
    trunk/configs/gtalk.conf.sample
    trunk/configs/h323.conf.sample
    trunk/configs/http.conf.sample
    trunk/configs/iax.conf.sample
    trunk/configs/iaxprov.conf.sample
    trunk/configs/indications.conf.sample
    trunk/configs/jabber.conf.sample
    trunk/configs/jingle.conf.sample
    trunk/configs/logger.conf.sample
    trunk/configs/manager.conf.sample
    trunk/configs/meetme.conf.sample
    trunk/configs/mgcp.conf.sample
    trunk/configs/minivm.conf.sample
    trunk/configs/misdn.conf.sample
    trunk/configs/modules.conf.sample
    trunk/configs/musiconhold.conf.sample
    trunk/configs/osp.conf.sample
    trunk/configs/oss.conf.sample
    trunk/configs/phone.conf.sample
    trunk/configs/phoneprov.conf.sample
    trunk/configs/queuerules.conf.sample
    trunk/configs/queues.conf.sample
    trunk/configs/res_odbc.conf.sample
    trunk/configs/res_snmp.conf.sample
    trunk/configs/rpt.conf.sample
    trunk/configs/rtp.conf.sample
    trunk/configs/say.conf.sample
    trunk/configs/sip.conf.sample
    trunk/configs/skinny.conf.sample
    trunk/configs/sla.conf.sample
    trunk/configs/telcordia-1.adsi
    trunk/configs/unistim.conf.sample
    trunk/configs/usbradio.conf.sample
    trunk/configs/users.conf.sample
    trunk/configs/voicemail.conf.sample

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]

Modified: trunk/configs/alarmreceiver.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/alarmreceiver.conf.sample?view=diff&rev=197535&r1=197534&r2=197535
==============================================================================
--- trunk/configs/alarmreceiver.conf.sample (original)
+++ trunk/configs/alarmreceiver.conf.sample Thu May 28 09:39:21 2009
@@ -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 @@
 
 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.
 ;
 
 sdtimeout = 200

Modified: trunk/configs/alsa.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/alsa.conf.sample?view=diff&rev=197535&r1=197534&r2=197535
==============================================================================
--- trunk/configs/alsa.conf.sample (original)
+++ trunk/configs/alsa.conf.sample Thu May 28 09:39:21 2009
@@ -39,23 +39,23 @@
 
 ;------------------------------ 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".
 ;-----------------------------------------------------------------------------------

Modified: trunk/configs/amd.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/amd.conf.sample?view=diff&rev=197535&r1=197534&r2=197535
==============================================================================
--- trunk/configs/amd.conf.sample (original)
+++ trunk/configs/amd.conf.sample Thu May 28 09:39:21 2009
@@ -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

Modified: trunk/configs/asterisk.adsi
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/asterisk.adsi?view=diff&rev=197535&r1=197534&r2=197535
==============================================================================
--- trunk/configs/asterisk.adsi (original)
+++ trunk/configs/asterisk.adsi Thu May 28 09:39:21 2009
@@ -35,39 +35,39 @@
 ; 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 @@
 ;
 
 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
 

Modified: trunk/configs/cdr.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/cdr.conf.sample?view=diff&rev=197535&r1=197534&r2=197535
==============================================================================
--- trunk/configs/cdr.conf.sample (original)
+++ trunk/configs/cdr.conf.sample Thu May 28 09:39:21 2009
@@ -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

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 at context 
-;
-; Enable echo cancellation 
+;mailbox=1234 at 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

Modified: trunk/configs/cli_permissions.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/cli_permissions.conf.sample?view=diff&rev=197535&r1=197534&r2=197535
==============================================================================
--- trunk/configs/cli_permissions.conf.sample (original)
+++ trunk/configs/cli_permissions.conf.sample Thu May 28 09:39:21 2009
@@ -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.
 ;

Modified: trunk/configs/console.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/console.conf.sample?view=diff&rev=197535&r1=197534&r2=197535
==============================================================================
--- trunk/configs/console.conf.sample (original)
+++ trunk/configs/console.conf.sample Thu May 28 09:39:21 2009
@@ -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 --------------------------

[... 5436 lines stripped ...]



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