[asterisk-commits] seanbright: trunk r197528 - /trunk/configs/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu May 28 09:32:29 CDT 2009


Author: seanbright
Date: Thu May 28 09:32:03 2009
New Revision: 197528

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=197528
Log:
Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.

Modified:
    trunk/configs/alarmreceiver.conf.sample
    trunk/configs/alsa.conf.sample
    trunk/configs/amd.conf.sample
    trunk/configs/asterisk.adsi
    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/extensions.ael.sample
    trunk/configs/extensions.conf.sample
    trunk/configs/extensions.lua.sample
    trunk/configs/features.conf.sample
    trunk/configs/func_odbc.conf.sample
    trunk/configs/gtalk.conf.sample
    trunk/configs/h323.conf.sample
    trunk/configs/iax.conf.sample
    trunk/configs/jabber.conf.sample
    trunk/configs/jingle.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/musiconhold.conf.sample
    trunk/configs/oss.conf.sample
    trunk/configs/phoneprov.conf.sample
    trunk/configs/queues.conf.sample
    trunk/configs/res_odbc.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/voicemail.conf.sample

Modified: trunk/configs/alarmreceiver.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/alarmreceiver.conf.sample?view=diff&rev=197528&r1=197527&r2=197528
==============================================================================
--- trunk/configs/alarmreceiver.conf.sample (original)
+++ trunk/configs/alarmreceiver.conf.sample Thu May 28 09:32:03 2009
@@ -10,7 +10,7 @@
 ;                                                                                                                                   
 ; Specify a timestamp format for the metadata section of the event files
 ; Default is %a %b %d, %Y @ %H:%M:%S %Z
-	        
+
 timestampformat = %a %b %d, %Y @ %H:%M:%S %Z
 
 ;

Modified: trunk/configs/alsa.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/alsa.conf.sample?view=diff&rev=197528&r1=197527&r2=197528
==============================================================================
--- trunk/configs/alsa.conf.sample (original)
+++ trunk/configs/alsa.conf.sample Thu May 28 09:32:03 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=197528&r1=197527&r2=197528
==============================================================================
--- trunk/configs/amd.conf.sample (original)
+++ trunk/configs/amd.conf.sample Thu May 28 09:32:03 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=197528&r1=197527&r2=197528
==============================================================================
--- trunk/configs/asterisk.adsi (original)
+++ trunk/configs/asterisk.adsi Thu May 28 09:32:03 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/chan_dahdi.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/chan_dahdi.conf.sample?view=diff&rev=197528&r1=197527&r2=197528
==============================================================================
--- trunk/configs/chan_dahdi.conf.sample (original)
+++ trunk/configs/chan_dahdi.conf.sample Thu May 28 09:32:03 2009
@@ -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
@@ -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".
 ;-----------------------------------------------------------------------------------

Modified: trunk/configs/cli_aliases.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/cli_aliases.conf.sample?view=diff&rev=197528&r1=197527&r2=197528
==============================================================================
--- trunk/configs/cli_aliases.conf.sample (original)
+++ trunk/configs/cli_aliases.conf.sample Thu May 28 09:32:03 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,

Modified: trunk/configs/cli_permissions.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/cli_permissions.conf.sample?view=diff&rev=197528&r1=197527&r2=197528
==============================================================================
--- trunk/configs/cli_permissions.conf.sample (original)
+++ trunk/configs/cli_permissions.conf.sample Thu May 28 09:32:03 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=197528&r1=197527&r2=197528
==============================================================================
--- trunk/configs/console.conf.sample (original)
+++ trunk/configs/console.conf.sample Thu May 28 09:32:03 2009
@@ -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 @@
 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.

Modified: trunk/configs/dnsmgr.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/dnsmgr.conf.sample?view=diff&rev=197528&r1=197527&r2=197528
==============================================================================
--- trunk/configs/dnsmgr.conf.sample (original)
+++ trunk/configs/dnsmgr.conf.sample Thu May 28 09:32:03 2009
@@ -1,5 +1,5 @@
 [general]
 ;enable=yes		; enable creation of managed DNS lookups
-			;   default is 'no'
+;   default is 'no'
 ;refreshinterval=1200	; refresh managed DNS lookups every <n> seconds
-			;   default is 300 (5 minutes)
+;   default is 300 (5 minutes)

Modified: trunk/configs/extensions.ael.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/extensions.ael.sample?view=diff&rev=197528&r1=197527&r2=197528
==============================================================================
--- trunk/configs/extensions.ael.sample (original)
+++ trunk/configs/extensions.ael.sample Thu May 28 09:32:03 2009
@@ -19,28 +19,28 @@
 //
 
 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 at provider
+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 at provider
 };
 
 //
@@ -110,61 +110,61 @@
 //
 //
 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
-	//
-	
-	switches { 
-		DUNDi/e164;
-	};
+//
+// Just a wrapper for the switch
+//
+
+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;
+};
+//
 };
 
 //
@@ -175,8 +175,8 @@
 //
 // ARG1 is the extension to Dial
 //
-	goto ${exten}|1;
-	return;
+goto ${exten}|1;
+return;
 };
 
 //
@@ -186,7 +186,7 @@
 // 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);
 };
 
 //
@@ -196,91 +196,91 @@
 // 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
-	//
-	
-	_91800NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
-	_91888NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
-	_91877NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
-	_91866NXXXXXX => Dial(${TRUNK}/${EXTEN:${TRUNKMSD}});
+//
+// 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}});
 };
 
 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;
+};
 };
 
 //
@@ -306,69 +306,69 @@
 
 
 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 at misery.digium.com/s at 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 at misery.digium.com/s at 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);
 };
 
 
@@ -383,9 +383,9 @@
 // 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

Modified: trunk/configs/extensions.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/extensions.conf.sample?view=diff&rev=197528&r1=197527&r2=197528
==============================================================================
--- trunk/configs/extensions.conf.sample (original)
+++ trunk/configs/extensions.conf.sample Thu May 28 09:32:03 2009
@@ -430,7 +430,7 @@
 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
 
@@ -459,7 +459,7 @@
 
 exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
 exten => _X.,n,Dial(${dev},20,p)			; Ring the interface, 20 seconds maximum, call screening 
-						; option (or use P for databased call _X.creening)
+; option (or use P for databased call _X.creening)
 exten => _X.,n,Goto(s-${DIALSTATUS},1)		; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
 
 exten => stdexten-NOANSWER,1,Voicemail(${mbx},u)	; If unavailable, send to voicemail w/ unavail announce
@@ -521,7 +521,7 @@
 ; voicemail, etc.
 ;
 exten => 1234,1,Playback(transfer,skip)		; "Please hold while..." 
-					; (but skip if channel is not up)
+; (but skip if channel is not up)
 exten => 1234,n,Gosub(stdexten(1234,${GLOBAL(CONSOLE)}))
 exten => 1234,n,Goto(default,s,1)		; exited Voicemail
 
@@ -640,11 +640,11 @@
 ;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)
 

Modified: trunk/configs/extensions.lua.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/extensions.lua.sample?view=diff&rev=197528&r1=197527&r2=197528
==============================================================================
--- trunk/configs/extensions.lua.sample (original)
+++ trunk/configs/extensions.lua.sample Thu May 28 09:32:03 2009
@@ -97,103 +97,103 @@
 --
 
 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;
+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;
+["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;
+["1000"] = function()
+app.goto("default", "s", 1)
+end;
 
-		["1234"] = function()
-			app.playback("transfer", "skip")
-			-- do a dial here
-		end;
+["1234"] = function()
+app.playback("transfer", "skip")
+-- do a dial here
+end;
 
-		["1235"] = function()
-			app.voicemail("1234", "u")
-		end;
+["1235"] = function()
+app.voicemail("1234", "u")
+end;
 
-		["1236"] = function()
-			app.dial("Console/dsp")
-			app.voicemail(1234, "b")
-		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;
+["#"] = demo_hangup;
+t = demo_hangup;
+i = function()
+app.playback("invalid")
+demo_instruct()
+end;
 
-		["500"] = function()
-			app.playback("demo-abouttotry")
-			app.dial("IAX2/guest at misery.digium.com/s at default")
-			app.playback("demo-nogo")
-			demo_instruct()
-		end;
+["500"] = function()
+app.playback("demo-abouttotry")
+app.dial("IAX2/guest at misery.digium.com/s at default")
+app.playback("demo-nogo")
+demo_instruct()
+end;
 
-		["600"] = function()
-			app.playback("demo-echotest")
-			app.echo()
-			app.playback("demo-echodone")
-			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;
+["8500"] = function()
+app.voicemailmain()
+demo_instruct()
+end;
 
-	};
+};
 
-	default = {
-		-- by default, do the demo
-		include = {"demo"};
-	};
+default = {
+-- by default, do the demo
+include = {"demo"};
+};
 
-	["local"] = {
-		["_NXXXXXX"] = outgoing_local;
-	};
+["local"] = {
+["_NXXXXXX"] = outgoing_local;
+};
 }
 

Modified: trunk/configs/features.conf.sample
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/configs/features.conf.sample?view=diff&rev=197528&r1=197527&r2=197528
==============================================================================
--- trunk/configs/features.conf.sample (original)
+++ trunk/configs/features.conf.sample Thu May 28 09:32:03 2009
@@ -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)
+; (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).
+; 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.
+; 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
+; 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

[... 2573 lines stripped ...]



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