[svn-commits] seanbright: trunk r197528 - /trunk/configs/
SVN commits to the Digium repositories
svn-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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