From a22b4735e5a6b8745a4915a260995886c56c7ffe Mon Sep 17 00:00:00 2001 From: seanbright Date: Thu, 28 May 2009 14:39:21 +0000 Subject: [PATCH] Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197535 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/agents.conf.sample | 14 +- configs/ais.conf.sample | 4 +- configs/alarmreceiver.conf.sample | 8 +- configs/alsa.conf.sample | 22 +- configs/amd.conf.sample | 10 +- configs/asterisk.adsi | 180 ++++---- configs/cdr.conf.sample | 22 +- configs/chan_dahdi.conf.sample | 130 +++--- configs/cli_aliases.conf.sample | 10 +- configs/cli_permissions.conf.sample | 2 +- configs/console.conf.sample | 36 +- configs/dnsmgr.conf.sample | 4 +- configs/dundi.conf.sample | 26 +- configs/extconfig.conf.sample | 2 +- configs/extensions.ael.sample | 414 ++++++++--------- configs/extensions.conf.sample | 74 +-- configs/extensions.lua.sample | 150 +++--- configs/extensions_minivm.conf.sample | 4 +- configs/features.conf.sample | 50 +- configs/festival.conf.sample | 12 +- configs/followme.conf.sample | 8 +- configs/func_odbc.conf.sample | 8 +- configs/gtalk.conf.sample | 12 +- configs/h323.conf.sample | 48 +- configs/http.conf.sample | 2 +- configs/iax.conf.sample | 86 ++-- configs/iaxprov.conf.sample | 2 +- configs/indications.conf.sample | 6 +- configs/jabber.conf.sample | 10 +- configs/jingle.conf.sample | 12 +- configs/logger.conf.sample | 4 +- configs/manager.conf.sample | 20 +- configs/meetme.conf.sample | 18 +- configs/mgcp.conf.sample | 64 +-- configs/minivm.conf.sample | 76 +-- configs/misdn.conf.sample | 24 +- configs/modules.conf.sample | 4 +- configs/musiconhold.conf.sample | 8 +- configs/osp.conf.sample | 56 +-- configs/oss.conf.sample | 182 ++++---- configs/phone.conf.sample | 4 +- configs/phoneprov.conf.sample | 62 +-- configs/queuerules.conf.sample | 14 +- configs/queues.conf.sample | 94 ++-- configs/res_odbc.conf.sample | 12 +- configs/res_snmp.conf.sample | 2 +- configs/rpt.conf.sample | 32 +- configs/rtp.conf.sample | 4 +- configs/say.conf.sample | 206 ++++----- configs/sip.conf.sample | 640 +++++++++++++------------- configs/skinny.conf.sample | 60 +-- configs/sla.conf.sample | 110 ++--- configs/telcordia-1.adsi | 70 +-- configs/unistim.conf.sample | 32 +- configs/usbradio.conf.sample | 22 +- configs/users.conf.sample | 4 +- configs/voicemail.conf.sample | 182 ++++---- 57 files changed, 1687 insertions(+), 1687 deletions(-) diff --git a/configs/agents.conf.sample b/configs/agents.conf.sample index a006d275a..3ac313322 100644 --- a/configs/agents.conf.sample +++ b/configs/agents.conf.sample @@ -32,14 +32,14 @@ persistentagents=yes ; Define autologoffunavail to have agents automatically logged ; out when the extension that they are at returns a CHANUNAVAIL ; status when a call is attempted to be sent there. -; Default is "no". +; Default is "no". ; ;autologoffunavail=yes ; ; Define ackcall to require a DTMF acknowledgement when ; an agent logs in using agentcallbacklogin. Default is "no". ; Can also be set to "always", which will also require AgentLogin -; agents to acknowledge calls. Use the acceptdtmf option to +; agents to acknowledge calls. Use the acceptdtmf option to ; configure what DTMF key press should be used to acknowledge the ; call. The default is '#'. ; @@ -70,14 +70,14 @@ persistentagents=yes ; ;goodbye => goodbye_file ; -; Define updatecdr. This is whether or not to change the source -; channel in the CDR record for this call to agent/agent_id so +; Define updatecdr. This is whether or not to change the source +; channel in the CDR record for this call to agent/agent_id so ; that we know which agent generates the call ; ;updatecdr=no ; ; Group memberships for agents (may change in mid-file) -; +; ;group=3 ;group=1,2 ;group= @@ -85,7 +85,7 @@ persistentagents=yes ; -------------------------------------------------- ; This section is devoted to recording agent's calls ; The keywords are global to the chan_agent channel driver -; +; ; Enable recording calls addressed to agents. It's turned off by default. ;recordagentcalls=yes ; @@ -100,7 +100,7 @@ persistentagents=yes ; /var/spool/asterisk/monitor ;savecallsin=/var/calls ; -; An optional custom beep sound file to play to always-connected agents. +; An optional custom beep sound file to play to always-connected agents. ;custom_beep=beep ; ; -------------------------------------------------- diff --git a/configs/ais.conf.sample b/configs/ais.conf.sample index f0bccc639..a4428891f 100644 --- a/configs/ais.conf.sample +++ b/configs/ais.conf.sample @@ -1,5 +1,5 @@ ; -; Sample configuration file for res_ais +; Sample configuration file for res_ais ; * SAForum AIS (Application Interface Specification) ; ; More information on the AIS specification is available from the SAForum. @@ -76,7 +76,7 @@ ; ; This example would be used for a node that has phones directly registered ; to it, but does not have direct access to voicemail. So, this node wants -; to be informed about MWI state changes on other voicemail server nodes, but +; to be informed about MWI state changes on other voicemail server nodes, but ; is not capable of publishing any state changes. ; ; [mwi] diff --git a/configs/alarmreceiver.conf.sample b/configs/alarmreceiver.conf.sample index 0ad23f8fc..796470181 100644 --- a/configs/alarmreceiver.conf.sample +++ b/configs/alarmreceiver.conf.sample @@ -7,7 +7,7 @@ [general] -; +; ; Specify a timestamp format for the metadata section of the event files ; Default is %a %b %d, %Y @ %H:%M:%S %Z @@ -32,7 +32,7 @@ timestampformat = %a %b %d, %Y @ %H:%M:%S %Z eventspooldir = /tmp -; +; ; The alarmreceiver app can either log the events one-at-a-time to individual ; files in the spool directory, or it can store them until the caller ; disconnects and write them all to one file. @@ -46,7 +46,7 @@ logindividualevents = no ; The timeout for receiving the first DTMF digit is adjustable from 1000 msec. ; to 10000 msec. The default is 2000 msec. Note: if you wish to test the ; receiver by entering digits manually, set this to a reasonable time out -; like 10000 milliseconds. +; like 10000 milliseconds. fdtimeout = 2000 @@ -54,7 +54,7 @@ fdtimeout = 2000 ; The timeout for receiving subsequent DTMF digits is adjustable from ; 110 msec. to 4000 msec. The default is 200 msec. Note: if you wish to test ; the receiver by entering digits manually, set this to a reasonable time out -; like 4000 milliseconds. +; like 4000 milliseconds. ; sdtimeout = 200 diff --git a/configs/alsa.conf.sample b/configs/alsa.conf.sample index 33c5a3fa8..f55030618 100644 --- a/configs/alsa.conf.sample +++ b/configs/alsa.conf.sample @@ -39,23 +39,23 @@ extension=s ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an -; ALSA channel. Defaults to "no". An enabled jitterbuffer will -; be used only if the sending side can create and the receiving -; side can not accept jitter. The ALSA channel can't accept jitter, -; thus an enabled jitterbuffer on the receive ALSA side will always -; be used if the sending side can create jitter. + ; ALSA channel. Defaults to "no". An enabled jitterbuffer will + ; be used only if the sending side can create and the receiving + ; side can not accept jitter. The ALSA channel can't accept jitter, + ; thus an enabled jitterbuffer on the receive ALSA side will always + ; be used if the sending side can create jitter. ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is -; resynchronized. Useful to improve the quality of the voice, with -; big jumps in/broken timestamps, usually sent from exotic devices -; and programs. Defaults to 1000. + ; resynchronized. Useful to improve the quality of the voice, with + ; big jumps in/broken timestamps, usually sent from exotic devices + ; and programs. Defaults to 1000. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP -; channel. Two implementations are currently available - "fixed" -; (with size always equals to jbmax-size) and "adaptive" (with -; variable size, actually the new jb of IAX2). Defaults to fixed. + ; channel. Two implementations are currently available - "fixed" + ; (with size always equals to jbmax-size) and "adaptive" (with + ; variable size, actually the new jb of IAX2). Defaults to fixed. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". ;----------------------------------------------------------------------------------- diff --git a/configs/amd.conf.sample b/configs/amd.conf.sample index e25c18e18..ce4808a0c 100644 --- a/configs/amd.conf.sample +++ b/configs/amd.conf.sample @@ -4,15 +4,15 @@ [general] initial_silence = 2500 ; Maximum silence duration before the greeting. -; If exceeded then MACHINE. + ; If exceeded then MACHINE. greeting = 1500 ; Maximum length of a greeting. If exceeded then MACHINE. after_greeting_silence = 800 ; Silence after detecting a greeting. -; If exceeded then HUMAN + ; If exceeded then HUMAN total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide -; on a HUMAN or MACHINE + ; on a HUMAN or MACHINE min_word_length = 100 ; Minimum duration of Voice to considered as a word between_words_silence = 50 ; Minimum duration of silence after a word to consider -; the audio what follows as a new word + ; the audio what follows as a new word maximum_number_of_words = 3 ; Maximum number of words in the greeting. -; If exceeded then MACHINE + ; If exceeded then MACHINE silence_threshold = 256 diff --git a/configs/asterisk.adsi b/configs/asterisk.adsi index 396de2c75..a58952589 100644 --- a/configs/asterisk.adsi +++ b/configs/asterisk.adsi @@ -35,39 +35,39 @@ DISPLAY "empty" IS "asdf" ; Begin soft key definitions ; KEY "callfwd" IS "CallFwd" OR "Call Forward" -OFFHOOK -VOICEMODE -WAITDIALTONE -SENDDTMF "*60" -GOTO "offHook" + OFFHOOK + VOICEMODE + WAITDIALTONE + SENDDTMF "*60" + GOTO "offHook" ENDKEY KEY "vmail_OH" IS "VMail" OR "Voicemail" -OFFHOOK -VOICEMODE -WAITDIALTONE -SENDDTMF "8500" + OFFHOOK + VOICEMODE + WAITDIALTONE + SENDDTMF "8500" ENDKEY KEY "vmail" IS "VMail" OR "Voicemail" -SENDDTMF "8500" + SENDDTMF "8500" ENDKEY KEY "backspace" IS "BackSpc" OR "Backspace" -BACKSPACE + BACKSPACE ENDKEY KEY "cwdisable" IS "CWDsble" OR "Disable Call Wait" -SENDDTMF "*70" -SETFLAG "nocallwaiting" -SHOWDISPLAY "cwdisabled" AT 4 -TIMERCLEAR -TIMERSTART 1 + SENDDTMF "*70" + SETFLAG "nocallwaiting" + SHOWDISPLAY "cwdisabled" AT 4 + TIMERCLEAR + TIMERSTART 1 ENDKEY KEY "cidblock" IS "CIDBlk" OR "Block Callerid" -SENDDTMF "*67" -SETFLAG "nocallwaiting" + SENDDTMF "*67" + SETFLAG "nocallwaiting" ENDKEY ; @@ -75,85 +75,85 @@ ENDKEY ; SUB "main" IS -IFEVENT NEARANSWER THEN -CLEAR -SHOWDISPLAY "titles" AT 1 NOUPDATE -SHOWDISPLAY "talkingto" AT 2 NOUPDATE -SHOWDISPLAY "callname" AT 3 -SHOWDISPLAY "callnum" AT 4 -GOTO "stableCall" -ENDIF -IFEVENT OFFHOOK THEN -CLEAR -CLEARFLAG "nocallwaiting" -CLEARDISPLAY -SHOWDISPLAY "titles" AT 1 -SHOWKEYS "vmail" -SHOWKEYS "cidblock" -SHOWKEYS "cwdisable" UNLESS "nocallwaiting" -GOTO "offHook" -ENDIF -IFEVENT IDLE THEN -CLEAR -SHOWDISPLAY "titles" AT 1 -SHOWKEYS "vmail_OH" -ENDIF -IFEVENT CALLERID THEN -CLEAR + IFEVENT NEARANSWER THEN + CLEAR + SHOWDISPLAY "titles" AT 1 NOUPDATE + SHOWDISPLAY "talkingto" AT 2 NOUPDATE + SHOWDISPLAY "callname" AT 3 + SHOWDISPLAY "callnum" AT 4 + GOTO "stableCall" + ENDIF + IFEVENT OFFHOOK THEN + CLEAR + CLEARFLAG "nocallwaiting" + CLEARDISPLAY + SHOWDISPLAY "titles" AT 1 + SHOWKEYS "vmail" + SHOWKEYS "cidblock" + SHOWKEYS "cwdisable" UNLESS "nocallwaiting" + GOTO "offHook" + ENDIF + IFEVENT IDLE THEN + CLEAR + SHOWDISPLAY "titles" AT 1 + SHOWKEYS "vmail_OH" + ENDIF + IFEVENT CALLERID THEN + CLEAR ; SHOWDISPLAY "titles" AT 1 NOUPDATE ; SHOWDISPLAY "incoming" AT 2 NOUPDATE -SHOWDISPLAY "callname" AT 3 NOUPDATE -SHOWDISPLAY "callnum" AT 4 -ENDIF -IFEVENT RING THEN -CLEAR -SHOWDISPLAY "titles" AT 1 NOUPDATE -SHOWDISPLAY "incoming" AT 2 -ENDIF -IFEVENT ENDOFRING THEN -SHOWDISPLAY "missedcall" AT 2 -CLEAR -SHOWDISPLAY "titles" AT 1 -SHOWKEYS "vmail_OH" -ENDIF -IFEVENT TIMER THEN -CLEAR -SHOWDISPLAY "empty" AT 4 -ENDIF + SHOWDISPLAY "callname" AT 3 NOUPDATE + SHOWDISPLAY "callnum" AT 4 + ENDIF + IFEVENT RING THEN + CLEAR + SHOWDISPLAY "titles" AT 1 NOUPDATE + SHOWDISPLAY "incoming" AT 2 + ENDIF + IFEVENT ENDOFRING THEN + SHOWDISPLAY "missedcall" AT 2 + CLEAR + SHOWDISPLAY "titles" AT 1 + SHOWKEYS "vmail_OH" + ENDIF + IFEVENT TIMER THEN + CLEAR + SHOWDISPLAY "empty" AT 4 + ENDIF ENDSUB SUB "offHook" IS -IFEVENT FARRING THEN -CLEAR -SHOWDISPLAY "titles" AT 1 NOUPDATE -SHOWDISPLAY "ringing" AT 2 NOUPDATE -SHOWDISPLAY "callname" at 3 NOUPDATE -SHOWDISPLAY "callnum" at 4 -ENDIF -IFEVENT FARANSWER THEN -CLEAR -SHOWDISPLAY "talkingto" AT 2 -GOTO "stableCall" -ENDIF -IFEVENT BUSY THEN -CLEAR -SHOWDISPLAY "titles" AT 1 NOUPDATE -SHOWDISPLAY "busy" AT 2 NOUPDATE -SHOWDISPLAY "callname" at 3 NOUPDATE -SHOWDISPLAY "callnum" at 4 -ENDIF -IFEVENT REORDER THEN -CLEAR -SHOWDISPLAY "titles" AT 1 NOUPDATE -SHOWDISPLAY "reorder" AT 2 NOUPDATE -SHOWDISPLAY "callname" at 3 NOUPDATE -SHOWDISPLAY "callnum" at 4 -ENDIF + IFEVENT FARRING THEN + CLEAR + SHOWDISPLAY "titles" AT 1 NOUPDATE + SHOWDISPLAY "ringing" AT 2 NOUPDATE + SHOWDISPLAY "callname" at 3 NOUPDATE + SHOWDISPLAY "callnum" at 4 + ENDIF + IFEVENT FARANSWER THEN + CLEAR + SHOWDISPLAY "talkingto" AT 2 + GOTO "stableCall" + ENDIF + IFEVENT BUSY THEN + CLEAR + SHOWDISPLAY "titles" AT 1 NOUPDATE + SHOWDISPLAY "busy" AT 2 NOUPDATE + SHOWDISPLAY "callname" at 3 NOUPDATE + SHOWDISPLAY "callnum" at 4 + ENDIF + IFEVENT REORDER THEN + CLEAR + SHOWDISPLAY "titles" AT 1 NOUPDATE + SHOWDISPLAY "reorder" AT 2 NOUPDATE + SHOWDISPLAY "callname" at 3 NOUPDATE + SHOWDISPLAY "callnum" at 4 + ENDIF ENDSUB SUB "stableCall" IS -IFEVENT REORDER THEN -SHOWDISPLAY "callended" AT 2 -ENDIF + IFEVENT REORDER THEN + SHOWDISPLAY "callended" AT 2 + ENDIF ENDSUB diff --git a/configs/cdr.conf.sample b/configs/cdr.conf.sample index 195f88f32..0c0413163 100644 --- a/configs/cdr.conf.sample +++ b/configs/cdr.conf.sample @@ -14,12 +14,12 @@ ;enable=yes ; Define whether or not to log unanswered calls. Setting this to "yes" will -; report every attempt to ring a phone in dialing attempts, when it was not +; report every attempt to ring a phone in dialing attempts, when it was not ; answered. For example, if you try to dial 3 extensions, and this option is "yes", ; you will get 3 CDR's, one for each phone that was rung. Default is "no". Some ; find this information horribly useless. Others find it very valuable. Note, in "yes" ; mode, you will see one CDR, with one of the call targets on one side, and the originating -; channel on the other, and then one CDR for each channel attempted. This may seem +; channel on the other, and then one CDR for each channel attempted. This may seem ; redundant, but cannot be helped. ;unanswered = no @@ -67,7 +67,7 @@ ; Normally, the 'billsec' field logged to the backends (text files or databases) ; is simply the end time (hangup time) minus the answer time in seconds. Internally, -; asterisk stores the time in terms of microseconds and seconds. By setting +; asterisk stores the time in terms of microseconds and seconds. By setting ; initiatedseconds to 'yes', you can force asterisk to report any seconds ; that were initiated (a sort of round up method). Technically, this is ; when the microsecond part of the end time is greater than the microsecond @@ -78,19 +78,19 @@ ; ; CHOOSING A CDR "BACKEND" (what kind of output to generate) ; -; To choose a backend, you have to make sure either the right category is -; defined in this file, or that the appropriate config file exists, and has the +; To choose a backend, you have to make sure either the right category is +; defined in this file, or that the appropriate config file exists, and has the ; proper definitions in it. If there are any problems, usually, the entry will ; silently ignored, and you get no output. -; -; Also, please note that you can generate CDR records in as many formats as you +; +; Also, please note that you can generate CDR records in as many formats as you ; wish. If you configure 5 different CDR formats, then each event will be logged ; in 5 different places! In the example config files, all formats are commented ; out except for the cdr-csv format. ; ; Here are all the possible back ends: ; -; csv, custom, manager, odbc, pgsql, radius, sqlite, tds +; csv, custom, manager, odbc, pgsql, radius, sqlite, tds ; (also, mysql is available via the asterisk-addons, due to licensing ; requirements) ; (please note, also, that other backends can be created, by creating @@ -104,7 +104,7 @@ ; backend is marked with XXX, you know that the "configure" command could not find ; the required libraries for that option. ; -; To get CDRs to be logged to the plain-jane /var/log/asterisk/cdr-csv/Master.csv +; To get CDRs to be logged to the plain-jane /var/log/asterisk/cdr-csv/Master.csv ; file, define the [csv] category in this file. No database necessary. The example ; config files are set up to provide this kind of output by default. ; @@ -126,7 +126,7 @@ ; shows that the modules are available, and the cdr_pgsql.conf file exists, and ; has a [global] section with the proper variables defined. ; -; For logging to radius databases, make sure all the proper libs are installed, that +; For logging to radius databases, make sure all the proper libs are installed, that ; "make menuselect" shows that the modules are available, and the [radius] ; category is defined in this file, and in that section, make sure the 'radiuscfg' ; variable is properly pointing to an existing radiusclient.conf file. @@ -135,7 +135,7 @@ ; which is usually /var/log/asterisk. Of course, the proper libraries should be available ; during the 'configure' operation. ; -; For tds logging, make sure the proper libraries are available during the 'configure' +; For tds logging, make sure the proper libraries are available during the 'configure' ; phase, and that cdr_tds.conf exists and is properly set up with a [global] category. ; ; Also, remember, that if you wish to log CDR info to a database, you will have to define diff --git a/configs/chan_dahdi.conf.sample b/configs/chan_dahdi.conf.sample index 76771fb3e..6d9847d2a 100644 --- a/configs/chan_dahdi.conf.sample +++ b/configs/chan_dahdi.conf.sample @@ -6,7 +6,7 @@ ; will reload the configuration file, but not all configuration options ; are re-configured during a reload (signalling, as well as PRI and ; SS7-related settings cannot be changed on a reload). -; +; ; This file documents many configuration variables. Normally unless you know ; what a variable means or that it should be changed, there's no reason to ; un-comment those lines. @@ -21,11 +21,11 @@ ; ; Trunk groups are used for NFAS or GR-303 connections. ; -; Group: Defines a trunk group. +; Group: Defines a trunk group. ; trunkgroup => ,[,...] ; ; trunkgroup is the numerical trunk group to create -; dchannel is the DAHDI channel which will have the +; dchannel is the DAHDI channel which will have the ; d-channel for the trunk. ; backup1 is an optional list of backup d-channels. ; @@ -85,7 +85,7 @@ ; example, if you set 'national', you will be unable to dial local or ; international numbers. ; -; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's +; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's ; numbering plan). In North America, the typical use is sending the 10 digit ; callerID number and setting the prilocaldialplan to 'national' (the default). ; Only VERY rarely will you need to change this. @@ -98,12 +98,12 @@ ; national: National ISDN ; international: International ISDN ; dynamic: Dynamically selects the appropriate dialplan -; redundant: Same as dynamic, except that the underlying number is not +; redundant: Same as dynamic, except that the underlying number is not ; changed (not common) ; ;pridialplan=unknown ;prilocaldialplan=national -; +; ; pridialplan may be also set at dialtime, by prefixing the dialled number with ; one of the following letters: ; U - Unknown @@ -133,27 +133,27 @@ ; ; PRI caller ID prefixes based on the given TON/NPI (dialplan) ; This is especially needed for EuroISDN E1-PRIs -; +; ; None of the prefix settings can be changed on reload. ; -; sample 1 for Germany +; sample 1 for Germany ;internationalprefix = 00 ;nationalprefix = 0 ;localprefix = 0711 ;privateprefix = 07115678 -;unknownprefix = +;unknownprefix = ; -; sample 2 for Germany +; sample 2 for Germany ;internationalprefix = + ;nationalprefix = +49 ;localprefix = +49711 ;privateprefix = +497115678 -;unknownprefix = +;unknownprefix = ; ; PRI resetinterval: sets the time in seconds between restart of unused ; B channels; defaults to 'never'. ; -;resetinterval = 3600 +;resetinterval = 3600 ; ; Overlap dialing mode (sending overlap digits) ; Cannot be changed on a reload. @@ -168,7 +168,7 @@ ; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work ; with all telcos. -; +; ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT ; inband: Signal Busy/Congestion using in-band tones (default) ; @@ -206,7 +206,7 @@ ; T203: Layer 2 max time without frames being exchanged (default 10000 ms) ; T305: Wait for DISCONNECT acknowledge (default 30000 ms) ; T308: Wait for RELEASE acknowledge (default 4000 ms) -; T309: Maintain active calls on Layer 2 disconnection (default -1, +; T309: Maintain active calls on Layer 2 disconnection (default -1, ; Asterisk clears calls) ; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s ; May vary in other ISDN standards (Q.931 1993 : 90000 ms) @@ -284,11 +284,11 @@ ; (see below). The 'signalling' format specified will be the inbound signalling ; format. If you only specify 'signalling', then it will be the format for ; both inbound and outbound. -; -; outsignalling can only be one of: +; +; outsignalling can only be one of: ; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd, ; featdmf, featdmf_ta, e911, fgccama, fgccamamf -; +; ; outsignalling cannot be changed on a reload. ; ;signalling=featdmf @@ -318,9 +318,9 @@ ; None of them will update on a reload. ; ; How long generated tones (DTMF and MF) will be played on the channel -; (in milliseconds). +; (in milliseconds). ; -; This is a global, rather than a per-channel setting. It will not be +; This is a global, rather than a per-channel setting. It will not be ; updated on a reload. ; ;toneduration=100 @@ -354,7 +354,7 @@ usecallerid=yes ; What signals the start of caller ID ; ring = a ring signals the start (default) ; polarity = polarity reversal signals the start -; polarity_IN = polarity reversal signals the start, for India, +; polarity_IN = polarity reversal signals the start, for India, ; for dtmf dialtone detection; using DTMF. ; (see doc/India-CID.txt) ; @@ -381,7 +381,7 @@ usecallerid=yes ; fsk,rpas - the FXO line is monitored for MWI FSK spills preceded ; by a ring pulse alert signal. ; neon - The fxo line is monitored for the presence of NEON pulses -; indicating MWI. +; indicating MWI. ; When detected, an internal Asterisk MWI event is generated so that any other ; part of Asterisk that cares about MWI state changes is notified, just as if ; the state change came from app_voicemail. @@ -432,7 +432,7 @@ usecallingpres=yes ; ; Some countries (UK) have ring tones with different ring tones (ring-ring), ; which means the caller ID needs to be set later on, and not just after -; the first ring, as per the default (1). +; the first ring, as per the default (1). ; ;sendcalleridafter = 2 ; @@ -472,10 +472,10 @@ cancallforward=yes ; callreturn=yes ; -; Stutter dialtone support: If a mailbox is specified without a voicemail -; context, then when voicemail is received in a mailbox in the default +; Stutter dialtone support: If a mailbox is specified without a voicemail +; context, then when voicemail is received in a mailbox in the default ; voicemail context in voicemail.conf, taking the phone off hook will cause a -; stutter dialtone instead of a normal one. +; stutter dialtone instead of a normal one. ; ; If a mailbox is specified *with* a voicemail context, the same will result ; if voicemail received in mailbox in the specified voicemail context. @@ -486,9 +486,9 @@ callreturn=yes ; ; for any other voicemail context, the following will produce the stutter tone: ; -;mailbox=1234@context +;mailbox=1234@context ; -; Enable echo cancellation +; Enable echo cancellation ; Use either "yes", "no", or a power of two from 32 to 256 if you wish to ; actually set the number of taps of cancellation. ; @@ -552,7 +552,7 @@ echocancelwhenbridged=yes ; ; There are several independent gain settings: ; rxgain: gain for the rx (receive - into Asterisk) channel. Default: 0.0 -; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel. +; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel. ; Default: 0.0 ; cid_rxgain: set the gain just for the caller ID sounds Asterisk ; emits. Default: 5.0 . @@ -581,9 +581,9 @@ pickupgroup=1 ; Channel variable to be set for all calls from this channel ;setvar=CHANNEL=42 ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will -; cause the given audio file to -; be played upon completion of -; an attended transfer. + ; cause the given audio file to + ; be played upon completion of + ; an attended transfer. ; ; Specify whether the channel should be answered immediately or if the simple @@ -600,10 +600,10 @@ pickupgroup=1 ; ; caller ID can be set to "asreceived" or a specific number if you want to ; override it. Note that "asreceived" only applies to trunk interfaces. -; fullname sets just the +; fullname sets just the ; ; fullname: sets just the name part. -; cid_number: sets just the number part: +; cid_number: sets just the number part: ; ;callerid = 123456 ; @@ -642,7 +642,7 @@ pickupgroup=1 ;smdiport=/dev/ttyS0 ; ; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D -; etc, it can be useful to perform busy detection either in an effort to +; etc, it can be useful to perform busy detection either in an effort to ; detect hangup or for detecting busies. This enables listening for ; the beep-beep busy pattern. ; @@ -685,8 +685,8 @@ pickupgroup=1 ; ;hanguponpolarityswitch=yes ; -; polarityonanswerdelay: minimal time period (ms) between the answer -; polarity switch and hangup polarity switch. +; polarityonanswerdelay: minimal time period (ms) between the answer +; polarity switch and hangup polarity switch. ; (default: 600ms) ; ; On trunk interfaces (FXS) it can be useful to attempt to follow the progress @@ -699,7 +699,7 @@ pickupgroup=1 ; with "progzone". ; ; progzone also affects the pattern used for buzydetect (unless -; busypattern is set explicitly). The possible values are: +; busypattern is set explicitly). The possible values are: ; us (default) ; ca (alias for 'us') ; cr (Costa Rica) @@ -741,7 +741,7 @@ pickupgroup=1 ;faxdetect=no ; ; When 'faxdetect' is used, one could use 'faxbuffers' to configure the DAHDI -; transmit buffer policy. The default is *OFF*. When this configuration +; transmit buffer policy. The default is *OFF*. When this configuration ; option is used, the faxbuffer policy will be used for the life of the call ; after a fax tone is detected. The faxbuffer policy is reverted after the ; call is torn down. The sample below will result in 6 buffers and a full @@ -792,23 +792,23 @@ pickupgroup=1 ; ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a -; DAHDI channel. Defaults to "no". An enabled jitterbuffer will -; be used only if the sending side can create and the receiving -; side can not accept jitter. The DAHDI channel can't accept jitter, -; thus an enabled jitterbuffer on the receive DAHDI side will always -; be used if the sending side can create jitter. + ; DAHDI channel. Defaults to "no". An enabled jitterbuffer will + ; be used only if the sending side can create and the receiving + ; side can not accept jitter. The DAHDI channel can't accept jitter, + ; thus an enabled jitterbuffer on the receive DAHDI side will always + ; be used if the sending side can create jitter. ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is -; resynchronized. Useful to improve the quality of the voice, with -; big jumps in/broken timestamps, usually sent from exotic devices -; and programs. Defaults to 1000. + ; resynchronized. Useful to improve the quality of the voice, with + ; big jumps in/broken timestamps, usually sent from exotic devices + ; and programs. Defaults to 1000. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a DAHDI -; channel. Two implementations are currently available - "fixed" -; (with size always equals to jbmax-size) and "adaptive" (with -; variable size, actually the new jb of IAX2). Defaults to fixed. + ; channel. Two implementations are currently available - "fixed" + ; (with size always equals to jbmax-size) and "adaptive" (with + ; variable size, actually the new jb of IAX2). Defaults to fixed. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". ;----------------------------------------------------------------------------------- @@ -834,7 +834,7 @@ pickupgroup=1 ; parameters that were specified above its declaration. ; ; For GR-303, CRV's are created like channels except they must start with the -; trunk group followed by a colon, e.g.: +; trunk group followed by a colon, e.g.: ; ; crv => 1:1 ; crv => 2:1-2,5-8 @@ -908,15 +908,15 @@ pickupgroup=1 ; A range of -1 will force it to always match. ; Anything lower than -1 would presumably cause it to never match. ; -;dring1=95,0,0 -;dring1context=internal1 +;dring1=95,0,0 +;dring1context=internal1 ;dring1range=10 -;dring2=325,95,0 -;dring2context=internal2 +;dring2=325,95,0 +;dring2context=internal2 ;dring2range=10 ; If no pattern is matched here is where we go. ;context=default -;channel => 1 +;channel => 1 ; ---------------- Options for use with signalling=ss7 ----------------- ; None of them can be changed by a reload. @@ -945,12 +945,12 @@ pickupgroup=1 ; ;ss7_calling_nai=dynamic ; -; -; sample 1 for Germany +; +; sample 1 for Germany ;ss7_internationalprefix = 00 ;ss7_nationalprefix = 0 -;ss7_subscriberprefix = -;ss7_unknownprefix = +;ss7_subscriberprefix = +;ss7_unknownprefix = ; ; This option is used to disable automatic sending of ACM when the call is started @@ -1056,7 +1056,7 @@ pickupgroup=1 ; 'stack' is for very verbose output of the channel and context call stack, only useful ; if you are debugging a crash or want to learn how the library works. The stack logging ; will be only enabled if the openr2 library was compiled with -DOR2_TRACE_STACKS -; You can mix up values, like: loglevel=error,debug,mf to log just error, debug and +; You can mix up values, like: loglevel=error,debug,mf to log just error, debug and ; multi frequency messages ; 'all' is a special value to log all the activity ; 'nothing' is a clean-up value, in case you want to not log any activity for @@ -1110,20 +1110,20 @@ pickupgroup=1 ; You most likely dont need this feature. Default is yes. ; When this is set to yes, all calls that are offered (incoming calls) which -; DNIS is valid (exists in extensions.conf) and pass collect call validation +; DNIS is valid (exists in extensions.conf) and pass collect call validation ; will be accepted with a Group B tone (either call with charge or not, depending on mfcr2_charge_calls) ; with this set to 'no' then the call will NOT be accepted on offered, and the call will start its ; execution in extensions.conf without being accepted until the channel is answered (either with Answer() or -; any other application resulting in the channel being answered). +; any other application resulting in the channel being answered). ; This can be set to 'no' if your telco or PBX needs the hangup cause to be set accurately ; when this option is set to no you must explicitly accept the call with DAHDIAcceptR2Call -; or implicitly through the Answer() application. +; or implicitly through the Answer() application. ; mfcr2_accept_on_offer=yes ; WARNING: advanced users only! I really mean it ; this parameter is commented by default because ; YOU DON'T NEED IT UNLESS YOU REALLY GROK MFC/R2 -; READ COMMENTS on doc/r2proto.conf in openr2 package +; READ COMMENTS on doc/r2proto.conf in openr2 package ; for more info ; mfcr2_advanced_protocol_file=/path/to/r2proto.conf @@ -1171,7 +1171,7 @@ pickupgroup=1 ; chan_dahdi.conf and [general] in users.conf - one section's configuration ; does not affect another one's. ; -; Instead of letting common configuration values "slide through" you can +; Instead of letting common configuration values "slide through" you can ; use configuration templates to easily keep the common part in one ; place and override where needed. ; diff --git a/configs/cli_aliases.conf.sample b/configs/cli_aliases.conf.sample index cc1e2e6d3..1d9cd9107 100644 --- a/configs/cli_aliases.conf.sample +++ b/configs/cli_aliases.conf.sample @@ -13,8 +13,8 @@ template = friendly ; By default, include friendly aliases ;template = asterisk12 ; Asterisk 1.2 style syntax ;template = asterisk14 ; Asterisk 1.4 style syntax ;template = individual_custom ; see [individual_custom] example below which -; includes a list of aliases from an external -; file + ; includes a list of aliases from an external + ; file ; Because the Asterisk CLI syntax follows a "module verb argument" syntax, @@ -70,7 +70,7 @@ pri intense debug span=pri set debug 2 span ; by Asterisk. If you wish to use the provided templates, simply define the ; context name which does not utilize the '_tpl' at the end. For example, ; if you would like to use the Asterisk 1.2 style syntax, define in the -; [general] section +; [general] section [asterisk12_tpl](!) show channeltypes=core show channeltypes @@ -92,7 +92,7 @@ show file formats=core show file formats show applications=core show applications show functions=core show functions show switches=core show switches -show hints=core show hints +show hints=core show hints show globals=core show globals show function=core show function show application=core show application @@ -102,7 +102,7 @@ show codecs=core show codecs show audio codecs=core show audio codecs show video codecs=core show video codecs show image codecs=core show image codecs -show codec=core show codec +show codec=core show codec moh classes show=moh show classes moh files show=moh show files agi no debug=agi debug off diff --git a/configs/cli_permissions.conf.sample b/configs/cli_permissions.conf.sample index 7cbad88f3..4a6973f50 100644 --- a/configs/cli_permissions.conf.sample +++ b/configs/cli_permissions.conf.sample @@ -23,7 +23,7 @@ [general] default_perm=permit ; To leave asterisk working as normal -; we should set this parameter to 'permit' + ; we should set this parameter to 'permit' ; ; Follows the per-users permissions configs. ; diff --git a/configs/console.conf.sample b/configs/console.conf.sample index d7e586a6b..9bd502696 100644 --- a/configs/console.conf.sample +++ b/configs/console.conf.sample @@ -5,7 +5,7 @@ [general] ; Set this option to "yes" to enable automatically answering calls on the -; console. This is very useful if the console is used as an intercom. +; console. This is very useful if the console is used as an intercom. ; The default value is "no". ; ;autoanswer = no @@ -21,7 +21,7 @@ ;extension = s ; Set the default CallerID for created channels. -; +; ;callerid = MyName Here <(256) 428-6000> ; Set the default language for created channels. @@ -34,7 +34,7 @@ ; The default is "no". ; ;overridecontext = no ; if 'no', the last @ will start the context -; if 'yes' the whole string is an extension. + ; if 'yes' the whole string is an extension. ; Default Music on Hold class to use when this channel is placed on hold in @@ -46,23 +46,23 @@ ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an -; Console channel. Defaults to "no". An enabled jitterbuffer will -; be used only if the sending side can create and the receiving -; side can not accept jitter. The Console channel can't accept jitter, -; thus an enabled jitterbuffer on the receive Console side will always -; be used if the sending side can create jitter. + ; Console channel. Defaults to "no". An enabled jitterbuffer will + ; be used only if the sending side can create and the receiving + ; side can not accept jitter. The Console channel can't accept jitter, + ; thus an enabled jitterbuffer on the receive Console side will always + ; be used if the sending side can create jitter. ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is -; resynchronized. Useful to improve the quality of the voice, with -; big jumps in/broken timestamps, usually sent from exotic devices -; and programs. Defaults to 1000. + ; resynchronized. Useful to improve the quality of the voice, with + ; big jumps in/broken timestamps, usually sent from exotic devices + ; and programs. Defaults to 1000. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a Console -; channel. Two implementations are currently available - "fixed" -; (with size always equals to jbmax-size) and "adaptive" (with -; variable size, actually the new jb of IAX2). Defaults to fixed. + ; channel. Two implementations are currently available - "fixed" + ; (with size always equals to jbmax-size) and "adaptive" (with + ; variable size, actually the new jb of IAX2). Defaults to fixed. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". ;----------------------------------------------------------------------------------- @@ -76,8 +76,8 @@ [default] input_device = default ; When configuring an input device and output device, output_device = default ; use the name that you see when you run the "console -; list available" CLI command. If you say "default", the -; system default input and output devices will be used. + ; list available" CLI command. If you say "default", the + ; system default input and output devices will be used. autoanswer = no context = default extension = s @@ -86,5 +86,5 @@ language = en overridecontext = no mohinterpret = default active = yes ; This option should only be set for one console. -; It means that it is the active console to be -; used from the Asterisk CLI. + ; It means that it is the active console to be + ; used from the Asterisk CLI. diff --git a/configs/dnsmgr.conf.sample b/configs/dnsmgr.conf.sample index a2939dc10..e34dbcf0a 100644 --- a/configs/dnsmgr.conf.sample +++ b/configs/dnsmgr.conf.sample @@ -1,5 +1,5 @@ [general] ;enable=yes ; enable creation of managed DNS lookups -; default is 'no' + ; default is 'no' ;refreshinterval=1200 ; refresh managed DNS lookups every seconds -; default is 300 (5 minutes) \ No newline at end of file + ; default is 300 (5 minutes) \ No newline at end of file diff --git a/configs/dundi.conf.sample b/configs/dundi.conf.sample index 3eb1bd320..1b6a174c0 100644 --- a/configs/dundi.conf.sample +++ b/configs/dundi.conf.sample @@ -1,6 +1,6 @@ ; ; DUNDi configuration file -; +; ; For more information about DUNDi, see http://www.dundi.com ; ; @@ -50,9 +50,9 @@ ttl=32 ; ; If we don't get ACK to our DPDISCOVER within 2000ms, and autokill is set -; to yes, then we cancel the whole thing (that's enough time for one +; to yes, then we cancel the whole thing (that's enough time for one ; retransmission only). This is used to keep things from stalling for a long -; time for a host that is not available, but would be ill advised for bad +; time for a host that is not available, but would be ill advised for bad ; connections. In addition to 'yes' or 'no' you can also specify a number ; of milliseconds. See 'qualify' for individual peers to turn on for just ; a specific peer. @@ -60,7 +60,7 @@ ttl=32 autokill=yes ; ; pbx_dundi creates a rotating key called "secret", under the family -; 'secretpath'. The default family is dundi (resulting in +; 'secretpath'. The default family is dundi (resulting in ; the key being held at dundi/secret). ; ;secretpath=dundi @@ -78,8 +78,8 @@ autokill=yes ; ; The "mappings" section maps DUNDi contexts ; to contexts on the local asterisk system. Remember -; that numbers that are made available under the e164 -; DUNDi context are regulated by the DUNDi General Peering +; that numbers that are made available under the e164 +; DUNDi context are regulated by the DUNDi General Peering ; Agreement (GPA) if you are a member of the DUNDi E.164 ; Peering System. ; @@ -108,14 +108,14 @@ autokill=yes ; ; Further options may include: ; -; nounsolicited: No unsolicited calls of any type permitted via this +; nounsolicited: No unsolicited calls of any type permitted via this ; route -; nocomunsolicit: No commercial unsolicited calls permitted via +; nocomunsolicit: No commercial unsolicited calls permitted via ; this route ; residential: This number is known to be a residence ; commercial: This number is known to be a business ; mobile: This number is known to be a mobile phone -; nocomunsolicit: No commercial unsolicited calls permitted via +; nocomunsolicit: No commercial unsolicited calls permitted via ; this route ; nopartial: Do not search for partial matches ; @@ -163,7 +163,7 @@ autokill=yes ; ; host - What their host is ; -; order - What search order to use. May be 'primary', 'secondary', +; order - What search order to use. May be 'primary', 'secondary', ; 'tertiary' or 'quartiary'. In large systems, it is beneficial ; to only query one up-stream host in order to maximize caching ; value. Adding one with primary and one with secondary gives you @@ -187,7 +187,7 @@ autokill=yes ; the local system. Set "all" to deny this host to ; lookup all contexts. ; -; model - inbound, outbound, or symmetric for whether we receive +; model - inbound, outbound, or symmetric for whether we receive ; requests only, transmit requests only, or do both. ; ; precache - Utilize/Permit precaching with this peer (to pre @@ -241,7 +241,7 @@ autokill=yes ;inkey = littleguy ;outkey = ourkey ;include = e164 ; In this case used only for precaching -;permit = e164 +;permit = e164 ;qualify = yes ; @@ -254,7 +254,7 @@ autokill=yes ;register = yes ;inkey = dhcp34 ;permit = all ; In this case used only for precaching -;include = all +;include = all ;qualify = yes ;outkey=foo diff --git a/configs/extconfig.conf.sample b/configs/extconfig.conf.sample index 2f1554f63..542bedb52 100644 --- a/configs/extconfig.conf.sample +++ b/configs/extconfig.conf.sample @@ -7,7 +7,7 @@ ; [settings] ; -; Static configuration files: +; Static configuration files: ; ; file.conf => driver,database[,table] ; diff --git a/configs/extensions.ael.sample b/configs/extensions.ael.sample index c7720290a..69f441d1e 100644 --- a/configs/extensions.ael.sample +++ b/configs/extensions.ael.sample @@ -3,49 +3,49 @@ // // // Static extension configuration file, used by -// the pbx_ael module. This is where you configure all your -// inbound and outbound calls in Asterisk. -// -// This configuration file is reloaded +// the pbx_ael module. This is where you configure all your +// inbound and outbound calls in Asterisk. +// +// This configuration file is reloaded // - With the "ael reload" command in the CLI // - With the "reload" command (that reloads everything) in the CLI // The "Globals" category contains global variables that can be referenced // in the dialplan by using the GLOBAL dialplan function: -// ${GLOBAL(VARIABLE)} +// ${GLOBAL(VARIABLE)} // ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid // Unix/Linux environmental variables are reached with the ENV dialplan // function: ${ENV(VARIABLE)} // globals { -CONSOLE="Console/dsp"; // Console interface for demo -//CONSOLE=DAHDI/1 -//CONSOLE=Phone/phone0 -IAXINFO=guest; // IAXtel username/password -//IAXINFO="myuser:mypass"; -TRUNK="DAHDI/G2"; // Trunk interface -// -// Note the 'G2' in the TRUNK variable above. It specifies which group (defined -// in dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in -// the specified group. The four possible options are: -// -// g: select the lowest-numbered non-busy DAHDI channel -// (aka. ascending sequential hunt group). -// G: select the highest-numbered non-busy DAHDI channel -// (aka. descending sequential hunt group). -// r: use a round-robin search, starting at the next highest channel than last -// time (aka. ascending rotary hunt group). -// R: use a round-robin search, starting at the next lowest channel than last -// time (aka. descending rotary hunt group). -// -TRUNKMSD=1; // MSD digits to strip (usually 1 or 0) -//TRUNK=IAX2/user:pass@provider + CONSOLE="Console/dsp"; // Console interface for demo + //CONSOLE=DAHDI/1 + //CONSOLE=Phone/phone0 + IAXINFO=guest; // IAXtel username/password + //IAXINFO="myuser:mypass"; + TRUNK="DAHDI/G2"; // Trunk interface + // + // Note the 'G2' in the TRUNK variable above. It specifies which group (defined + // in dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in + // the specified group. The four possible options are: + // + // g: select the lowest-numbered non-busy DAHDI channel + // (aka. ascending sequential hunt group). + // G: select the highest-numbered non-busy DAHDI channel + // (aka. descending sequential hunt group). + // r: use a round-robin search, starting at the next highest channel than last + // time (aka. ascending rotary hunt group). + // R: use a round-robin search, starting at the next lowest channel than last + // time (aka. descending rotary hunt group). + // + TRUNKMSD=1; // MSD digits to strip (usually 1 or 0) + //TRUNK=IAX2/user:pass@provider }; // -// Any category other than "General" and "Globals" represent -// extension contexts, which are collections of extensions. +// Any category other than "General" and "Globals" represent +// extension contexts, which are collections of extensions. // // Extension names may be numbers, letters, or combinations // thereof. If an extension name is prefixed by a '_' @@ -56,12 +56,12 @@ TRUNKMSD=1; // MSD digits to strip (usually 1 or 0) // Z - any digit from 1-9 // N - any digit from 2-9 // [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9) -// . - wildcard, matches anything remaining (e.g. _9011. matches +// . - wildcard, matches anything remaining (e.g. _9011. matches // anything starting with 9011 excluding 9011 itself) // ! - wildcard, causes the matching process to complete as soon as // it can unambiguously determine that no other matches are possible // -// For example the extension _NXXXXXX would match normal 7 digit dialings, +// For example the extension _NXXXXXX would match normal 7 digit dialings, // while _1NXXNXXXXXX would represent an area code plus phone number // preceded by a one. // @@ -72,8 +72,8 @@ TRUNKMSD=1; // MSD digits to strip (usually 1 or 0) // The priority "same" or "s" means the same as the previously specified // priority, again regardless of whether the previous entry was for the // same extension. Priorities may be immediately followed by a plus sign -// and another integer to add that amount (most useful with 's' or 'n'). -// Priorities may then also have an alias, or label, in +// and another integer to add that amount (most useful with 's' or 'n'). +// Priorities may then also have an alias, or label, in // parenthesis after their name which can be used in goto situations // // Contexts contain several lines, one for each step of each @@ -87,11 +87,11 @@ TRUNKMSD=1; // MSD digits to strip (usually 1 or 0) // exten-name => { // application(arg1,arg2,...); // -// Timing list for includes is +// Timing list for includes is // //