[asterisk-commits] tilghman: branch 1.2 r42716 -
/branches/1.2/configs/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Mon Sep 11 09:39:06 MST 2006
Author: tilghman
Date: Mon Sep 11 11:39:06 2006
New Revision: 42716
URL: http://svn.digium.com/view/asterisk?rev=42716&view=rev
Log:
Spelling/grammar fixes (Issue 7929)
Modified:
branches/1.2/configs/alsa.conf.sample
branches/1.2/configs/cdr.conf.sample
branches/1.2/configs/dundi.conf.sample
branches/1.2/configs/enum.conf.sample
branches/1.2/configs/extensions.conf.sample
branches/1.2/configs/features.conf.sample
branches/1.2/configs/iax.conf.sample
branches/1.2/configs/misdn.conf.sample
branches/1.2/configs/osp.conf.sample
branches/1.2/configs/phone.conf.sample
branches/1.2/configs/sip.conf.sample
branches/1.2/configs/skinny.conf.sample
branches/1.2/configs/voicemail.conf.sample
branches/1.2/configs/vpb.conf.sample
branches/1.2/configs/zapata.conf.sample
Modified: branches/1.2/configs/alsa.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/alsa.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/alsa.conf.sample (original)
+++ branches/1.2/configs/alsa.conf.sample Mon Sep 11 11:39:06 2006
@@ -20,7 +20,7 @@
;
;language=en
;
-; Silence supression can be enabled when sound is over a certain threshold.
+; Silence suppression can be enabled when sound is over a certain threshold.
; The value for the threshold should probably be between 500 and 2000 or so,
; but your mileage may vary. Use the echo test to evaluate the best setting.
;silencesuppression = yes
Modified: branches/1.2/configs/cdr.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/cdr.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/cdr.conf.sample (original)
+++ branches/1.2/configs/cdr.conf.sample Mon Sep 11 11:39:06 2006
@@ -35,7 +35,7 @@
;time=300
; The CDR engine uses the internal asterisk scheduler to determine when to post
-; records. Posting can either occure inside the scheduler thread, or a new
+; records. Posting can either occur inside the scheduler thread, or a new
; thread can be spawned for the submission of every batch. For small batches,
; it might be acceptable to just use the scheduler thread, so set this to "yes".
; For large batches, say anything over size=10, a new thread is recommended, so
Modified: branches/1.2/configs/dundi.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/dundi.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/dundi.conf.sample (original)
+++ branches/1.2/configs/dundi.conf.sample Mon Sep 11 11:39:06 2006
@@ -89,7 +89,7 @@
;
; 'weight' is the weight to use for the responses provided from this
; mapping. The number must be >= 0 and < 60000. Since it is totally
-; valid to receive multiple reponses to a query, responses received
+; valid to receive multiple responses to a query, responses received
; with a lower weight are tried first. Note that the weight has a
; special meaning in the e164 context - see the GPA for more details.
;
@@ -144,7 +144,7 @@
; '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
-; redundancy without sacraficing performance.
+; redundancy without sacrificing performance.
;
; include - Includes this peer when searching a particular context
; for lookup (set "all" to perform all lookups with that
Modified: branches/1.2/configs/enum.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/enum.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/enum.conf.sample (original)
+++ branches/1.2/configs/enum.conf.sample Mon Sep 11 11:39:06 2006
@@ -11,7 +11,7 @@
;
search => e164.arpa
;
-; If you'd like to use the E.164.org public ENUM registery in addition
+; If you'd like to use the E.164.org public ENUM registry in addition
; to the official e164.arpa one, uncomment the following line
;
;search => e164.org
Modified: branches/1.2/configs/extensions.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/extensions.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/extensions.conf.sample (original)
+++ branches/1.2/configs/extensions.conf.sample Mon Sep 11 11:39:06 2006
@@ -107,7 +107,7 @@
;
; For example the extension _NXXXXXX would match normal 7 digit dialings,
; while _1NXXNXXXXXX would represent an area code plus phone number
-; preceeded by a one.
+; preceded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension. The priority
@@ -214,7 +214,7 @@
exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
;
-; The SWITCH statement permits a server to share the dialplain with
+; The SWITCH statement permits a server to share the dialplan with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
@@ -397,7 +397,7 @@
exten => 500,n,Goto(s,6) ; Return to the start over message.
;
-; Create an extension, 600, for evaulating echo latency.
+; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
Modified: branches/1.2/configs/features.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/features.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/features.conf.sample (original)
+++ branches/1.2/configs/features.conf.sample Mon Sep 11 11:39:06 2006
@@ -10,7 +10,7 @@
context => parkedcalls ; Which context parked calls are in
;parkingtime => 45 ; Number of seconds a call can be parked for
; (default is 45 seconds)
-;transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call
+;transferdigittimeout => 3 ; Number of seconds to wait between digits when transferring a call
;courtesytone = beep ; Sound file to play to the parked caller
; when someone dials a parked call
;xfersound = beep ; to indicate an attended transfer is complete
Modified: branches/1.2/configs/iax.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/iax.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/iax.conf.sample (original)
+++ branches/1.2/configs/iax.conf.sample Mon Sep 11 11:39:06 2006
@@ -246,7 +246,7 @@
;
; caller - Consider the callers preferred order ahead of the host's.
; host - Consider the host's preferred order ahead of the caller's.
-; disabled - Disable the consideration of codec preference alltogether.
+; disabled - Disable the consideration of codec preference altogether.
; (this is the original behaviour before preferences were added)
; reqonly - Same as disabled, only do not consider capabilities if
; the requested format is not available the call will only
@@ -371,7 +371,7 @@
;mask=255.255.255.255
;qualify=yes ; Make sure this peer is alive
;qualifysmoothing = yes ; use an average of the last two PONG
- ; results to reduce falsly detected LAGGED hosts
+ ; results to reduce falsely detected LAGGED hosts
; Default: Off
;qualifyfreqok = 60000 ; how frequently to ping the peer when
; everything seems to be ok, in milliseconds
Modified: branches/1.2/configs/misdn.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/misdn.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/misdn.conf.sample (original)
+++ branches/1.2/configs/misdn.conf.sample Mon Sep 11 11:39:06 2006
@@ -74,7 +74,7 @@
;
stop_tone_after_first_digit=yes
-; wether to append overlapdialed Digits to Extension or not
+; whether to append overlapdialed Digits to Extension or not
;
; default value: yes
;
@@ -82,7 +82,7 @@
;;; CRYPTION STUFF
-; Wether to look for dynamic crypting attempt
+; Whether to look for dynamic crypting attempt
;
; default value: no
;
@@ -104,7 +104,7 @@
; users sections:
;
; name your sections as you which but not "general" !
-; the secions are Groups, you can dial out in extensions.conf
+; the sections are Groups, you can dial out in extensions.conf
; with Dial(mISDN/g:extern/101) where extern is a section name,
; chan_misdn tries every port in this section to find a
; new free channel
@@ -134,7 +134,7 @@
musicclass=default
;
-; Either if we should produce DTMF Tones ourselve
+; Either if we should produce DTMF Tones ourselves
;
senddtmf=yes
@@ -166,7 +166,7 @@
rxgain=0
txgain=0
-; some telcos espacially in NL seem to need this set to yes, also in
+; some telcos especially in NL seem to need this set to yes, also in
; switzerland this seems to be important
;
; default value: no
@@ -177,7 +177,7 @@
;
; This option defines, if chan_misdn should check the L1 on a PMP
-; before makeing a group call on it. The L1 may go down for PMP Ports
+; before making a group call on it. The L1 may go down for PMP Ports
; so we might need this.
; But be aware! a broken or plugged off cable might be used for a group call
; as well, since chan_misdn has no chance to distinguish if the L1 is down
@@ -282,7 +282,7 @@
; Pickup and Callgroup
;
-; deafult values: not set = 0
+; default values: not set = 0
;
;callgroup=1
;pickupgroup=1
@@ -295,7 +295,7 @@
; s=0, p=0 -> callerid presented not screened
; s=1, p=1 -> callerid presented but screened (the remote end does not see it!)
;
-; defaule values s=-1, p=-1
+; default values s=-1, p=-1
presentation=-1
screen=-1
@@ -343,11 +343,11 @@
[first_extern]
; again port defs
ports=4
-; again a context for incomming calls
+; again a context for incoming calls
context=Extern1
; msns for te ports, listen on those numbers on the above ports, and
; indicate the incoming calls to asterisk
-; here you can give a comma seperated list or simply an '*' for
+; here you can give a comma separated list or simply an '*' for
; any msn.
msns=*
Modified: branches/1.2/configs/osp.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/osp.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/osp.conf.sample (original)
+++ branches/1.2/configs/osp.conf.sample Mon Sep 11 11:39:06 2006
@@ -10,7 +10,7 @@
;
[general]
;
-; Should hardware accelleration be enabled? May not be changed
+; Should hardware acceleration be enabled? May not be changed
; on a reload.
;
;accelerate=yes
Modified: branches/1.2/configs/phone.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/phone.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/phone.conf.sample (original)
+++ branches/1.2/configs/phone.conf.sample Mon Sep 11 11:39:06 2006
@@ -28,7 +28,7 @@
;
echocancel=medium
;
-; You can optionally use VAD/CNG silence supression
+; You can optionally use VAD/CNG silence suppression
;
;silencesupression=yes
;
@@ -38,7 +38,7 @@
;
; You can set txgain and rxgain for each device in the same way as context.
; If you want to change default gain value (1.0 =~ 100%) for device, simple
-; add txgain or rxgain line before device line. But rememeber, if you change
+; add txgain or rxgain line before device line. But remember, if you change
; volume all cards listed below will be affected by these values. You can
; use float values (1.0, 0.5, 2.0) or percentage values (100%, 150%, 50%).
;
Modified: branches/1.2/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/sip.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/sip.conf.sample (original)
+++ branches/1.2/configs/sip.conf.sample Mon Sep 11 11:39:06 2006
@@ -59,7 +59,7 @@
;tos=184 ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
;maxexpiry=3600 ; Max length of incoming registration we allow
-;defaultexpiry=120 ; Default length of incoming/outoing registration
+;defaultexpiry=120 ; Default length of incoming/outgoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
@@ -338,7 +338,7 @@
;
; For local phones, type=friend works most of the time
;
-; If you have one-way audio, you propably have NAT problems.
+; If you have one-way audio, you probably have NAT problems.
; If Asterisk is on a public IP, and the phone is inside of a NAT device
; you will need to configure nat option for those phones.
; Also, turn on qualify=yes to keep the nat session open
@@ -428,7 +428,7 @@
;
;callgroup=1,3-4 ; We are in caller groups 1,3,4
;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
-;defaultip=192.168.0.60 ; IP address to use if peer has not registred
+;defaultip=192.168.0.60 ; IP address to use if peer has not registered
;[cisco1]
;type=friend
Modified: branches/1.2/configs/skinny.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/skinny.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/skinny.conf.sample (original)
+++ branches/1.2/configs/skinny.conf.sample Mon Sep 11 11:39:06 2006
@@ -24,7 +24,7 @@
; Typical config for a 7910
;[duba] ; Device name
;model=7910 ; Device model
-;device=SEP0007EB463101 ; Offical identifier
+;device=SEP0007EB463101 ; Official identifier
;version=P002F202 ; Firmware version identifier
;host=192.168.1.144 ;
;permit=192.168.0/24 ; Optional, used for authentication
Modified: branches/1.2/configs/voicemail.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/voicemail.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/voicemail.conf.sample (original)
+++ branches/1.2/configs/voicemail.conf.sample Mon Sep 11 11:39:06 2006
@@ -41,7 +41,7 @@
;minmessage=3
; Maximum length of greetings in seconds
;maxgreet=60
-; How many miliseconds to skip forward/back when rew/ff in message playback
+; How many milliseconds to skip forward/back when rew/ff in message playback
skipms=3000
; How many seconds of silence before we end the recording
maxsilence=10
@@ -81,7 +81,7 @@
; limitation in the Asterisk configuration subsystem.
;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
; The following definition is very close to the default, but the default shows
-; just the CIDNAME, if it is not null, otherise just the CIDNUM, or "an unknown
+; just the CIDNAME, if it is not null, otherwise just the CIDNUM, or "an unknown
; caller", if they are both null.
;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n
;
@@ -142,11 +142,11 @@
; if the e-mail is specified, a message will be sent when a message is
; received, to the given mailbox. If pager is specified, a message will be
; sent there as well. If the password is prefixed by '-', then it is
-; considered to be unchangable.
+; considered to be unchangeable.
;
; Advanced options example is extension 4069
; NOTE: All options can be expressed globally in the general section, and
-; overriden in the per-mailbox settings, unless listed otherwise.
+; overridden in the per-mailbox settings, unless listed otherwise.
;
; tz=central ; Timezone from zonemessages above. Irrelevant if envelope=no.
; attach=yes ; Attach the voicemail to the notification email *NOT* the pager email
Modified: branches/1.2/configs/vpb.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/vpb.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/vpb.conf.sample (original)
+++ branches/1.2/configs/vpb.conf.sample Mon Sep 11 11:39:06 2006
@@ -42,7 +42,7 @@
; txhwgain => Transmit hardware gain (-12 => 12)
; rxhwgain => Receive Hardware gain (-12 => 12)
;
-; These are advanced settings and only mentioned for fullnes.
+; These are advanced settings and only mentioned for completeness.
; bal1 => Hybrid balance codec register 1
; bal2 => Hybrid balance codec register 2
; bal3 => Hybrid balance codec register 3
Modified: branches/1.2/configs/zapata.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.2/configs/zapata.conf.sample?rev=42716&r1=42715&r2=42716&view=diff
==============================================================================
--- branches/1.2/configs/zapata.conf.sample (original)
+++ branches/1.2/configs/zapata.conf.sample Mon Sep 11 11:39:06 2006
@@ -208,7 +208,7 @@
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; How long generated tones (DTMF and MF) will be played on the channel
-; (in miliseconds)
+; (in milliseconds)
;toneduration=100
;
; Whether or not to do distinctive ring detection on FXO lines
@@ -291,7 +291,7 @@
; stutter dialtone instead of a normal one.
;
; If a mailbox is specified *with* a voicemail context, the same will result
-; if voicemail recieved in mailbox in the specified voicemail context.
+; if voicemail received in mailbox in the specified voicemail context.
;
; for default voicemail context, the example below is fine:
;
@@ -350,7 +350,7 @@
;
; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing
; and it is a member of a group which is one of your pickup groups, then
-; you can answer it by picking up and dialing *8#. For simple offices, just
+; you can answer it by picking up and dialling *8#. For simple offices, just
; make these both the same. Groups range from 0 to 63.
;
callgroup=1
@@ -566,7 +566,7 @@
;
-; Used for distintive ring support for x100p.
+; Used for distinctive ring support for x100p.
; You can see the dringX patterns is to set any one of the dringXcontext fields
; and they will be printed on the console when an inbound call comes in.
;
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