More spelling fixes (issue 7930)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

doc/CODING-GUIDELINES

@@ -476,7 +476,7 @@ preceding them.
 ------------------------------------------
 
 - Look at the code once more
-When you achieve your desired functionalty, make another few refactor
+When you achieve your desired functionality, make another few refactor
 passes over the code to optimize it.
 
 - Read the patch

doc/ael.txt

@@ -13,7 +13,7 @@ functionality.
 
 AEL is really the merger of 4 different 'languages', or syntaxes:
 
-    * The first and most obvious is the AEL syntax itselft. A BNF is
+    * The first and most obvious is the AEL syntax itself. A BNF is
       provided near the end of this document.
 
     * The second syntax is the Expression Syntax, which is normally
@@ -92,7 +92,7 @@ lines together.
 
 The extensions can also contain "goto" or "jump" commands to skip to
 extensions in other contexts. Conditionals provide the ability to
-react to different stimiuli, and there you have it.
+react to different stimuli, and there you have it.
 
 Macros
 ------
@@ -931,7 +931,7 @@ statements. It means the "first" statement in the extension. I would
 not advise trying to use numeric labels other than "1" in goto's or
 jumps, nor would I advise declaring a "1" label anywhere! As a matter
 of fact, it would be bad form to declare a numeric label, and it might
-confllict with the priority numbers used internally by asterisk.
+conflict with the priority numbers used internally by asterisk.
 
 The syntax of the jump statement is: jump
 extension[,priority][@context] If priority is absent, it defaults to
@@ -1006,7 +1006,7 @@ macro std-exten( ext , dev ) {
 }
 
 
-A macro is then called by preceeding the macro name with an
+A macro is then called by preceding the macro name with an
 ampersand. Empty arguments can be passed simply with nothing between
 comments(0.11).
 
@@ -1130,7 +1130,7 @@ These checks will be:
 Differences with the original version of AEL
 ============================================
 
-   1. The $[...] expressions have been enhanced to inlcude the ==, ||,
+   1. The $[...] expressions have been enhanced to include the ==, ||,
       and && operators. These operators are exactly equivalent to the
       =, |, and & operators, respectively. Why? So the C, Java, C++
       hackers feel at home here.
@@ -1174,10 +1174,10 @@ Differences with the original version of AEL
   13. Added the optional time spec to the contexts in the includes
       construct. See examples above.
   14. You don't have to wrap a single "true" statement in curly
-      braces, as in the orignal AEL. An "else" is attached to the
+      braces, as in the original AEL. An "else" is attached to the
       closest if. As usual, be careful about nested if statements!
       When in doubt, use curlies!
-  15. Added the syntax [regexten] [hint(channel)] to preceed an
+  15. Added the syntax [regexten] [hint(channel)] to precede an
       extension declaration. See examples above, under
       "Extension". The regexten keyword will cause the priorities in
       the extension to begin with 2 instead of 1. The hint keyword
@@ -1216,7 +1216,7 @@ Hints and Bugs
      trouble with the old way, is that, if x contains any spaces, then
      problems occur, usually syntax errors. It is better practice and
      safer wrap all such tests with double quotes! Also, there are now
-     some functions that can be used in a variable referenece,
+     some functions that can be used in a variable reference,
      ISNULL(), and LEN(), that can be used to test for an empty string:
      ${ISNULL(${x})} or $[ ${LEN(${x}) = 0 ].
 

doc/app-sms.txt

@@ -266,7 +266,7 @@ Using smsq
    Create queue, but do not dial to send message
    --no-wait
    Do not wait if a call appears to be in progress
-   This could have a small window where a mesdsage is queued but not
+   This could have a small window where a message is queued but not
    sent, so regular calls to smsq should be done to pick up any missed
    messages
    --concurrent
@@ -314,7 +314,7 @@ Using smsq
    Note that when smsq attempts to make a file in
    /var/spool/asterisk/outgoing, it checks if there is already a call
    queued for that queue. It will try several filenames, up to the
-   --concorrent setting. If these files exists, then this means Asterisk
+   --concurrent setting. If these files exist, then this means Asterisk
    is already queued to send all messages for that queue, and so Asterisk
    should pick up the message just queued. However, this alone could
    create a race condition, so if the files exist then smsq will wait up
@@ -359,13 +359,13 @@ Using smsq
    $ude
    User data, escaped UTF-8, including all characters, but control
    characters \n, \r, \t, \f, \xxx and \ is escaped as \\
-   Useful fGuaranteed one line printable text, so useful in Subject lines
+   Useful guaranteed one line printable text, so useful in Subject lines
    of emails, etc
    $ud8
    Hex UCS-1 coding of user data (2 hex digits per character)
    Present only if all user data is in range U+0000 to U+00FF
    $ud16
-   Hex UCS-2 coding of user data (4 hex digits per chartacter)
+   Hex UCS-2 coding of user data (4 hex digits per character)
    other
    Other fields set using their field name, e.g. mr, pid, dcs, etc. udh
    is a hex byte string
@@ -388,7 +388,7 @@ File formats
    line starts with a keyword and an = and then data. udh and ud have
    options for hex encoding, see below.
    UTF-8. The user data (ud) field is treated as being UTF-8 encoded
-   unless the DCS is specified indicating 8 bit formart. If 8 bit format
+   unless the DCS is specified indicating 8 bit format. If 8 bit format
    is specified then the user data is sent as is.
    The keywords are as follows:-
 

doc/asterisk-conf.txt

@@ -8,7 +8,7 @@ and does not touch it when it already exists.
 ---------------
 
 [directories]
-; Make sure these directoriess have the right permissions if not
+; Make sure these directories have the right permissions if not
 ; running Asterisk as root 
 
 ; Where the configuration files (except for this one) are located
@@ -51,7 +51,7 @@ dumpcore = yes | no				; Dump core on failure (-g)
 quiet = yes | no				; Run quietly (-q)
 timestamp = yes | no				; Force timestamping in CLI verbose output (-T)
 runuser = asterisk				; User to run asterisk as (-U) NOTE: will require changes to
-						; directory and device permisions
+						; directory and device permissions
 rungroup = asterisk				; Group to run asterisk as (-G)
 internal_timing = yes | no			; Enable internal timing support (-I)
 

doc/backtrace.txt

@@ -24,7 +24,7 @@ The interesting information is located in the last column.
 
 Second, your copy of Asterisk must have been built without
 optimization or the backtrace will be (nearly) unusable. This can be
-done by using 'make dont-optimize' intead of 'make install' to build
+done by using 'make dont-optimize' instead of 'make install' to build
 and install the Asterisk binary and modules.
 
 After Asterisk crashes, a core file will be "dumped" in your /tmp/

doc/billing.txt

@@ -116,5 +116,5 @@ The following format string will emulate the regular cdr file format:
 
 format => "${CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}"
 
-You can put anything you want as the value of format incuding new cdr vars you make up or any global variables.
+You can put anything you want as the value of format including new cdr vars you make up or any global variables.
 

doc/callfiles.txt

@@ -27,7 +27,7 @@ The call file syntax
 The call file consists of <Key>: <value> pairs; one per line.
 
 Comments are indicated by a '#' character that begins a line, or follows a space
-or tab character.  To be consistant with the configuration files in Asterisk,
+or tab character.  To be consistent with the configuration files in Asterisk,
 comments can also be indicated by a semicolon.  However, the multiline comments
 (;-- --;) used in Asterisk configuration files are not supported. Semicolons can
 be escaped by a backslash.

doc/chaniax.txt

@@ -180,7 +180,7 @@ connections to be over a LAN.
 * maxexcessbuffer is the maximum amount of excess jitter buffer 
   that is permitted before the jitter buffer is slowly shrunk to eliminate 
   latency.
-* minexcessbuffer is the minimum amout of excess jitter buffer
+* minexcessbuffer is the minimum amount of excess jitter buffer
 
 > accountcode = <code>
 > amaflags = [default|omit|billing|documentation]

doc/configuration.txt

@@ -64,7 +64,7 @@ Including other files
 ---------------------
 In all of the configuration files, you may include the content of another
 file with the #include statement. The content of the other file will be
-included at the row that the #include statement occured.
+included at the row that the #include statement occurred.
 	
 	#include myusers.conf
 

doc/cygwin.txt

@@ -1,4 +1,6 @@
-Cygwin support is completely experimental and usupported at this time.  The current state of cygwin support is that it will compile, and start the cli, but will not yet take calls properly.
+Cygwin support is completely experimental and unsupported at this time.  The
+current state of cygwin support is that it will compile, and start the cli,
+but will not yet take calls properly.
 
 To compile with cygwin, you will need at least a standard base cygwin install plus the following packages:
 

doc/dundi.txt

@@ -19,7 +19,7 @@ You can find the PEERING agreement in the doc directory.
 
 Much less dramatically, DUNDi can also be used within a private
 enterprise to share a dialplan efficiently between multiple nodes,
-without incuring a risk of a single point of failure.  In this way,
+without incurring a risk of a single point of failure.  In this way,
 administrators can locally add extensions which become immediately
 available to the other nodes in the system.
 

doc/extensions.txt

@@ -15,7 +15,7 @@ of characters (a-z), digits, #, and *. Please note that when matching a
 pattern, "N", "X", and "Z" are interpreted as classes of digits.
 
 For each extension, several actions may be listed and must be given a unique
-priority. When each action completes, the call continunes at the next priority
+priority. When each action completes, the call continues at the next priority
 (except for some modules which use explicitly GOTO's). 
 
 When each action completes, it generally moves to the next priority (except for
@@ -34,7 +34,7 @@ functions in your Asterisk, use the "show functions" command.
 The example dial plan, in the configs/extensions.conf.sample file
 is installed as extensions.conf if you run "make samples" after
 installation of Asterisk. This file includes many more instructions
-and examples than this file, so it's worthwile to read it.
+and examples than this file, so it's worthwhile to read it.
 	
 * Special extensions
 

doc/imapstorage.txt

@@ -137,7 +137,7 @@ for deletion when a user exits the voicemail system (hangs up the phone).
 UW IMAP-2006 Development Branch
 -------------------------------
 This version supports UIDPLUS, which allows UID_EXPUNGE capabilities.  This
-feature allow the system to expunge ONLY pretainent messages, instead of the
+feature allow the system to expunge ONLY pertinent messages, instead of the
 default behavior, which is to expunge ALL messages marked for deletion when
 EXPUNGE is called.  The IMAP storage mechanism is this version of Asterisk
 will check if the UID_EXPUNGE feature is supported by the server, and use it

doc/ip-tos.txt

@@ -39,11 +39,11 @@ The tos* parameters also take numeric values.
 
 The lowdelay, throughput, reliability, mincost, and none values are
 deprecated because they set the IP TOS using the outdated "IP
-prececence" model as defined in RFC 791 and RFC 1349. They still
+precedence" model as defined in RFC 791 and RFC 1349. They still
 work in this version of Asterisk, but will be removed in future releases.
 
 ===========================================
-Configuation	Parameter	Recommended
+Configuration	Parameter	Recommended
 File				Setting
 -------------------------------------------
 sip.conf	tos_sip		cs3

doc/jabber.txt

@@ -12,4 +12,4 @@ the dial plan, jabberstatus, jabbersend, and soon jabberrecv.  res_jabber is als
 to provide the connection interface for chan_jingle.
 
 The maintainer of res_jabber is Matthew O'Gorman <mogorman@digum.com> or
-mog_work on irc or prefered mogorman@astjab.org over jabber.
+mog_work on irc or (preferred) mogorman@astjab.org over jabber.

doc/manager.txt

@@ -247,7 +247,7 @@ Some standard AMI headers:
  Paused: <num>			-- Queue member paused status
  Peer: <tech/name>		-- "channel" specifier :-)
  PeerStatus: <tech/name>	-- Peer status code 
-					"Unregistred", "Registred", "Lagged", "Reachable"
+					"Unregistered", "Registered", "Lagged", "Reachable"
  Penalty: <num>			-- Queue penalty
  Priority:			-- Extension priority
  Privilege: <privilege>		-- AMI authorization class (system, call, log, verbose, command, agent, user)
@@ -284,7 +284,7 @@ Some standard AMI headers:
  Time: <sec>			-- Roundtrip time (latency)
  Timeout:			-- Parking timeout time
  Timeout:			-- Timeout for call setup (Originate)
- Timeout: <seconds>		-- Timout for call
+ Timeout: <seconds>		-- Timeout for call
  Uniqueid:			-- Channel Unique ID
  Uniqueid1:			-- Channel 1 Unique ID (Link event)
  Uniqueid2:			-- Channel 2 Unique ID (Link event)

doc/misdn.txt

@@ -122,7 +122,7 @@ contain the ports variable which mean the mISDN Ports. Here you can add
 multiple ports, comma separated.
 
 Espacially for TE-Mode Ports there is a msns option. This option tells the
-chan_misdn driver to listen for incomming calls with the given msns, you can
+chan_misdn driver to listen for incoming calls with the given msns, you can
 insert a '*' as single msn, which leads in getting every incoming call (if you
 want to share on PMP TE S0 with a asterisk and a phone or isdn card you should
 insert here the msns which you'll like to give the Asterisk).  Finally a
@@ -148,7 +148,7 @@ The available Optchars are:
 	n - don't detect dtmf tones on called channel
 	h - make digital outgoing call
 	c - make crypted outgoing call, param is keyindex
-	e - perform echo cancelation on this channel, 
+	e - perform echo cancellation on this channel, 
 	    takes taps as arguments (32,64,128,256)
 	s - send Non Inband DTMF as inband
 	vr - rxgain control
@@ -203,7 +203,7 @@ Now you should see the misdn cli commands:
 	-> config	(shows the configuration options)
 	-> channels	(shows the current active misdn channels)
 	-> channel	(shows details about the given misdn channels)
-	-> stacks	(shows the currend ports, there protocols and states)
+	-> stacks	(shows the current ports, their protocols and states)
 	-> fullstacks	(shows the current active and inactive misdn channels)
 
 - restart 
@@ -255,7 +255,7 @@ exten => _0X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1})
 exten => _1X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1}/:dHello)
 exten => _1X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1}/:dHello Test:n)
 
-In the last line you will notice the last argument (Hello), this is sended
+On the last line, you will notice the last argument (Hello); this is sent
 as Display Message to the Phone.
 
 Known Problems
@@ -265,11 +265,11 @@ Known Problems
 
 -> You need to use ztdummy as dummy zaptel interface for the iax timing in
 trunking mode, simply grab libpri, zaptel and compile them (i think you need
-to modify the makefile in zaptel to add ztdummy to the defaultly compiled
+to modify the makefile in zaptel to add ztdummy to the default compiled
 modules) then modprobe ztdummy, this resolves the problem.
 
 
-* I cannot hear any tone after succesfull CONNECT to other end
+* I cannot hear any tone after a successful CONNECT to the other end
 
 -> you forgot to load mISDNdsp, which is now needed by chan_misdn for switching
 and dtmf tone detection

doc/osp.txt

@@ -314,7 +314,7 @@ TransNexus and OSP Secured are trademarks of TransNexus, Inc.
   ;
   [general]
   ;
-  ; Should hardware accelleration be enabled?  May not be changed
+  ; Should hardware acceleration be enabled?  May not be changed
   ; on a reload.
   ;
   accelerate=no

doc/queues-with-callback-members.txt

@@ -66,7 +66,7 @@ call.
 "joinempty" set to "strict" will keep incoming callers from
 being placed in queues where there are no agents to take calls.
 The Queue() application will return, and the dial plan can 
-detemine what to do next.
+determine what to do next.
 
 If there are calls queued, and the last agent logs out, the
 remaining incoming callers will immediately be removed from

doc/sms.txt

@@ -85,7 +85,7 @@ specified. oa is ignored.
 
 When generating files to send to a phone, only oa and ud need be specified. da is ignored.
 
-When receing a message as a service centre, only the destination address is
+When receiving a message as a service centre, only the destination address is
 sent, so the originating address is set to the callerid.
 
 EXAMPLES
@@ -114,7 +114,7 @@ The script sends
 You put the message as the name of the caller ID (messy, I know), the
 originating number and hence queue name as the number of the caller ID and the
 exten as the number to which the sms is to be sent. The context uses SMS to
-create the message in the queue and then SMS to communicate iwth 17094009 to
+create the message in the queue and then SMS to communicate with 17094009 to
 actually send the message.
 
 Note that the 9 on the end of 17094009 is the sub address 9 meaning no sub

doc/speechrec.txt

@@ -179,7 +179,7 @@ The module res_speech.so exports a C based API that any developer can use to spe
 recognize enable their application. The API gives greater control, but requires the 
 developer to do more on their end in comparison to the dialplan speech utilities.
 
-For all API calls that return an integer value a non-zero value indicates an error has occured.
+For all API calls that return an integer value, a non-zero value indicates an error has occurred.
 
 - Creating a speech structure: