Index: doc/channelvariables.txt =================================================================== --- doc/channelvariables.txt (revision 42715) +++ doc/channelvariables.txt (working copy) @@ -109,7 +109,7 @@ ;Remove the first character of extension, save in "number" variable exten => _9X.,1,Set(number=${EXTEN:1}) -Assuming we've dialed 918005551234, the value saved to the 'number' variable +Assuming we've dialled 918005551234, the value saved to the 'number' variable would be 18005551234. This is useful in situations when we require users to dial a number to access an outside line, but do not wish to pass the first digit. @@ -117,9 +117,9 @@ If you use a negative offset number, Asterisk starts counting from the end of the string and then selects everything after the new position. The following example will save the numbers 1234 to the 'number' variable, still assuming -we've dialed 918005551234. +we've dialled 918005551234. - ;Remove everything before the last four digits of the dialed string + ;Remove everything before the last four digits of the dialled string exten => _9X.,1,Set(number=${EXTEN:-4}) We can also limit the number of characters from our offset position that we Index: doc/speechrec.txt =================================================================== --- doc/speechrec.txt (revision 42715) +++ doc/speechrec.txt (working copy) @@ -179,7 +179,7 @@ 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: Index: doc/sms.txt =================================================================== --- doc/sms.txt (revision 42715) +++ doc/sms.txt (working copy) @@ -85,7 +85,7 @@ 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 @@ 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 Index: doc/enum.txt =================================================================== --- doc/enum.txt (revision 42715) +++ doc/enum.txt (working copy) @@ -78,7 +78,7 @@ these hostnames or SIP proxies are imaginary. Of course, the tel: replies go to directory assistance for New York City and San Francisco...) Also note that the complex SIP NAPTR at weight 30 will -strip off the leading "+" from the dialed string if it exists. This +strip off the leading "+" from the Dialed string if it exists. This is probably a better NAPTR than hard-coding the number into the NAPTR, and it is included as a more complex regexp example, though other simpler NAPTRs will work just as well. @@ -151,7 +151,7 @@ ensure your lookup includes the leading plus sign. Other DNS trees may or may not require a leading "+" - check before using those trees, as it is possible the parsed NAPTRs will not provide correct - results unless you have the correct dialed string. If you get + results unless you have the correct Dialed string. If you get console messages like "WARNING[24907]: enum.c:222 parse_naptr: NAPTR Regex match failed." then it is very possible that the returned NAPTR expects a leading "+" in the search string (or the returned @@ -228,7 +228,7 @@ All examples below except where noted use "e164.arpa" as the referenced domain, which is the default domain name for ENUMLOOKUP. -All numbers are assumed to not have a leading "+" as dialed by the +All numbers are assumed to not have a leading "+" as Dialed by the inbound channel, so that character is added where necessary during ENUMLOOKUP function calls. Index: doc/misdn.txt =================================================================== --- doc/misdn.txt (revision 42715) +++ doc/misdn.txt (working copy) @@ -122,7 +122,7 @@ 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 @@ 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 @@ -> 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, there protocols and states) -> fullstacks (shows the current active and inactive misdn channels) - restart @@ -255,7 +255,7 @@ 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 +In 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 @@ -> 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 successful CONNECT to other end -> you forgot to load mISDNdsp, which is now needed by chan_misdn for switching and dtmf tone detection Index: doc/chaniax.txt =================================================================== --- doc/chaniax.txt (revision 42715) +++ doc/chaniax.txt (working copy) @@ -180,7 +180,7 @@ * 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 = > amaflags = [default|omit|billing|documentation] @@ -188,7 +188,7 @@ These parameters affect call detail record generation. The first sets the account code for records received with IAX. The account code can be overridden on a per-user basis for incoming calls (see below). The -amaflags controls how the record is labeled ("omit" causes no record to be +amaflags controls how the record is labelled ("omit" causes no record to be written. "billing" and "documentation" label the records as billing or documentation records respectively, and "default" selects the system default. Index: doc/ip-tos.txt =================================================================== --- doc/ip-tos.txt (revision 42715) +++ doc/ip-tos.txt (working copy) @@ -39,11 +39,11 @@ 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 +precedences" 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 Index: doc/app-sms.txt =================================================================== --- doc/app-sms.txt (revision 42715) +++ doc/app-sms.txt (working copy) @@ -26,10 +26,10 @@ a suitable text capable landline phone, or a separate texting box such as a Magic Messenger on your phone line. This sends a message to a message centre your telco provides by making a normal call and sending - the data using 1200 Baud FSK signaling according to the ETSI spec. To + the data using 1200 Baud FSK signalling according to the ETSI spec. To receive a message the message centre calls the line with a specific calling number, and the text capable phone answers the call and - receives the data using 1200 Baud FSK signaling. This works + receives the data using 1200 Baud FSK signalling. This works particularly well in the UK as the calling line identity is sent before the first ring, so no phones in the house would ring when a message arrives. @@ -147,7 +147,7 @@ connected to a Magic Messenger. It could however by that you are operating Asterisk as a message centre for calls from outside. Either way, you look at the called number and goto smsmorx. In the UK, the - SMSC number that would be dialed is 1709400X where X is the caller sub + SMSC number that would be dialled is 1709400X where X is the caller sub address. As such typical usage in extension.config at the point of handling a call from a sip phone is:- exten = 17094009,1,Goto(smsmorx,${CALLERIDNUM},1) @@ -266,7 +266,7 @@ 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 @@ 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 exists, 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 @@ $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 @@ 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:- Index: doc/extensions.txt =================================================================== --- doc/extensions.txt (revision 42715) +++ doc/extensions.txt (working copy) @@ -15,7 +15,7 @@ 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 @@ 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 Index: doc/cygwin.txt =================================================================== --- doc/cygwin.txt (revision 42715) +++ doc/cygwin.txt (working copy) @@ -1,4 +1,4 @@ -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: Index: doc/osp.txt =================================================================== --- doc/osp.txt (revision 42715) +++ doc/osp.txt (working copy) @@ -10,7 +10,7 @@ 2.1.2 Preparing to build the OSP Toolkit 2.1.3 Building the OSP Toolkit 2.1.4 Installing the OSP Toolkit -2.1.5 Building the Enrollment Utility +2.1.5 Building the Enrolment Utility 2.2 Obtain Crypto Files 3 Asterisk 3.1 OSP Support Implementation @@ -129,11 +129,11 @@ The "Development" option is recommended for a first time build. The CFLAGS definition in the Makefile must be modified to build in development mode. -2.1.5 Building the Enrollment Utility - Device enrollment is the process of establishing a trusted cryptographic +2.1.5 Building the Enrolment Utility + Device enrolment is the process of establishing a trusted cryptographic relationship between the VoIP device and the OSP Server. The Enroll program is a utility application for establishing a trusted relationship between and - OSP client and an OSP server. Please see the document "Device Enrollment" at + OSP client and an OSP server. Please see the document "Device Enrolment" at www.sipfoundry.org/OSP/OSPclient for more information about the enroll application. 10) From within the OSP Toolkit directory (/usr/src/TK-3_3_4-20051103), @@ -153,10 +153,10 @@ present, the OSP module will not start and the Asterisk will not support the OSP protocol. Use the enroll.sh script from the toolkit distribution to enroll the Asterisk OSP module with an OSP server to obtain the crypto files. - Documentation explaining how to use the enroll.sh script (Device Enrollment) + Documentation explaining how to use the enroll.sh script (Device Enrolment) to enroll with an OSP server is available at www.sipfoundry.org/OSP/ospclient. Copy the files file generated by the - enrollment process to the Asterisk configuration directory. + enrolment process to the Asterisk configuration directory. Note: The osptestserver.transnexus.com is configured only for sending and receiving non-SSL messages, and issuing signed tokens. If you need help, post a message on the OSP mailing list of www.sipfoundry.org or send an e-mail to @@ -314,7 +314,7 @@ ; [general] ; - ; Should hardware accelleration be enabled? May not be changed + ; Should hardware acceleration be enabled? May not be changed ; on a reload. ; accelerate=no Index: doc/dundi.txt =================================================================== --- doc/dundi.txt (revision 42715) +++ doc/dundi.txt (working copy) @@ -19,7 +19,7 @@ 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. Index: doc/manager.txt =================================================================== --- doc/manager.txt (revision 42715) +++ doc/manager.txt (working copy) @@ -247,7 +247,7 @@ Paused: -- Queue member paused status Peer: -- "channel" specifier :-) PeerStatus: -- Peer status code - "Unregistred", "Registred", "Lagged", "Reachable" + "Unregistred", "Registered", "Lagged", "Reachable" Penalty: -- Queue penalty Priority: -- Extension priority Privilege: -- AMI authorization class (system, call, log, verbose, command, agent, user) @@ -284,7 +284,7 @@ Time: -- Roundtrip time (latency) Timeout: -- Parking timeout time Timeout: -- Timeout for call setup (Originate) - Timeout: -- Timout for call + Timeout: -- Timeout for call Uniqueid: -- Channel Unique ID Uniqueid1: -- Channel 1 Unique ID (Link event) Uniqueid2: -- Channel 2 Unique ID (Link event) Index: doc/CODING-GUIDELINES =================================================================== --- doc/CODING-GUIDELINES (revision 42715) +++ doc/CODING-GUIDELINES (working copy) @@ -476,7 +476,7 @@ ------------------------------------------ - 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 Index: doc/backtrace.txt =================================================================== --- doc/backtrace.txt (revision 42715) +++ doc/backtrace.txt (working copy) @@ -24,7 +24,7 @@ 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/ Index: doc/billing.txt =================================================================== --- doc/billing.txt (revision 42715) +++ doc/billing.txt (working copy) @@ -116,5 +116,5 @@ 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. Index: doc/jabber.txt =================================================================== --- doc/jabber.txt (revision 42715) +++ doc/jabber.txt (working copy) @@ -12,4 +12,4 @@ to provide the connection interface for chan_jingle. The maintainer of res_jabber is Matthew O'Gorman or -mog_work on irc or prefered mogorman@astjab.org over jabber. +mog_work on irc or preferred mogorman@astjab.org over jabber. Index: doc/jitterbuffer.txt =================================================================== --- doc/jitterbuffer.txt (revision 42715) +++ doc/jitterbuffer.txt (working copy) @@ -124,7 +124,7 @@ 2) Take a look and see what iax2 show netstats is saying about the call, and if it makes sense. -3) a tcpdump of the frames, (or, tethereal output from), so we can see the timestamps and delivery +3) a tcpdump of the frames, (or, ethereal output from), so we can see the timestamps and delivery times of the frames you're receiving. You can make such a tcpdump with: tcpdump -s 2048 -w /tmp/example.dump udp and port 4569 [and host ] Index: doc/callfiles.txt =================================================================== --- doc/callfiles.txt (revision 42715) +++ doc/callfiles.txt (working copy) @@ -27,7 +27,7 @@ The call file consists of : 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. Index: doc/ael.txt =================================================================== --- doc/ael.txt (revision 42715) +++ doc/ael.txt (working copy) @@ -13,7 +13,7 @@ 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 @@ 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 ------ @@ -616,7 +616,7 @@ } In the above, the 819/7079953345 extension will only be matched if the -CallerID is 7079953345, and the dialed number is 819. Hopefully you have +CallerID is 7079953345, and the dialled number is 819. Hopefully you have another 819 extension defined for all those who wish 819, that are not so lucky as to have 7079953345 as their CallerID! @@ -931,7 +931,7 @@ 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 @@ } -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 @@ 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. @@ -1142,7 +1142,7 @@ file, line number, and column. 4. It checks the contents of $[ ] expressions (or what will end up being $[ ] expressions!) for syntax errors. It also does - matching paren/bracket counts. + matching parent/bracket counts. 5. It runs several semantic checks after the parsing is over, but before the compiling begins, see the list above. 6. It handles #include "filepath" directives. -- ALMOST @@ -1174,10 +1174,10 @@ 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 @@ -1214,9 +1214,9 @@ "" ] The old way would do as shell scripts often do, and append something on both sides, like this: $[ ${x}foo = foo ]. The trouble with the old way, is that, if x contains any spaces, then - problems occur, usually syntax errors. It is better practice and + problems occur, usually syntax errors. It is better practise 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 ]. Index: doc/queues-with-callback-members.txt =================================================================== --- doc/queues-with-callback-members.txt (revision 42715) +++ doc/queues-with-callback-members.txt (working copy) @@ -66,7 +66,7 @@ "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 Index: doc/configuration.txt =================================================================== --- doc/configuration.txt (revision 42715) +++ doc/configuration.txt (working copy) @@ -64,7 +64,7 @@ --------------------- 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 Index: doc/asterisk-conf.txt =================================================================== --- doc/asterisk-conf.txt (revision 42715) +++ doc/asterisk-conf.txt (working copy) @@ -8,7 +8,7 @@ --------------- [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 @@ 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) Index: doc/PEERING =================================================================== --- doc/PEERING (revision 42715) +++ doc/PEERING (working copy) @@ -147,7 +147,7 @@ (e.g., for the purpose of locating facsimile, modem services, or systematic telemarketing). - (e) Initial control signaling for all communication sessions that + (e) Initial control signalling for all communication sessions that utilize Routes obtained from the Peering System must be sent from a member of the Peering System to the Service or Egress Gateway identified in the selected Route. For example, 'SIP Index: doc/imapstorage.txt =================================================================== --- doc/imapstorage.txt (revision 42715) +++ doc/imapstorage.txt (working copy) @@ -137,7 +137,7 @@ 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 Index: configs/extensions.ael.sample =================================================================== --- configs/extensions.ael.sample (revision 42715) +++ configs/extensions.ael.sample (working copy) @@ -61,9 +61,9 @@ // ! - 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 diallings, // 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 @@ -101,7 +101,7 @@ // // ignorepat => 9; // -// so that dialtone remains even after dialing a 9. +// so that dialtone remains even after dialling a 9. //}; @@ -189,10 +189,10 @@ }; // -// 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. +// to be on-line or else dialling can be severely delayed. // context iaxprovider { switches { @@ -228,7 +228,7 @@ context trunklocal { // - // Local seven-digit dialing accessed through trunk interface + // Local seven-digit dialling accessed through trunk interface // _9NXXXXXX => { Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}); Index: configs/skinny.conf.sample =================================================================== --- configs/skinny.conf.sample (revision 42715) +++ configs/skinny.conf.sample (working copy) @@ -25,7 +25,7 @@ ;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, usualy sent from exotic devices + ; 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 @@ -48,7 +48,7 @@ ; Typical config for a 7910 ;[duba] ; Device name -;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 Index: configs/alsa.conf.sample =================================================================== --- configs/alsa.conf.sample (revision 42715) +++ configs/alsa.conf.sample (working copy) @@ -27,7 +27,7 @@ ; ;mohinterpret=default ; -; 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 @@ -49,11 +49,11 @@ ; 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, usualy sent from exotic devices + ; 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 implementation are currenlty available - "fixed" + ; channel. Two implementation 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. Index: configs/iax.conf.sample =================================================================== --- configs/iax.conf.sample (revision 42715) +++ configs/iax.conf.sample (working copy) @@ -231,7 +231,7 @@ ; ; 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 @@ -359,7 +359,7 @@ ;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 Index: configs/followme.conf.sample =================================================================== --- configs/followme.conf.sample (revision 42715) +++ configs/followme.conf.sample (working copy) @@ -41,14 +41,14 @@ number=>01233456,25 ; The a follow-me number to call. The format is: ; number=> [, [, ] ] -; You can specify as many of these numbers as you like. They will be dialed in the +; You can specify as many of these numbers as you like. They will be dialled in the ; order that you specify them in the config file OR as specified with the order field -; on the number prompt. As you can see from the example, forked dialing of multiple +; on the number prompt. As you can see from the example, forked dialling of multiple ; numbers in the same step is supported with this application if you'd like to dial ; multiple numbers in the same followme step. ; It's also important to note that the timeout value is not the same ; as the timeout value you would use in app_dial. This timeout value is the amount of -; time allowed between the time the dialing step starts and the callee makes a choice +; time allowed between the time the dialling step starts and the callee makes a choice ; on whether to take the call or not. That being the case, you may want to account for ; this time, and make this timeout longer than a timeout you might specify in app_dial. takecall=>1 Index: configs/mgcp.conf.sample =================================================================== --- configs/mgcp.conf.sample (revision 42715) +++ configs/mgcp.conf.sample (working copy) @@ -21,11 +21,11 @@ ; 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, usualy sent from exotic devices + ; 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 MGCP - ; channel. Two implementations are currenlty available - "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. Index: configs/sip.conf.sample =================================================================== --- configs/sip.conf.sample (revision 42715) +++ configs/sip.conf.sample (working copy) @@ -60,7 +60,7 @@ ;maxexpiry=3600 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) ;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) -;defaultexpiry=120 ; Default length of incoming/outoing registration +;defaultexpiry=120 ; Default length of incoming/outgoing registration ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts ; Defaults to 100 ms ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY @@ -294,7 +294,7 @@ ; For non-realtime peers, when their registration expires, the ; information will _not_ be removed from memory or the Asterisk database ; if you attempt to place a call to the peer, the existing information - ; will be used in spiteof it having expired + ; will be used in spite of it having expired ; ; For realtime peers, when the peer is retrieved from realtime storage, ; the registration information will be used regardless of whether @@ -353,11 +353,11 @@ ; 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, usualy sent from exotic devices + ; 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 currenlty available - "fixed" + ; channel. Two implementations are currently available - "fixed" ; (with size always equals to jbmaxsize) and "adaptive" (with ; variable size, actually the new jb of IAX2). Defaults to fixed. @@ -450,7 +450,7 @@ ;host=box.provider.com ;usereqphone=yes ; This provider requires ";user=phone" on URI ;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer -;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer +;outboundproxy=proxy.provider.domain ; send outbound signalling to this proxy, not directly to the peer ; Call-limits will not be enforced on real-time peers, ; since they are not stored in-memory @@ -465,7 +465,7 @@ ; ; 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 @@ -566,7 +566,7 @@ ; ;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 ;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address ;permit=192.168.0.60/255.255.255.0 Index: configs/osp.conf.sample =================================================================== --- configs/osp.conf.sample (revision 42715) +++ configs/osp.conf.sample (working copy) @@ -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 Index: configs/dundi.conf.sample =================================================================== --- configs/dundi.conf.sample (revision 42715) +++ configs/dundi.conf.sample (working copy) @@ -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 Index: configs/oss.conf.sample =================================================================== --- configs/oss.conf.sample (revision 42715) +++ configs/oss.conf.sample (working copy) @@ -58,11 +58,11 @@ ; 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, usualy sent from exotic devices + ; 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 an OSS - ; channel. Two implementations are currenlty available - "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. Index: configs/enum.conf.sample =================================================================== --- configs/enum.conf.sample (revision 42715) +++ configs/enum.conf.sample (working copy) @@ -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 Index: configs/queues.conf.sample =================================================================== --- configs/queues.conf.sample (revision 42715) +++ configs/queues.conf.sample (working copy) @@ -24,7 +24,7 @@ ; probably more along the lines of how a queue should work and ; in most cases, you will want to enable this behavior. If you ; do not specify or comment out this option, it will default to no -; to keep backward compatability with the old behavior. +; to keep backward compatibility with the old behavior. ; autofill = yes ; @@ -35,7 +35,7 @@ ; the concept of "joining/mixing" the in/out files now goes away ; when this is enabled. You can set the default type for all queues ; here, and then also change monitor-type for individual queues within -; queue by using the same configuation parameter within a queue +; queue by using the same configuration parameter within a queue ; configuration block. If you do not specify or comment out this option, ; it will default to the old 'Monitor' behavior to keep backward ; compatibility. Index: configs/voicemail.conf.sample =================================================================== --- configs/voicemail.conf.sample (revision 42715) +++ configs/voicemail.conf.sample (working copy) @@ -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 @@ -54,7 +54,7 @@ ; called when a voicemail is left, delivered, or your voicemailbox ; is checked, uncomment this. It can also be set to 'smdi' to use ; smdi for external notification. If it is 'smdi', smdiport should -; be set to a valid port as specfied in smdi.conf. +; be set to a valid port as specified in smdi.conf. ;externnotify=/usr/bin/myapp ;smdiport=/dev/ttyS0 @@ -87,7 +87,7 @@ ; 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 ; @@ -147,11 +147,11 @@ ; 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 below. Irrelevant if envelope=no. ; attach=yes ; Attach the voicemail to the notification email *NOT* the pager email @@ -167,7 +167,7 @@ ; sayduration=no ; Turn on/off the duration information before the message. [ON by default] ; saydurationm=2 ; Specify the minimum duration to say. Default is 2 minutes ; dialout=fromvm ; Context to dial out from [option 4 from the advanced menu] - ; if not listed, dialing out will not be permitted + ; if not listed, dialling out will not be permitted sendvoicemail=yes ; Context to Send voicemail from [option 5 from the advanced menu] ; if not listed, sending messages from inside voicemail will not be ; permitted Index: configs/cdr.conf.sample =================================================================== --- configs/cdr.conf.sample (revision 42715) +++ configs/cdr.conf.sample (working copy) @@ -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 Index: configs/misdn.conf.sample =================================================================== --- configs/misdn.conf.sample (revision 42715) +++ configs/misdn.conf.sample (working copy) @@ -89,7 +89,7 @@ ; 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 ; @@ -97,7 +97,7 @@ ;;; CRYPTION STUFF -; Wether to look for dynamic crypting attempt +; Whether to look for dynamic crypting attempt ; ; default value: no ; @@ -119,7 +119,7 @@ ; 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 @@ -149,7 +149,7 @@ musicclass=default ; -; Either if we should produce DTMF Tones ourselve +; Either if we should produce DTMF Tones ourselves ; senddtmf=yes @@ -181,7 +181,7 @@ 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 @@ -192,7 +192,7 @@ ; ; 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 @@ -298,7 +298,7 @@ ; Pickup and Callgroup ; -; deafult values: not set = 0 +; default values: not set = 0 ; range: 0-63 ; ;callgroup=1 @@ -312,7 +312,7 @@ ; 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 @@ -364,7 +364,7 @@ ; ; defines the maximum amount of incoming calls per port for ; this group. Calls which exceed the maximum will be marked with -; the channel varible MAX_OVERFLOW. It will contain the amount of +; the channel variable MAX_OVERFLOW. It will contain the amount of ; overflowed calls ; max_incoming=-1 @@ -392,11 +392,11 @@ [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=* Index: configs/features.conf.sample =================================================================== --- configs/features.conf.sample (revision 42715) +++ configs/features.conf.sample (working copy) @@ -22,7 +22,7 @@ ; as long as the class is not set on the channel directly ; using Set(CHANNEL(musicclass)=whatever) in the dialplan -;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 ; (default is 3 seconds) ;xfersound = beep ; to indicate an attended transfer is complete ;xferfailsound = beeperr ; to indicate a failed transfer Index: configs/zapata.conf.sample =================================================================== --- configs/zapata.conf.sample (revision 42715) +++ configs/zapata.conf.sample (working copy) @@ -112,7 +112,7 @@ ; ;resetinterval = 3600 ; -; Overlap dialing mode (sending overlap digits) +; Overlap dialling mode (sending overlap digits) ; ;overlapdial=yes ; @@ -207,7 +207,7 @@ ; format. If you only specify 'signalling', then it will be the format for ; both inbound and outbound. ; -; signaling=featdmf +; signalling=featdmf ; outsignalling=featb ; ; For Feature Group D Tandem access, to set the default CIC and OZZ use these @@ -229,7 +229,7 @@ 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 @@ -316,7 +316,7 @@ ; 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: ; @@ -380,7 +380,7 @@ ; ; 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 @@ -419,7 +419,7 @@ ; ; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel ; basis if you would like that channel to behave like an SMDI message desk. -; The SMDI port specfied should have already been defined in smdi.conf. The +; The SMDI port specified should have already been defined in smdi.conf. The ; default port is /dev/ttyS0. ; ;usesmdi=yes @@ -547,11 +547,11 @@ ; 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, usualy sent from exotic devices + ; 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 ZAP - ; channel. Two implementations are currenlty available - "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. @@ -641,7 +641,7 @@ ; -; 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. ; Index: configs/vpb.conf.sample =================================================================== --- configs/vpb.conf.sample (revision 42715) +++ configs/vpb.conf.sample (working copy) @@ -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 fullness. ; bal1 => Hybrid balance codec register 1 ; bal2 => Hybrid balance codec register 2 ; bal3 => Hybrid balance codec register 3 Index: configs/smdi.conf.sample =================================================================== --- configs/smdi.conf.sample (revision 42715) +++ configs/smdi.conf.sample (working copy) @@ -11,7 +11,7 @@ ;twostopbits = no -; Character size or bit length is the size of each character sent accross the +; Character size or bit length is the size of each character sent across the ; link. Character size can be 7 or 8. The default is 7. ;charsize = 7 @@ -34,7 +34,7 @@ ; Occasionally Asterisk and the SMDI switch may become out of sync. If this ; happens, Asterisk will appear one or several calls behind as it processes -; voicemail requests. To prevent this from hapening adjust the msgexpirytime. +; voicemail requests. To prevent this from happening adjust the msgexpirytime. ; This will make Asterisk discard old SMDI messages that have not yet been ; processed. The default expiry time is 30000 milliseconds. Index: configs/extensions.conf.sample =================================================================== --- configs/extensions.conf.sample (revision 42715) +++ configs/extensions.conf.sample (working copy) @@ -31,7 +31,7 @@ ; or HANGUP depending on Asterisk's best guess. This is the default. ; ; If autofallthrough is not set, then if an extension runs out of -; things to do, Asterisk will wait for a new extension to be dialed +; things to do, Asterisk will wait for a new extension to be dialled ; (this is the original behavior of Asterisk 1.0 and earlier). ; ;autofallthrough=no @@ -108,9 +108,9 @@ ; ! - 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 diallings, ; 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 @@ -151,7 +151,7 @@ ; ;ignorepat => 9 ; -; so that dialtone remains even after dialing a 9. +; so that dialtone remains even after dialling a 9. ; ; @@ -228,10 +228,10 @@ exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel) ; -; The SWITCH statement permits a server to share the dialplain with +; The SWITCH statement permits a server to share the dialplan with ; another server. Use with care: Reciprocal switch statements are not ; allowed (e.g. both A -> B and B -> A), and the switched server needs -; to be on-line or else dialing can be severly delayed. +; to be on-line or else dialling can be severely delayed. ; [iaxprovider] ;switch => IAX2/user:[key]@myserver/mycontext @@ -252,7 +252,7 @@ [trunklocal] ; -; Local seven-digit dialing accessed through trunk interface +; Local seven-digit dialling accessed through trunk interface ; exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) @@ -388,7 +388,7 @@ exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions -exten => s,n,WaitExten ; Wait for an extension to be dialed. +exten => s,n,WaitExten ; Wait for an extension to be dialled. exten => 2,1,BackGround(demo-moreinfo) ; Give some more information. exten => 2,n,Goto(s,instruct) @@ -423,7 +423,7 @@ exten => i,1,Playback(invalid) ; "That's not valid, try again" ; -; Create an extension, 500, for dialing the +; Create an extension, 500, for dialling the ; Asterisk demo. ; exten => 500,1,Playback(demo-abouttotry); Let them know what's going on @@ -432,7 +432,7 @@ 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 Index: configs/modem.conf.sample =================================================================== --- configs/modem.conf.sample (revision 42715) +++ configs/modem.conf.sample (working copy) @@ -24,12 +24,12 @@ ;type=autodetect ;type=aopen ; -; We can strip a given number of digits on outgoing dialing, so, for example +; We can strip a given number of digits on outgoing dialling, so, for example ; you can have it dial "8871042" when given "98871042". ; stripmsd=0 ; -; Type of dialing +; Type of dialling ; dialtype=tone ;dialtype=pulse Index: configs/rpt.conf.sample =================================================================== --- configs/rpt.conf.sample (revision 42715) +++ configs/rpt.conf.sample (working copy) @@ -21,7 +21,7 @@ ;;dphone_functions = functions-dphone ; (optional) different functions for 'D' mode ;;nodes = nodes-different ; (optional) different node list ;tonezone = us ; use US tones (default) -;context = default ; dialing context for phone +;context = default ; dialling context for phone ;callerid = "WB6NIL Repeater" <(213) 555-0123> ; Callerid for phone calls ;idrecording = wb6nil ; id recording ;accountcode=RADIO ; account code (optional) @@ -62,7 +62,7 @@ ;;dphone_functions = functions-dphone ; (optional) different functions for 'D' mode ;;nodes = nodes-different ; (optional) different node list ;tonezone = us ; use US tones (default) -;context = default ; dialing context for phone +;context = default ; dialling context for phone ;callerid = "WB6NIL Repeater" <(213) 555-0123> ; Callerid for phone calls ;idrecording = wb6nil ; id recording ;accountcode=RADIO ; account code (optional) Index: configs/phone.conf.sample =================================================================== --- configs/phone.conf.sample (revision 42715) +++ configs/phone.conf.sample (working copy) @@ -30,7 +30,7 @@ ; echocancel=medium ; -; You can optionally use VAD/CNG silence supression +; You can optionally use VAD/CNG silence suppression ; ;silencesupression=yes ; @@ -40,7 +40,7 @@ ; ; 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%). ;