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Fix escaping and some of the formattting (closes issue #10285)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Tilghman Lesher
2007-07-24 18:19:18 +00:00
parent cf551decc5
commit 95bee48cde
13 changed files with 588 additions and 574 deletions
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36
doc/tex/ael.tex
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@@ -584,7 +584,7 @@ name/number with a slash (/) and the number to match against the Caller ID:
\begin{verbatim}
context zoombo
{
819/7079953345 => { NoOp(hello, 3345); }
819/7079953345 => { NoOp(hello, 3345); }
}
\end{verbatim}
@@ -727,8 +727,8 @@ You can declare variables in Macros, as so:
\begin{verbatim}
Macro myroutine(firstarg, secondarg)
{
Myvar=1;
NoOp(Myvar is set to ${myvar});
Myvar=1;
NoOp(Myvar is set to ${myvar});
}
\end{verbatim}
@@ -747,8 +747,8 @@ before setting them to a value;
\begin{verbatim}
Macro myroutine(firstarg, secondarg)
{
local Myvar=1;
NoOp(Myvar is set to ${Myvar}, and firstarg is ${firstarg}, and secondarg is ${secondarg});
local Myvar=1;
NoOp(Myvar is set to ${Myvar}, and firstarg is ${firstarg}, and secondarg is ${secondarg});
}
\end{verbatim}
@@ -760,8 +760,8 @@ If you need to make a local variable within the Set() application, you can do it
\begin{verbatim}
Macro myroutine(firstarg, secondarg)
{
Set(LOCAL(Myvar)=1);
NoOp(Myvar is set to ${Myvar}, and firstarg is ${firstarg}, and secondarg is ${secondarg});
Set(LOCAL(Myvar)=1);
NoOp(Myvar is set to ${Myvar}, and firstarg is ${firstarg}, and secondarg is ${secondarg});
}
\end{verbatim}
@@ -787,9 +787,9 @@ context loops {
}
\end{verbatim}
NOTE: The conditional expression (the "\${y} >= 0" above) is wrapped in
NOTE: The conditional expression (the "\${y} $>$= 0" above) is wrapped in
\$[ ] so it can be evaluated. NOTE: The for loop test expression
(the "\${x} < 3" above) is wrapped in \$[ ] so it can be evaluated.
(the "\${x} $<$ 3" above) is wrapped in \$[ ] so it can be evaluated.
@@ -1134,9 +1134,7 @@ These checks will be:
currently known functions. A warning is issued if it is not.
\end{itemize}
Differences with the original version of AEL
============================================
\section{Differences with the original version of AEL}
\begin{enumerate}
\item The \$[...] expressions have been enhanced to include the ==, ||,
@@ -1174,8 +1172,8 @@ Differences with the original version of AEL
AEL README.
\item Added the "return" keyword, which will jump to the end of an
extension/Macro.
\item Added the ifTime (<time range>|<days of week>|<days of
month>|<months> ) {} [else {}] construct, which executes much
\item Added the ifTime ($<$time range$>$|$<$days of week$>$|$<$days of
month$>$|$<$months$>$ ) {} [else {}] construct, which executes much
like an if () statement, but the decision is based on the
current time, and the time spec provided in the ifTime. See the
example above. (Note: all the other time-dependent Applications
@@ -1200,7 +1198,7 @@ Differences with the original version of AEL
NoOp() added, to make sure the label exists in the extension on
Asterisk. (0.6)
\item (0.9) the semicolon is no longer required after a closing brace!
(i.e. "];" ===> "\}". You can have them there if you like, but
(i.e. "];" ===$>$ "\}". You can have them there if you like, but
they are not necessary. Someday they may be rejected as a syntax
error, maybe.
\item (0.9) the // comments are not recognized and removed in the
@@ -1208,11 +1206,11 @@ Differences with the original version of AEL
arguments. You may have to move a comment if you get errors in
existing files.
\item (0.10) the random statement has been added. Syntax: random (
<expr> ) <lucky-statement> [ else <unlucky-statement> ]. The
probability of the lucky-statement getting executed is <expr>,
$<$expr$>$ ) $<$lucky-statement$>$ [ else $<$unlucky-statement$>$ ]. The
probability of the lucky-statement getting executed is $<$expr$>$,
which should evaluate to an integer between 0 and 100. If the
<lucky-statement> isn't so lucky this time around, then the
<unlucky-statement> gets executed, if it is present.
$<$lucky-statement$>$ isn't so lucky this time around, then the
$<$unlucky-statement$>$ gets executed, if it is present.
\end{enumerate}

24
doc/tex/asterisk.tex
View File

@@ -61,10 +61,10 @@ reference purposes.
\chapter{Channel Variables}
\input{channelvariables.tex}
\chapter{AEL, Asterisk Extension Language}
\chapter{AEL: Asterisk Extension Language}
\input{ael.tex}
\chapter{SLA (Shared Line Appearances)}
\chapter{SLA: Shared Line Appearances}
\input{sla.tex}
\chapter{Channel Drivers}
@@ -96,6 +96,7 @@ reference purposes.
\chapter{Voicemail}
\section{ODBC Storage}
\label{odbcstorage}
\input{odbcstorage.tex}
\section{IMAP Storage}
\input{imapstorage.tex}
@@ -112,18 +113,19 @@ reference purposes.
%
%Misc
%----
%asterisk-mib.txt SNMP mib for Asterisk (net-snmp)
%digium-mib.txt SNMP mib for Asterisk (net-snmp)
%asterisk-mib.txt SNMP mib for Asterisk (net-snmp)
%digium-mib.txt SNMP mib for Asterisk (net-snmp)
%
%For developers
%--------------
%See http://www.asterisk.org/developers for more information
%
%backtrace.txt How to produce a backtrace when Asterisk crashes
%CODING-GUIDELINES Guidelines for developers
%externalivr.txt Documentation of the protocol used in externalivr()
%modules.txt How Asterisk modules work
%datastores.txt About channel data stores
%speechrec.txt The Generic Speech Recognition API
%backtrace.txt How to produce a backtrace when Asterisk crashes
%callfiles.txt Asterisk callfiles using instruction
%CODING-GUIDELINES Guidelines for developers
%externalivr.txt Documentation of the protocol used in externalivr()
%modules.txt How Asterisk modules work
%datastores.txt About channel data stores
%speechrec.txt The Generic Speech Recognition API
\enddocument
\end{document}

91
doc/tex/billing.tex
View File

@@ -1,48 +1,47 @@
\section{Applications}
\begin{itemize}
\item SetAccount - Set account code for billing
\item SetAMAFlags - Sets AMA flags
\item NoCDR - Make sure no CDR is saved for a specific call
\item ResetCDR - Reset CDR
\item ForkCDR - Save current CDR and start a new CDR for this call
\item Authenticate - Authenticates and sets the account code
\item SetCDRUserField - Set CDR user field
\item AppendCDRUserField - Append data to CDR User field
\item SetAccount - Set account code for billing
\item SetAMAFlags - Sets AMA flags
\item NoCDR - Make sure no CDR is saved for a specific call
\item ResetCDR - Reset CDR
\item ForkCDR - Save current CDR and start a new CDR for this call
\item Authenticate - Authenticates and sets the account code
\item SetCDRUserField - Set CDR user field
\item AppendCDRUserField - Append data to CDR User field
\end{itemize}
For more information, use the "core show application <application>" command.
You can set default account codes and AMA flags for devices in
channel configuration files, like sip.conf, iax.conf etc.
\section{Fields of the CDR in Asterisk}
\begin{itemize}
\item accountcode: What account number to use, (string, 20 characters)
\item src: Caller*ID number (string, 80 characters)
\item dst: Destination extension (string, 80 characters)
\item dcontext: Destination context (string, 80 characters)
\item clid: Caller*ID with text (80 characters)
\item channel: Channel used (80 characters)
\item dstchannel: Destination channel if appropriate (80 characters)
\item lastapp: Last application if appropriate (80 characters)
\item lastdata: Last application data (arguments) (80 characters)
\item start: Start of call (date/time)
\item answer: Answer of call (date/time)
\item end: End of call (date/time)
\item duration: Total time in system, in seconds (integer), from dial to hangup
\item billsec: Total time call is up, in seconds (integer), from answer to hangup
\item disposition: What happened to the call: ANSWERED, NO ANSWER, BUSY
\item amaflags: What flags to use: DOCUMENTATION, BILL, IGNORE etc,
specified on a per channel basis like accountcode.
\item user field: A user-defined field, maximum 255 characters
\item accountcode: What account number to use, (string, 20 characters)
\item src: Caller*ID number (string, 80 characters)
\item dst: Destination extension (string, 80 characters)
\item dcontext: Destination context (string, 80 characters)
\item clid: Caller*ID with text (80 characters)
\item channel: Channel used (80 characters)
\item dstchannel: Destination channel if appropriate (80 characters)
\item lastapp: Last application if appropriate (80 characters)
\item lastdata: Last application data (arguments) (80 characters)
\item start: Start of call (date/time)
\item answer: Answer of call (date/time)
\item end: End of call (date/time)
\item duration: Total time in system, in seconds (integer), from dial to hangup
\item billsec: Total time call is up, in seconds (integer), from answer to hangup
\item disposition: What happened to the call: ANSWERED, NO ANSWER, BUSY
\item amaflags: What flags to use: DOCUMENTATION, BILL, IGNORE etc,
specified on a per channel basis like accountcode.
\item user field: A user-defined field, maximum 255 characters
\end{itemize}
In some cases, uniqueid is appended:
\begin{itemize}
\item uniqueid: Unique Channel Identifier (32 characters)
\item uniqueid: Unique Channel Identifier (32 characters)
This needs to be enabled in the source code at compile time
\end{itemize}
@@ -61,24 +60,24 @@ can be accessed just like channel variables. The following builtin variables
are available.
\begin{verbatim}
${CDR(clid)} Caller ID
${CDR(src)} Source
${CDR(dst)} Destination
${CDR(dcontext)} Destination context
${CDR(channel)} Channel name
${CDR(dstchannel)} Destination channel
${CDR(lastapp)} Last app executed
${CDR(lastdata)} Last app's arguments
${CDR(start)} Time the call started.
${CDR(answer)} Time the call was answered.
${CDR(end)} Time the call ended.
${CDR(duration)} Duration of the call.
${CDR(billsec)} Duration of the call once it was answered.
${CDR(disposition)} ANSWERED, NO ANSWER, BUSY
${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc
${CDR(accountcode)} The channel's account code.
${CDR(uniqueid)} The channel's unique id.
${CDR(userfield)} The channels uses specified field.
${CDR(clid)} Caller ID
${CDR(src)} Source
${CDR(dst)} Destination
${CDR(dcontext)} Destination context
${CDR(channel)} Channel name
${CDR(dstchannel)} Destination channel
${CDR(lastapp)} Last app executed
${CDR(lastdata)} Last app's arguments
${CDR(start)} Time the call started.
${CDR(answer)} Time the call was answered.
${CDR(end)} Time the call ended.
${CDR(duration)} Duration of the call.
${CDR(billsec)} Duration of the call once it was answered.
${CDR(disposition)} ANSWERED, NO ANSWER, BUSY
${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc
${CDR(accountcode)} The channel's account code.
${CDR(uniqueid)} The channel's unique id.
${CDR(userfield)} The channels uses specified field.
\end{verbatim}
In addition, you can set your own extra variables by using Set(CDR(name)=value).

466
doc/tex/channelvariables.tex
View File

@@ -22,8 +22,8 @@ will be passed down to the Background command, in this example.
Also, characters special to variable substitution, expression evaluation, etc
(see below), can be quoted. For example, to literally use a \$ on the
string "\$1231", quote it with a preceding \\. Special characters that must
be quoted to be used, are [ ] \$ " \\. (to write \\ itself, use \\).
string "\$1231", quote it with a preceding \textbackslash. Special characters that must
be quoted to be used, are [ ] \$ " \textbackslash. (to write \textbackslash itself, use \textbackslash).
These Double quotes and escapes are evaluated at the level of the
asterisk config file parser.
@@ -35,7 +35,7 @@ Double quotes can also be used inside expressions, as discussed below.
Parameter strings can include variables. Variable names are arbitrary strings.
They are stored in the respective channel structure.
To set a variable to a particular value, do :
To set a variable to a particular value, do:
\begin{verbatim}
exten => 1,2,Set(varname=value)
\end{verbatim}
@@ -52,8 +52,8 @@ requires a variable), just write the name. To refer to the variable's value,
enclose it inside \${}. For example, Set takes as the first argument
(before the =) a variable name, so:
\begin{verbatim}
exten => 1,2,Set(koko=lala)
exten => 1,3,Set(${koko}=blabla)
exten => 1,2,Set(koko=lala)
exten => 1,3,Set(${koko}=blabla)
\end{verbatim}
stores to the variable "koko" the value "lala" and to variable "lala" the
value "blabla".
@@ -91,14 +91,14 @@ NoOp(${__FOO}) is identical to NoOp(${FOO})
The format for selecting characters from a variable can be expressed as:
\begin{verbatim}
${variable_name[:offset[:length]]}
${variable_name[:offset[:length]]}
\end{verbatim}
If you want to select the first N characters from the string assigned
to a variable, simply append a colon and the number of characters to
skip from the beginning of the string to the variable name.
\begin{verbatim}
;Remove the first character of extension, save in "number" variable
exten => _9X.,1,Set(number=${EXTEN:1})
; Remove the first character of extension, save in "number" variable
exten => _9X.,1,Set(number=${EXTEN:1})
\end{verbatim}
Assuming we've dialed 918005551234, the value saved to the 'number' variable
would be 18005551234. This is useful in situations when we require users to
@@ -110,16 +110,16 @@ of the string and then selects everything after the new position. The following
example will save the numbers 1234 to the 'number' variable, still assuming
we've dialed 918005551234.
\begin{verbatim}
;Remove everything before the last four digits of the dialed string
exten => _9X.,1,Set(number=${EXTEN:-4})
; Remove everything before the last four digits of the dialed string
exten => _9X.,1,Set(number=${EXTEN:-4})
\end{verbatim}
We can also limit the number of characters from our offset position that we
wish to use. This is done by appending a second colon and length value to the
variable name. The following example will save the numbers 555 to the 'number'
variable.
\begin{verbatim}
;Only save the middle numbers 555 from the string 918005551234
exten => _9X.,1,Set(number=${EXTEN:5:3})
; Only save the middle numbers 555 from the string 918005551234
exten => _9X.,1,Set(number=${EXTEN:5:3})
\end{verbatim}
The length value can also be used in conjunction with a negative offset. This
may be useful if the length of the string is unknown, but the trailing digits
@@ -127,14 +127,14 @@ are. The following example will save the numbers 555 to the 'number' variable,
even if the string starts with more characters than expected (unlike the
previous example).
\begin{verbatim}
;Save the numbers 555 to the 'number' variable
exten => _9X.,1,Set(number=${EXTEN:-7:3})
; Save the numbers 555 to the 'number' variable
exten => _9X.,1,Set(number=${EXTEN:-7:3})
\end{verbatim}
If a negative length value is entered, Asterisk will remove that many characters
from the end of the string.
\begin{verbatim}
;Set pin to everything but the trailing #.
exten => _XXXX#,1,Set(pin=${EXTEN:0:-1})
; Set pin to everything but the trailing #.
exten => _XXXX#,1,Set(pin=${EXTEN:0:-1})
\end{verbatim}
\section{Expressions}
@@ -258,15 +258,15 @@ with equal precedence are grouped within { } symbols.
expr1 ? expr2 :: expr3
Traditional Conditional operator. If expr1 is a number
that evaluates to 0 (false), expr3 is result of the this
expression evaluation. Otherwise, expr2 is the result.
If expr1 is a string, and evaluates to an empty string,
or the two characters (""), then expr3 is the
result. Otherwise, expr2 is the result. In Asterisk, all
3 exprs will be "evaluated"; if expr1 is "true", expr2
will be the result of the "evaluation" of this
expression. expr3 will be the result otherwise. This
operator has the lowest precedence.
that evaluates to 0 (false), expr3 is result of the this
expression evaluation. Otherwise, expr2 is the result.
If expr1 is a string, and evaluates to an empty string,
or the two characters (""), then expr3 is the
result. Otherwise, expr2 is the result. In Asterisk, all
3 exprs will be "evaluated"; if expr1 is "true", expr2
will be the result of the "evaluation" of this
expression. expr3 will be the result otherwise. This
operator has the lowest precedence.
\end{verbatim}
Parentheses are used for grouping in the usual manner.
@@ -274,7 +274,7 @@ Parentheses are used for grouping in the usual manner.
Operator precedence is applied as one would expect in any of the C
or C derived languages.
subsection{Floating Point Numbers}
\subsection{Floating Point Numbers}
In 1.6 and above, we shifted the \$\[...\] expressions to be calculated
via floating point numbers instead of integers. We use 'long double' numbers
@@ -290,13 +290,13 @@ on integer behavior. If you were counting on 1/4 evaluating to 0, you need to no
TRUNC(1/4). For a list of all the truncation/rounding capabilities, see the next section.
subsection{Functions}
\subsection{Functions}
In 1.6 and above, we upgraded the $[] expressions to handle floating point numbers.
In 1.6 and above, we upgraded the \$[] expressions to handle floating point numbers.
Because of this, folks counting on integer behavior would be disrupted. To make
the same results possible, some rounding and integer truncation functions have been
added to the core of the Expr2 parser. Indeed, dialplan functions can be called from
$[..] expressions without the ${...} operators. The only trouble might be in the fact that
\$[..] expressions without the \$\{...\} operators. The only trouble might be in the fact that
the arguments to these functions must be specified with a comma. If you try to call
the MATH function, for example, and try to say 3 + MATH(7*8), the expression parser will
evaluate 7*8 for you into 56, and the MATH function will most likely complain that its
@@ -312,7 +312,7 @@ are available by simply calling them (read-only). In other words, you don't need
surround function calls in \$\[...\] expressions with \$\{...\}. Don't jump to conclusions,
though! -- you still need to wrap variable names in curly braces!
\begin{Enumerate}
\begin{enumerate}
\item COS(x) x is in radians. Results vary from -1 to 1.
\item SIN(x) x is in radians. Results vary from -1 to 1.
\item TAN(x) x is in radians.
@@ -334,94 +334,94 @@ If this quotient is 1/2, it is rounded to the nearest even number.
\item LOG(x) returns the natural logarithm of x.
\item LOG2(x) returns the base 2 log of x.
\item LOG10(x) returns the base 10 log of x.
\end{Enumerate}
\end{enumerate}
subsection{Examples}
\subsection{Examples}
\begin{verbatim}
"One Thousand Five Hundred" =~ "(T[^ ]+)"
returns: Thousand
returns: Thousand
"One Thousand Five Hundred" =~ "T[^ ]+"
returns: 8
returns: 8
"One Thousand Five Hundred" : "T[^ ]+"
returns: 0
returns: 0
"8015551212" : "(...)"
returns: 801
returns: 801
"3075551212":"...(...)"
returns: 555
returns: 555
! "One Thousand Five Hundred" =~ "T[^ ]+"
returns: 0 (because it applies to the string, which is non-null,
returns: 0 (because it applies to the string, which is non-null,
which it turns to "0", and then looks for the pattern
in the "0", and doesn't find it)
!( "One Thousand Five Hundred" : "T[^ ]+" )
returns: 1 (because the string doesn't start with a word starting
returns: 1 (because the string doesn't start with a word starting
with T, so the match evals to 0, and the ! operator
inverts it to 1 ).
2 + 8 / 2
returns 6. (because of operator precedence; the division is done first, then the addition).
returns 6. (because of operator precedence; the division is done first, then the addition).
2+8/2
returns 6. Spaces aren't necessary.
returns 6. Spaces aren't necessary.
(2+8)/2
returns 5, of course.
returns 5, of course.
(3+8)/2
returns 5.5 now.
returns 5.5 now.
TRUNC((3+8)/2)
returns 5.
returns 5.
FLOOR(2.5)
returns 2
returns 2
FLOOR(-2.5)
returns -3
returns -3
CEIL(2.5)
returns 3.
returns 3.
CEIL(-2.5)
returns -2.
returns -2.
ROUND(2.5)
returns 3.
returns 3.
ROUND(3.5)
returns 4.
returns 4.
ROUND(-2.5)
returns -3
returns -3
RINT(2.5)
returns 2.
returns 2.
RINT(3.5)
returns 4.
returns 4.
RINT(-2.5)
returns -2.
returns -2.
RINT(-3.5)
returns -4.
returns -4.
TRUNC(2.5)
returns 2.
returns 2.
TRUNC(3.5)
returns 3.
returns 3.
TRUNC(-3.5)
returns -3.
returns -3.
\begin{verbatim}
\end{verbatim}
Of course, all of the above examples use constants, but would work the
same if any of the numeric or string constants were replaced with a
@@ -441,7 +441,7 @@ case.
There is one conditional application - the conditional goto :
\begin{verbatim}
exten => 1,2,gotoif(condition?label1:label2)
exten => 1,2,GotoIf(condition?label1:label2)
\end{verbatim}
If condition is true go to label1, else go to label2. Labels are interpreted
@@ -453,7 +453,7 @@ This is designed to be used together with the expression syntax described
above, eg :
\begin{verbatim}
exten => 1,2,gotoif($[${CALLERID} = 123456]?2|1:3|1)
exten => 1,2,GotoIf($[${CALLERID} = 123456]?2|1:3|1)
\end{verbatim}
Example of use :
@@ -494,9 +494,9 @@ marked with the "\^" character.
\subsection{NULL Strings}
Testing to see if a string is null can be done in one of two different ways:
\begin{verbatim}
exten => _XX.,1,GotoIf($["${calledid}" != ""]?3)
exten => _XX.,1,GotoIf($["${calledid}" != ""]?3)
exten => _XX.,1,GotoIf($[foo${calledid} != foo]?3)
exten => _XX.,1,GotoIf($[foo${calledid} != foo]?3)
\end{verbatim}
The second example above is the way suggested by the WIKI. It will
@@ -574,7 +574,7 @@ of possible concern with "legacy" extension.conf files:
quotes, it was no reason for concern. It is now!
\item LE, GE, NE operators removed. The code supported these operators,
but they were not documented. The symbolic operators, <=, >=, and !=
but they were not documented. The symbolic operators, $<$=, $>$=, and !=
should be used instead.
\item Added the unary '-' operator. So you can 3+ -4 and get -1.
@@ -670,43 +670,45 @@ only read in the dialplan. Writes to such variables are silently
ignored.
\begin{verbatim}
${ACCOUNTCODE} * Account code (if specified) (Deprecated; use ${CDR(accountcode)})
${BLINDTRANSFER} The name of the channel on the other side of a blind transfer
${BRIDGEPEER} Bridged peer
${CALLERANI} * Caller ANI (PRI channels) (Deprecated; use ${CALLERID(ani)})
${CALLERID} * Caller ID (Deprecated; use ${CALLERID(all)})
${CALLERIDNAME} * Caller ID Name only (Deprecated; use ${CALLERID(name)})
${CALLERIDNUM} * Caller ID Number only (Deprecated; use ${CALLERID(num)})
${CALLINGANI2} * Caller ANI2 (PRI channels)
${CALLINGPRES} * Caller ID presentation for incoming calls (PRI channels)
${CALLINGTNS} * Transit Network Selector (PRI channels)
${CALLINGTON} * Caller Type of Number (PRI channels)
${CHANNEL} * Current channel name
${CONTEXT} * Current context
${DATETIME} * Current date time in the format: DDMMYYYY-HH:MM:SS (Deprecated; use ${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)})
${DB_RESULT} Result value of DB_EXISTS() dial plan function
${DNID} * Dialed Number Identifier (Deprecated; use ${CALLERID(dnid)})
${EPOCH} * Current unix style epoch
${EXTEN} * Current extension
${ENV(VAR)} Environmental variable VAR
${GOTO_ON_BLINDXFR} Transfer to the specified context/extension/priority
after a blind transfer (use ^ characters in place of
| to separate context/extension/priority when setting
this variable from the dialplan)
${HANGUPCAUSE} * Asterisk cause of hangup (inbound/outbound)
${HINT} * Channel hints for this extension
${HINTNAME} * Suggested Caller*ID name for this extension
${INVALID_EXTEN} The invalid called extension (used in the "i" extension)
${LANGUAGE} * Current language (Deprecated; use ${LANGUAGE()})
${LEN(VAR)} * String length of VAR (integer)
${PRIORITY} * Current priority in the dialplan
${PRIREDIRECTREASON} Reason for redirect on PRI, if a call was directed
${RDNIS} * Redirected Dial Number ID Service (Deprecated; use ${CALLERID(rdnis)})
${TIMESTAMP} * Current date time in the format: YYYYMMDD-HHMMSS (Deprecated; use ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
${TRANSFER_CONTEXT} Context for transferred calls
${FORWARD_CONTEXT} Context for forwarded calls
${UNIQUEID} * Current call unique identifier
${SYSTEMNAME} * value of the systemname option of asterisk.conf
${CDR(accountcode)} * Account code (if specified)
${BLINDTRANSFER} The name of the channel on the other side of a blind transfer
${BRIDGEPEER} Bridged peer
${CALLERID(ani)} * Caller ANI (PRI channels)
${CALLERID(all)} * Caller ID
${CALLERID(dnid)} * Dialed Number Identifier
${CALLERID(name)} * Caller ID Name only
${CALLERID(num)} * Caller ID Number only
${CALLERID(rdnis)} * Redirected Dial Number ID Service
${CALLINGANI2} * Caller ANI2 (PRI channels)
${CALLINGPRES} * Caller ID presentation for incoming calls (PRI channels)
${CALLINGTNS} * Transit Network Selector (PRI channels)
${CALLINGTON} * Caller Type of Number (PRI channels)
${CHANNEL} * Current channel name
${CONTEXT} * Current context
${DATETIME} * Current date time in the format: DDMMYYYY-HH:MM:SS
(Deprecated; use ${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)})
${DB_RESULT} Result value of DB_EXISTS() dial plan function
${EPOCH} * Current unix style epoch
${EXTEN} * Current extension
${ENV(VAR)} Environmental variable VAR
${GOTO_ON_BLINDXFR} Transfer to the specified context/extension/priority
after a blind transfer (use ^ characters in place of
| to separate context/extension/priority when setting
this variable from the dialplan)
${HANGUPCAUSE} * Asterisk cause of hangup (inbound/outbound)
${HINT} * Channel hints for this extension
${HINTNAME} * Suggested Caller*ID name for this extension
${INVALID_EXTEN} The invalid called extension (used in the "i" extension)
${LANGUAGE} * Current language (Deprecated; use ${LANGUAGE()})
${LEN(VAR)} * String length of VAR (integer)
${PRIORITY} * Current priority in the dialplan
${PRIREDIRECTREASON} Reason for redirect on PRI, if a call was directed
${TIMESTAMP} * Current date time in the format: YYYYMMDD-HHMMSS
(Deprecated; use ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
${TRANSFER_CONTEXT} Context for transferred calls
${FORWARD_CONTEXT} Context for forwarded calls
${UNIQUEID} * Current call unique identifier
${SYSTEMNAME} * value of the systemname option of asterisk.conf
\end{verbatim}
\subsection{Application return values}
@@ -715,189 +717,189 @@ In Asterisk 1.2, many applications return the result in a variable
instead of, as in Asterisk 1.0, changing the dial plan priority (+101).
For the various status values, see each application's help text.
\begin{verbatim}
${AGISTATUS} * agi()
${AQMSTATUS} * addqueuemember()
${AVAILSTATUS} * chanisavail()
${CHECKGROUPSTATUS} * checkgroup()
${CHECKMD5STATUS} * checkmd5()
${CPLAYBACKSTATUS} * controlplayback()
${DIALSTATUS} * dial()
${DBGETSTATUS} * dbget()
${ENUMSTATUS} * enumlookup()
${HASVMSTATUS} * hasnewvoicemail()
${LOOKUPBLSTATUS} * lookupblacklist()
${OSPAUTHSTATUS} * ospauth()
${OSPLOOKUPSTATUS} * osplookup()
${OSPNEXTSTATUS} * ospnext()
${OSPFINISHSTATUS} * ospfinish()
${PARKEDAT} * parkandannounce()
${PLAYBACKSTATUS} * playback()
${PQMSTATUS} * pausequeuemember()
${PRIVACYMGRSTATUS} * privacymanager()
${QUEUESTATUS} * queue()
${RQMSTATUS} * removequeuemember()
${SENDIMAGESTATUS} * sendimage()
${SENDTEXTSTATUS} * sendtext()
${SENDURLSTATUS} * sendurl()
${SYSTEMSTATUS} * system()
${TRANSFERSTATUS} * transfer()
${TXTCIDNAMESTATUS} * txtcidname()
${UPQMSTATUS} * unpausequeuemember()
${VMSTATUS} * voicmail()
${VMBOXEXISTSSTATUS} * vmboxexists()
${WAITSTATUS} * waitforsilence()
${AGISTATUS} * agi()
${AQMSTATUS} * addqueuemember()
${AVAILSTATUS} * chanisavail()
${CHECKGROUPSTATUS} * checkgroup()
${CHECKMD5STATUS} * checkmd5()
${CPLAYBACKSTATUS} * controlplayback()
${DIALSTATUS} * dial()
${DBGETSTATUS} * dbget()
${ENUMSTATUS} * enumlookup()
${HASVMSTATUS} * hasnewvoicemail()
${LOOKUPBLSTATUS} * lookupblacklist()
${OSPAUTHSTATUS} * ospauth()
${OSPLOOKUPSTATUS} * osplookup()
${OSPNEXTSTATUS} * ospnext()
${OSPFINISHSTATUS} * ospfinish()
${PARKEDAT} * parkandannounce()
${PLAYBACKSTATUS} * playback()
${PQMSTATUS} * pausequeuemember()
${PRIVACYMGRSTATUS} * privacymanager()
${QUEUESTATUS} * queue()
${RQMSTATUS} * removequeuemember()
${SENDIMAGESTATUS} * sendimage()
${SENDTEXTSTATUS} * sendtext()
${SENDURLSTATUS} * sendurl()
${SYSTEMSTATUS} * system()
${TRANSFERSTATUS} * transfer()
${TXTCIDNAMESTATUS} * txtcidname()
${UPQMSTATUS} * unpausequeuemember()
${VMSTATUS} * voicmail()
${VMBOXEXISTSSTATUS} * vmboxexists()
${WAITSTATUS} * waitforsilence()
\end{verbatim}
\subsection{Various application variables}
\begin{verbatim}
${CURL} * Resulting page content for curl()
${ENUM} * Result of application EnumLookup
${EXITCONTEXT} Context to exit to in IVR menu (app background())
or in the RetryDial() application
${MONITOR} * Set to "TRUE" if the channel is/has been monitored (app monitor())
${MONITOR_EXEC} Application to execute after monitoring a call
${MONITOR_EXEC_ARGS} Arguments to application
${MONITOR_FILENAME} File for monitoring (recording) calls in queue
${QUEUE_PRIO} Queue priority
${QUEUE_MAX_PENALTY} Maximum member penalty allowed to answer caller
${QUEUESTATUS} Status of the call, one of:
(TIMEOUT | FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL)
${RECORDED_FILE} * Recorded file in record()
${TALK_DETECTED} * Result from talkdetect()
${TOUCH_MONITOR} The filename base to use with Touch Monitor (auto record)
${TOUCH_MONITOR_FORMAT} The audio format to use with Touch Monitor (auto record)
${CURL} * Resulting page content for curl()
${ENUM} * Result of application EnumLookup
${EXITCONTEXT} Context to exit to in IVR menu (app background())
or in the RetryDial() application
${MONITOR} * Set to "TRUE" if the channel is/has been monitored (app monitor())
${MONITOR_EXEC} Application to execute after monitoring a call
${MONITOR_EXEC_ARGS} Arguments to application
${MONITOR_FILENAME} File for monitoring (recording) calls in queue
${QUEUE_PRIO} Queue priority
${QUEUE_MAX_PENALTY} Maximum member penalty allowed to answer caller
${QUEUESTATUS} Status of the call, one of:
(TIMEOUT | FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL)
${RECORDED_FILE} * Recorded file in record()
${TALK_DETECTED} * Result from talkdetect()
${TOUCH_MONITOR} The filename base to use with Touch Monitor (auto record)
${TOUCH_MONITOR_FORMAT} The audio format to use with Touch Monitor (auto record)
${TOUCH_MONITOR_OUTPUT} * Recorded file from Touch Monitor (auto record)
${TXTCIDNAME} * Result of application TXTCIDName
${VPB_GETDTMF} chan_vpb
${TXTCIDNAME} * Result of application TXTCIDName
${VPB_GETDTMF} chan_vpb
\end{verbatim}
\subsection{The MeetMe Conference Bridge}
\begin{verbatim}
${MEETME_RECORDINGFILE} Name of file for recording a conference with
the "r" option
${MEETME_RECORDINGFORMAT} Format of file to be recorded
${MEETME_EXIT_CONTEXT} Context for exit out of meetme meeting
${MEETME_AGI_BACKGROUND} AGI script for Meetme (zap only)
${MEETMESECS} * Number of seconds a user participated in a MeetMe conference
${MEETME_RECORDINGFILE} Name of file for recording a conference with
the "r" option
${MEETME_RECORDINGFORMAT} Format of file to be recorded
${MEETME_EXIT_CONTEXT} Context for exit out of meetme meeting
${MEETME_AGI_BACKGROUND} AGI script for Meetme (zap only)
${MEETMESECS} * Number of seconds a user participated in a MeetMe conference
\end{verbatim}
\subsection{The VoiceMail() application}
\begin{verbatim}
${VM_CATEGORY} Sets voicemail category
${VM_NAME} * Full name in voicemail
${VM_DUR} * Voicemail duration
${VM_MSGNUM} * Number of voicemail message in mailbox
${VM_CALLERID} * Voicemail Caller ID (Person leaving vm)
${VM_CIDNAME} * Voicemail Caller ID Name
${VM_CIDNUM} * Voicemail Caller ID Number
${VM_DATE} * Voicemail Date
${VM_MESSAGEFILE} * Path to message left by caller
${VM_CATEGORY} Sets voicemail category
${VM_NAME} * Full name in voicemail
${VM_DUR} * Voicemail duration
${VM_MSGNUM} * Number of voicemail message in mailbox
${VM_CALLERID} * Voicemail Caller ID (Person leaving vm)
${VM_CIDNAME} * Voicemail Caller ID Name
${VM_CIDNUM} * Voicemail Caller ID Number
${VM_DATE} * Voicemail Date
${VM_MESSAGEFILE} * Path to message left by caller
\end{verbatim}
\subsection{The VMAuthenticate() application}
\begin{verbatim}
${AUTH_MAILBOX} * Authenticated mailbox
${AUTH_CONTEXT} * Authenticated mailbox context
${AUTH_MAILBOX} * Authenticated mailbox
${AUTH_CONTEXT} * Authenticated mailbox context
\end{verbatim}
\subsection{DUNDiLookup()}
\begin{verbatim}
${DUNDTECH} * The Technology of the result from a call to DUNDiLookup()
${DUNDDEST} * The Destination of the result from a call to DUNDiLookup()
${DUNDTECH} * The Technology of the result from a call to DUNDiLookup()
${DUNDDEST} * The Destination of the result from a call to DUNDiLookup()
\end{verbatim}
\subsection{chan\_zap}
\begin{verbatim}
${ANI2} * The ANI2 Code provided by the network on the incoming call.
(ie, Code 29 identifies call as a Prison/Inmate Call)
${CALLTYPE} * Type of call (Speech, Digital, etc)
${CALLEDTON} * Type of number for incoming PRI extension
i.e. 0=unknown, 1=international, 2=domestic, 3=net_specific,
4=subscriber, 6=abbreviated, 7=reserved
${CALLINGSUBADDR} * Called PRI Subaddress
${FAXEXTEN} * The extension called before being redirected to "fax"
${PRIREDIRECTREASON} * Reason for redirect, if a call was directed
${SMDI_VM_TYPE} * When an call is received with an SMDI message, the 'type'
of message 'b' or 'u'
${ANI2} * The ANI2 Code provided by the network on the incoming call.
(ie, Code 29 identifies call as a Prison/Inmate Call)
${CALLTYPE} * Type of call (Speech, Digital, etc)
${CALLEDTON} * Type of number for incoming PRI extension
i.e. 0=unknown, 1=international, 2=domestic, 3=net_specific,
4=subscriber, 6=abbreviated, 7=reserved
${CALLINGSUBADDR} * Called PRI Subaddress
${FAXEXTEN} * The extension called before being redirected to "fax"
${PRIREDIRECTREASON} * Reason for redirect, if a call was directed
${SMDI_VM_TYPE} * When an call is received with an SMDI message, the 'type'
of message 'b' or 'u'
\end{verbatim}
\subsection{chan\_sip}
\begin{verbatim}
${SIPCALLID} * SIP Call-ID: header verbatim (for logging or CDR matching)
${SIPDOMAIN} * SIP destination domain of an inbound call (if appropriate)
${SIPUSERAGENT} * SIP user agent (deprecated)
${SIPURI} * SIP uri
${SIP_CODEC} Set the SIP codec for a call
${SIP_URI_OPTIONS} * additional options to add to the URI for an outgoing call
${RTPAUDIOQOS} RTCP QoS report for the audio of this call
${RTPVIDEOQOS} RTCP QoS report for the video of this call
${SIPCALLID} * SIP Call-ID: header verbatim (for logging or CDR matching)
${SIPDOMAIN} * SIP destination domain of an inbound call (if appropriate)
${SIPUSERAGENT} * SIP user agent (deprecated)
${SIPURI} * SIP uri
${SIP_CODEC} Set the SIP codec for a call
${SIP_URI_OPTIONS} * additional options to add to the URI for an outgoing call
${RTPAUDIOQOS} RTCP QoS report for the audio of this call
${RTPVIDEOQOS} RTCP QoS report for the video of this call
\end{verbatim}
\subsection{chan\_agent}
\begin{verbatim}
${AGENTMAXLOGINTRIES} Set the maximum number of failed logins
${AGENTUPDATECDR} Whether to update the CDR record with Agent channel data
${AGENTGOODBYE} Sound file to use for "Good Bye" when agent logs out
${AGENTACKCALL} Whether the agent should acknowledge the incoming call
${AGENTAUTOLOGOFF} Auto logging off for an agent
${AGENTWRAPUPTIME} Setting the time for wrapup between incoming calls
${AGENTNUMBER} * Agent number (username) set at login
${AGENTSTATUS} * Status of login ( fail | on | off )
${AGENTEXTEN} * Extension for logged in agent
${AGENTMAXLOGINTRIES} Set the maximum number of failed logins
${AGENTUPDATECDR} Whether to update the CDR record with Agent channel data
${AGENTGOODBYE} Sound file to use for "Good Bye" when agent logs out
${AGENTACKCALL} Whether the agent should acknowledge the incoming call
${AGENTAUTOLOGOFF} Auto logging off for an agent
${AGENTWRAPUPTIME} Setting the time for wrapup between incoming calls
${AGENTNUMBER} * Agent number (username) set at login
${AGENTSTATUS} * Status of login ( fail | on | off )
${AGENTEXTEN} * Extension for logged in agent
\end{verbatim}
\subsection{The Dial() application}
\begin{verbatim}
${DIALEDPEERNAME} * Dialed peer name
${DIALEDPEERNUMBER} * Dialed peer number
${DIALEDTIME} * Time for the call (seconds)
${ANSWEREDTIME} * Time from dial to answer (seconds)
${DIALSTATUS} * Status of the call, one of:
(CHANUNAVAIL | CONGESTION | BUSY | NOANSWER
| ANSWER | CANCEL | DONTCALL | TORTURE)
${DYNAMIC_FEATURES} * The list of features (from the [applicationmap] section of
features.conf) to activate during the call, with feature
names separated by '#' characters
${LIMIT_PLAYAUDIO_CALLER} Soundfile for call limits
${LIMIT_PLAYAUDIO_CALLEE} Soundfile for call limits
${LIMIT_WARNING_FILE} Soundfile for call limits
${LIMIT_TIMEOUT_FILE} Soundfile for call limits
${LIMIT_CONNECT_FILE} Soundfile for call limits
${OUTBOUND_GROUP} Default groups for peer channels (as in SetGroup)
* See "show application dial" for more information
${DIALEDPEERNAME} * Dialed peer name
${DIALEDPEERNUMBER} * Dialed peer number
${DIALEDTIME} * Time for the call (seconds)
${ANSWEREDTIME} * Time from dial to answer (seconds)
${DIALSTATUS} * Status of the call, one of:
(CHANUNAVAIL | CONGESTION | BUSY | NOANSWER
| ANSWER | CANCEL | DONTCALL | TORTURE)
${DYNAMIC_FEATURES} * The list of features (from the [applicationmap] section of
features.conf) to activate during the call, with feature
names separated by '#' characters
${LIMIT_PLAYAUDIO_CALLER} Soundfile for call limits
${LIMIT_PLAYAUDIO_CALLEE} Soundfile for call limits
${LIMIT_WARNING_FILE} Soundfile for call limits
${LIMIT_TIMEOUT_FILE} Soundfile for call limits
${LIMIT_CONNECT_FILE} Soundfile for call limits
${OUTBOUND_GROUP} Default groups for peer channels (as in SetGroup)
* See "show application dial" for more information
\end{verbatim}
\subsection{The chanisavail() application}
\begin{verbatim}
${AVAILCHAN} * the name of the available channel if one was found
${AVAILORIGCHAN} * the canonical channel name that was used to create the channel
${AVAILSTATUS} * Status of requested channel
${AVAILCHAN} * the name of the available channel if one was found
${AVAILORIGCHAN} * the canonical channel name that was used to create the channel
${AVAILSTATUS} * Status of requested channel
\end{verbatim}
\subsection{Dialplan Macros}
\begin{verbatim}
${MACRO_EXTEN} * The calling extensions
${MACRO_CONTEXT} * The calling context
${MACRO_PRIORITY} * The calling priority
${MACRO_OFFSET} Offset to add to priority at return from macro
${MACRO_EXTEN} * The calling extensions
${MACRO_CONTEXT} * The calling context
${MACRO_PRIORITY} * The calling priority
${MACRO_OFFSET} Offset to add to priority at return from macro
\end{verbatim}
\subsection{The ChanSpy() application}
\begin{verbatim}
${SPYGROUP} * A ':' (colon) separated list of group names.
(To be set on spied on channel and matched against the g(grp) option)
${SPYGROUP} * A ':' (colon) separated list of group names.
(To be set on spied on channel and matched against the g(grp) option)
\end{verbatim}
\subsection{OSP}
\begin{verbatim}
${OSPINHANDLE} OSP handle of in_bound call
${OSPINTIMELIMIT} Duration limit for in_bound call
${OSPOUTHANDLE} OSP handle of out_bound call
${OSPTECH} OSP technology
${OSPDEST} OSP destination
${OSPCALLING} OSP calling number
${OSPOUTTOKEN} OSP token to use for out_bound call
${OSPOUTTIMELIMIT} Duration limit for out_bound call
${OSPRESULTS} Number of remained destinations
${OSPINHANDLE} OSP handle of in_bound call
${OSPINTIMELIMIT} Duration limit for in_bound call
${OSPOUTHANDLE} OSP handle of out_bound call
${OSPTECH} OSP technology
${OSPDEST} OSP destination
${OSPCALLING} OSP calling number
${OSPOUTTOKEN} OSP token to use for out_bound call
${OSPOUTTIMELIMIT} Duration limit for out_bound call
${OSPRESULTS} Number of remained destinations
\end{verbatim}

10
doc/tex/hardware.tex
View File

@@ -1,8 +1,8 @@
\subsection{Introduction}
A PBX is only really useful if you can get calls into it. Of course, you
can use Asterisk with VoIP calls (SIP, H.323, IAX), but you can also talk
to the real PSTN through various cards.
can use Asterisk with VoIP calls (SIP, H.323, IAX, etc.), but you can also
talk to the real PSTN through various cards.
Supported Hardware is divided into two general groups: Zaptel devices and
non-zaptel devices. The Zaptel compatible hardware supports pseudo-TDM
@@ -31,7 +31,7 @@ compatible hardware may have different features.
\item TE210P - The TE210P improves performance and scalability through bus mastering architecture. It supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis.
\item TE207P - The TE207P offers an on-board DSP-based echo cancellation module. It supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis.
\item TE205P - The TE205P improves performance and scalability through bus mastering architecture. It supports both E1 and T1/J1 environments and is selectable on a per-card or per-port basis.
\item TE120P - The TE120P is a single span, selectable T1, E1, or J1 card and utilizes Digium's VoiceBus™ technology. It supports both voice and data modes.
\item TE120P - The TE120P is a single span, selectable T1, E1, or J1 card and utilizes Digium's VoiceBus\texttrademark technology. It supports both voice and data modes.
\item TE110P - The TE110P brings a high-performance, cost-effective, and flexible single span togglable T1, E1, J1 interface to the Digium line-up of telephony interface devices.
\end{itemize}
\end{itemize}
@@ -60,7 +60,7 @@ chan\_misdn. See the mISDN section for more information.
\begin{itemize}
\item Digium, Inc. (Primary Developer of Asterisk)
http://www.digium.com
http://www.digium.com
\begin{itemize}
\item B410P - 4 Port BRI card (TE/NT)
\end{itemize}
@@ -68,7 +68,7 @@ chan\_misdn. See the mISDN section for more information.
\begin{itemize}
\item beroNet
http://www.beronet.com
http://www.beronet.com
\begin{itemize}
\item BN4S0 - 4 Port BRI card (TE/NT)

2
doc/tex/jitterbuffer.tex
View File

@@ -43,7 +43,7 @@ backwards compatible architecture. Look in the source code for instructions.
\subsubsection{Testing and monitoring}
You can test the effectiveness of PLC and the new jitterbuffer's detection of loss by using
the new CLI command "iax2 test losspct <n>". This will simulate n percent packet loss
the new CLI command "iax2 test losspct $<$n$>$". This will simulate n percent packet loss
coming \_in\_ to chan\_iax2. You should find that with PLC and the new JB, 10 percent packet
loss should lead to just a tiny amount of distortion, while without PLC, it would lead to
silent gaps in your audio.

266
doc/tex/manager.tex
View File

@@ -41,7 +41,7 @@ subscriptions and call limits)
\section{Command Syntax}
Management communication consists of tags of the form "header: value",
terminated with an empty newline (\\r\\n) in the style of SMTP, HTTP, and
terminated with an empty newline (\textbackslash r\textbackslash n) in the style of SMTP, HTTP, and
other headers.
The first tag MUST be one of the following:
@@ -106,143 +106,143 @@ http://www.asterisk.org web site for more information.
\section{Some standard AMI headers}
\begin{verbatim}
Account: -- Account Code (Status)
AccountCode: -- Account Code (cdr_manager)
ACL: <Y | N> -- Does ACL exist for object ?
Action: <action> -- request or notification of a particular action
Address-IP: -- IPaddress
Address-Port: -- IP port number
Agent: <string> -- Agent name
AMAflags: -- AMA flag (cdr_manager, sippeers)
AnswerTime: -- Time of answer (cdr_manager)
Append: <bool> -- CDR userfield Append flag
Application: -- Application to use
Async: -- Whether or not to use fast setup
AuthType: -- Authentication type (for login or challenge)
"md5"
BillableSeconds: -- Billable seconds for call (cdr_manager)
CallerID: -- Caller id (name and number in Originate & cdr_manager)
CallerID: -- CallerID number
Number or "<unknown>" or "unknown"
(should change to "<unknown>" in app_queue)
CallerID1: -- Channel 1 CallerID (Link event)
CallerID2: -- Channel 2 CallerID (Link event)
CallerIDName: -- CallerID name
Name or "<unknown>" or "unknown"
(should change to "<unknown>" in app_queue)
Callgroup: -- Call group for peer/user
CallsTaken: <num> -- Queue status variable
Cause: <value> -- Event change cause - "Expired"
Cause: <value> -- Hangupcause (channel.c)
CID-CallingPres: -- Caller ID calling presentation
Channel: <channel> -- Channel specifier
Channel: <dialstring> -- Dialstring in Originate
Account: -- Account Code (Status)
AccountCode: -- Account Code (cdr_manager)
ACL: <Y | N> -- Does ACL exist for object ?
Action: <action> -- Request or notification of a particular action
Address-IP: -- IPaddress
Address-Port: -- IP port number
Agent: <string> -- Agent name
AMAflags: -- AMA flag (cdr_manager, sippeers)
AnswerTime: -- Time of answer (cdr_manager)
Append: <bool> -- CDR userfield Append flag
Application: -- Application to use
Async: -- Whether or not to use fast setup
AuthType: -- Authentication type (for login or challenge)
"md5"
BillableSeconds: -- Billable seconds for call (cdr_manager)
CallerID: -- Caller id (name and number in Originate & cdr_manager)
CallerID: -- CallerID number
Number or "<unknown>" or "unknown"
(should change to "<unknown>" in app_queue)
CallerID1: -- Channel 1 CallerID (Link event)
CallerID2: -- Channel 2 CallerID (Link event)
CallerIDName: -- CallerID name
Name or "<unknown>" or "unknown"
(should change to "<unknown>" in app_queue)
Callgroup: -- Call group for peer/user
CallsTaken: <num> -- Queue status variable
Cause: <value> -- Event change cause - "Expired"
Cause: <value> -- Hangupcause (channel.c)
CID-CallingPres: -- Caller ID calling presentation
Channel: <channel> -- Channel specifier
Channel: <dialstring> -- Dialstring in Originate
Channel: <tech/[peer/username]> -- Channel in Registry events (SIP, IAX2)
Channel: <tech> -- Technology (SIP/IAX2 etc) in Registry events
ChannelType: -- Tech: SIP, IAX2, ZAP, MGCP etc
Channel1: -- Link channel 1
Channel2: -- Link channel 2
ChanObjectType: -- "peer", "user"
Codecs: -- Codec list
CodecOrder: -- Codec order, separated with comma ","
Command: -- Cli command to run
Context: -- Context
Count: <num> -- Number of callers in queue
Data: -- Application data
Default-addr-IP: -- IP address to use before registration
Default-Username: -- Username part of URI to use before registration
Destination: -- Destination for call (Dialstring ) (dial, cdr_manager)
DestinationContext: -- Destination context (cdr_manager)
DestinationChannel: -- Destination channel (cdr_manager)
DestUniqueID: -- UniqueID of destination (dial event)
Disposition: -- Call disposition (CDR manager)
Domain: <domain> -- DNS domain
Duration: <secs> -- Duration of call (cdr_manager)
Dynamic: <Y | N> -- Device registration supported?
Endtime: -- End time stamp of call (cdr_manager)
EventList: <flag> -- Flag being "Start", "End", "Cancelled" or "ListObject"
Events: <eventmask> -- Eventmask filter ("on", "off", "system", "call", "log")
Exten: -- Extension (Redirect command)
Extension: -- Extension (Status)
Family: <string> -- ASTdb key family
File: <filename> -- Filename (monitor)
Format: <format> -- Format of sound file (monitor)
From: <time> -- Parking time (ParkedCall event)
Hint: -- Extension hint
Incominglimit: -- SIP Peer incoming limit
Channel: <tech> -- Technology (SIP/IAX2 etc) in Registry events
ChannelType: -- Tech: SIP, IAX2, ZAP, MGCP etc
Channel1: -- Link channel 1
Channel2: -- Link channel 2
ChanObjectType: -- "peer", "user"
Codecs: -- Codec list
CodecOrder: -- Codec order, separated with comma ","
Command: -- Cli command to run
Context: -- Context
Count: <num> -- Number of callers in queue
Data: -- Application data
Default-addr-IP: -- IP address to use before registration
Default-Username: -- Username part of URI to use before registration
Destination: -- Destination for call (Dialstring ) (dial, cdr_manager)
DestinationContext: -- Destination context (cdr_manager)
DestinationChannel: -- Destination channel (cdr_manager)
DestUniqueID: -- UniqueID of destination (dial event)
Disposition: -- Call disposition (CDR manager)
Domain: <domain> -- DNS domain
Duration: <secs> -- Duration of call (cdr_manager)
Dynamic: <Y | N> -- Device registration supported?
Endtime: -- End time stamp of call (cdr_manager)
EventList: <flag> -- Flag being "Start", "End", "Cancelled" or "ListObject"
Events: <eventmask> -- Eventmask filter ("on", "off", "system", "call", "log")
Exten: -- Extension (Redirect command)
Extension: -- Extension (Status)
Family: <string> -- ASTdb key family
File: <filename> -- Filename (monitor)
Format: <format> -- Format of sound file (monitor)
From: <time> -- Parking time (ParkedCall event)
Hint: -- Extension hint
Incominglimit: -- SIP Peer incoming limit
Key:
Key: -- ASTdb Database key
LastApplication: -- Last application executed (cdr_manager)
LastCall: <num> -- Last call in queue
LastData: -- Data for last application (cdr_manager)
Link: -- (Status)
ListItems: <number> -- Number of items in Eventlist (Optionally sent in "end" packet)
Location: -- Interface (whatever that is -maybe tech/name in app_queue )
Loginchan: -- Login channel for agent
Logintime: <number> -- Login time for agent
Mailbox: -- VM Mailbox (id@vmcontext) (mailboxstatus, mailboxcount)
MD5SecretExist: <Y | N> -- Whether secret exists in MD5 format
Membership: <string> -- "Dynamic" or "static" member in queue
Message: <text> -- Text message in ACKs, errors (explanation)
Mix: <bool> -- Boolean parameter (monitor)
NewMessages: <count> -- Count of new Mailbox messages (mailboxcount)
Newname:
ObjectName: -- Name of object in list
OldName: -- Something in Rename (channel.c)
OldMessages: <count> -- Count of old mailbox messages (mailboxcount)
Outgoinglimit: -- SIP Peer outgoing limit
Paused: <num> -- Queue member paused status
Peer: <tech/name> -- "channel" specifier :-)
PeerStatus: <tech/name> -- Peer status code
"Unregistered", "Registered", "Lagged", "Reachable"
Penalty: <num> -- Queue penalty
Priority: -- Extension priority
Privilege: <privilege> -- AMI authorization class (system, call, log, verbose, command, agent, user)
Pickupgroup: -- Pickup group for peer
Position: <num> -- Position in Queue
Queue: -- Queue name
Reason: -- "Autologoff"
Reason: -- "Chanunavail"
Response: <response> -- response code, like "200 OK"
"Success", "Error", "Follows"
Restart: -- "True", "False"
RegExpire: -- SIP registry expire
RegExpiry: -- SIP registry expiry
Reason: -- Originate reason code
Seconds: -- Seconds (Status)
Secret: <password> -- Authentication secret (for login)
SecretExist: <Y | N> -- Whether secret exists
Shutdown: -- "Uncleanly", "Cleanly"
Key: -- ASTdb Database key
LastApplication: -- Last application executed (cdr_manager)
LastCall: <num> -- Last call in queue
LastData: -- Data for last application (cdr_manager)
Link: -- (Status)
ListItems: <number> -- Number of items in Eventlist (Optionally sent in "end" packet)
Location: -- Interface (whatever that is -maybe tech/name in app_queue )
Loginchan: -- Login channel for agent
Logintime: <number> -- Login time for agent
Mailbox: -- VM Mailbox (id@vmcontext) (mailboxstatus, mailboxcount)
MD5SecretExist: <Y | N> -- Whether secret exists in MD5 format
Membership: <string> -- "Dynamic" or "static" member in queue
Message: <text> -- Text message in ACKs, errors (explanation)
Mix: <bool> -- Boolean parameter (monitor)
NewMessages: <count> -- Count of new Mailbox messages (mailboxcount)
Newname:
ObjectName: -- Name of object in list
OldName: -- Something in Rename (channel.c)
OldMessages: <count> -- Count of old mailbox messages (mailboxcount)
Outgoinglimit: -- SIP Peer outgoing limit
Paused: <num> -- Queue member paused status
Peer: <tech/name> -- "channel" specifier :-)
PeerStatus: <tech/name> -- Peer status code
"Unregistered", "Registered", "Lagged", "Reachable"
Penalty: <num> -- Queue penalty
Priority: -- Extension priority
Privilege: <privilege> -- AMI authorization class (system, call, log, verbose, command, agent, user)
Pickupgroup: -- Pickup group for peer
Position: <num> -- Position in Queue
Queue: -- Queue name
Reason: -- "Autologoff"
Reason: -- "Chanunavail"
Response: <response> -- response code, like "200 OK"
"Success", "Error", "Follows"
Restart: -- "True", "False"
RegExpire: -- SIP registry expire
RegExpiry: -- SIP registry expiry
Reason: -- Originate reason code
Seconds: -- Seconds (Status)
Secret: <password> -- Authentication secret (for login)
SecretExist: <Y | N> -- Whether secret exists
Shutdown: -- "Uncleanly", "Cleanly"
SIP-AuthInsecure:
SIP-FromDomain: -- Peer FromDomain
SIP-FromUser: -- Peer FromUser
SIP-FromDomain: -- Peer FromDomain
SIP-FromUser: -- Peer FromUser
SIP-NatSupport:
SIPLastMsg:
Source: -- Source of call (dial event, cdr_manager)
SrcUniqueID: -- UniqueID of source (dial event)
StartTime: -- Start time of call (cdr_manager)
State: -- Channel state
Status: -- Registration status (Registry events SIP)
Status: -- Extension status (Extensionstate)
Status: -- Peer status (if monitored) ** Will change name **
"unknown", "lagged", "ok"
Status: <num> -- Queue Status
Status: -- DND status (DNDState)
Time: <sec> -- Roundtrip time (latency)
Timeout: -- Parking timeout time
Timeout: -- Timeout for call setup (Originate)
Timeout: <seconds> -- Timeout for call
Uniqueid: -- Channel Unique ID
Uniqueid1: -- Channel 1 Unique ID (Link event)
Uniqueid2: -- Channel 2 Unique ID (Link event)
User: -- Username (SIP registry)
UserField: -- CDR userfield (cdr_manager)
Val: -- Value to set/read in ASTdb
Variable: -- Variable AND value to set (multiple separated with | in Originate)
Variable: <name> -- For channel variables
Value: <value> -- Value to set
VoiceMailbox: -- VM Mailbox in SIPpeers
Waiting: -- Count of mailbox messages (mailboxstatus)
Source: -- Source of call (dial event, cdr_manager)
SrcUniqueID: -- UniqueID of source (dial event)
StartTime: -- Start time of call (cdr_manager)
State: -- Channel state
Status: -- Registration status (Registry events SIP)
Status: -- Extension status (Extensionstate)
Status: -- Peer status (if monitored) ** Will change name **
"unknown", "lagged", "ok"
Status: <num> -- Queue Status
Status: -- DND status (DNDState)
Time: <sec> -- Roundtrip time (latency)
Timeout: -- Parking timeout time
Timeout: -- Timeout for call setup (Originate)
Timeout: <seconds> -- Timeout for call
Uniqueid: -- Channel Unique ID
Uniqueid1: -- Channel 1 Unique ID (Link event)
Uniqueid2: -- Channel 2 Unique ID (Link event)
User: -- Username (SIP registry)
UserField: -- CDR userfield (cdr_manager)
Val: -- Value to set/read in ASTdb
Variable: -- Variable AND value to set (multiple separated with | in Originate)
Variable: <name> -- For channel variables
Value: <value> -- Value to set
VoiceMailbox: -- VM Mailbox in SIPpeers
Waiting: -- Count of mailbox messages (mailboxstatus)
\end{verbatim}
** Please try to re-use existing headers to simplify manager message parsing in clients.

52
doc/tex/misdn.tex
View File

@@ -129,15 +129,15 @@ The Optionsstring looks Like:
the ":" character is the delimiter.
The available Optchars are:
d - Send display text on called phone, text is the optparam
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 cancellation on this channel,
takes taps as arguments (32,64,128,256)
s - send Non Inband DTMF as inband
vr - rxgain control
vt - txgain control
d - Send display text on called phone, text is the optparam
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 cancellation on this channel,
takes taps as arguments (32,64,128,256)
s - send Non Inband DTMF as inband
vr - rxgain control
vt - txgain control
\end{verbatim}
chan\_misdn registers a new dial plan application "misdn\_set\_opt" when
@@ -182,24 +182,24 @@ Now you should see the misdn cli commands:
\begin{verbatim}
- clean
-> pid (cleans a broken call, use with care, leads often
to a segmentation fault)
-> pid (cleans a broken call, use with care, leads often
to a segmentation fault)
- send
-> display (sends a Text Message to a Asterisk channel,
this channel must be an misdn channel)
-> display (sends a Text Message to a Asterisk channel,
this channel must be an misdn channel)
- set
-> debug (sets debug level)
-> debug (sets debug level)
- show
-> config (shows the configuration options)
-> channels (shows the current active misdn channels)
-> channel (shows details about the given misdn channels)
-> stacks (shows the current ports, their protocols and states)
-> fullstacks (shows the current active and inactive misdn channels)
-> config (shows the configuration options)
-> channels (shows the current active misdn channels)
-> channel (shows details about the given misdn channels)
-> stacks (shows the current ports, their protocols and states)
-> fullstacks (shows the current active and inactive misdn channels)
- restart
-> port (restarts given port (L2 Restart) )
-> port (restarts given port (L2 Restart) )
- reload (reloads misdn.conf)
- reload (reloads misdn.conf)
\end{verbatim}
You can only use "misdn send display" when an Asterisk channel is created and
@@ -218,8 +218,8 @@ msn (callerid) of the Phone to send the text to.
mISDN Exports/Imports a few Variables:
\begin{verbatim}
- MISDN_ADDRESS_COMPLETE : Is either set to 1 from the Provider, or you
can set it to 1 to force a sending complete.
- MISDN_ADDRESS_COMPLETE : Is either set to 1 from the Provider, or you
can set it to 1 to force a sending complete.
\end{verbatim}
@@ -258,9 +258,7 @@ as Display Message to the Phone.
\subsection{Known Problems}
\begin{verbatim}
* I cannot hear any tone after a successful CONNECT to the other end
Q: 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
A: You forgot to load mISDNdsp, which is now needed by chan\_misdn for switching
and dtmf tone detection
\end{verbatim}

2
doc/tex/odbcstorage.tex
View File

@@ -1,3 +1,5 @@
ODBC Storage allows you to store voicemail messages within a database
instead of using a file. This is *not* a full realtime engine and
*only* supports ODBC. The table description for the "voicemessages"

88
doc/tex/privacy.tex
View File

@@ -268,25 +268,27 @@ option):
\begin{verbatim}
[home-introduction]
exten => s,1,Background,intro-options ;; Script: To hear your Introduction, dial 1.
;; to record a new introduction, dial 2.
;; to return to the main menu, dial 3.
;; to hear what this is all about, dial 4.
exten => s,1,Background,intro-options ;; Script:
;; To hear your Introduction, dial 1.
;; to record a new introduction, dial 2.
;; to return to the main menu, dial 3.
;; to hear what this is all about, dial 4.
exten => 1,1,Playback,priv-callerintros/${CALLERIDNUM}
exten => 1,2,Goto(s,1)
exten => 2,1,Goto(home-introduction-record,s,1)
exten => 3,1,Goto(homeline,s,7)
exten => 4,1,Playback,intro-intro ;; Script:
;; This may seem a little strange, but it really is a neat
;; thing, both for you and for us. I've taped a short introduction
;; for many of the folks who normally call us. Using the Caller ID
;; from each incoming call, the system plays the introduction
;; for that phone number over a speaker, just as the call comes in.
;; This helps the folks
;; here in the house more quickly determine who is calling.
;; and gets the right ones to gravitate to the phone.
;; You can listen to, and record a new intro for your phone number
;; using this menu.
exten => 4,1,Playback,intro-intro
;; Script:
;; This may seem a little strange, but it really is a neat
;; thing, both for you and for us. I've taped a short introduction
;; for many of the folks who normally call us. Using the Caller ID
;; from each incoming call, the system plays the introduction
;; for that phone number over a speaker, just as the call comes in.
;; This helps the folks
;; here in the house more quickly determine who is calling.
;; and gets the right ones to gravitate to the phone.
;; You can listen to, and record a new intro for your phone number
;; using this menu.
exten => 4,2,Goto(s,1)
exten => t,1,Goto(s,1)
exten => i,1,Background,invalid
@@ -295,37 +297,37 @@ exten => o,1,Goto(s,1)
[home-introduction-record]
exten => s,1,Background,intro-record-choices ;; Script:
;; If you want some advice about recording your
;; introduction, dial 1.
;; otherwise, dial 2, and introduce yourself after
;; the beep.
;; If you want some advice about recording your
;; introduction, dial 1.
;; otherwise, dial 2, and introduce yourself after
;; the beep.
exten => 1,1,Playback,intro-record
;; Your introduction should be short and sweet and crisp.
;; Your introduction will be limited to 4 seconds.
;; This is NOT meant to be a voice mail message, so
;; please, don't say anything about why you are calling.
;; After we are done making the recording, your introduction
;; will be saved for playback.
;; If you are the only person that would call from this number,
;; please state your name. Otherwise, state your business
;; or residence name instead. For instance, if you are
;; friend of the family, say, Olie McPherson, and both
;; you and your kids might call here a lot, you might
;; say: "This is the distinguished Olie McPherson Residence!"
;; If you are the only person calling, you might say this:
;; "This is the illustrious Kermit McFrog! Pick up the Phone, someone!!"
;; If you are calling from a business, you might pronounce a more sedate introduction,like,
;; "Fritz from McDonalds calling.", or perhaps the more original introduction:
;; "John, from the Park County Morgue. You stab 'em, we slab 'em!".
;; Just one caution: the kids will hear what you record every time
;; you call. So watch your language!
;; I will begin recording after the tone.
;; When you are done, hit the # key. Gather your thoughts and get
;; ready. Remember, the # key will end the recording, and play back
;; your intro. Good Luck, and Thank you!"
;; Your introduction should be short and sweet and crisp.
;; Your introduction will be limited to 4 seconds.
;; This is NOT meant to be a voice mail message, so
;; please, don't say anything about why you are calling.
;; After we are done making the recording, your introduction
;; will be saved for playback.
;; If you are the only person that would call from this number,
;; please state your name. Otherwise, state your business
;; or residence name instead. For instance, if you are
;; friend of the family, say, Olie McPherson, and both
;; you and your kids might call here a lot, you might
;; say: "This is the distinguished Olie McPherson Residence!"
;; If you are the only person calling, you might say this:
;; "This is the illustrious Kermit McFrog! Pick up the Phone, someone!!"
;; If you are calling from a business, you might pronounce a more sedate introduction,like,
;; "Fritz from McDonalds calling.", or perhaps the more original introduction:
;; "John, from the Park County Morgue. You stab 'em, we slab 'em!".
;; Just one caution: the kids will hear what you record every time
;; you call. So watch your language!
;; I will begin recording after the tone.
;; When you are done, hit the # key. Gather your thoughts and get
;; ready. Remember, the # key will end the recording, and play back
;; your intro. Good Luck, and Thank you!"
exten => 1,2,Goto(2,1)
exten => 2,1,Background,intro-start
;; OK, here we go! After the beep, please give your introduction.
;; OK, here we go! After the beep, please give your introduction.
exten => 2,2,Background,beep
exten => 2,3,Record,priv-callerintros/${CALLERIDNUM}:gsm|4
exten => 2,4,Background,priv-callerintros/${CALLERIDNUM}

57
doc/tex/qos.tex
View File

@@ -7,7 +7,7 @@ congested with other traffic.
Also asterisk running on Linux can set 802.1p CoS marks in VLAN packets
for all used VoIP protocols. It is useful when you are working in switched
enviropment. For maping skb->priority and VLAN CoS mark you need to use
enviropment. For maping skb-$>$priority and VLAN CoS mark you need to use
command "vconfig set\_egress\_map [vlan-device] [skb-priority] [vlan-qos]".
\subsubsection{SIP}
@@ -64,30 +64,30 @@ integer values from 0 to 7.
\begin{verbatim}
===========================================
Configuration Parameter Recommended
File Setting
-------------------------------------------
sip.conf tos\_sip cs3
sip.conf tos\_audio ef
sip.conf tos\_video af41
sip.conf tos\_text af41
sip.conf cos\_sip 4
sip.conf cos\_audio 6
sip.conf cos\_video 5
sip.conf cos\_text 0
-------------------------------------------
iax.conf tos ef
iax.conf cos 6
-------------------------------------------
iaxprov.conf tos ef
-------------------------------------------
mgcp.conf tos ef
mgcp.conf cos 6
-------------------------------------------
h323.conf tos ef
h323.conf cos 6
===========================================
+==============+============+==============+
|Configuration | Parameter | Recommended |
|File | Setting | |
+--------------+------------+--------------+
| | tos_sip | cs3 |
| | tos_audio | ef |
| | tos_video | af41 |
| sip.conf | tos_text | af41 |
| | cos_sip | 4 |
| | cos_audio | 6 |
| | cos_video | 5 |
| | cos_text | 0 |
+--------------+------------+--------------+
| iax.conf | tos | ef |
| | cos | 6 |
+--------------+------------+--------------+
| iaxprov.conf | tos | ef |
+--------------+------------+--------------+
| mgcp.conf | tos | ef |
| | cos | 6 |
+--------------+------------+--------------+
| h323.conf | tos | ef |
| | cos | 6 |
+==============+============+==============+
\end{verbatim}
\subsubsection{Reference}
@@ -108,12 +108,11 @@ To get the most out of setting the TOS on packets generated by
Asterisk, you will need to ensure that your network handles packets
with a TOS properly. For Cisco devices, see the previously mentioned
"Enterprise QoS Solution Reference Network Design Guide". For Linux
systems see the "Linux Advanced Routing \& Traffic Control HOWTO" at
<http://www.lartc.org/>.
systems see the "Linux Advanced Routing \& Traffic Control HOWTO" at
http://www.lartc.org/.
For more information on Quality of
Service for VoIP networks see the "Enterprise QoS Solution Reference
Network Design Guide" version 3.3 from Cisco at:
<http://www.cisco.com/application/pdf/en/us/guest/netsol/ns432/c649/ccmigration\_09186a008049b062.pdf>
http://www.cisco.com/application/pdf/en/us/guest/netsol/ns432/c649/ccmigration\_09186a008049b062.pdf

54
doc/tex/queuelog.tex
View File

@@ -6,59 +6,71 @@ queue log, located (by default) in /var/log/asterisk/queue\_log.
These are the events (and associated information) in the queue log:
ABANDON(position|origposition|waittime)
\textbf{ABANDON(position|origposition|waittime)}
The caller abandoned their position in the queue. The position is the
caller's position in the queue when they hungup, the origposition is
the original position the caller was when they first entered the
queue, and the waittime is how long the call had been waiting in the
queue at the time of disconnect.
AGENTDUMP
\textbf{AGENTDUMP}
The agent dumped the caller while listening to the queue announcement.
AGENTLOGIN(channel)
\textbf{AGENTLOGIN(channel)}
The agent logged in. The channel is recorded.
AGENTCALLBACKLOGIN(exten@context)
\textbf{AGENTCALLBACKLOGIN(exten@context)}
The callback agent logged in. The login extension and context is recorded.
AGENTLOGOFF(channel|logintime)
\textbf{AGENTLOGOFF(channel|logintime)}
The agent logged off. The channel is recorded, along with the total time
the agent was logged in.
AGENTCALLBACKLOGOFF(exten@context|logintime|reason)
\textbf{AGENTCALLBACKLOGOFF(exten@context|logintime|reason)}
The callback agent logged off. The last login extension and context is
recorded, along with the total time the agent was logged in, and the
reason for the logoff if it was not a normal logoff
(e.g., Autologoff, Chanunavail)
COMPLETEAGENT(holdtime|calltime|origposition)
\textbf{COMPLETEAGENT(holdtime|calltime|origposition)}
The caller was connected to an agent, and the call was terminated normally
by the *agent*. The caller's hold time and the length of the call are both
recorded. The caller's original position in the queue is recorded in
origposition.
COMPLETECALLER(holdtime|calltime|origposition)
\textbf{COMPLETECALLER(holdtime|calltime|origposition)}
The caller was connected to an agent, and the call was terminated normally
by the *caller*. The caller's hold time and the length of the call are both
recorded. The caller's original position in the queue is recorded in
origposition.
CONFIGRELOAD
\textbf{CONFIGRELOAD}
The configuration has been reloaded (e.g. with asterisk -rx reload)
CONNECT(holdtime|bridgedchanneluniqueid)
\textbf{CONNECT(holdtime|bridgedchanneluniqueid)}
The caller was connected to an agent. Hold time represents the amount
of time the caller was on hold. The bridged channel unique ID contains
the unique ID of the queue member channel that is taking the call. This
is useful when trying to link recording filenames to a particular
call in the queue.
ENTERQUEUE(url|callerid)
\textbf{ENTERQUEUE(url|callerid)}
A call has entered the queue. URL (if specified) and Caller*ID are placed
in the log.
EXITEMPTY(position|origposition|waittime)
\textbf{EXITEMPTY(position|origposition|waittime)}
The caller was exited from the queue forcefully because the queue had no
reachable members and it's configured to do that to callers when there
are no reachable members. The position is the caller's position in the
@@ -66,29 +78,35 @@ queue when they hungup, the origposition is the original position the
caller was when they first entered the queue, and the waittime is how
long the call had been waiting in the queue at the time of disconnect.
EXITWITHKEY(key|position|origposition|waittime)
\textbf{EXITWITHKEY(key|position|origposition|waittime)}
The caller elected to use a menu key to exit the queue. The key and
the caller's position in the queue are recorded. The caller's entry
position and amoutn of time waited is also recorded.
EXITWITHTIMEOUT(position|origposition|waittime)
\textbf{EXITWITHTIMEOUT(position|origposition|waittime)}
The caller was on hold too long and the timeout expired. The position in the
queue when the timeout occurred, the entry position, and the amount of time
waited are logged.
QUEUESTART
\textbf{QUEUESTART}
The queueing system has been started for the first time this session.
RINGNOANSWER(ringtime)
\textbf{RINGNOANSWER(ringtime)}
After trying for ringtime ms to connect to the available queue member,
the attempt ended without the member picking up the call. Bad queue
member!
SYSCOMPAT
\textbf{SYSCOMPAT}
A call was answered by an agent, but the call was dropped because the
channels were not compatible.
TRANSFER(extension|context|holdtime|calltime)
\textbf{TRANSFER(extension|context|holdtime|calltime)}
Caller was transferred to a different extension. Context and extension
are recorded. The caller's hold time and the length of the call are both
recorded. PLEASE remember that transfers performed by SIP UA's by way

14
doc/tex/realtime.tex
View File

@@ -6,9 +6,6 @@ functions implemented in Asterisk.
The benefits of this architecture are many, both from a code management
standpoint and from an installation perspective.
Additional information on the configuration of Realtime with Asterisk
can be found in doc/extconfig.txt
The ARA is designed to be independent of storage. Currently, most
drivers are based on SQL, but the architecture should be able to handle
other storage methods in the future, like LDAP.
@@ -77,7 +74,7 @@ databases and reload it whenever you want. You can force a reload over
the AMI, Asterisk Manager Interface or by calling Asterisk from a
shell script with
asterisk -rx "reload"
asterisk -rx "reload"
You may also dynamically add SIP and IAX devices and extensions
and making them available without a reload, by using the realtime
@@ -87,7 +84,7 @@ objects and the realtime switch.
You configure the ARA in extconfig.conf (yes, it's a strange name, but
is was defined in the early days of the realtime architecture and kind
of stuck). Please see doc/extconfig.txt for database schemas.
of stuck).
The part of Asterisk that connects to the ARA use a well defined family
name to find the proper database driver. The syntax is easy:
@@ -110,11 +107,8 @@ Defined well-known family names are:
\item extensions - Realtime extensions (switch)
\end{itemize}
There is documentation of the SQL database in the file
doc/extconfig.txt in your Asterisk source code tree.
For voicemail storage with the support of ODBC, there is a
doc/odbcstorage.txt documentation file.
Voicemail storage with the support of ODBC described in file
docs/odbcstorage.tex (\ref{odbcstorage}).
\subsubsection{Limitations}