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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76874 65c4cc65-6c06-0410-ace0-fbb531ad65f3

doc/tex/ael.tex

@@ -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}
 
-
+\section{Differences with the original version of AEL}
-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
+  \item Added the ifTime ($<$time range$>$|$<$days of week$>$|$<$days of
-      month>|<months> ) {} [else {}] construct, which executes much
+      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
+      $<$expr$>$ ) $<$lucky-statement$>$ [ else $<$unlucky-statement$>$ ]. The
-      probability of the lucky-statement getting executed is <expr>,
+      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
+      $<$lucky-statement$>$ isn't so lucky this time around, then the
-      <unlucky-statement> gets executed, if it is present.
+      $<$unlucky-statement$>$ gets executed, if it is present.
 \end{enumerate}
 
 

doc/tex/asterisk.tex

@@ -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}
@@ -120,10 +121,11 @@ reference purposes.
 %See http://www.asterisk.org/developers for more information
 %
 %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}

doc/tex/billing.tex

@@ -15,7 +15,6 @@ 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}

doc/tex/channelvariables.tex

@@ -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
+string "\$1231", quote it with a preceding \textbackslash. Special characters that must
-be quoted to be used, are [ ] \$ " \\. (to write \\ itself, use \\). 
+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. 
@@ -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.
@@ -334,9 +334,9 @@ 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[^ ]+)"
@@ -421,7 +421,7 @@ TRUNC(3.5)
 TRUNC(-3.5)
    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 : 
@@ -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.
@@ -670,22 +670,24 @@ only read in the dialplan.  Writes to such variables are silently
 ignored.
 
 \begin{verbatim}
-${ACCOUNTCODE} 	 	* Account code (if specified) (Deprecated; use ${CDR(accountcode)})
+${CDR(accountcode)}    * Account code (if specified)
 ${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(ani)}       * Caller ANI (PRI channels)
-${CALLERID}	 	* Caller ID (Deprecated; use ${CALLERID(all)})
+${CALLERID(all)}       * Caller ID
-${CALLERIDNAME}	 	* Caller ID Name only (Deprecated; use ${CALLERID(name)})
+${CALLERID(dnid)}      * Dialed Number Identifier
-${CALLERIDNUM}	 	* Caller ID Number only (Deprecated; use ${CALLERID(num)})
+${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)})
+${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
@@ -701,8 +703,8 @@ ${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 
-${TIMESTAMP}	 	* Current date time in the format: YYYYMMDD-HHMMSS (Deprecated; use ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
+                         (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

doc/tex/hardware.tex

@@ -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
+can use Asterisk with VoIP calls (SIP, H.323, IAX, etc.), but you can also 
-to the real PSTN through various cards.
+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}

doc/tex/jitterbuffer.tex

@@ -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.

doc/tex/manager.tex

@@ -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:
@@ -109,7 +109,7 @@ http://www.asterisk.org web site for more information.
  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
+ Action: <action>         -- Request or notification of a particular action
  Address-IP:              -- IPaddress 
  Address-Port:            -- IP port number
  Agent: <string>          -- Agent name

doc/tex/misdn.tex

@@ -258,9 +258,7 @@ as Display Message to the Phone.
 
 \subsection{Known Problems}
 
-\begin{verbatim}
+Q: I cannot hear any tone after a successful CONNECT to the other end
-* I cannot hear any tone after a successful CONNECT to the other end
 
--> you forgot to load mISDNdsp, which is now needed by chan\_misdn for switching
+A: You forgot to load mISDNdsp, which is now needed by chan\_misdn for switching
 and dtmf tone detection
-\end{verbatim}

doc/tex/odbcstorage.tex

@@ -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" 

doc/tex/privacy.tex

@@ -268,7 +268,8 @@ option):
 
 \begin{verbatim}
 [home-introduction]
-exten => s,1,Background,intro-options   ;; Script: To hear your Introduction, dial 1.
+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.
@@ -276,7 +277,8 @@ 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:
+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

doc/tex/qos.tex

@@ -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
+|Configuration | Parameter  | Recommended  |
-File				Setting
+|File          | Setting    |              |
--------------------------------------------
++--------------+------------+--------------+
-sip.conf	tos\_sip	cs3
+|              | tos_sip    | cs3          |
-sip.conf	tos\_audio	ef
+|              | tos_audio  | ef           |
-sip.conf	tos\_video	af41
+|              | tos_video  | af41         |
-sip.conf	tos\_text	af41
+| sip.conf     | tos_text   | af41         |
-sip.conf	cos\_sip	4
+|              | cos_sip    | 4            |
-sip.conf	cos\_audio	6
+|              | cos_audio  | 6            |
-sip.conf	cos\_video	5
+|              | cos_video  | 5            |
-sip.conf	cos\_text	0
+|              | cos_text   | 0            |
--------------------------------------------
++--------------+------------+--------------+
-iax.conf	tos		ef
+| iax.conf     | tos        | ef           |
-iax.conf	cos		6	
+|              | cos        | 6            |
--------------------------------------------
++--------------+------------+--------------+
-iaxprov.conf	tos		ef
+| iaxprov.conf | tos        | ef           |
--------------------------------------------
++--------------+------------+--------------+
-mgcp.conf	tos		ef
+| mgcp.conf    | tos        | ef           |
-mgcp.conf	cos		6
+|              | cos        | 6            |
--------------------------------------------
++--------------+------------+--------------+
-h323.conf	tos		ef
+| h323.conf    | tos        | ef           |
-h323.conf	cos		6
+|              | cos        | 6            |
-===========================================
++==============+============+==============+
 \end{verbatim}
 
 \subsubsection{Reference}
@@ -109,11 +109,10 @@ 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/>.
+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>
 

doc/tex/queuelog.tex

@@ -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

doc/tex/realtime.tex

@@ -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.
@@ -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
+Voicemail storage with the support of ODBC described in file 
-doc/extconfig.txt in your Asterisk source code tree.
+docs/odbcstorage.tex (\ref{odbcstorage}).
-
-For voicemail storage with the support of ODBC, there is a 
-doc/odbcstorage.txt documentation file.
 
 \subsubsection{Limitations}