From 15eb8220b5c2a3b63f8dde5888ad057b039ec845 Mon Sep 17 00:00:00 2001
From: rmudgett <rmudgett@f38db490-d61c-443f-a65b-d21fe96a405b>
Date: Sat, 18 Jul 2009 02:09:13 +0000
Subject: Merged revisions 145293,158010 from
 https://origsvn.digium.com/svn/asterisk/branches/1.4 to make merging easier. 
 These changes are already on trunk.

................
  r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines

  channels/chan_misdn.c
  channels/misdn/isdn_lib.c
  *  Miscellaneous other fixes from trunk to make merging easier later.

  ........
  r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines

  *  Miscellaneous formatting changes to make v1.4 and trunk
  more merge compatible in the mISDN area.

  channels/chan_misdn.c
  *  Eliminated redundant code in cb_events() EVENT_SETUP

  ........
  r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines

  improved helptext of misdn_set_opt.
  ........
  r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line

  Cleaned up comment

  ........
  r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines

  channels/chan_misdn.c
  *  Made bearer2str() use allowed_bearers_array[]
  *  Made use the causes.h defines instead of hardcoded numbers.
  *  Made use Asterisk presentation indicator values if either of the
  mISDN presentation or screen options are negative.
  *  Updated the misdn_set_opt application option descriptions.
  *  Renamed the awkward Caller ID presentation misdn_set_opt
  application option value not_screened to restricted.
  Deprecated the not_screened option value.

  channels/misdn/isdn_lib.c
  *  Made use the causes.h defines instead of hardcoded numbers.
  *  Fixed some spelling errors and typos.
  *  Added all defined facility code strings to fac2str().

  channels/misdn/isdn_lib.h
  *  Added doxygen comments to struct misdn_bchannel.

  channels/misdn/isdn_lib_intern.h
  *  Added doxygen comments to struct misdn_stack.

  channels/misdn_config.c
  configs/misdn.conf.sample
  *  Updated the mISDN presentation and screen parameter descriptions.

  doc/misdn.txt (doc/tex/misdn.tex)
  *  Updated the misdn_set_opt application option descriptions.
  *  Fixed some spelling errors and typos.
................
  r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines

  Merged revision 157977 from
  https://origsvn.digium.com/svn/asterisk/team/group/issue8824

  ........
  Fixes JIRA ABE-1726

  The dial extension could be empty if you are using MISDN_KEYPAD
  to control ISDN provider features.
................


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@207287 f38db490-d61c-443f-a65b-d21fe96a405b
---
 doc/tex/misdn.tex | 106 +++++++++++++++++++++++++++++-------------------------
 1 file changed, 58 insertions(+), 48 deletions(-)

(limited to 'doc/tex/misdn.tex')

diff --git a/doc/tex/misdn.tex b/doc/tex/misdn.tex
index a42b152ff..2d5c9413c 100644
--- a/doc/tex/misdn.tex
+++ b/doc/tex/misdn.tex
@@ -1,28 +1,27 @@
 \subsection{Introduction}
 
 This package contains the mISDN Channel Driver for the Asterisk PBX. It
-supports every mISDN Hardware and provides an interface for asterisk.
+supports every mISDN Hardware and provides an interface for Asterisk.
 
 \subsection{Features}
 
 \begin{itemize}
-\item  NT and TE mode
-\item  PP and PMP mode
-\item  BRI and PRI (with BNE1 and BN2E1 Cards)
-\item  Hardware Bridging
-\item  DTMF Detection in HW+mISDNdsp
-\item  Display Messages on Phones (on those that support display msg)
-\item  app\_SendText
-\item  HOLD/RETRIEVE/TRANSFER on ISDN Phones : )
-\item  Screen/ Not Screen User Number
-\item  EchoCancellation
-\item  Volume Control
-\item  Crypting with mISDNdsp (Blowfish)
-\item  Data (HDLC) callthrough
-\item  Data Calling (with app\_ptyfork +pppd)
-\item  Echo cancellation
-\item  CallDeflection
-\item Some other
+\item NT and TE mode
+\item PP and PMP mode
+\item BRI and PRI (with BNE1 and BN2E1 Cards)
+\item Hardware bridging
+\item DTMF detection in HW+mISDNdsp
+\item Display messages on phones (on those that support it)
+\item app\_SendText
+\item HOLD/RETRIEVE/TRANSFER on ISDN phones : )
+\item Allow/restrict user number presentation
+\item Volume control
+\item Crypting with mISDNdsp (Blowfish)
+\item Data (HDLC) callthrough
+\item Data calling (with app\_ptyfork +pppd)
+\item Echo cancellation
+\item Call deflection
+\item Some others
 \end{itemize}
 
 \subsection{Fast Installation Guide}
@@ -31,7 +30,7 @@ It is easy to install mISDN and mISDNuser. This can be done by:
 \begin{itemize}
      \item You can download latest stable releases from \url{http://www.misdn.org/downloads/}
 
-     \item Just fetch the newest head of the GIT (mISDN provect moved from CVS)
+     \item Just fetch the newest head of the GIT (mISDN project moved from CVS)
      In details this process described here: \url{http://www.misdn.org/index.php/GIT}
 \end{itemize}
 
@@ -50,7 +49,7 @@ cd mISDNuser ;
 make && make install
 \end{verbatim}
 \end{astlisting}
-Now you can compile chan\_misdn, just by making asterisk:
+Now you can compile chan\_misdn, just by making Asterisk:
 \begin{astlisting}
 \begin{verbatim}
 cd asterisk ;
@@ -66,8 +65,7 @@ Modules. Also install process described in \url{http://www.misdn.org/index.php/I
 
 To compile and install this driver, you'll need at least one mISDN Driver and
 the mISDNuser package. Chan\_misdn works with both, the current release version
-and the development (svn trunk) version of Asterisk. mISDNuser and mISDN must
-be fetched from cvs.isdn4linux.de.
+and the development (svn trunk) version of Asterisk.
 
 You should use Kernels $>$= 2.6.9
 
@@ -84,7 +82,7 @@ script is:
 \end{verbatim}
 \end{astlisting}
 Now you will want to configure the misdn.conf file which resides in the
-asterisk config directory (normally /etc/asterisk).
+Asterisk config directory (normally /etc/asterisk).
 
 \subsubsection{misdn.conf: [general]}
 The misdn.conf file contains a "general" subsection, and user subsections which
@@ -93,13 +91,13 @@ contain misdn port settings and different Asterisk contexts.
 In the general subsection you can set options that are not directly port
 related. There is for example the very important debug variable which you can
 set from the Asterisk cli (command line interface) or in this configuration
-file, bigger numbers will lead to more debug output. There's also a tracefile
+file, bigger numbers will lead to more debug output. There's also a trace file
 option, which takes a path+filename where debug output is written to.
 
 \subsubsection{misdn.conf: [default] subsection}
 
 The default subsection is another special subsection which can contain all the
-options available in the user/port subsections. the user/port subsection inherit
+options available in the user/port subsections. The user/port subsections inherit
 their parameters from the default subsection.
 
 \subsubsection{misdn.conf: user/port subsections}
@@ -108,13 +106,13 @@ The user subsections have names which are unequal to "general". Those subsection
 contain the ports variable which mean the mISDN Ports. Here you can add
 multiple ports, comma separated.
 
-Espacially for TE-Mode Ports there is a msns option. This option tells the
+Especially for TE-Mode Ports there is a msns option. This option tells the
 chan\_misdn driver to listen for incoming calls with the given msns, you can
-insert a '*' as single msn, which leads in getting every incoming call (if you
-want to share on PMP TE S0 with a asterisk and a phone or isdn card you should
-insert here the msns which you'll like to give the Asterisk).  Finally a
-context variable resides in the user subsections, which tells chan\_misdn where to
-send incoming calls to in the Asterisk dial plan (extension.conf).
+insert a '*' as single msn, which leads to getting every incoming call. If you
+want to share on PMP TE S0 with Asterisk and a phone or ISDN card you should
+insert here the msns which you assign to Asterisk.  Finally a context variable
+resides in the user subsections, which tells chan\_misdn where to send incoming
+calls to in the Asterisk dial plan (extension.conf).
 
 
 \subsubsection{Dial and Options String}
@@ -127,20 +125,32 @@ so the generic dial string looks like:
 mISDN/<port>[:bchannel]|g:<group>/<extension>[/<OPTIONSSTRING>]
 
 The Optionsstring looks Like:
-:<optchar1><OptParam1>:<optchar2><OptParam2>
+:<optchar><optarg>:<optchar><optarg>...
 
 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
+The available options are:
+  a - Have Asterisk detect DTMF tones on called channel
+  c - Make crypted outgoing call, optarg is keyindex
+  d - Send display text to called phone, text is the optarg
+  e - Perform echo cancelation on this channel,
+      takes taps as optarg (32,64,128,256)
+ e! - Disable echo cancelation on this channel
+  f - Enable fax detection
+  h - Make digital outgoing call
+ h1 - Make HDLC mode digital outgoing call
+  i - Ignore detected DTMF tones, don't signal them to Asterisk,
+      they will be transported inband.
+ jb - Set jitter buffer length, optarg is length
+ jt - Set jitter buffer upper threshold, optarg is threshold
+ jn - Disable jitter buffer
+  n - Disable mISDN DSP on channel.
+      Disables: echo cancel, DTMF detection, and volume control.
+  p - Caller ID presentation,
+      optarg is either 'allowed' or 'restricted'
+  s - Send Non-inband DTMF as inband
+ vr - Rx gain control, optarg is gain
+ vt - Tx gain control, optarg is gain
 \end{verbatim}
 \end{astlisting}
 
@@ -161,7 +171,7 @@ Phone1 --> * Box 1 --> PSTN_TE
 PSTN_TE --> * Box 2 --> Phone2
 \end{verbatim}
 
-The Encryption must be done on the PSTN sides, so the dialplan on the boxes
+The encryption must be done on the PSTN sides, so the dialplan on the boxes
 are:
 
 \begin{verbatim}
@@ -210,7 +220,7 @@ Now you should see the misdn cli commands:
 
 You can only use "misdn send display" when an Asterisk channel is created and
 isdn is in the correct state. "correct state" means that you have established a
-call to another phone (mustn't be isdn though).
+call to another phone (must not be isdn though).
 
 Then you use it like this:
 
@@ -232,8 +242,8 @@ mISDN Exports/Imports a few Variables:
 \subsection{Debugging and sending bug reports}
 
 If you encounter problems, you should set up the debugging flag, usually
-debug=2 should be enough. the messages are divided in asterisk and misdn
-parts.  Misdn Debug messages begin with an 'I', asterisk messages begin with
+debug=2 should be enough. The messages are divided into Asterisk and mISDN
+parts.  mISDN Debug messages begin with an 'I', Asterisk messages begin with
 an '*', the rest is clear I think.
 
 Please take a trace of the problem and open a report in the Asterisk issue
@@ -266,7 +276,7 @@ as Display Message to the Phone.
 
 \subsection{Known Problems}
 
-Q: 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.
 
 A: You forgot to load mISDNdsp, which is now needed by chan\_misdn for switching
-and dtmf tone detection
+and DTMF tone detection.
-- 
cgit v1.2.3

