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diff --git a/doc/tex/misdn.tex b/doc/tex/misdn.tex deleted file mode 100644 index 9b2c705cb..000000000 --- a/doc/tex/misdn.tex +++ /dev/null @@ -1,282 +0,0 @@ -\subsection{Introduction} - -This package contains the mISDN Channel Driver for the Asterisk PBX. It -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 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} - -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 project moved from CVS) - In details this process described here: \url{http://www.misdn.org/index.php/GIT} -\end{itemize} - - -then compile and install both with: -\begin{astlisting} -\begin{verbatim} -cd mISDN ; -make && make install -\end{verbatim} -\end{astlisting} -(you will need at least your kernel headers to compile mISDN). -\begin{astlisting} -\begin{verbatim} -cd mISDNuser ; -make && make install -\end{verbatim} -\end{astlisting} -Now you can compile chan\_misdn, just by making Asterisk: -\begin{astlisting} -\begin{verbatim} -cd asterisk ; -./configure && make && make install -\end{verbatim} -\end{astlisting} -That's all! - -Follow the instructions in the mISDN Package for how to load the Kernel -Modules. Also install process described in \url{http://www.misdn.org/index.php/Installing_mISDN} - -\subsection{Pre-Requisites} - -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. - -You should use Kernels $>$= 2.6.9 - - -\subsection{Configuration} - -First of all you must configure the mISDN drivers, please follow the -instructions in the mISDN package to do that, the main config file and config -script is: -\begin{astlisting} -\begin{verbatim} -/etc/init.d/misdn-init and -/etc/misdn-init.conf -\end{verbatim} -\end{astlisting} -Now you will want to configure the misdn.conf file which resides in the -Asterisk config directory (normally /etc/asterisk). - -\subsubsection{misdn.conf: [general]} -The misdn.conf file contains a "general" subsection, and user subsections which -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 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 subsections inherit -their parameters from the default subsection. - -\subsubsection{misdn.conf: user/port subsections} - -The user subsections have names which are unequal to "general". Those subsections -contain the ports variable which mean the mISDN Ports. Here you can add -multiple ports, comma separated. - -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 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} - -The dial string of chan\_misdn got more complex, because we added more features, -so the generic dial string looks like: - -\begin{astlisting} -\begin{verbatim} -mISDN/<port>[:bchannel]|g:<group>/<extension>[/<OPTIONSSTRING>] - -The Optionsstring looks Like: -:<optchar><optarg>:<optchar><optarg>... - -the ":" character is the delimiter. - -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} - -chan\_misdn registers a new dial plan application "misdn\_set\_opt" when -loaded. This application takes the Optionsstring as argument. The Syntax is: - -\begin{verbatim} -misdn_set_opt(<OPTIONSSTRING>) -\end{verbatim} - -When you set options in the dialstring, the options are set in the external -channel. When you set options with misdn\_set\_opt, they are set in the current -incoming channel. So if you like to use static encryption, the scenario looks -as follows: - -\begin{verbatim} -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 -are: - -\begin{verbatim} -* Box 1: -exten => _${CRYPT_PREFIX}X.,1,Dial(mISDN/g:outbound/:c1) - -* Box 2: -exten => ${CRYPT_MSN},1,misdn_set_opt(:c1) -exten => ${CRYPT_MSN},2,dial(${PHONE2}) -\end{verbatim} - - -\subsection{mISDN CLI commands} - -At the Asterisk cli you can try to type in: - -\begin{verbatim} -misdn <tab> <tab> -\end{verbatim} - -Now you should see the misdn cli commands: - -\begin{astlisting} -\begin{verbatim} -- clean - -> 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) -- set - -> 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) - -- restart - -> port (restarts given port (L2 Restart) ) - -- reload (reloads misdn.conf) -\end{verbatim} -\end{astlisting} - -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 (must not be isdn though). - -Then you use it like this: - -misdn send display mISDN/1/101 "Hello World!" - -where 1 is the Port of the Card where the phone is plugged in, and 101 is the -msn (callerid) of the Phone to send the text to. - -\subsection{mISDN Variables} - -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. -\end{verbatim} - - -\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 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 -tracker at \url{https://issues.asterisk.org} in the "channel drivers" project, -"chan\_misdn" category. Read the bug guidelines to make sure you -provide all the information needed. - - -\subsection{Examples} - -Here are some examples of how to use chan\_misdn in the dialplan -(extensions.conf): - -\begin{astlisting} -\begin{verbatim} -[globals] -OUT_PORT=1 ; The physical Port of the Card -OUT_GROUP=ExternE1 ; The Group of Ports defined in misdn.conf - -[misdnIn] -exten => _X.,1,Dial(mISDN/${OUT_PORT}/${EXTEN}) -exten => _0X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1}) -exten => _1X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1}/:dHello) -exten => _1X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1}/:dHello Test:n) -\end{verbatim} -\end{astlisting} - -On the last line, you will notice the last argument (Hello); this is sent -as Display Message to the Phone. - -\subsection{Known Problems} - -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. |