267 lines
8.3 KiB
TeX
267 lines
8.3 KiB
TeX
\subsection{Introduction}
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This package contains the mISDN Channel Driver for the Asterisk PBX. It
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supports every mISDN Hardware and provides an interface for asterisk.
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\subsection{Features}
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\begin{itemize}
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\item NT and TE mode
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\item PP and PMP mode
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\item BRI and PRI (with BNE1 and BN2E1 Cards)
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\item Hardware Bridging
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\item DTMF Detection in HW+mISDNdsp
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\item Display Messages on Phones (on those that support display msg)
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\item app\_SendText
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\item HOLD/RETRIEVE/TRANSFER on ISDN Phones : )
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\item Screen/ Not Screen User Number
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\item EchoCancellation
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\item Volume Control
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\item Crypting with mISDNdsp (Blowfish)
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\item Data (HDLC) callthrough
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\item Data Calling (with app\_ptyfork +pppd)
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\item Echo cancellation
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\item CallDeflection
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\item Some other
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\end{itemize}
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\subsection{Fast Installation Guide}
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It is easy to install mISDN and mISDNuser. This can be done by:
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\begin{itemize}
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\item You can download latest stable releases from \url{http://www.misdn.org/downloads/}
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\item Just fetch the newest head of the GIT (mISDN provect moved from CVS)
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In details this process described here: \url{http://www.misdn.org/index.php/GIT}
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\end{itemize}
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then compile and install both with:
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\begin{verbatim}
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cd mISDN ;
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make && make install
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\end{verbatim}
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(you will need at least your kernel headers to compile mISDN).
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\begin{verbatim}
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cd mISDNuser ;
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make && make install
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\end{verbatim}
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Now you can compile chan\_misdn, just by making asterisk:
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\begin{verbatim}
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cd asterisk ;
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./configure && make && make install
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\end{verbatim}
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That's all!
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Follow the instructions in the mISDN Package for how to load the Kernel
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Modules. Also install process described in \url{http://www.misdn.org/index.php/Installing_mISDN}
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\subsection{Pre-Requisites}
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To compile and install this driver, you'll need at least one mISDN Driver and
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the mISDNuser package. Chan\_misdn works with both, the current release version
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and the development (svn trunk) version of Asterisk. mISDNuser and mISDN must
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be fetched from cvs.isdn4linux.de.
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You should use Kernels $>$= 2.6.9
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\subsection{Configuration}
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First of all you must configure the mISDN drivers, please follow the
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instructions in the mISDN package to do that, the main config file and config
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script is:
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\begin{verbatim}
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/etc/init.d/misdn-init and
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/etc/misdn-init.conf
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\end{verbatim}
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Now you will want to configure the misdn.conf file which resides in the
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asterisk config directory (normally /etc/asterisk).
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\subsubsection{misdn.conf: [general]}
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The misdn.conf file contains a "general" subsection, and user subsections which
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contain misdn port settings and different Asterisk contexts.
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In the general subsection you can set options that are not directly port
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related. There is for example the very important debug variable which you can
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set from the Asterisk cli (command line interface) or in this configuration
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file, bigger numbers will lead to more debug output. There's also a tracefile
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option, which takes a path+filename where debug output is written to.
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\subsubsection{misdn.conf: [default] subsection}
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The default subsection is another special subsection which can contain all the
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options available in the user/port subsections. the user/port subsection inherit
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their parameters from the default subsection.
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\subsubsection{misdn.conf: user/port subsections}
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The user subsections have names which are unequal to "general". Those subsections
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contain the ports variable which mean the mISDN Ports. Here you can add
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multiple ports, comma separated.
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Espacially for TE-Mode Ports there is a msns option. This option tells the
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chan\_misdn driver to listen for incoming calls with the given msns, you can
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insert a '*' as single msn, which leads in getting every incoming call (if you
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want to share on PMP TE S0 with a asterisk and a phone or isdn card you should
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insert here the msns which you'll like to give the Asterisk). Finally a
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context variable resides in the user subsections, which tells chan\_misdn where to
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send incoming calls to in the Asterisk dial plan (extension.conf).
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\subsubsection{Dial and Options String}
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The dial string of chan\_misdn got more complex, because we added more features,
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so the generic dial string looks like:
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\begin{verbatim}
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mISDN/<port>|g:<group>/<extension>[/<OPTIONSSTRING>]
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The Optionsstring looks Like:
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:<optchar1><OptParam1>:<optchar2><OptParam2>
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the ":" character is the delimiter.
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The available Optchars are:
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d - Send display text on called phone, text is the optparam
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n - don't detect dtmf tones on called channel
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h - make digital outgoing call
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c - make crypted outgoing call, param is keyindex
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e - perform echo cancellation on this channel,
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takes taps as arguments (32,64,128,256)
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s - send Non Inband DTMF as inband
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vr - rxgain control
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vt - txgain control
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\end{verbatim}
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chan\_misdn registers a new dial plan application "misdn\_set\_opt" when
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loaded. This application takes the Optionsstring as argument. The Syntax is:
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\begin{verbatim}
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misdn_set_opt(<OPTIONSSTRING>)
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\end{verbatim}
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When you set options in the dialstring, the options are set in the external
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channel. When you set options with misdn\_set\_opt, they are set in the current
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incoming channel. So if you like to use static encryption, the scenario looks
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as follows:
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\begin{verbatim}
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Phone1 --> * Box 1 --> PSTN_TE
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PSTN_TE --> * Box 2 --> Phone2
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\end{verbatim}
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The Encryption must be done on the PSTN sides, so the dialplan on the boxes
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are:
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\begin{verbatim}
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* Box 1:
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exten => _${CRYPT_PREFIX}X.,1,Dial(mISDN/g:outbound/:c1)
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* Box 2:
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exten => ${CRYPT_MSN},1,misdn_set_opt(:c1)
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exten => ${CRYPT_MSN},2,dial(${PHONE2})
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\end{verbatim}
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\subsection{mISDN CLI commands}
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At the Asterisk cli you can try to type in:
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\begin{verbatim}
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misdn <tab> <tab>
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\end{verbatim}
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Now you should see the misdn cli commands:
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\begin{verbatim}
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- clean
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-> pid (cleans a broken call, use with care, leads often
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to a segmentation fault)
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- send
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-> display (sends a Text Message to a Asterisk channel,
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this channel must be an misdn channel)
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- set
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-> debug (sets debug level)
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- show
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-> config (shows the configuration options)
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-> channels (shows the current active misdn channels)
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-> channel (shows details about the given misdn channels)
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-> stacks (shows the current ports, their protocols and states)
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-> fullstacks (shows the current active and inactive misdn channels)
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- restart
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-> port (restarts given port (L2 Restart) )
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- reload (reloads misdn.conf)
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\end{verbatim}
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You can only use "misdn send display" when an Asterisk channel is created and
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isdn is in the correct state. "correct state" means that you have established a
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call to another phone (mustn't be isdn though).
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Then you use it like this:
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misdn send display mISDN/1/101 "Hello World!"
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where 1 is the Port of the Card where the phone is plugged in, and 101 is the
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msn (callerid) of the Phone to send the text to.
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\subsection{mISDN Variables}
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mISDN Exports/Imports a few Variables:
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\begin{verbatim}
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- MISDN_ADDRESS_COMPLETE : Is either set to 1 from the Provider, or you
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can set it to 1 to force a sending complete.
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\end{verbatim}
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\subsection{Debugging and sending bug reports}
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If you encounter problems, you should set up the debugging flag, usually
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debug=2 should be enough. the messages are divided in asterisk and misdn
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parts. Misdn Debug messages begin with an 'I', asterisk messages begin with
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an '*', the rest is clear I think.
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Please take a trace of the problem and open a report in the Asterisk issue
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tracker at \url{http://bugs.digium.com} in the "channel drivers" project,
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"chan\_misdn" category. Read the bug guidelines to make sure you
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provide all the information needed.
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\subsection{Examples}
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Here are some examples of how to use chan\_misdn in the dialplan
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(extensions.conf):
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\begin{verbatim}
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[globals]
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OUT_PORT=1 ; The physical Port of the Card
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OUT_GROUP=ExternE1 ; The Group of Ports defined in misdn.conf
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[misdnIn]
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exten => _X.,1,Dial(mISDN/${OUT_PORT}/${EXTEN})
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exten => _0X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1})
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exten => _1X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1}/:dHello)
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exten => _1X.,1,Dial(mISDN/g:${OUT_GROUP}/${EXTEN:1}/:dHello Test:n)
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\end{verbatim}
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On the last line, you will notice the last argument (Hello); this is sent
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as Display Message to the Phone.
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\subsection{Known Problems}
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Q: I cannot hear any tone after a successful CONNECT to the other end
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A: You forgot to load mISDNdsp, which is now needed by chan\_misdn for switching
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and dtmf tone detection
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