path: root/doc/misdn.txt

mISDN Channel Driver for Asterisk PBX
======================================


This package contains the mISDN Channel Driver for the Asterisk PBX. It 
supports every mISDN Hardware and provides an interface for asterisk. 

Features: 
---------

* NT and TE mode
* PP and PMP mode
* BRI and PRI (with BNE1 and BN2E1 Cards)
* Hardware Bridging
* DTMF Detection in HW+mISDNdsp 
* Display Messages on Phones (on those that support display msg)
* app_SendText
* HOLD/RETRIEVE/TRANSFER on ISDN Phones : )
* Screen/ Not Screen User Number
* EchoCancellation 
* Volume Control 
* Crypting with mISDNdsp (Blowfish)
* Data (HDLC) callthrough 
* Data Calling (with app_ptyfork +pppd)
* Echo cancellation
* CallDeflection
* Some other

Supported Hardware:
-------------------

chan_misdn supports any mISDN compatible Hardware.

Overview
--------

- Fast Installation Guide
- Pre-Requisites
- Configuration
- Dial and Options String
- misdn cli commands
- mISDN Variables
- Debugging and sending Bugreports
- Examples
- Known Problems
- Changes


Fast Installation Guide
-----------------------

It is easy to install mISDN and mISDNuser. Just fetch the newest head of the
cvs, this can be done by:

cvs -d:pserver:anonymous:readonly@cvs.isdn4linux.de:/i4ldev co mISDN mISDNuser

the compile and install both with:

cd mISDN ;
make && make install

(you will need at least your kernel headers to compile mISDN).

cd mISDNuser ;
make && make install

Now you can compile chan_misdn, just by making asterisk: 

cd asterisk ; 
make && make install

That's all!


Follow the instructions in the mISDN Package for howto loading the Kernel
Modules.

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. mISDNuser and mISDN must
be fetched from cvs.isdn4linux.de.

You should use Kernels >= 2.6.9 


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:

/etc/init.d/misdn-init  and
/etc/misdn-init.conf


Now you will want to configure the misdn.conf file which resides in the
asterisk config directory (normally /etc/asterisk).

- misdn.conf: [general]
The misdn.conf file contains a "general" Section, and user sections which
contain misdn port settings and different Asterisk contexts.

In the general Section 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
option, which takes a path+filename where debug output is written to.

- misdn.conf: [default] section 

The default section is another special section which can contain all the
options available in the user/port sections. the user/port section inherit
their parameters from the default section.

- misdn.conf: user/port sections

The user sections have names which are unequal to "general". Those sections
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
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 sections, which tells chan_misdn where to
send incoming calls to in the Asterisk dial plan (extension.conf).


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:

mISDN/<port>|g:<group>/<extension>[/<OPTIONSSTRING>]

The Optionsstring looks Like:
:<optchar1><OptParam1>:<optchar2><OptParam2>

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


chan_misdn registers a new dial plan application "misdn_set_opt" when
loaded. This application takes the Optionsstring as argument. The Syntax is:

misdn_set_opt(<OPTIONSSTRING>)


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:

Phone1 --> * Box 1 --> PSTN_TE 
PSTN_TE --> * Box 2 --> Phone2

The Encryption must be done on the PSTN sides, so the dialplan on the boxes
are:

* 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})




misdn cli commands
------------------

At the Asterisk cli you can try to type in: 

misdn <tab> <tab>

Now you should see the misdn cli commands: 

- 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)

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).

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.


mISDN Variables
---------------

mISDN Exports/Imports a few Variables: 

- MISDN_ADDRESS_COMPLETE : 	Is either set to 1 from the Provider, or you 
				can set it to 1 to force a sending complete.



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
an '*', the rest is clear I think.

Please take a trace of the problem and open a report in the Asterisk issue
tracker at http://bugs.digium.com in the "channel drivers" project,
"chan_misdn" category. Read the bug guidelines to make sure you
provide all the information needed.


Examples
--------

Here are some examples of how to use chan_misdn in the dialplan
(extensions.conf):


[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)

On the last line, you will notice the last argument (Hello); this is sent
as Display Message to the Phone.

Known Problems
--------------

* When I use mISDN->IAX I cannot make Trunk calls

-> You need to use ztdummy as dummy zaptel interface for the iax timing in
trunking mode, simply grab libpri, zaptel and compile them (i think you need
to modify the makefile in zaptel to add ztdummy to the default compiled
modules) then modprobe ztdummy, this resolves the problem.


* 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
and dtmf tone detection


Changes
-------
in the Changes File