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-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 it)
-* app_SendText
-* HOLD/RETRIEVE/TRANSFER on ISDN phones : )
-* Allow/restrict user number presentation
-* Volume control
-* Crypting with mISDNdsp (Blowfish)
-* Data (HDLC) callthrough
-* Data calling (with app_ptyfork +pppd)
-* Echo cancellation
-* Call deflection
-* Some others
-
-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. This can be done by:
- * You can download latest stable releases from http://www.misdn.org/downloads
- * Just fetch the newest head of the GIT (mISDN project moved from CVS)
- In details this process described here: http://www.misdn.org/index.php/GIT
-
-then 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 ;
-./configure && make && make install
-
-That's all!
-
-
-Follow the instructions in the mISDN Package for how to load the Kernel
-Modules. Also install process described in http://www.misdn.org/index.php/Installing_mISDN
-
-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
-
-
-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" 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.
-
-- 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.
-
-- 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).
-
-
-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>[: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
-
-
-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 (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.
-
-
-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 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 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
-