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---  ******* IMPORTANT NOTE ***********
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---  This module is currently unsupported.  Use it at your own risk.
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 Copyright (C) 2005, Objective Systems, Inc.
 
 Objective Systems asterisk-ooh323c driver for supporting H323 on asterisk

 Thank you for downloading Objective Systems H323 driver for asterisk. This 
 package contains a driver which will enable ooh323c, a C based open source 
 H.323 stack from Objective Systems(www.obj-sys.com/open), to be used with 
 asterisk, an open source IP-PBX solution, from Digium, Inc.(www.asterisk.org)

 Capabilities supported: ulaw, gsm, g728, g729, g729a, g723.1, rfc2833

 Package Contents:

    asterisk-ooh323c-0.5
    |
    |-src [H323 channel driver code]
    |
    |_ooh323c [ooh323c stack code]


 How to build:
 -------------
 1. If you hadn't installed asterisk yet, To build this package successfully, 
    you have to have latest version of asterisk installed on your system. 
    You can skip to step 2, if you have already installed the asterisk.

      Following precedure help you download and install asterisk 1.2.x

      To get latest asterisk sources from svn branch 1.2:
      >svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2

      To install asterisk.(You will have to be in 'super user' mode for this)
      >cd asterisk
      >make install
      >make samples

 2. To build asterisk-ooh323c addons

      To update or if you hadn't downloaded the asterisk-ooh323c channel driver
      code, you can do so using following command (To get the latest 
      asterisk-addons 1.2 branch source from svn):
      >svn checkout http://svn.digium.com/svn/asterisk-addons/branches/1.2 asterisk-addons-1.2

      To build & install:
      >cd asterisk-addons-1.2/asterisk-ooh323c
      >./configure
      >make

      For debugging purpose, instead of "make", run "make debug"

      The library will be generated at 
      asterisk-addons-1.2/asterisk-ooh323c/.libs/libchan_h323.so.1.0.1 
 
      To install libchan_h323.so in /usr/libs/modules/ directory, change to 
      'super user' mode and then run:
      >make install

 3. Open h323.conf.sample in asterisk-ooh323c-0.5 directory. Under [general] 
    you will see global configuration setting. Modify IP addresses of asterisk 
    server "bindaddr" to match your configuration.

 4. To install sample h323.conf in /etc/asterisk
      >make sample

 5. Now run asterisk as
      >/usr/sbin/asterisk -vvvc

 6. From asterisk console
    To check whether H323 channel is registered properly
       CLI>show channeltypes

    To see all the defined H323 users.
       CLI>ooh323 show users
   
    To see details of a specific user
       CLI>ooh323 show user myuser1

    To see all the defined peers
       CLI>ooh323 show peers
    
    To see details of a specific peer
       CLI>ooh323 show peer mypeer1


Getting Started with some simple examples:
-----------------------------------------
 First stop asterisk as follows:
      CLI>stop now

 Now to run tests, you will have to copy extensions.conf.sample in 
 asterisk-ooh323c-0.5 directory to /etc/asterisk as extensions.conf. Make
 sure to save a backup copy of your existing extensions.conf. Now restart
 the pbx as follows:
      >/usr/sbin/asterisk -vvvc

 Now you have asterisk with our sample dial plan running.     

 Test1: 
--------
    Someone from outside calls into asterisk server and a recorded message is
    playedback to him.

 1. Now you need an h323 phone(phone1) to test. You can use any H323 phones 
    including NetMeeting or OpenH323 based ohphone.
    You can download ohphone executable
    from http://www.openh323.org/code.html#linux    or 
         http://www.openh323.org/code.html#windows

 2. Now run ohphone as follows:
     >ohphone -n -ttttt -o trace.log <ip address of asterisk server>

 Test2:
--------
   A registered users calls into asterisk server at a particular extension and
   a different messages is played back to him.

  1. Use ohphone as a registered user myuser1 to call extension 100 in 
      asterisk as follows:

      >./ohphone -n -u myuser1 -tttt -o trace.log 100@<ip of asterisk server>

      The ohphone will send call to asterisk, which will identify the user as 
      myuser1, and use the users context to handle extension 100, which 
      playbacks a pre-recorded message.

 Test3:
--------
  A registered user calls an extension which belongs to a peer and the call is
  routed to the peer's phone

 1. Use one more h323 phone (phone2)as a peer.

 2. Stop asterisk as:
       CLI>stop now

 3. Open /etc/asterisk/h323.conf. Modify ip/port of mypeer1 to match those of
     the h323 phone you are going to use as peer.

 4. Run your peer phone phone2 as follows:

     >./ohphone -n -u mypeer1 -tttttt -o trace.log -l

     This keeps phone2 in listen mode.

 5. Now run asterisk as follows:
    
     >/usr/sbin/asterisk -vvvc

 6. Check details of peer for ip/port
    
     CLI>ooh323 show peer mypeer1

 7. Now use phone1 to make a call to asterisk at extension 101 as follows.

     >./ohphone -n -u myuser1 -ttttt -o trace.log 101@<ip of asterisk>
 

Reporting Problems:

If you have any further questions or comments on what you would like to
see in the product or what is difficult to use or understand, please 
communicate them to us. Your feedback is important to us. Please let us
know how it works out for you - either good or bad. 

For support please use ooh323c mailing list at http://lists.sourceforge.net/lists/listinfo/ooh323c-devel