Difference between revisions of "Assignment 2: H323 (by Fabio & Anneke)"

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We use '''gnomemeeting''' softphones and a '''gnugk''' gatekeeper.
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<big>'''2.1 H323'''</big>
=== Configuration of Gnomemeeting ===
 
'''Edit -> Preferences'''
 
  
* General: You can fill in your personal data and leave everything default
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'''[[H323 analysis]]'''
* H323 Settings: Gatekeeper:
 
  
Registering method: Gatekeeper host
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'''[[Our Architecture]]'''
Gatekeeper host: 129.125.71.173:1719
 
Gatekeeper alias: 601
 
 
''Gatekeeper host'' is the IP address and port of the gatekeeper. Port 1720 is the standard h323 port but in our case the h323 client was on the same computer so the 1720 port was already occupied.
 
  
''Gatekeeper alias'' is the primary alias with the gatekeeper, also the extension number in Asterisk to keep things simple.
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'''[[HOWTO SET UP TRIXBOX AS H323 GATEWAY]]'''
  
Tick the "Register this alias as the primary alias with the gatekeeper" box and Apply (sorry, this is of course after you've installed the gk) :).
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<big>'''2.2 Connection to regular telephone network'''</big>
* H323 Settings: Gateway - Do not use a gateway
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* [[VoIPBuster]]
* Codecs: Audio Codecs - We enabled all available codecs.
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* [[ZapTel]]
* Codecs: Video Codecs - We have not been able to spend time on this
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<big>'''2.3 Configuring videoconferencing'''</big>
* Devices:
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* [[Videoswitch support for MeetMe]]
** Audio Devices:
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* [[app_conference]]
*** Audio Plugin ALSA
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<big>'''2.4 Intercom'''</big>
*** Output device: Intel ICH5
 
*** Input device: Intel ICH5
 
**'''Detect Devices'''
 
  
(All the audio and video devices settings are of course depending on your local system, this is more for our own reference) :)
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<big>'''2.5 Updating to newer versions of Trixbox'''</big>
 
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* [[Regarding H323]]
=== GnuGk Gatekeeper ===
 
Packages for this gatekeeper are not available for CentOS. We found out that this gatekeeper was already installed in Etch. (System administration provided us with two machines with etch so we could get the sound working).
 
 
 
Almost no configuring here. Just enable PROXY.
 
 
 
We are still playing with the gatekeeper so this section is under construction.
 
 
 
=== Configuration of Trixbox 1.2.3 ===
 
 
 
We followed mostly the instructions from these two forums on the [http://www.trixbox.org/ Trixbox] site:
 
* [http://www.trixbox.org/modules/newbb/viewtopic.php?viewmode=flat&order=ASC&topic_id=2865&forum=6 here is the working configuration & tips for oh323]
 
* [http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=3228&forum=2#forumpost41741 H323, SIP and Gatekeeper]
 
 
 
There are more forums where people say things about h323 but these seem to be the most helpful.
 
 
 
Following the suggestions we used Trixbox 1.2.3 with asterisk-addons-1.2.3. In order to do this we deleted the current addons:
 
 
 
rpm -qa | grep asterisk-addons
 
rpm -e asterisk-addons-1.2.4_1.2.12.1-1.294
 
 
 
And loaded the older addons: (these can be found ...............)
 
 
 
rpm -i asterisk-addons-1.2.3-1.219.i386.rpm
 
 
 
If this is really necessary we can't tell right now. We got things working when changing other things as well so later on we can try if it still works without this step.
 
 
 
==== ooh323.conf ====
 
 
 
Next thing was to edit the ooh323.conf file. This file needs to be copied from the samples first:
 
cp /etc/asterisk-1.2.8-samples/ooh323.conf /etc/asterisk
 
 
 
Here is our ooh323.conf:
 
 
 
; Objective System's H323 Configuration example for Asterisk
 
; ooh323c driver configuration
 
;
 
; [general] section defines global parameters
 
;
 
; This is followed by profiles which can be of three types - user/peer/friend
 
; Name of the user profile should match with the h323id of the user device.
 
; For peer/friend profiles, host ip address must be provided as "dynamic" is
 
; not supported as of now.
 
;
 
; Syntax for specifying a H323 device in extensions.conf is
 
; For Registered peers/friends profiles:
 
;        OOH323/name where name is the name of the peer/friend profile.
 
;
 
; For unregistered H.323 phones:
 
;        OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be any H323
 
;                          alias
 
;
 
; For dialing into another asterisk peer at a specific exten
 
;      OOH323/exten/peer OR OOH323/exten@ip
 
;
 
; Domain name resolution is not yet supported.
 
;
 
; When a H.323 user calls into asterisk, his H323ID is matched with the profile
 
; name and context is determined to route the call
 
;
 
; The channel driver will register all global aliases and aliases defined in
 
; peer profiles with the gatekeeper, if one exists. So, that when someone
 
; outside our pbx (non-user) calls an extension, gatekeeper will route that
 
; call to our asterisk box, from where it will be routed as per dial plan.
 
 
 
[general]
 
;Define the asetrisk server h323 endpoint
 
 
;The port asterisk should listen for incoming H323 connections.
 
;Default - 1720
 
;port=1720
 
;CHANGED
 
port=1720
 
 
;The dotted IP address asterisk should listen on for incoming H323
 
;connections
 
;Default - tries to find out local ip address on it's own
 
bindaddr=0.0.0.0
 
 
;This parameter indicates whether channel driver should register with
 
;gatekeeper as a gateway or an endpoint.
 
;Default - no
 
;gateway=no
 
;CHANGED
 
gateway=yes
 
 
;Whether asterisk should use fast-start and tunneling for H323 connections.
 
;Default - yes
 
;faststart=no
 
h245tunneling=no
 
;CHANGED
 
;h245tunneling=yes
 
 
 
;H323-ID to be used for asterisk server
 
;Default - Asterisk PBX
 
h323id=ObjSysAsterisk
 
;e164=100
 
;CHANGED
 
e164=501
 
 
;CallerID to use for calls
 
;Default - Same as h323id
 
callerid=asterisk
 
 
;Whether this asterisk server will use gatekeeper.
 
;Default - DISABLE
 
;gatekeeper = DISCOVER
 
;gatekeeper = DISABLE
 
;CHANGED
 
gatekeeper = 129.125.71.173
 
 
;Location for H323 log file
 
;Default - /var/log/asterisk/h323_log
 
;CHANGED
 
logfile=/var/log/asterisk/h323_log
 
 
 
;Following values apply to all users/peers/friends defined below, unless
 
;overridden within their client definition
 
 
 
;Sets default context all clients will be placed in.
 
;Default - default
 
context=default
 
 
;Sets rtptimeout for all clients, unless overridden
 
;Default - 60 seconds
 
;rtptimeout=60      ; Terminate call if 60 seconds of no RTP activity
 
                    ; when we're not on hold
 
;CHANGED
 
rtptimeout=3
 
;Type of Service
 
;Default - none (lowdelay, thoughput, reliability, mincost, none)
 
;tos=lowdelay
 
 
;amaflags = default
 
 
;The account code used by default for all clients.
 
;accountcode=h3230101
 
 
;The codecs to be used for all clients.Only ulaw and gsm supported as of now.
 
;Default - ulaw
 
; ONLY ulaw, gsm, g729 and g7231 supported as of now
 
;disallow=all    ;Note order of disallow/allow is important.
 
;allow=gsm
 
;allow=ulaw
 
;;CHANGED
 
;allow=g729
 
;allow=g723
 
 
;CHANGED
 
allow=all
 
 
; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad
 
; h245alphanumeric, h245signal.
 
;Default - rfc 2833
 
dtmfmode=rfc2833
 
 
; User/peer/friend definitions:
 
; User config options                    Peer config options
 
; ------------------                    -------------------
 
; context
 
; disallow                              disallow
 
; allow                                  allow
 
; accountcode                            accountcode
 
; amaflags                              amaflags
 
; dtmfmode                              dtmfmode
 
; rtptimeout                            ip
 
;                                        port
 
;                                        h323id
 
;                                        email
 
;                                        url
 
;                                        e164
 
;                                        rtptimeout
 
 
;
 
 
;Define users here
 
;Section header is extension
 
[myuser1]
 
type=user
 
context=context1
 
disallow=all
 
allow=gsm
 
allow=ulaw
 
 
 
 
[mypeer1]
 
type=peer
 
context=context2
 
ip=a.b.c.d  ; UPDATE with appropriate ip address
 
port=1720    ; UPDATE with appropriate port
 
e164=101
 
 
 
 
;[myfriend1]
 
 
[friend-Anneke]
 
type=friend
 
;CHANGED
 
context=from-internal
 
ip=129.125.71.172  ; UPDATE with appropriate ip address
 
;port=1820          ; UPDATE with appropriate port
 
port=1720
 
disallow=all
 
;allow=ulaw
 
;CHANGED
 
allow=gsm
 
allow=ulaw
 
allow=g729
 
allow=g723
 
 
;CHANGED
 
;e164=12345
 
e164=601
 
 
rtptimeout=3
 
dtmfmode=rfc2833
 
 
 
[friend-Fabio]
 
type=friend
 
context=default
 
ip=129.125.71.173  ; UPDATE with appropriate ip address
 
;port=1820          ; UPDATE with appropriate port
 
port=1720
 
disallow=all
 
;allow=ulaw
 
;CHANGED
 
allow=gsm
 
allow=ulaw
 
allow=g729
 
allow=g723
 
 
;CHANGED
 
;e164=12345
 
 
rtptimeout=3
 
dtmfmode=rfc2833
 
 
;[601]
 
;type=peer
 
;context=default
 
;ip=129.125.71.173:1719
 
;e164=100
 
;OH323/129.125.71.173:1719
 
;gatekeeper=ENABLE
 
;OOH323/601
 
 
 
 
 
(Still needs some cleaning up, will do later)
 
 
 
Comments on this file:
 
* All parameters that have been changed by us are preceded with a comment line ''';CHANGED'''
 
* The Trixbox is going to act as gateway for the gatekeeper hence '''gateway = yes'''
 
* h245tunneling still under investigation
 
* For some reason it seems that you have to set e164 to the SIP extension number ??? (how to handle multiple extension???)
 
* We will be using a gatekeeper so '''gatekeeper = 129.125.71.173''' (ip-address of the gatekeeper)
 
* The user profile section is a bit messy right now
 
 
 
==== Adding H323 extensions ====
 
 
 
A H323 extension can be added in FreePBX as a custom extension. For example:
 
Displayname: Anneke h323
 
Dial: 600
 
 
 
Asterisk saves these extensions in the file
 
 
 
 
 
 
 
 
 
amportal stop
 
amportal start
 
 
then it works perfect.
 

Latest revision as of 15:55, 16 March 2007

2.1 H323

H323 analysis

Our Architecture

HOWTO SET UP TRIXBOX AS H323 GATEWAY

2.2 Connection to regular telephone network

2.3 Configuring videoconferencing

2.4 Intercom

2.5 Updating to newer versions of Trixbox