H.323 for trixbox

i've never been able to get this result!!! I'd like to make something like this and to find h323 in extensions to let me callfromh323 to sip and reverse mode.

HELP!!!

tnx a lot
grazios72

Just one quick thing guys: I wanted to install GnuGK as an h323 gatekeeper. I trried the pwlib /openh323 / gnugk compile but got a zillion errors! I downloaded the old asterisk@home addon and, after a few restarts all is fine now. Did I do anything wrong? I just hope that installing the asteriskathome-h323-1.0.zip didn't messup trixbox's existing h323 implementation!

:-o

Get back to you AS soon as I've got it working! At present, my H323 works between terminals and from terminals to the pots (ISDN pri card) and to asterisk extensions.

The directory /etc/asterisk-1.2.7.1_samples on my system is empty. Where can I get these files?

Hi,
Could you tell me what should I do to use some H.323 terminal (ex. NetMeetiug) as an external extension number. I mean how to configure the chanel to register H.323 phone as for example 4000 extension and to have a possibility to call that ext form some SIP ext.

Do I need some Gatekeeper? Or where should I use appropriate configuration?

Currently I'm using Asterisk@Home v.1.2.4 and ooh323 channel.
I succeded to setup H323 -> SIP call.

Thanks in advance.

I too have been struggling to get H323 to work. We have a few older systems out there (most around AAH 1.5 from memory) where you could download and install the addon pack no problems.

We need to connect to a provider via H323 for out-going toll calls, however I can't get ooh323 configured correctly, though I can at least install it. The other route I took was trying to compile new pwlib / oh323 and get the old version that worked with AAH going.

Nothing seems to work and there is very little if any information floating around about setting up H323 as trunks to providers.

Hi there,

Is it possible to post your h323.conf file and a sample of dialplan for setting the H323 trunk? I have searched everywhere but I couldn't find any sample configuration.
Any help would be appraciated as now hitting head on the wall.

Thanks.

Hello,
I have same your problems on port 1720 and my configuration 00h323.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
;port=3729

;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=10.0.0.115

;This parameter indicates whether channel driver should register with
;gatekeeper as a gateway or an endpoint.
;Default - no
gateway=no

;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes

faststart=yes
h245tunneling=yes

;H323-ID to be used for asterisk server
;Default - Asterisk PBX
h323id=ObjSysAsterisk
e164=100

;CallerID to use for calls
;Default - Same as h323id
callerid=asterisk

;Whether this asterisk server will use gatekeeper.
;Default - DISABLE
;gatekeeper = DISCOVER

;gatekeeper = DISABLE

;Location for H323 log file
;Default - /var/log/asterisk/h323_log
;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 - 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

;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=g729
allow=gsm
allow=ulaw
allow=g723

; 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=default
;disallow=all
;allow=gsm
;allow=ulaw
;allow=g729

[505]
type=peer
context=default
ip=10.0.0.112 ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
e164=100
OH323/10.0.0.112:1720
gatekeeper=ENABLE
OOH323/505

;[myfriend1]
;type=friend
;context=default
;ip=10.0.0.82 ; UPDATE with appropriate ip address
;port=1820 ; UPDATE with appropriate port
;disallow=all
;allow=ulaw
;e164=12345
;rtptimeout=60
;dtmfmode=rfc2833

I have exactly the same problem, sip to h323 no problem, only able to make one call for h323 to sip and then server hangs, I'm still able to make sip to sip calls but the ooh323 seems to have crashed. I have to restart trixbox to make another call? Has anybosy else had similar problems?

Thanks in Advance

Hi
can we get free pbx to work with h323 and record all the call through h323 like in free PBX ??

does this have problems in 729 and 723 codecs ??

Hi Carlo.

Can you send me the scrip that you make to fix the problem using ooh323. I have the same problem. My asterisk crash when I make a call using h323.

My email is dmonzontsd@hotmail.com

thank you

Danil

I have basic call functionality between TB 2.2 and Avaya including dtmf, using H.323.

** the only items I've not completed yet is turning MWI (red light) on Avaya extensions and using Avaya as a tandem switch to reach the pstn.

If anyone is interested in Asterisk to Avaya setup, respond and I'll post general settings...

or

email = matt@tycal.com
aim= matt mitchell

I'm about ready to start testing with IP office. So far I've only worked this with the G3r. I'll let you know how I do if your still interested...

Once I installed the G729 codec, got ooh323 (the one in asterisk addons) to compile without errors, and fiddled a little with the IP Office I now have a working connection. I did have a few small challenges. Calls would ring and be disconnected on answer. Turned out to be codec related. The link could not negotiate a common codec.

For the record, in the IP Office on the Line VoIP setting select g729 as the codec. Disable H450 and check fast start. Don't forget to setup the short code in both the Line and the Short Codes section. In asterisk add the asterisk IP Address to ooh323.conf and create a custom trunk and dial string (OOH323/$OUTNUM$@XX.XX.XX.XX:1720).

That's it!! If you want my configs let me know.

Hi mmitchel,

I used g729 because TB and the IPO couldn't or wouldn't negotiate a codec when set to auto negotiate on the IPO. Initially the call didn't succeed, but after setting both ends to g729 TB -> IPO worked but IPO -> TB didn't work because of short codes (see my previous post). Once I fixed that the call cleared on answer.

I can call from IPO -> TB and use the IPO as a PSTN gateway, ie TB -> IPO -> PSTN. I have only tested call duration TB -> IPO and the call was not torn down until I hung up one end. I had the call up for 20 minutes.

I have only use an IAX softphone on TB but expect to be able to test SIP soon. (Currently only run IAX softphones.)

Cheers

http://www.trixbox.org/forums/trixbox-forums/h-323/here-h323-pack...