Dears;
I am having same problem, that when I place a call from the H323 end point
(even if it is not added in the ooh323.conf), then asterisk handle the call and
play the wave file in the default context. Also I added endpoint to the
ooh323.conf and same thing, it keep goes for default context whatever the
context placed.
My Asterisk vesion is 1.4.25
My Asterisk add-on version is: 1.4.8
What I have to do to capture the call from the IP Phone and route it using the
correct context that I configured it in the [ ] of the ooh323.conf? Any
specific thing need to be done?
Regards
Bilal
--------------
> > Hi guys,
> >
> > I'm trying out ooh323 and couldn't bridge ooh323 and
> sip/zap.
> > I'm using netmeeting and set gateway to my asterisk.
> >
> > Here's my CLI dump:
> >
> > == Spawn extension (h323, 9999, 1)
> exited non-zero on
> > 'OOH323/(null)-8c76'
> > -- Executing [9...@h323:1]
> Dial("OOH323/(null)-3074",
> > "Zap/8/604xxxxxxx") in new stack
> > -- Called 8/604xxxxxxx
> > -- Zap/8-1 is ringing
> > [2008-07-02 15:48:55] WARNING[21544]: channel.c:2390
> ast_indicate_data:
> > Unable to handle indication 3 for
> 'OOH323/(null)-3074'
> > -- Zap/8-1 is ringing
> > -- Zap/8-1 answered
> OOH323/(null)-3074
> > [2008-07-02 15:49:08] WARNING[21544]:
> chan_ooh323.c:1053
> > ooh323_indicate: Don't know how to indicate condition
> 20 on ooh323c_5
> >
> > My ooh323.conf:
> >
> > [general]
> > bindaddr=192.168.1.9
> > h323id=ObjSysAsterisk
> > e164=100
> > callerid=asterisk
> > gatekeeper = DISABLE
> > gateway = yes
> > context = h323
> > disallow = all
> > allow = ulaw
> > dtmfmode = rfc2833
> >
> >
> > extensions.conf
> > [h323]
> > Exten => 9999,1,Dial(Zap/8/604xxxxxxx)
> > Exten => 9999,n,Hangup
> >
> > 604xxxxxxx goes to my cell. it rings fine but no
> audio. After I picked
> > up from cell, netmeeting still shows "watiting for
> 9999 to answer"
> > message.
> >
> > Any ideas?
>
> I don't like the look of the (null) in the channel names.
>
> If what you quoted was the whole of your ooh323.conf file,
> you don't have
> any peer, user or friend sections. Try adding something
> like:
>
> [h323gw]
> type=friend
> context=h323
> ip=192.168.1.200 (or
> whatever the IP of your remote H323 endpoint is)
> port=1720
>
> If that still doesn't help, please mention what versions of
> asterisk and
> asterisk-addons you are using.
>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: t...@softins.co.uk
> - http://www.softins.co.uk
> Play: t...@mountifield.org
> - http://tony.mountifield.org
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