On Fri, 2007-02-23 at 11:47 +0000, Tim King wrote:
> Probably a codec problem. Try only having one codec choice (and make
> sure that it is supported by both ends)

It's bad, confusing and undocumented that calls hang up just after being
answered when you've got "allow=all".

I've spent 2 days trying to get H.323 working on Asterisk, chan_h323
doesn't and can't support g723 and g729 because it needs OpenH323 codecs
which don't exist, chan_oh323 is no longer developed and doesn't exist
for Asterisk 1.4, woomera doesn't exist for Asterisk 1.4 and when I
hacked it to work on 1.4 it segfaults asterisk on hangup, and
chan_ooh323 works on 1.4, supports g723 and g729, and scales well with
large numbers of calls, making it the best Asterisk H.323 channel at the
moment. Can't we fix these last few little bugs and make it perfect?

Other examples I've found:

http://www.mail-archive.com/ooh323c-
[EMAIL PROTECTED]/msg00429.html

(Only set one codec, and set that in [general]. I use ulaw. If you use
two it uses one for RX and a different one for TX unless it negotiates
being the Master when it works. Seems to ignore codec in other
contexts.)

http://www.mail-archive.com/ooh323c-
[EMAIL PROTECTED]/msg00385.html

(my experiments so far have shown that you should have the codec
disallow/allow statements under [general] in ooh323.conf and not in the
individual user/peer sections. somehow the codecs in the individual
user/peer sections are not honoured.
also, even for disallow/allow, you can only have one codec selected like so,

disallow=all
allow=ulaw ; or alaw or gsm

having more than one allow line seems to confuse ooh323 as it uses one
codec for the transmit leg and another for the receive leg.)

Damjan


> Damjan Jovanovic wrote: 
> > On Fri, 2007-02-23 at 01:20 -0800, George Melika wrote:
> >   
> > > This has nothing to do with chan_ooh323, this is a dialplan design 
> > > problem.  You need to "Answer" the call before you start playing back 
> > > audio.
> > > 
> > > exten => s,1,Answer
> > > 
> > > George
> > >     
> > 
> > I did:
> > 
> > [dj1]
> > exten => 07,1,Answer()
> > exten => 07,2,Playback(demo-congrats)
> > exten => 07,3,Hangup()
> > 
> > The problem is (/var/log/asterisk/h323_log) that "local endpoint does
> > not have any audio capabilities" unless you put disallow=all,
> > allow=<something> in /etc/asterisk/ooh323.conf - why is that? How can I
> > help fix it (I'm a good programmer)?
> > 
> > Damjan
> > 
> > 
> > 
> >   
> > > ----- Original Message ----
> > > From: Damjan Jovanovic <[EMAIL PROTECTED]>
> > > To: ooh323c-devel@lists.sourceforge.net
> > > Sent: Friday, February 23, 2007 1:10:17 AM
> > > Subject: Re: [ooh323c-devel] asterisk 1.4.0 + chan_ooh323 = segfaults + 
> > > memory corruption
> > > 
> > > 
> > > On Thu, 2007-02-22 at 16:25 +0200, Damjan Jovanovic wrote:
> > >     
> > > > Hi
> > > > 
> > > > I'm using asterisk 1.4.0 and chan_ooh323 from asterisk-addons-1.4.0, and
> > > > I am experiencing segmentation faults and serious memory corruption.
> > > >       
> > > Sorry, there was something wrong with my setup. When I set it up on a
> > > diffent box it no longer has those problem - but there are others.
> > > 
> > > When I dial into asterisk using either GnomeMeeting or Netmeeting, the
> > > dialplan starts being executed and audio starts being played, even
> > > though the client believes the phone is still ringing.
> > > 
> > > Any thoughts?
> > > 
> > > Damjan
> > > 
> > > 
> > > 
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