T. Chan tommy.chan at utimail.com
Sat Jun 26 14:38:39 MST 2004

Hi, Jeremy, thanks for your help and dedication in resolving the problem.

There must be something that could have caused the problem. Why don't I
provide detailed information on what hardware I use and how I installed the
Asterisk and I would suggest that other colleagues who had or are having
this problem might want to do the same in order for Jeremy to help us.

I have tried two different hardware configuration with the same result. The
first Asterisk server I use a Pentium Xeon 2.4G with 512M Ram without any
digium card, I use Redhat 7.3 with Kernel upgraded to 2.4.20-28.7smp, ie.
enabling Hyperthreading. The second Asterisk server, I use a Pentium4 3.0G
with 512M Ram with same OS version and Kernel version. I read somewhere that
the system should be more stable without hyperthreading, I have tried using
2.4.20-28.7 Kernel but do not find any difference in terms of stability nor
voice quality at all.

I have tried many many times the following steps on both servers.

1. Get pwlib 1.5.2 and openh323 1.12.2 (ones as suggested by Jeremy) and
under pwlib, do ./configure, make clean, and then make both (I even tried
doing just a make opt here), and then openh323, do ./configure, make clean,
and then make opt.
1. Obtain asterisk, libpri, zaptel (although I don't need without digium
card) from cvs development head by doing CVS checkout asterisk libpri
zaptel. Everytime when I do this step, I will erase old directories to make
sure I have everything cleaned.
2. Do, make clean and make install on all directories, except that for
asterisk directory, I will go in ../asterisk/channels/h323 and do a make
clean and then make (without the install) before doing a make install under
the asterisk directory.
3. Asterisk ready.

I tried calling from another Asterisk running a January cvs into one of
these servers and out to cisco (or quintum or yet another Asterisk with
digium), but I got no audio on both servers. I tried calling from SJPhone
into one of these servers and out to cisco or quintum or another Asterisk
with digium and same thing with no audio both ends on both servers. I tried
calling from cisco, passing through one of these servers and out to another
cisco or endpoints above, same thing. No matter what I did, there was just
no audio on both ends at all.

Now, I have kept everything the same except that I changed to the stable CVS
and did the same thing as described above, and I could get audio now.

Jeremy, I hope that would give you some idea if I did anything wrong, and
probably most colleagues out there were doing similar to what I did and have
unknowingly made some mistakes somewhere. By the way, I was using a
h323.conf that is practically the same as your sample. Meantime, can you
tell us if you will be incorporating options like fast start, h245
tunneling, early alerting...into the driver? Thanks, and I hope you can
resolve this as soon as possible, thanks again for your support.


-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Jeremy
Sent: Saturday, June 26, 2004 4:28 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS

T. Chan wrote:

  Jeremy, any way to fix that? Thanks again.

I've spent many many days trying to duplicate any of these problems and
absolutely cannot.

I have tried everything from my mini-itx to my celeron based laptop to
my dual xeon dell 1750s and every single one of them work 100%
successfully in both directions with the cvs -head and chan_h323.

I've also very successfully tested interop with 5300s, Quintium A800,
some multi-tech box someone in IRC let me push a few calls thru (sorry
forgot your nick) and even my 7960 here now runs the H.323 load and then
OpenPhone works perfectly... I simply cannot duplicate any such problems.

I even manage a few different production systems with 5300s and they are
running absolutely perfectly on asterisk cvs -head with chan_h323.

Jeremy McNamara
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