What is the difference between H323 and OH323 in Asterisk? I need Asterisk
to have basic H.323 support so we can offer some simple H323 termination
for some of our Cisco and Quintim hardware. Our upstream provider uses
SIP, so I figured I'd use Asterisk as the go-between. I already setup
Asterisk so it can push calls out through our providers via SIP. I just
need a good/solid/very simple H323 support in Asterisk, to make the simple
conversion without any quality issues.
I went to /usr/local/src/asterisk/channels/h323/ and read the instructions
and did something like this:
cd /path/to/pwlib
./configure
make clean opt
cd /path/to/openh323
./configure
make clean opt
cd /path/to/asterisk/channels/h323
make
cd /path/to/asterisk
make install
This seemed to work fine, in /usr/lib/asterisk/modules/ I have a
chan_h323.so and I created an /etc/asterisk/h323.conf file:
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
dtmfmode=rfc2833
gatekeeper = DISABLE
AllowGKRouted = yes
context=unauthenticated
[test]
type=friend
host=209.237.227.185
context=termination
incominglimit=10
accountcode=008
However, when I have the Asterisk at 209.237.227.185 push a call through
via H323, the Asterisk on the receiving end of course does not bother to
match the source IP, and decides to put them in the "unauthenticated"
context, instead of "termination". No matter, I implemented a 4 digit
tech
prefix thing in unauthenticated, so that I can set the accountcode via the
dial-plan, and now they're able to push calls, but guess what, no ringing
sound is heard while it rings the phones. I've traced the issue down to
the H323 part, SIP to SIP and IAX to SIP work fine, but H323 to SIP has
ringing issues. I implement the ",r" feature of the dial statement and
now
they hear fake ringing, but they found out it's fake and aren't happy
about it.
Should I give up and just go to OH323? What's the difference? Is there
another H323 to SIP conversion device I need to be looking to? Something
more solid or reliable?
Best Regards,
Deon
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