[Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting

[Bruno Hertz]

Jesse Guardiani <jesse at wingnet.net> writes:

>> Wait a sec...  COME TO THINK OF IT!
>> Why not run asterisk on your linux box that you are running GnomeMeeting 
>> on, and use it to convert between H.323 and IAX and SIP???
>> 
>> After all, it is a penguin...
>
> That's certainly a good alternative. I'm currently in the process of
> hacking up the latest linphone (1.0.1) to fix a few personal
> show-stoppers. If I can get it to the point that I like it, then I'll
> probably just go with linphone. But you're right. If it's took much work,
> then I'll probably just start running asterisk on my laptop to do H.323 to
> SIP conversions. Thanks for the suggestion! I hadn't thought of that yet.
> I'd been looking at things like the commercial sip323 program, but I
> hadn't thought of doing it with a local copy of asterisk.

If your only reason to stick to H323 is Gnomemeeting you could try
other softphones as well. Especially, the XLite beta for Linux looks
promising, and some people like SJphone for Linux.

Also, SIP support for Gnomemeeting is underway, but development is
slow. I'm constantly pointing out to them how much interest there is,
but things still seem to take their time ...

Finally, on a recent discussion about the future design of GM on their
list, I was surprised to learn that quite a few people really use it
for direct PC to PC video calls over the internet. So somehow, after
extensive NAT and router fiddling I guess, direct calls apparently
work even with H323 (there is already support built into GM for
external IP address discovery, as you know, so those remarks about
transmission of bogus IP addresses on H323 level probably don't really
apply in this case).

Anyway, I myself use the setup recommended above, i.e. local * server
as protocol translator, and it works reasonably well.

Regards, Bruno.