[asterisk-users] Asterisk and H.323

[Cesc]

On 7/24/06, Aaron Anderson <webmaster at psphacks.net> wrote:
>
>  I have been messing with both all day.  I think what might be tripping me
> up is the extensions.conf.
>
i do think so too :)


>  I was able to receive an incoming connection from the client, but all the
> system returned was a busy signal.  This call was to a known good number (my
> phone) so I'm not sure what's wrong.
>
if you got the incoming connection but then got a busy tone, it is
because the "peer info" in your asterisk conf is not good ...
try the "standard" demo setup in asterisk ... there you can call
extension 600 (i think) and you get an echo test ...
later, extend this to call whoever you want ... but you'll need to
enter the "location" info somehow (i never did it ... ). I used
sjphone for testing ... try it ... you can add an extension to dial
directly to the IP address of the sjphone. So, you dial the MMM
extension on your phone, get to asterisk and then asterisk dials the
IP address of sjphone ... nice and easy for testing.

>  Will gnugk do a "translation" from h.323 to sip so I don't have to make any
> major modifications?  Is there an example gnugk.ini and h323.conf file I can
> look at to get this all running?

gnugk has no clue of sip.
gnugk is a gatekeeper for h323 ... you can do stuff with the dialled
numbers ... forwarding the call to various [asterisk] h323 gateways
and the like ... but the conversion of h323 to sip is done in
asterisk.


Cesc


>
>  Thank you in advance
>  Aaron
>
>
>  harrygaillac-sip at yahoo.fr wrote:
>
>  Hello,
>
> Try both chan_oh323 and gnugk .
>
> Harry
> --- Aaron Anderson <webmaster at psphacks.net> a écrit :
>
>
>
>  I have been scouring the net the last couple of days
> looking for some
> kind of tutorial or walkthrough on setting up a
> h.323 channel in asterisk.
>
> What I need to do is basically this:
>
> I have a client who wants to be able to connect to
> me via h.323 and make
> a local phone call (local to me, he is in a
> different country). The
> call is an automated process and no callee
> interaction is required. My
> client simply wants to be able to call a user and
> give them a
> verification number and then hang up. He's using
> some in-house software
> so unfortunately, h.323 is his only option.
>
> Can someone point me to a doc or perhaps give me a
> simple breakdown of
> what I need to add to asterisk in order to be able
> to do this? I am on
> a tight deadline and my searches have not revealed
> the information I am
> looking for. I have built chan_h323 and it is
> loaded but I'm not sure
> how to set it up beyond that.
>
> Any help would be much appreciated.
>
> Thank you
> Aaron
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