Hello,<br>
<br>
yate is great for h.323<->SIP, it even has a config on its web to do it. I've used it a lot.<br>
<br>
have fun!<br>
<br><br><br><div class="gmail_quote">On Mon, Jun 1, 2009 at 12:55 PM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@voice-system.ro">bogdan@voice-system.ro</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="im">Wayne Scholar wrote:<br>
> A bit more info.<br>
><br>
> We use opensips as a gateway to know about all our asterisk servers<br>
> behind it. We do not use it for media just for signaling and so we<br>
> don't have to register IPs with our carriers other than our gateway.<br>
> The opensips will dist the calls across our asterisk servers.<br>
><br>
> I am looking to build the same setup with h323 protocol. So i don't<br>
> know if i need a gateway or gatekeeper or what software can perform<br>
> this function. freeswitch is a lot like asterisk as a read it?<br>
<br>
</div>I think yate has some nice h323 support - you may look around to see if<br>
it can do what you need.<br>
<br>
Regards,<br>
<font color="#888888">Bogdan<br>
</font><div class="im">><br>
> Wayne<br>
><br>
> On Fri, 2009-05-29 at 15:28 -0500, Brett Nemeroff wrote:<br>
>> freeswitch<br>
</div>> ------------------------------------------------------------------------<br>
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