David,<div>What capacities are you able to achieve with yate and h323? roughly?</div><div><br></div><div><br><div class="gmail_quote">On Thu, Jun 4, 2009 at 7:15 AM, David Villasmil <span dir="ltr"><<a href="mailto:david.villasmil.work@gmail.com">david.villasmil.work@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">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!<div><div></div><div class="h5"><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" target="_blank">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>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>><br>
> Wayne<br>
><br>
> On Fri, 2009-05-29 at 15:28 -0500, Brett Nemeroff wrote:<br>
>> freeswitch<br>
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