Thanks,<br><br>My background is old analog telephony systems and I know a little bit of Asterisk.<br>With Asterisk we do not allow reinvites and it does most things we want. And it is relatively easy to secure an Asterisk server. Reinvites are to make sure nat issues do not show up.<br>
<br>What I am trying to do is set up a proxy and registrar for 100.000 subscribers. It's behaviour should be stateful. It should allow anonymous invites for subscribers (extensions) and alias database. So you could call form Ekiga.net to our domain. Our subscribers (extensions) should be authenticated and allowed to dial other domains and pstn. Because of nat issues every call must be send to Asterisk.<br>
<br>So after an invite from subscriber 101 to subscriber 102 (usrloc) dialplan rewrites to usrloc, usrloc is then translated to loadbalancer and send out to an Asterisk server.<br><br>No matter where you are dialing to you will always be forwarded to a media server for rtp.<br>
That's why I asked about the t_onreply function. <br><br>I am sorry that I am not so good in this programming language. :(<br><br><br>Albert<br><br><br><div class="gmail_quote">On Wed, Jun 2, 2010 at 8:48 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;">Hi Albert,<br>
<br>
I do not fully understand what you want to achieve - maybe describing a<br>
simple flow (logical point of view) will really help in helping you.<br>
<br>
Best regards,<br>
Bogdan<br>
<div><div></div><div class="h5"><br>
Albert Paijmans wrote:<br>
> Hi,<br>
><br>
> I am a bit at a loss right now, I have a script wich does dialplan<br>
> translations but permission and group modules are loaded, registration<br>
> fails.<br>
> The dialplan module and avpops should give an attribute to a number.<br>
> Also I am wondering (since the script does not work I can''t test<br>
> this) if a t_relpy route on usrloc would work.<br>
> So after OpenSIPS does database lookup for extensions and db_alias<br>
> sends an invite and relay both extensions to an Asterisk server via<br>
> gateway list.<br>
><br>
> I have send the opensips.cfg as attachement<br>
><br>
> Thanks<br>
><br>
> Albert<br>
><br>
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<br>
<br>
--<br>
Bogdan-Andrei Iancu<br>
<a href="http://www.voice-system.ro" target="_blank">www.voice-system.ro</a><br>
<br>
<br>
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