Our system is:<div>SIPClients <-> Kamailio <->Asterisk<->external access points</div><div>SIP clients are connected to Kamailio. All calls (from clients) are forwarded to Asterisk, and Asterisk either send call back to Kamailio, or send call to external world through paid SIP trunk.</div>
<div>One of Asterisk servers is registered with VoIP provider. Calls to access points are going directly to Asterisk server, although it is behind Kamailio - Asterisk servers have only private IP address. Some of calls are forwarded to Kamailio and clients connected to it.</div>
<div>Kamailio itself, as I understand is not prepared as gateway, but as sip router with some other functions.</div><div>We use also rtpproxy module, and we do not need STUN for clients.</div><div>PS</div><div>I can send you some pieces of code, or even whole conf file.</div>
<div>It is far far away from perfectness because I am working from very short time with Kamailio, until last 5 months we used only Asterisk.</div><div><br><br><div class="gmail_quote">On Tue, Apr 3, 2012 at 4:00 PM, Rob Watkin <span dir="ltr"><<a href="mailto:robwatkin@gmail.com">robwatkin@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Stoyan,<br><br>Does that mean that you use Asterisk as pure SIP PSTN Gateways? I imagined Asterisk as a physical PSTN gateway but thought that Kamailio/RTPProxy would scale out better for pure SIP. I was planning to use Asterisk or FreeSWITCH as a media server for hold, VM, conference and IVR.<span class="HOEnZb"><font color="#888888"><br>
<br>Rob</font></span><div class="HOEnZb"><div class="h5"><br><br><div class="gmail_quote">On 3 April 2012 13:23, Stoyan Mihaylov <span dir="ltr"><<a href="mailto:stoyan.v.mihaylov@gmail.com" target="_blank">stoyan.v.mihaylov@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
We do just that way - Kamailo to handle load balancing and clients, and Asterisk servers for routing, gateway etc....<div>You can forward calls to your main gateway which then will work with other gateways or calls from gateways to clients registered in Kamailio.<br>
<br><div class="gmail_quote"><div><div>On Tue, Apr 3, 2012 at 3:15 PM, Rob Watkin <span dir="ltr"><<a href="mailto:robwatkin@gmail.com" target="_blank">robwatkin@gmail.com</a>></span> wrote:<br></div></div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>
Hi,<br><br>I am just getting started with Kamailio and have been following the book "Building Telephony Systems with OpenSER" by Flavio E. Goncalves. The book describes an architecture with a SIP Proxy handling registrations and handing calls to a PSTN Gateway. I now have a basic test network running where calls are routed via the SIP Proxy (Kamailio) to a third party PSTN Gateway. I feel that a better design would be to implement my own PSTN Gateway using Kamailio. This single gateway would then handle all third party PSTN gateways. Thus one Kamailio server would be facing my clients while another would be facing my suppliers.<br>
<br>Is this a sensible architecture and are there any sample configurations for Kamailio performing this role?<br><br>Thanks<span><font color="#888888"><br>Rob Watkin<br><br><br>
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