In Asterisk - sip.conf<div><div>[MyKamailioUser]</div><div>type = friend</div><div>host = 192.168.2.2 - internal IP of Kamailio</div><div>insecure = port,invite</div><div>context = {where you would like to get calls - you can use default, or not use this part at all}</div>
<div>In Asterisk - only authentication is based on IP of Kamailio server.</div><div><br></div><div>In Kamailio, I am using:</div><div>ds_select_dst("1","4");</div><div>to get Asterisk from pool (if you have more then one asterisk behind Kamailio)</div>
<div>and then - t_relay, instead of forward, because my Asterisk has only internal IP. Calls are sent back to Kamailio in case of calling other user, or directly to provider. Asterisk also is connected via registrar to provider for incomming calls.</div>
<div>In my configuration, I am using some dirty solutions for problems with ACK and BYE, but more or less everything works fine. rtpproxy - solves problems with NAT, and users must not use STUN.</div><div><br></div><div><br>
</div><div class="gmail_quote">On Tue, Jan 31, 2012 at 8:50 AM, Sammy Govind <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@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,<div><br></div><div>Kamailio is definitely the exact tool for this purpose, I have exactly the same setup running as yours and for scalability we started using Kamailio in front of our asterisk servers. Long story short, read these articles.</div>
<div><br></div><div><a href="http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb" target="_blank">http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb</a> </div><div><a href="http://saevolgo.blogspot.com/2011/11/increasing-voip-services-capacity.html" target="_blank">http://saevolgo.blogspot.com/2011/11/increasing-voip-services-capacity.html</a> </div>
<div><br></div><div>Once after setting up the environments you don't need Public IPs on your asterisk servers. Just only for Kamailio server(s)</div><div><br></div><div>Regards,</div><div>Sammy.<div><div><br>
<br><div class="gmail_quote">
On Tue, Jan 31, 2012 at 11:16 AM, Greg Mannie <span dir="ltr"><<a href="mailto:greg@latigi.com" target="_blank">greg@latigi.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>
We have been using Asterisk for sometime and over the last year have started hosting instances for our clients on a vmware platform. These virtual pbx are located on public ip addresses and each customer has their own SIP trunk arrangements with various providers. We have decided to pool our resources and would like to start aggregating traffic.<br>
<br>