Tandem Switch Question


I had posted in the the wrong forum, so I apologize for cross posting.

My Voip provider (thinktel.ca) has requested we install a proxy server for aggregating our sip trunks.
I have used asterisk for some time with freepbx as my ui. We now have several asterisk servers in different locations and I would like to aggregate my sip traffic to save money.

I have been told I am talking about a tandem switch configuration where I can configure trunks on my proxy with my provider and direct inbound and accepting outbound from my remote asterisk servers? I want to make sure I can keep track of call use and would prefer to use the proxy for this if possible. Does anyone have any documentation on this type of setup, or should I be looking to some other software?

Thank you


Configure one Asterisk as the interface point. Configure all the other Asterisks as though that Asterisk was their SIP provider, but using your VPN addresses (I hope you have a unified VPN!).

Configure the others as peers on the hub. On the hub, use a very short dialiplan - one line may be enough, to forward non-local calls to the other nodes. Having discrete extension number ranges will make this less of a hassle.

Thank you for your help. I am going through the Kamailio Knowledge Base and it’s a serious learning curve for me. I’ll read up on the Asterisk configurations as I probably have a much better chance of having it running in the next week.

Do you know of any good examples or documents on people who have configured their Asterisk to be a tandem switch?