Hi Mike, <br><br>Lets suppose we have:<br><br><ul><li>2 machines configured for high availability (LAN HA) in a master/slave configuration with a floating public address on the master. (<a href="http://www.ultramonkey.org/3/topologies/ha-overview.html">http://www.ultramonkey.org/3/topologies/ha-overview.html</a>)<br>
</li><li>freeswitch installed on every machine configured to use mysql in the core via odbc</li><li>both freeswitch have identical dialplan and directory configuration<br></li><li>mysql installed on every machine (with replication between the DBs)</li>
<li>SIP Trunks towards the upper provider (without registration but i should work with registration)</li><li>SIP Phones/Terminals registering to the active freeswitch<br></li></ul><br>When a terminal registers to the active freeswitch, the registration is propagated to the inactive one via DB replication. Now, lets suppose we have a switchover ... of course we will lose the ongoing calls but new calls (from SIP Phones) should be able to establish. The same applies to incoming calls from the upper provider.<br>
<br><br>Im just talking about HA here not loadbalancing and performance scaling...<br><br>what do you think about that?<br><br><br><br><br><br><br><div class="gmail_quote">On Fri, Dec 4, 2009 at 1:56 AM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com">mike@jerris.com</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;">so your registering to the provider to get the calls? If so, this gets tricky, the provider likely does not support multiple registrations, even if they did they probably send the call to both registered endpoints. With this big unknown its not very easy to suggest a good solution. If I were looking to set this up without needing proxies I would want to use srv records and naptr records and a provider that would balance using these including failiover.<br>
<br>
Mike<br>
<div><div></div><div class="h5"><br>
<br>
On Dec 3, 2009, at 3:40 PM, Tim Uckun wrote:<br>
<br>
> On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>> wrote:<br>
>> The easiest place to do this is at the point you send the calls to FreeSWITCH. How are the calls coming in?<br>
>><br>
><br>
> From an as of now unkown SIP trunk provider (we are still in<br>
> negotiations with a couple of companies).<br>
><br>
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