On Dec 3, 2009, at 5:05 PM, Matt Riddell wrote: > On 4/12/09 9:28 AM, Scott L. Lykens wrote: >> Apologize for not directly answering your questions, however, I'm >> considering playing with Remus and Xen in the future to deal with high >> availability without dropping calls. >> >> See http://dsg.cs.ubc.ca/remus/ for some details. >> >> I have no idea if it will work or what the implications are but I >> noticed that in doing research for some other projects and made a note >> of it to try in the future. > > Yeah, I was looking at that too - haven't had much time to work on it > further, but if it can handle running Asterisk cleanly it looks like a > pretty nice solution. > > I'm just not 100% convinced that it will work in a real time environment > rather than for hosting web sites. > > I'm sure that it would be perfect if you wanted failover web sites > without any downtime, but wonder how it would work with Asterisk. > > Post your progress as you move through it :) >
If you're using just SIP to SIP, a better option would be a pure sip proxy, ala Kamailio/SER, etc. They can survive a failover without a drop. If you're using Asterisk during the call for media, recording, etc., I couldn't think of a solution where that would not result in the calls on the specific instance ending. ---fred http://qxork.com _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users