A little off topic, but SipX has built in redudancy. if it is so important to you, you should have a look.
On 9/25/07, Atis Lezdins <[EMAIL PROTECTED]> wrote: > [snip] > > http://lists.digium.com/pipermail/asterisk-users/2007-August/195339.html > > >> could provide you with some answers. > > > > > > > > >Hi, > > >This seems nice way of sharing settings, however it wouldn't take over > > >calls in progress. For us, currently the greatest problem is that > > >whenever Asterisk crashes, calls are lost, and that means - lost > > >money. Are there any ideas? > > > > You might want to take Asterisk out of the media path then. If it crashes, > > calls will stay up, although your CDR's will be screwed. If screwed CDR's > > still means lost money... your still screwed! > > Nop, i can't stay out of media path, as there are essential features > depending on it - hell, that's why i need asterisk - transfers, > chanspy, monitoring.. Of course in case of crash - monitoring and CDR > can be lost - that would be minor problem comparing to lost calls. > > I'm thinking about some mechanism how asterisk could communicate with > second asterisk and report all state operations made with SIP. So if > asterisk fails, redundancy asterisk performs IP takeover and > continues. Unfortunately my SIP knowledge is nearly minimal (as are my > C skills), and i don't have any ideas how to implement this. > > A simplest method could be something like SIP proxy, that sends calls > to asterisk, but if asterisk stops responding, it plays some message > and tries to send call to redundancy server - however then problem can > occur with redundancy server. And this would have some major drawbacks > - calls wouldn't be matched to corresponding agents in queue. > > Hmm, thinking a bit more about topic - maybe redundancy mechanism > would have enough to keep state of channels, bridges, and > corresponding dialplan location (assuming that config is identical). > Too much of duplicating everything would mean that second asterisk > could have the same crash. > > Regards, > Atis > > -- > Atis Lezdins > VoIP Developer, > IQ Labs Inc. > [EMAIL PROTECTED] > Skype: atis.lezdins > Cell Phone: +371 28806004 > Work phone: +1 800 7502835 > > _______________________________________________ > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > --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 > -- Guilherme Loch Góes MSN:[EMAIL PROTECTED] (48) 99115299 _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
