Can you look at ticket # 702556000194? This is very simple:
Asterisk is down, I am simulating that with the command "stop now," Calls should then go to the failover SIP address, but they do not. I have been back and forth for weeks with your support and they do not figure it out. I am not even sure they understand what I am saying. On Tue, Dec 9, 2008 at 15:34, Suzanne Bowen <[email protected]> wrote: > There is a Northwest Florida organization http://www.linkingarms.org who > wants to have a telethon using open source telephony technologies. If anyone > reading this is interested in talking with the executive director Kenny > about this, please email me OFF the list so we won't bother the list with > further details. > > -- > Thank you, > Suzanne Bowen > Blogs: http://blogs.didx.net, http://suzanne.supertec.com, > http://www.tmcnet.com/tmcnet/blogs.htm, > IP communications events we recommend and sponsor at > http://www.didx.net/events > Media channel: http://www.tmcnet.com/channels/did-ddi/ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-biz > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
