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 <suzanne@supertec.com> 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
No comments:
Post a Comment