[asterisk-users] 回覆︰ "Simple" failover configuration

Michelle Dupuis mdupuis at ocg.ca
Thu Nov 15 18:08:28 CST 2012


Or...you could use HAAST (www.generationd.com<http://www.generationd.com>) - it detects failure, switches IP to an Asterisk peer, updates routes, updates ARP tables, synchronizes settings between peers, etc.

-M-

P.S. I work for generationd, so I think the product is amazing :)

________________________________
From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of kingman chui [chuikingman at yahoo.com.hk]
Sent: Thursday, November 15, 2012 6:34 PM
To: Asterisk Users List
Subject: [asterisk-users] 回覆︰ "Simple" failover configuration

I think you can use virtual IP with a group of sip server that run in HA .
so, only one of sipserver is handle the call and other is standby ...
Is this what you want ...?????
Regard/chui king man

寄件人︰ Danny Nicholas <danny at debsinc.com>
收件人︰ 'Asterisk Users Mailing List - Non-Commercial Discussion' <asterisk-users at lists.digium.com>
傳送日期︰ 2012年11月15日 (週四) 11:31 PM
主題︰ Re: [asterisk-users] "Simple" failover configuration

You can actually configure at least some Polycom phones to 3 or more SIP
servers.  Your problem is going to be that when one of your servers is down
for whatever reason, the "line key" attached to that server will be "off".
In a "Dual Server" environment, I would lean toward putting something like
Kamailio (sp) in line so it can determine which server is the active one.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>
[mailto:asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] On Behalf Of Eric Wieling
Sent: Thursday, November 15, 2012 9:27 AM
To: chris at acsdi.com<mailto:chris at acsdi.com>; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] "Simple" failover configuration

Polycom phones after firmware 2.x register to BOTH the primary and backup
servers.

On Thu, Nov 15, 2012 at 8:59 AM, Chris Nighswonger
<cnighswonger at foundations.edu<mailto:cnighswonger at foundations.edu>> wrote:


    Would the simplest approach to failover be to just configure my
    primary asterisk server as the first SIP server and my backup as the
    second?

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com/ -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com/ --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121115/77567d42/attachment.htm>


More information about the asterisk-users mailing list