At present I have two hardware identically freepbx/asterisk boxes. The mysql db on one is slaved to the other and all config files are rsync'd once every 24 hours (we have few configuration changes). We use Polycom 321/331/550/650 phones, and I notice that these phones can be configured with two SIP servers. 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? Kind Regards, Chris
Christopher Harrington
2012-Nov-15 15:25 UTC
[asterisk-users] "Simple" failover configuration
On Thu, Nov 15, 2012 at 8:59 AM, Chris Nighswonger < cnighswonger at foundations.edu> wrote:> At present I have two hardware identically freepbx/asterisk boxes. The > mysql db on one is slaved to the other and all config files are > rsync'd once every 24 hours (we have few configuration changes). > > We use Polycom 321/331/550/650 phones, and I notice that these phones > can be configured with two SIP servers. > > 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? > > Unless your Polycom phones automatically detect that the primary Asteriskserver has returned after an outage, you will likely end up with a partition, won't you?> Kind Regards, > Chris > > -- > _____________________________________________________________________ > -- 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 >-- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121115/9d590e44/attachment.htm>
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> 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?
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] On Behalf Of Eric Wieling
Sent: Thursday, November 15, 2012 9:27 AM
To: 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> 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?
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On 11/15/2012 10:27 AM, Eric Wieling wrote: What I have found most difficult in any failover situation is having everything decide at the same time something has failed. (this applies to anything not just asterisk) For example how does the polycom react if it can make the sip connection, but no outbound routes are available on the primary server for some reason ? Is your setup smart enough to actually shut down asterisk completely if its upstream network interface or route is dead to prevent local connections ? what if both servers are in that situation ? would they both shut down, neither ? what would you want to happen in that case ? They are not trivial questions to answer and the answers depend on your setup, there is no univeral right way of handling it.> 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> 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
Take a look at this doc from Polycom...it answers your question I think. https://encrypted.google.com/url?sa=t&rct=j&q=polycom%20redundant%20server&source=web&cd=1&cad=rja&ved=0CEUQFjAA&url=http%3A%2F%2Fsupport.polycom.com%2Fglobal%2Fdocuments%2Fsupport%2Ftechnical%2Fproducts%2Fvoice%2FConfiguring_Optional.pdf&ei=TjGtUMuDD86E0QHPpYCQDA&usg=AFQjCNGL4uuttNHorfaTnTGcqxCQAZrwCQ&sig2=-HbRXBZJR1nqEtT0VmYq1A On Thu, Nov 15, 2012 at 6:59 AM, Chris Nighswonger < cnighswonger at foundations.edu> wrote:> At present I have two hardware identically freepbx/asterisk boxes. The > mysql db on one is slaved to the other and all config files are > rsync'd once every 24 hours (we have few configuration changes). > > We use Polycom 321/331/550/650 phones, and I notice that these phones > can be configured with two SIP servers. > > 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? > > Kind Regards, > Chris > > -- > _____________________________________________________________________ > -- 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/20121121/346a20e9/attachment.htm>