[Asterisk-Users] Hypothetical : Working across multipleservers??

Lee Goodman lee.goodman at comcast.net
Thu Oct 9 10:13:32 MST 2003


You might be better off using the "emergency proxy" feature in the Cisco
7960 phones. If one Asterisk server fails , the phone will use "emergency
proxy" and send the call to the 2nd Asterisk server.

This won't help a call already in progress, but would allow a user to dial
again

Lee

PS. SIP devices that support either SRV records or DNS round robin would
work to


----- Original Message -----
From: "Steven Critchfield" <critch at basesys.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, October 08, 2003 12:18 PM
Subject: Re: [Asterisk-Users] Hypothetical : Working across
multipleservers??


> On Wed, 2003-10-08 at 10:52, WipeOut wrote:
> > Hypothetical question..
> >
> > Lets say there is a situation where you are using the highest
> > compression codecs for all extensions (I guess that would be G.729) and
> > the load on a single server is overpowering the most powerful single
> > processor(lets say SMP is not an option).. So two or more servers are
> > required..
> >
> > Or
> >
> > The situation is that you need fault tolerance so want to have two
> > Asterisk servers serving the same population of users and all users are
> > able to authenticate to both servers (I guess you simply copy the same
> > sip.conf to both servers for this) but you also want all other
> > extensions to be availible to all users irrespective of which server
> > they have connected to..
> >
> > I guess in a way it would be like an Asterisk cluster of some sort..
> >
> > What would the solution be? Has anyone got an install similar that they
> > would like to share how it was done?
>
> Wouldn't this be the perfect case for a switch statement, especially if
> re-invites worked reliably for SIP? You could then have one machine that
> accepted authentication and directed call management via it's
> extensions.conf file. The other machines would then do the negotiation
> with the endpoint and then do the codec work also.
> --
> Steven Critchfield  <critch at basesys.com>
>
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> Asterisk-Users at lists.digium.com
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