[OpenSIPS-Users] best way to use one active one hot spare asterisk server behind OpenSIPS
    Ovidiu Sas 
    osas at voipembedded.com
       
    Wed Nov  9 22:03:04 CET 2011
    
    
  
I suggest start reading the opensips documentation page:
http://www.opensips.org/Resources/Documentation
The routes are described here.:
http://www.opensips.org/Resources/DocsCoreRoutes17
Check the default opensips config file, try to understand it and then
you will be able to build and debug your own configs.
Regards,
Ovidiu Sas
On Wed, Nov 9, 2011 at 3:51 PM, gwallis <gary.wallis at voicecarrier.com> wrote:
> Where/how is the failover route configured?
>
> Any more details or manual/book or other doc links hints appreciated.
>
> (Sorry in advance for the newbie -or RTFM avoidance- questions)
>
> --
> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/best-way-to-use-one-active-one-hot-spare-asterisk-server-behind-OpenSIPS-tp6979044p6979499.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
    
    
More information about the Users
mailing list