Gareth Blades <[email protected]> schrieb am Fre, 06. Sep 13:21:
> No asterisk will always use the first SRV record and wont load
> balance or switch to a backup if its not reachable.
hmm okay :O
> What we do is have each endpoint defined in sip.conf with
> qualify=yes and then in the dialplan use the ${SIPPEER(x)} variable
> to pull out the status of each peer and pass it into an AGI
> application to perform the load balancing etc...
>
> If you are happy with wone being a primary and one being a backup
> then if you have qualify=yes set for both you could just dial using
> the first one and then an execif hangupcause=20 then try dialing the
> backup.
okay, then i must try this way. ^^
thank you for your help and information.
greetings
dominique
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