Gareth Blades <mailinglist+aster...@dns99.co.uk> 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 -- _____________________________________________________________________ -- 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