Hi! > > I am just curious because I was having problems with dropped calls as > > well
Maybe rtptimeout in sip.conf is involved (and not behaving as expected)? > All extensions are hard-coded. We only have a handful of > phones that don't change. This last sentence is a wounderful example of a sentence that can be interpreted in two, and very opposite, ways. :-) Philipp -- _____________________________________________________________________ -- 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
