On Friday 11 July 2008 09:17:37 Artie Gold wrote: > In updating to 1.4.21 recently, we've encountered a problem, when running > over a satellite connection (where the latency is considerable; a "regular" > internet connection did not exhibit this problem), where incoming calls are > being dropped as a result of the sip handshake timing out (dropping down to > 1.4.18.1 solved the problem for us). From reading the change logs and other > posts, it seems that some work has been done in this area recently to get > it "right"; it appears that, at least in the satellite case, things may > have gotten a little too "tight"... > > If this rings a bell for anyone, any insight would be appreciated.
Try setting t1min to something higher than the default, 100 (ms). This value is settable globally, as well as per-peer. -- Tilghman _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
