This is a quite promising idea. Many thanks. I'll post my results to the list...
Cheers, --ag On Fri, Jul 11, 2008 at 11:22 AM, Tilghman Lesher < [EMAIL PROTECTED]> wrote: > 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 > -- Artie Gold F4W, Inc.
_______________________________________________ -- 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