On Tue, 2007-12-18 at 17:43 +0100, Administrateur www.jeremy-salmon.org wrote: > Randomly I have dropped calls during communication. No absolutetimeout or > other > calling limitation options. > > Any ideas on how to solve this problem?
The first place I'd look would be the Asterisk CLI. Make sure you turn up the CLI verbosity first by typing "core set verbose 5" before the call. If that doesn't offer any clues, I'd next look at the SIP signaling. You can see that by typing "sip set debug" at the Asterisk CLI, or by using a network tool such as Wireshark, tcpdump, or ngrep. --- Jared Smith Community Relations Manager Digium, Inc. _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
