Hi Joshua, > Since you are having media go directly the only thing that can be > monitored is the signaling of the call itself. This can be > accomplished using SIP session timers. There is a section "SIP > Session-Timers" in the sip.conf.sample file which has the various > configuration options relating to it.
I have been reading about it and it looks like an effective way to mitigate this problem. I'll test it and see how it goes with my providers. Thank you!! -- _____________________________________________________________________ -- 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
