On Mon, 2010-02-22 at 16:13 -0500, JT wrote: > Is this something that is fixed in an update? (Currently running 1.2)
Yes... modern versions of Asterisk support SIP session timers. (If I remember correctly, Asterisk 1.2 could tear down a call based on lack of RTP data, but I never found it worked well enough in my tests to warrant its use.) -- Jared Smith 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
