I am using Asterisk 12 svn, from today, and after a few thousand calls, I run out of ports. This happens eith PJSIOP and regular old SIP. I think it is RTP related. Any idea how can I troblshoot this. It happened teh same with Asterisk 11. On the other end there is a freeswitch. My guess is that there is an incompatibility. Thanks in advance for your thoughts
-- _____________________________________________________________________ -- 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