check your ulimits :)
On 26 September 2014 17:15, CDR <vene...@gmail.com> wrote: > 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 -- _____________________________________________________________________ -- 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
