Check the sip.conf.sample file. I think it is the call-limit parameter that is getting you. The sample file should tell you what the default is. Another possibility is that your rtp range is set too low; the "normal" range is 10000-20000, which allows for 2500 calls(or 5000 if you set other things "correctly").
-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes Sent: Friday, March 30, 2012 7:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] concurrent channels limit On 30 Mar 2012, at 10:14, Syco wrote: > Finally the problem is: I cannot manage more than 80 concurrent calls. What happens on the 81st call?.. S -- _____________________________________________________________________ -- 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