On 22.05.2013, at 16:18, Tommy Cooper <[email protected]> wrote: > Thank you for your help I finally solved this issue. Is it possible that my > setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core > using 3.5 GHz, and 1Gb of RAM?
Easily, as long as you have no media :) Use -sn uac_pcap instead of -sn uac to test with RTP (and watch your call count drop). Add recording (MixMonitor()) to your dialplan and watch the call count go down even more. ;) A rough way to see if call quality is deteriorating would be to call your Asterisk box while the SIPP test is running and listen to some message played via Background(). > > ----- Forwarded Message ----- > From: Marie Fischer <[email protected]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Sent: Wednesday, May 22, 2013 1:16 PM > Subject: Re: [asterisk-users] Stress testing Asterisk > > > On 21.05.2013, at 0:05, Tommy Cooper <[email protected]> wrote: > > > Hi, > > I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is > > generating are failing. I am trying to run Sipp on the same machine as > > Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command. > > Do you have a peer and extension configured for SIPP in your Asterisk > configuration? You also needat least the -s <extension_to_dial> option on > your sipp command line. > http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has > some simple instructions which should get you started. > If the calls still fail, Asterisk console output would be helpful. -- _____________________________________________________________________ -- 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
