Hello Marie, Increasing the rate got us up 2 folds, Thank you so much for your help. We have a clustered asterisk setup, and it seems like 200 concurrent calls at 70% cpu is how we can keep these machine humming comfortably.
Kind Regards, Nick On 4/9/13, Marie Fischer <ma...@vtl.ee> wrote: > On 09.04.2013, at 23:43, Nick Khamis <sym...@gmail.com> wrote: > >> That's just it! Nothing! It just does not pass the 91 mark. There are >> no failed calls during the test: >> >> Successful call | 0 | 20802 >> Failed call | 0 | 0 >> >> It's locked on 91 calls. I think I have a channel limit or call limit >> thing set somewhere by accident? >> >>>> >>>> The SIPP Results >>>> >>>> >>>> ------------------------------ Scenario Screen -------- [1-9]: Change >>>> Screen -- >>>> Call-rate(length) Port Total-time Total-calls Remote-host >>>> 10.0(0 ms)/1.000s 5060 2089.21 s 20802 >>>> 192.168.2.10:5060(UDP) > > So you have total calls = 20802. Does this number grow over time? > > >>>> 0 new calls during 0.000 s period 0 ms scheduler resolution >>>> 0 calls (limit 100) Peak was 91 calls, after 9 s > > IIRC, peak shows maximum concurrent calls. > What command line do you use to start SIPP? I see your call rate is 10 > calls/sec and maximum calls set to 100. Have you tried experimenting with > increasing the call rate (-r command line parameter)? How long is the > recording you are playing or have you set a call length for SIPP (-d command > line option) - that is, how long are your calls? > > SIPP generates just as many calls as specified - if you have 10 calls per > sec, it's quite logical to have ~90 ongoing calls after 9 secs. If your > recording is about 9 secs, then the first calls will end at that time and > you will never have more than ~90 concurrent calls. > > -- > marie > > > -- > _____________________________________________________________________ > -- 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