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Using asterisk 1.4 with 100M or 1000M ethernet and 230 SIP clients and a 64 bit 4200+ box would there be any noticable lag or delay to bring each one of them into a PAGE mode. so one speaker can talk out on all 230 SIP clients for a message. Would this work? Thanks, Jerry

My test case must relative with burst traffic such data or real time traffic somebody can advise me how to generate traffic data like ICMP to be burst and about real time traffic too ,.. i don''t have any idea to make burst traffic for test HTB and HFSC which can manage that situation good. Express yourself instantly with MSN Messenger! MSN Messenger Download today it''s FREE!

On Mon, Jun 29, 2020 at 6:46 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Sun, Jun 28, 2020 at 2:26 PM Dovid Bender <dovid at telecurve.com> wrote: > >> Hi, >> >> We have a box up and we are starting to see a lot of "Exceptionally long >> queue length queuing" in the logs. From all the research so far it seems >> like this leads to

Hello, I''m trying to get lossless VoIP traffic over my 3000k/500k ADSL line. Shaping outgoing traffic is no problem: I set total ceil for outgiong device (ppp0) to 450kbit and put VoIP into highest prio class. Even during full upload the voice is clean on the other end. Now I tried to get the same result for incoming data. I attached HTB to eth1 where the incoming voip traffic is

Hello, First of all, I already contacted Martin Devera, the developer of HTB, and he told me to search for help on this mailinglist, thus I am describing my problems here now... I am kind of seriously annoyed by QoS as I have been trying for over 3 years to get it working properly - first I did not understand how it works, then it seemed not to work, then it was working perfectly for half a