similar to: htb quantum/r2q problem/question

Hello Stef, I am working on a Bandwidth management project as part of my college curriculum.I am running TCNG 10-b but I am not able to classify packets based on MAC address using tcng, so I am running tc commands in /tcng/bin and they are working. But is there a way out to classify packets based on MAC address using TCNG. And, I am successful in throttling down the Bandwidth to 8kbit

Bernard Łabno wrote: > Hello, > What does this line mean : > > HTB: quantum of class 10200 is small. Consider r2q change. > > > And how should I fix it ? I have dsl 1mb connection and 2 cards > (realtek-8139C and realtek-8100B/8139D) > > Could the above line cause this ? : > NETDEV WATCHDOG: eth0: transmit timed out > eth0: Transmit timeout, status 0c 0005

Hello! One simple question regaring htb2->htb3 way. I''m using 2.4.25 kernel with tc3 from Devik''s site, RH9. My data flow is about 100Mbit duplex and is subject to grow, so I''m creating root class with 200Mbit rate, with r2q=10, quantum = rate/r2q, and 1500 < quantum < 200000 (sch_htb.c) with such r2q my default shaping window will be from 120Kbit to 16Mbit,

Hi, I have this message in my syslog: HTB: quantum of class 10001 is big. Consider r2q change. I don''t know why it''s there. I think all my setup is right. I am shaping traffic from my web server. It lives at three IP addresses, first is fast (and most important - aaa.bbb.ccc.1), second slower (aaa.bbb.ccc.2), third slowest (aaa.bbb.ccc.3). Total bandwidth is 2700kBps. Other

hi. When I execute follow class setting qdisc 1:0 root htb default 2 parent 1:0 classid 1:1 htb rate 10Mbit ceil 10Mbit parent 1:1 classid 1:2 htb rate 10kbit ceil 100Mbit HTB: quantum of class 10001 is big. Consider r2q change. <7>htb*g j=1014xxxxx HTB: quantum of class 10001 is big. Consider r2q change. <7>htb*g j=1014xxxxx HTB: quantum of class 10001 is big. Consider r2q change.

Hi, We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI> show channels Channel Location

If you look at the specs on the Dlink box that Primus gives you, you will see that it is SIP. I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP.

I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP. David >>> asterisk-users@eol.ca 1/21/2004 6:39:34 AM >>> I'm not sure Primus

Hello, I have 2 Dell 1955 blade servers, running RHEL5-Xen. I''m testing the migrate functionality from one blade to another. I can start the domain, move it to one blade (minor delay/packet loss) and everything is fine. When I try to move it back to the original blade the migration fails and the DomU crashes c1b1 = Blade 1 (192.168.131.201) c1b2 = Blade 2

Hi All I’m facing problem in bridging two SIP channels . I’m having SIP Trunk with Service Provider. Whenever I make any international call it get through and after some seconds it give error “Bridge stops bridging channels SIP/XXXX-3be3 and SI P/primus-9381” & call drops. May 13 17:25:19 VERBOSE[7491]: -- SIP/primus-9381 is making progress passing it to SIP/7125-3be3 May 13

Hi All I facing problem in bridging two SIP channels . I having SIP Trunk with Service Provider. Whenever I make any international call it get through and after some seconds it give error "Bridge stops bridging channels SIP/XXXX-3be3 and SI P/primus-9381"#12539;& call drops. May 13 17:25:19 VERBOSE[7491]: -- SIP/primus-9381 is making progress passing it to

Salut lartc (lartc@mailman.ds9a.nl), 18:12:52 kernel HTB init, kernel part version 3.10 18:12:52 kernel HTB: quantum of class 100001 is small. Consider r2q change. 18:12:52 kernel HTB: quantum of class 100002 is small. Consider r2q change. 18:12:52 kernel HTB: quantum of class 100003 is small. Consider r2q change. 18:12:52 kernel HTB: quantum of class 100004 is small. Consider r2q change.

Hi, My system always complain with the warning below : HTB: quantum of class 10099 is small. Consider r2q change. I simply do not understand why... Below my tc commands : /sbin/tc qdisc del dev eth0 root /sbin/tc qdisc add dev eth0 root handle 1 htb default 99 /sbin/tc class add dev eth0 parent 1: classid 1:2 htb rate 128Kbit /sbin/tc class add dev eth0 parent 1:2 classid 1:10 htb rate 80Kbit

Primus in Canada has launched a SIP-based service to replace your business and residential POTS lines with a VoIP version. It's called TalkBroadband and it looks killer: http://www.primus.ca/en/residential/talkbroadband/index.html Basic service for $20 Cdn a month!! Local number portability!! Cheapo Primus LD rates!! They don't care where geographically you plug it in!! When you sign

Hi, I have the setup of my xp100 plugging into my dlink gateway that i use with a voip provider. I notice that when someone calls my pstn # that goes to the asterisk box it works but when they hang up asterisk does not recognize the hangup. What needs to be done to make it work with a dlink gateway. when i plug the xp100 into my land line jack everything works fine and the xp100 reconizes

Hi all, A while back, there was a short thread on using the FXS interface from a Primus TalkBroadBand device (a DLink ATA) as a incoming line for the FXO interface on the TDM400P: Primus <--> DLink ATA FXS <--> TDM400P FXO <--> Asterisk In that thread, a couple of people suggested that this does not work reliabley, and the ATA FXS <--> TDM FXO link 'goes

Hey, I've checked all over and can't find what I need to know, so I'm posting here. I want to use Asterisk with my Primus VoIP service but it seems I need a username and password to authenticate with at Primus. Has anyone had any experience with this? How did you get it? Is it stored somewhere in the D-Link gateway they gave me? Thanks in advance and sorry if this makes no sense.

Hi list, For a client I am setting up a system which will use T1 PRI from Primus, who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only used switchtypes euroISDN and National. Although the documentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If

https://bugs.freedesktop.org/show_bug.cgi?id=75199 Priority: medium Bug ID: 75199 Assignee: nouveau at lists.freedesktop.org Summary: unsupported GT620M in Samsung All-in-One PC DP500A2D QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified OS: Linux (All) Reporter: slappinjohn

Hello, I have a problem with my GNU/Linux router. I mean, I am trying to configure a VPN conection for the clients of the LAN and allow to connect them to the Internet trought the router. I have installed in the server a QoS policy and I have configured the firewall for allowing all the clients to connect. I attach the script. The idea is that when a client connect this pc the dhcp gives him an