similar to: SLA/Shaping Question

I''m trying to fill a token bucket with enough tokens to burst several gigs of data. However, it doesn''t seem to get any higher than ~3.9GB: >tc qdisc add dev eth0 root tbf rate 1440kbit latency 50ms \ burst 16000000000 >tc qdisc show dev eth0 qdisc tbf 800b: rate 1440Kbit burst 3908420240b lat 2197.8s A smaller attempt of ~1.6 gigs works just fine: >tc qdisc

Hi All, I read the advanced networking howto and find a useful way to shape my adsl link. is says: "tc qdisc add dev ppp0 root tbf rate 220kbit latency 50ms burst 1500 [...] Change 220kbit to your uplinks *actual* speed, minus a few percent. If you have a really fast modem, raise ''burst'' a bit. " My adsl''s upload brandwidth is about 120 kbits/sec,

Hi, I am currently using TC for setting up user MAC based filters,and I am seeing a problem when I try to assign different priorities to the MAC Match filter apart from the root qdisc and parent class. Here is my script. /*****************************************************/ First of all I create the qdisc and class based on priority 2 and also the Hash mask tc qdisc add dev eth1 root handle

Hello, i have a server (486sx and 16ram). It gives me this error : NETDEVICE WATCHDOG eth1 : ...timeout. I checked in many places for solutions. Even wrote to this mailinglist. But lately i recived information, that too much upload may mix my adsl modem up. So..... my question is how should my script look while the situation is : eth0 - connected to adsl modem and thus to internet eth1 -

Hello, i have a server (486sx and 16ram). My pc is providing internet to 12 other computers. Ethernet cards are realteks 8139 (drivers builtin to the kernel 2.6.8). It gives me this error : NETDEVICE WATCHDOG eth1 : ...timeout. I checked in many places for solutions. Even wrote to this mailinglist. But lately i recived information, that too much upload may mix my adsl modem up. So..... my

Hi all, I tried last week to do traffic shaping on a production system, object of the exercise was simply to throttle the outgoing traffic. tc qdisc add dev eth4 root tbf rate 300mbit burst 300kb latency 50ms But the server became rather instable, crashing repeatedly without anything in the logs. Can anybody spot glaring mistakes in the tc command above, or tell me what I should have done

Hello all, My company is sharing it''s internet access with a neighbor and the neighbor in question was eating our bandwidth (both in upload and download) most of the time. I finally set up a linux router to limit the incoming/outgoing bandwidth to his network by adding a tbf on eth0 interface and eth1 interface of the router (neighbor is behind eth1 and we are in-between him and the

Hi all! I just started to experiment with bandwidth management using tbf. I have the folowing problems with token understanding: 1) What unit is token? Is "1 token = 1 byte" ? If this is so, then with rate=0.5mbit and HZ=100 the bucket gets filled with 0,64kb (655 tokens) every 10ms? 2) How much tokens are used for one packet (1500)? 3) And token unrelated question. Does

Hi All. I've been experimenting with SLA on Asterisk 1.4.13 (patched up to 1.4.14). I am using a SIP channel for my "trunk" line. On the whole things are good, but I have noticed that if I misdial an outgoing call, i.e. I get 404 "Not Found" in the SIP trace, then the trunk line just drops, rather than presenting an error tone or message to the user.

Fellow asterisk users, I am trying to get Single Line Appearance functionality working on a set of Linksys SPA942 phones and have not been successful. It looks like sla.conf is not getting read, only one phone reads as registered for the shared line, and a busy tone every time the shared extension is dialed. I have followed the documentation [1] and followed through other threads I saw

Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should

I have htb on 486 sx with 16mb ram. Slackware 9.1. Connection : dsl 1Mbit. 486 works as router and trafic shaper for network made of 12 pc''s. it does the job quite well, but when i play Counter-Strike or connect to my 486 via ssh (on lan), i get huge lag every 11-20 sec. when i connect to 486 via ssh and run iptraf program i see all the trafic, and after a while when lag comes the

I configured my sla.conf to use with a Polycom phone. I have no idea if I did it right, however, none of the console "sla" commands exist. Do I have to something special to compile in this support, or should it just work out of the box? ~jay

I understand that the Asterisk SLA implementation is somewhat different from most key systems and PBX systems. I also understand that in Asterisk, one does not put an SLA line on hold since it is just a MeetMe conference. However, is there any way to make the BLF flash when the answering party on the Asterisk system presses the hold key on their set and leaves the calling party alone in the

Has anyone done SLA with Polycom phones? I've got a large project coming up where the customer is keen on SLA for trunks and extensions. Trunks will be on a PRI. We may do this with Cisco phones if they work better. Mark Willis

Greetings, Back in September someone asked about documentation for the new SLA feature in 1.4, however they received no replies. I thought I might ask the same question now in December. Apart from sla.conf.sample and a few comments in app_meetme.c I have been unable to find useful documentation. Is anyone using this feature right now? Is there a helpful source for information this highly

Has anyone successfully gotten the features of sla.conf to work with Polycom phones? I would like to get 501's to show 3 shared lines on each of the keys. Particularly I am looking for a sample configuration for the phone, the Asterisk configuration seems simple enough. I've searched and searched and searched and have not found anything, general info about SLA yes, but nothing specific

Hello, every body Anyone set up, the sla sharing line appearances, in asterisk, I'm setting, tha but, don't, work, I change the sla.conf, extensions.conf, and sip.cfg, but don't work fine. Any one, could setup, tha? Regards Carlos Rojas -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to record all calls including calls that are part of a SLA. Using both monitor and mixmonitor the recording appears to happen (that is, asterisk logging shows it happening) however the file is never written to. It doesn't seem to be possible to use the recording that is part of the meetme app because when the SLA app creates the conference it passes NULL in the options field

Anyone have an example of using SLA with SIP handsets in Asterisk 1.6? I'm looking at the sla.tex file and wondering when it was last updated... Thanks, -Philip