Hello.
Recently I installed voipmonitor and voipmonitor-gui trial version.
After examining it, I was amazed at the abundance of useful information
that can and should be obtained from the work of Asterisk.
1. The cost voipmonitor-gui too expensive and not justified in my case.
For this reason,
On Tuesday 14 January 2014, richard.seg...@marisec.ca wrote:
> I asked this on the list over the weekend, and likely missed a few people
> inboxes.
>
> I'm having a problem pulling data from RTPAUDIOQOS.For testing purposes
> I have asterisk sending QOS data to the console. It seems I get QO
4 10:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk QOS
I asked this on the list over the weekend, and likely missed a few people
inboxes.
I'm having a problem pulling data from RTPAUDIOQOS.For testing purposes I
have asterisk sending QOS data to the cons
I asked this on the list over the weekend, and likely missed a few people
inboxes.
I'm having a problem pulling data from RTPAUDIOQOS.For testing purposes I
have asterisk sending QOS data to the console. It seems I get QOS data only
if the caller hangs up, with the variable being empty
On Thursday 05 August 2004 06:56, Andrew Kohlsmith wrote:
> #!/bin/bash
... well that got bitched up sufficiently...
http://www.mixdown.ca/~andrew/dump/rc.tc is a copy of the script -- kmail is
trying to be smart and substituting soft line breaks for hard ones... ugh.
-A.
_
On Wednesday 04 August 2004 05:26, lists-jmhunter wrote:
> # lower the MTU to decrease latency
> #$IP link set dev $DEV mtu $MTU
Just a note -- you're not lowering your MTU to 1492 to reduce latency (the
default is 1500), you are reducing it because you're running PPPoE and those
8 bytes are the
On Wednesday 04 August 2004 05:26, lists-jmhunter wrote:
> As seen on my post at:
> http://www.sveasoft.com/modules/phpBB2/viewtopic.php?p=28112#28112
> This works very well... It does NOT work with stable 4.0! sveasoft
> will be issuing a bug fix for this (4.1) in the near future.
I've been usin
1) I would think pfifo would be a better choice than sfq for your voip
qdisc. Something like:
$TC qdisc add dev $DEV parent 1:10 handle 10: pfifo limit 10
2) Marking packets worked better for me. I could never get it to work
any other way. (Hey, I'm not arguing. I'm jealous.)
3) Shouldn't y
As seen on my post at:
http://www.sveasoft.com/modules/phpBB2/viewtopic.php?p=28112#28112
This works very well... It does NOT work with stable 4.0! sveasoft
will be issuing a bug fix for this (4.1) in the near future.
Final Rev of working script w/ asterisk support
I'm not going to run alchemy
