Displaying 20 results from an estimated 4000 matches similar to: "Using Asterisk to measure call quality: Introducing Recqual"
2008 Dec 31
2
Friday VUC 12 Noon ET with Kristian Kielhofner: Identifying Asterisk Quality Issues
Happy New Year in advance by a few ticks for the northern hemisphere.
Here's the first topic and guest for 2009:
In any voice path there are several potential sources of quality
problems, ranging from
echo to voice dropouts and everything in between. With VoIP systems
the potential for
quality problems increases dramatically, often times making it very difficult to
identify the source of
2009 Feb 25
2
SheevaPlug Development Kit
Hello everyone,
I just ordered one of these:
http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp
Just over $110 with shipping but they are expecting the price to
come down quite a bit:
- 1.2Ghz ARM5
- 512MB RAM
- Multiple flash storage options
- Gigabit ethernet
- USB 2.0
- 5 watt power usage
They probably won't be shipping until late March but I
2008 Nov 01
1
VoIP traffic shaping
This was so interesting I had to move it to its own thread!
Is anyone using this script? How does it perform compared to the older
WonderShaper script?
-M-
==================
Thanks Kristian I will checkout the new script and see how it goes!
Jonn
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at
2008 Nov 10
6
changing the size of voice packets
Dear,
is any way to change , the size of voice packets?
I want to increase the quality by decreasing the size of each packets, because of bandwidth failure.
?
thanks in advance
Mani
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2013 Feb 06
2
Somewhat OT: Specific SIP packets can cause ethernet controller reset
While not strictly Asterisk related this issue could certainly affect
some of you:
http://blog.krisk.org/2013/02/packets-of-death.html
--
Kristian Kielhofner
2010 Jan 28
1
Use of "603 Declined"
Hello everyone,
I've had the time to examine some specific serial/parallel forking
scenarios with Asterisk lately. Looking at chan_sip it appears that
anytime Asterisk wants to tear down a call before it's brought up, it
sends a 603 Declined:
} else { /* Incoming call, not up */
const char *res;
2005 May 28
1
Pictures of the Digium booth at ISPCon 2005
Hello everyone,
Even though a lot of it was a bit last minute, several of us from the
commnunity made it to Baltimore to help Digium with their booth at
ISPCon. It was a great time.
Gregory Boehnlein, Brian Capouch, Christian Savinovich, Kristian
Kielhofner (me), and John Todd (not pictured) were there (as well as
others), and some pictures were taken (the up close ones of me were very
2009 Oct 19
3
delay in processing dtmf
Hi,
I'm new to this list
I'm developing asterisk application where users can call and control volume
up and down in music player.
Problem I'm getting is if users press 222228 in fast speed, system will
process all those 2s and then process 8, so there is few seconds ( around
4-5) processing key press 8 , therefore users will feel unresponsiveness in
system.(in other words users will
2006 Feb 06
12
Asterisk native sounds now available!
Hello everyone,
As I promised at eTel last week, I have finished up work on my
"Asterisk Native Sounds" project. Here's a little diddy from astlinux.org:
-----------------------------------
Asterisk Native Sounds are a collection of audio prompts for Asterisk.
They will improve quality, reduce CPU usage, reduce latency, and (in
some cases) eliminate the need for G729
2003 Jul 31
3
using vcut on split ogg files
hi,
I am recording a continuous 24/7 broadcast using ecasound, oggenc and
cronolog.
because the broadcast is continuous and to not lose any data, I split up
the recorded stream into a file for every 30 minutes using cronolog
(http://www.cronolog.org).
o the chain looks like the following:
ecasound | oggenc | cronolog
the problem is the following, because the oggfiles are split, they lack
a
2004 Oct 13
4
cpu usage for ices and oggenc
I'm running ices-kh to stream from jack at 64kbps, and also using oggenc
(with ecasound via jack) to record the audio to disk at the same time.
This is also running at 64kbps.
ices is using virtually no cpu (0.0%), but oggenc is using 15-16%. I can't
see why there should be such a difference - both are recording the same
audio stream in real time at the same bitrate. oggenc is getting its
2004 Oct 13
4
cpu usage for ices and oggenc
I'm running ices-kh to stream from jack at 64kbps, and also using oggenc
(with ecasound via jack) to record the audio to disk at the same time.
This is also running at 64kbps.
ices is using virtually no cpu (0.0%), but oggenc is using 15-16%. I can't
see why there should be such a difference - both are recording the same
audio stream in real time at the same bitrate. oggenc is getting its
2008 Nov 10
3
directrtpsetup without reinvite
Hi,
I want to be able to bridge two sip channels using direct RTP
between my endpoints (Audio IP : not local) but without
using reinvites. So I set up my asterisk sip endpoints as follows:
[test1]
type=friend
host=dynamic
username=test1
dtmfmode=info
context=test_rtp
allow=all
canreinvite=no
directrtpsetup=yes
[test2]
type=friend
host=dynamic
username=test2
dtmfmode=info
context=test_rtp
2006 Jun 07
1
asterisk-1.2.9 / res_snmp.so
--- hgaillac-sip@yahoo.fr a ?crit :
> hello,
>
> How asterisk could support res_snmp even this module
> don't help to monitor all asterisk features?
>
> monitoring asterisk with snmp would be a good
> thing.
> Which solution ?
>
> Harry
> --- Kristian Kielhofner <kris@krisk.org> a ?crit :
>
> > hgaillac-sip@yahoo.fr wrote:
> > > I
2004 Oct 18
1
Re: ices-kh dropping jack ports unexpectedly
On Mon, 18 Oct 2004 16:15:44 +0100, EvilOverlord wrote:
>> The setup I'm working with is an ecasound session relaying audio between
>> its input and output ports, with its output connected to all the ices
>> clients for the different streams we're running. Then I'm
>> switching the ecasound input between different stream sources (line in,
>> recorded
2008 Dec 11
5
Linux Software to monitor quality of bandwidth for carrying voip traffic - suggestions please?
Hi,
Would like to run the software to monitor the quality of the bandwidth.
Suggestions welcome?
Thank you.
Shaun
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2005 Mar 19
2
More HEAD wierdness (chan_sip, jitterbuffer/PLC problems)
Hello,
After checking out CVS HEAD from yesterday (for those new
PLC/Jitterbuffer patches), I was affected by bug 3795 with my Polycom
IP600's. After seing it resolved as of this morning (thanks Mark), I
decided to try again...
I can answer incoming calls. No problem there. Putting calls on hold,
however, results in my Polycom IP600 indicating the call on hold, but
the caller does
2004 Oct 18
9
ices-kh dropping jack ports unexpectedly
I've been having a problem where ices-kh (the jack'ified version)
disconnects from its jack input source unexpectedly. This happens mainly
while other jack clients are being started/stopped, or
connected/disconnected, but also at other times (e.g. switching between
different X sessions). I'm planning to do a bit more work on tuning up the
jack setup to see if I can get rid of the
2013 Sep 20
1
Somewhat-OT: Stupid NAT tricks to learn from Apple?
I've been spending some time looking at some of the significant
changes Apple has made to Facetime in iOS 7. I'm far from an Apple
fanboy but some of them are pretty interesting:
- multiplexing everything over a single UDP port
- deflate compression with SIP
- various /slight/ protocol violations ;)
More here:
http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html
As
2008 Oct 22
6
fax / t38 gateway
I'm trying to figure out how to handle our fax line when we switch to
our asterisk for voice. After a lot of reading and poking about I have
concluded, as have many others it would seem, that the best thing to
do is either to have a separate pstn fax line or use some sort of
internet faxing service rather than try and make faxing work in a way
it's not meant to over voip lines.