Hi Patrick,
You are welcome to try our tools out for active and passive voice quality
measurement tools. It's waveform analysis (like PESQ or POLQA) and VoIP
metrics analysis (like G.107 E-model and other metrics).
You can read more at http://www.sevana.biz
or older site http://www.sevana.fi
On
Thanks for the suggestions guys. I’ll try to have a play with Voipmonitor
in the near future.
So can I assume from the lack of discussion nobody is using the “sip show
channelstats” stuff?
Regards,
Patrick.
On 31/03/2015 08:23, "Olivier" wrote:
>Some SIP hardphones (Polycom) or softphones (Cou
Some SIP hardphones (Polycom) or softphones (Counterpath) embed a
module that metter MOS.
Regards
2015-03-25 14:21 GMT+01:00 Patrick Beaumont :
> Hi everyone.
>
> We regularly get customers complaining about call quality issues. Most of
> the time it turns out to be their own broadband. Very occ
tions.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Weiler
Sent: Thursday, 26 March 2015 7:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Quality Measuring
Hi Pa
Hi Patrick,
try voipmon, there it's free and you can even track MOS.
Markus
Am 25.03.2015 um 14:21 schrieb Patrick Beaumont:
Hi everyone.
We regularly get customers complaining about call quality issues. Most of
the time it turns out to be their own broadband. Very occasionally server
load.
Have you tried using tcpdump? Then analyze the pcap on wireshark?
Marlon Araujo
> On Mar 25, 2015, at 13:00, asterisk-users-requ...@lists.digium.com wrote:
>
> 1. Re: Call Quality Measuring (Laszlo)
--
_
-- Bandwidth and Co
On Wed, Mar 25, 2015 at 2:21 PM, Patrick Beaumont <
p.beaum...@hatsoffsoftware.co.uk> wrote:
> Hi everyone.
>
> We regularly get customers complaining about call quality issues. Most of
> the time it turns out to be their own broadband. Very occasionally server
> load. Does anyone have any advice
users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 June 2009 16:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug
On 2 Jun 2009, at 14:14, Adrian Marsh wrote:
> I'
t step, memory upgrade and the A*k upgrade.
Thanks,
Adrian
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: 02 June 2009 16:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
ists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Gibbons
Sent: 02 June 2009 15:09
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call quality - how to debug
2mb is small potatoes... unless you mean MegaBytes ins
digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh
Sent: Tuesday, June 02, 2009 10:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug
I don't need QoS.
The voice network here is seperated
On 2 Jun 2009, at 14:14, Adrian Marsh wrote:
> I’m at a loss of how to debug the voice issue further, without
> putting a wireshark PC on the switch, port-mirroring the server and
> then capturing all of the traffic in a round-robin-type capture and
> even then I’m not sure what that will ac
On Tue, 2009-06-02 at 15:49 +0100, Adrian Marsh wrote:
> However - my question would still stand, how exactly would I be able to
> debug whats going on in the RTP stream? And why its stuttering
> (sometimes halfway through a call).
>
> Any tips or tricks for actually debugging within Asterisk ?
W
lf Of Darrick
Hartman
Sent: 02 June 2009 15:22
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug
Do you have any idea the number of bugs that have been fixed since
1.4.15? Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug
From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
> Howes
> Sent: 02 June 2009 14:23
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Call quality - how to debug
>
>
> On 2 Jun
: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug
Hi,
It's a 2mb dedicated leased fibre line, with <50% utilisation.
My first thoughts were the internet link, but that wouldn't explain why
the client transmit (other chann
AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug
Hi,
It's a 2mb dedicated leased fibre line, with <50% utilisation.
My first thoughts were the internet link, but that wouldn't explain why
the client transmit (other cha
m] On Behalf Of David
Gibbons
Sent: 02 June 2009 15:09
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call quality - how to debug
2mb is small potatoes... unless you mean MegaBytes instead of
Megabits...
I am assuming you've already impl
Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 June 2009 14:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality - how to debug
On 2 Jun 2009, at 1
- "Steve Howes" wrote:
> On 2 Jun 2009, at 14:14, Adrian Marsh wrote:
>
> > Hi All,
> >
> > I’ve a 1.4.15 A*k server supporting several users (approx 80 total,
> > but <10 sim calls usually). I’ve one user who complains of
> > intermittent bad calls, though I suspect the bad calls are a
On 2 Jun 2009, at 14:14, Adrian Marsh wrote:
> Hi All,
>
> I’ve a 1.4.15 A*k server supporting several users (approx 80 total,
> but <10 sim calls usually). I’ve one user who complains of
> intermittent bad calls, though I suspect the bad calls are across
> the board, but intermittent.
>
>
On Mon, 2008-11-03 at 13:17 -0500, Lincoln King-Cliby wrote:
> It's conceivable, but how would I verify this and how would I change
> it if that was the problem?
There's a few things you can do here.
1) Check the sip.conf on both sides to see what is defined there for the
trunk. Look for some dis
D] [mailto:[EMAIL PROTECTED] On Behalf Of Lincoln
King-Cliby
Sent: Monday, November 03, 2008 1:17 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP
Bob,
It's conceivable, but how would I verify th
L PROTECTED] On Behalf Of Bob Pierce
Sent: Monday, November 03, 2008 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP
On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote:
> Any ideas why the a
On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote:
> Any ideas why the audio quality would be so markedly different when
> the only thing that seems to be different is where the call is
> originating from (POTS line vs. SIP phone)?
Is it possible that calls from your POTS line are going
Hello,
this is the case. Idle goes to 0% and IRQ goes to 100%.
I have a Junghanns ISDN card (bristuff) card. And I guess it is using
that Echo Canceler.
Best regards,
Loic Didelot.
On Thu, 2008-07-03 at 14:52 +1200, Matt Riddell wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Loi
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Loic Didelot wrote:
> Hi,
> I am using g711a everywhere.
>
> I checked on a completely idle system (no calls at all) and idle CPU is
> dropping from 100% to 0% more than once per minute.
If you run top, and the idle goes to 0% is it the IRQ that is u
2008/7/2 Loic Didelot <[EMAIL PROTECTED]>:
> Depends on the phone.
>
> On many devices you can setup buttons to call a url. Thats what I did.
Ah, yes. Would be a good thing to implement here. Then you can do
anything, like a support ticket etc.
Cheers.
___
Hi its me again.
Here is the output of zttest of a completely idle system (no calls).
Acoording to some documents those values do not seem to be good.
The IRQ of my zaptel card is shared with other devices. But not sure if
this causes a problem.
lspci -v | grep "IRQ 22" -B4
00:0c.0 ISDN contr
Hi,
I am using g711a everywhere.
I checked on a completely idle system (no calls at all) and idle CPU is
dropping from 100% to 0% more than once per minute.
procs ---memory-- ---swap-- -io -system-- cpu
r b swpd free buff cache si sobibo in
Depends on the phone.
On many devices you can setup buttons to call a url. Thats what I did.
Loic
On Tue, 2008-07-01 at 21:19 +0100, Gavin Henry wrote:
> What did you do to setup a button for alerts?
>
> Thanks.
>
> ___
> -- Bandwidth and Colocation
What did you do to setup a button for alerts?
Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNS
On Tue, Jul 01, 2008 at 03:22:07PM -0400, Steve Totaro wrote:
> Run top along with the tool that indicated the high I/O and see what
> is going on. Are you doing G729 or anything like that?
vmstat will probably provide more useful data (vmstat 1 etc. for a
continous run).
--
Tzaf
Run top along with the tool that indicated the high I/O and see what
is going on. Are you doing G729 or anything like that?
Thanks,
Steve T
On Tue, Jul 1, 2008 at 3:06 PM, Loic Didelot <[EMAIL PROTECTED]> wrote:
> Yes,
> but they get like 10 voicemails per day. That feature isnt really used
> al
Yes,
but they get like 10 voicemails per day. That feature isnt really used
alot.
Loic
On Tue, 2008-07-01 at 14:58 -0400, Steve Totaro wrote:
> I don't think your issue is the VIA CPU but the I/O of your flash
> drive. Voicemail is what I suspect being the I/O bottleneck.
>
> Thanks,
> Steve T
I don't think your issue is the VIA CPU but the I/O of your flash
drive. Voicemail is what I suspect being the I/O bottleneck.
Thanks,
Steve T
On Tue, Jul 1, 2008 at 2:52 PM, Loic Didelot <[EMAIL PROTECTED]> wrote:
> Yes, most calls are SIP-PSTN calls.
>
> Thanks for your help.
>
> I will try a
Yes, most calls are SIP-PSTN calls.
Thanks for your help.
I will try a faster box. Are VIA CPUs known to cause problems?
Loic
On Tue, 2008-07-01 at 18:50 +0300, Tzafrir Cohen wrote:
> On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote:
> > The problem appears mostly on outgoing calls
On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote:
> The problem appears mostly on outgoing calls SIP-PSTN but not only. 10%
> of all alerts are internal calls.
Any chance that omst of the calls are outgoing SIP->PSTN calls?
> I had the chance to notice the problem
> once myself but I
Try IOSTAT
http://www.linuxquestions.org/linux/articles/Jeremys_Magazine_Articles/Hunting_I_O_Bottlenecks_with_iostat
Maybe you can correlate VM and/or emailing of VM to your IO spikes.
Have you watched top and the Asterisk CLI when someone hits the panic button?
Thanks,
Steve T
On Tue, Jul 1,
The problem appears mostly on outgoing calls SIP-PSTN but not only. 10%
of all alerts are internal calls. I had the chance to notice the problem
once myself but I could never again reproduce.
Best regards,
Loic Didelot.
On Tue, 2008-07-01 at 18:03 +0300, Tzafrir Cohen wrote:
> On Tue, Jul 01, 2
On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote:
> Hello,
> one of my customers complained about bad voice quality on several calls,
> so I programmed a button on each phone which users can hit if they have
> audio drops and echo.
>
> I did this to check if there is a common recurrent
Recording the calls may or may not reveal an issue. I have personally
done this exact same method of troubleshooting only to find the
recordings were perfect but not the actual calls.
I think you should try just putting a regular server in place of your
"appliance" and then test.
I have a feelin
I considered that,
but I fear that this would load the machine even more. So I guess I
should take a more powerful box with a good harddrive (at the moment I
have a solid state flash card) and start recording calls.
Best regards,
Loic Didelot.
On Tue, 2008-07-01 at 09:10 -0400, David Backeberg
Hello,
I forgot to include CPU information
[EMAIL PROTECTED]:/usr/src/bristuff-0.4.0-RC2# cat /proc/cpuinfo
processor : 0
vendor_id : CentaurHauls
cpu family : 6
model : 10
model name : VIA Esther processor 1000MHz
stepping: 9
cpu MHz : 1000.127
cac
On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot <[EMAIL PROTECTED]> wrote:
> Maybe someone can help me to track down the problem. What should I
> check, monitor test. Any ideas are welcome.
If there are no legal reasons not to, consider recording all calls for
a limited time. It's easier for engineer
Hi,
its a new installation in a new office. Customer moved in, so right
moment to get a new PBX.
The box is running asterisk, nothing else:
- asterisk
- postfix just to send out voicemails
- no realtime
- som AGIS at call setup and call end
- Asterisk 1.4.19.1-BRIstuffed-0.4.0-RC2
- zaptel-1
I/O wait is very suspicious. What is your hardware platform? Is this
just a plain Jane PBX or are you doing anything unusual?
Thanks,
Steve T
On Tue, Jul 1, 2008 at 8:57 AM, Loic Didelot <[EMAIL PROTECTED]> wrote:
> I tried to get a little into cpu utilization and found the following
> results.
On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot <[EMAIL PROTECTED]> wrote:
> Hello,
> one of my customers complained about bad voice quality on several calls,
> so I programmed a button on each phone which users can hit if they have
> audio drops and echo.
>
> I did this to check if there is a common
I tried to get a little into cpu utilization and found the following
results.
Can they help me to come to a conclusion?
Best regards,
Loic Didelot.
[EMAIL PROTECTED]:~# mpstat 1
Linux 2.6.22-14-server (ppsite1)07/01/2008
02:54:40 PM CPU %user %nice%sys %iowait%irq %soft
2008/7/1 Loic Didelot <[EMAIL PROTECTED]>:
> Hello,
> one of my customers complained about bad voice quality on several calls,
> so I programmed a button on each phone which users can hit if they have
> audio drops and echo.
>
> I did this to check if there is a common recurrent problem to a given
P Please consider the environment before you print this e-mail or any
> attachments.
>
>
>
>
>
> __
> From: Paul Hales [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, 12 December 2007 4:40 PM
> To: Daniel Cole
> Subject: RE: [asterisk-users] Call Quality Issues With 2 Trixb
PROTECTED]
Sent: Wednesday, 12 December 2007 4:40 PM
To: Daniel Cole
Subject: RE: [asterisk-users] Call Quality Issues With 2 Trixbox's - Router
Issue?
Hmmm..wierd
Are you getting an weird jitter/latency figures in the CLI?
PaulH
On Wed, 2007-12-12 at 16:37 +1100, Daniel Cole wrote:
G72
any computer.
P Please consider the environment before you print this e-mail or any
attachments.
From: Paul Hales [mailto:[EMAIL PROTECTED]
Sent: Wednesday, 12 December 2007 4:10 PM
To: Daniel Cole
Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbo
What codec are you using?
PaulH
On Wed, 2007-12-12 at 13:00 +1100, Daniel Cole wrote:
> Hello Everyone,
>
> We have recently installed a pair of Trixbox servers in for a client
> of our. They have two locations, with one server each. The servers
> terminate 3 standard POTS lines into a Sangoma
lto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Wednesday, 12 December 2007 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbox's -
RouterIssue?
How are the calls being transferred from Box
Do an RTP analysis with Wireshark of a sample call. That could
probably narrow down the source of the problem. I would suspect you
will either see some jitter or packets out of order.
Daniel Cole wrote:
> Hello Everyone,
>
> We have recently installed a pair of Trixbox servers in for a clien
How are the calls being transferred from Box A to Box B?
On what box is the receptionist registered too?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Cole
Sent: Tuesday, December 11, 2007 9:00 PM
To: asterisk-users@lists.digiu
Funny you mention this because I've run into some voice degradation
problems with IAX2 myself recently...
When I have an external call come in on a DiD I frequently have to
send it back out to the PSTN (i.e. to a cell phone). When this
happens I don't want my server in the media path, I want to
Try running the echo test from both the house side and the co (outside)
side. That will let us know where the problem is.
Post results.
Alex
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Barry Fawthrop
> Sent: Monday, October 02
try "iax2 show netstats"
On 6/23/06, Dr. Michael J. Chudobiak <[EMAIL PROTECTED]> wrote:
Is it possible to set up some sort of call-quality statistics
reporting/logging for IAX2 calls? Something that can keep track of
dropped packet / jitter trends?
(I know "iax2 show channels" shows this info
Doug Lytle wrote:
I think the only time you need a timing source is if you are mixing
audio streams, i.e. meetme, MOH. In which case you'd probably need to
run ztdummy.
Yes , ztdummy is running.
I'm going to (temporarily) put a TDM card in the system just to
eliminate that possibility.
Michael Welter wrote:
I'm not on site, but I remember 1.6.4.
I had in place 1.6.2, and had way to many problems with it. I reverted
back to 1.5.2 and things cleared up.
Is the phone (or Asterisk) performing echo suppression that drops the
last part of the tone?
I believe the phone does
Doug Lytle wrote:
Michael Welter wrote:
Doug Lytle wrote:
Michael Welter wrote:
The machine is totally idle.
The T1 vendor noticed 2% packet loss during a ping flood originating
from outside. We changed the Cisco IAD, and there is no longer packet
I've noted from employees that the volum
Michael Welter wrote:
Doug Lytle wrote:
Michael Welter wrote:
The machine is totally idle.
The T1 vendor noticed 2% packet loss during a ping flood originating
from outside. We changed the Cisco IAD, and there is no longer packet
I've noted from employees that the volumes levels on the phon
Doug Lytle wrote:
Michael Welter wrote:
I'm having difficulty with an Asterisk system. The external party has
very good call quality, but the internal party hears clipping and drop
outs.
RX Gains too high
IRQ sharing of the of the ZAP device
There is no ZAP device (it is a SIP-only imp
Michael Welter wrote:
I'm having difficulty with an Asterisk system. The external party has
very good call quality, but the internal party hears clipping and drop
outs.
RX Gains too high
IRQ sharing of the of the ZAP device
High load of the machine
Are a few that come to mind.
Doug
--
Be
On 03:57, Sat 20 Aug 05, Ing. Marlo R. Beltran G wrote:
> Hi
>
> i just implemented asterisk and is such a grate solution...i am using
> polycom 301 and 501 phoneson lan a iam using g.711 and i have a
> 16 port linksys switch...
>
> the problem come when somebody inside the network is m
Hi
Some basic mailing lists ethics:
1. writing in CAPITAL LETTERS usually indicates SHOUTING. Please don't
do that.
2. when you want to start a new message to the list, write a new
message, and don't just reply to an existing list message.
3. Proper English is also preffered, so readers spend
Thanks for the reply, Adam.
If this is the case, it would seem to me (because the degradation
happens only after a period of time, and quite suddenly) that the issue
lies with digium's implementation of g729.
As an interesting note, I had the same problems using ulaw -> ulaw over
the local
Thanks for the reply, Adam.
If this is the case, it would seem to me (because the degradation
happens only after a period of time, and quite suddenly) that the issue
lies with digium's implementation of g729.
As an interesting note, I had the same problems using ulaw -> ulaw over
the local n
On Thu, 2005-07-21 at 15:56 -0400, Adam Dobrin wrote:
> I'm using Polycom 501's; with stable1.0.8, g729 and a very decent
> machine; we have a PRI interface to a T1.
>
> Many users complain that after a given amount of time, say, 30 or 40
> minutes on a call, the outside party complains that th
At 3:32 PM +0200 on 3/17/05, Calin Serbanescu wrote:
Hello,
I need some help setting up statistics per call. I need to store in a
database call quality details such as jitter, packets lost and other
informations. Is there any way to do this?
I'd really appreciate some links or any other kind of inf
Chris Icide <[EMAIL PROTECTED]> writes:
> Satellite links can be pretty tough to troubleshoot. It sounds like
> you are running into a uplink buffer issue. On heavily loaded
> uplinks, the input buffers can get quite large, and if the satellite
> provider isn't using some form of buffer handling
mjr,
Satellite links can be pretty tough to troubleshoot. It sounds like
you are running into a uplink buffer issue. On heavily loaded
uplinks, the input buffers can get quite large, and if the satellite
provider isn't using some form of buffer handling that prioritizes udp
traffic, it may be th
Lane Hoskins wrote:
> Our basic system is as follows:
>
> P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH 9.0 OS, * from CVS
> several weeks ago, working OK for routing, VM, and AA, calls in on
> separate PSTN lines to Adtran TSU 600, into * server through T100P
> card. The hardware is not taxed a
Thanks...
-Original Message-
From: mattf [mailto:[EMAIL PROTECTED]
Sent: Fri 1/30/2004 5:16 PM
To: '[EMAIL PROTECTED]'
Cc:
Subject: RE: [Asterisk-Users] Call quality questions
Hello,
Hello,
Did you set the flag in the makefile for zaptel for SMP kernels?
1. I have a couple Snom200 phones on my system running redhat with a P4 HT
and haven't had any issues with horrible sound quality using 711ulaw.
2. As for the speakerphone cutout, that's to be expected, The snom200s are
just
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