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
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 or links to advice on measuring call
quality?
I’ve been playing around with “sip show channelstats” bu
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.
>
>
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.
Inbound calls are via in IAX trunk from Gradwell. CPU stats
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
Hi All,
Got a strange (at least IMHO) issue that doesn't make much sense to me.
Basic configuration is two sites with a site-to-site (aka router-to-router)
VPN. Both sites have Cisco 7961G phones [with SIP firmware] on users' desks,
and the only VoIP is internal - all of our outward telecom is
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
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
destination or just for one user etc... But till now I
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
@lists.digium.com
Subject: [asterisk-users] Call Quality Issues With 2 Trixbox's -
RouterIssue?
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 A200D
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 A200D card. The servers are IBM x3250 1RU servers
(1GB Ram, Raid 1 160GB HDD, Dual Core Xenon Pr
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
Hey all,
I recently got a message from my provider about IAX:
> We do not recommend the use of IAX. It is a lossy protocol that is
> known to cause crackling, loss of audio and other issues. You can
> use IAX if you want, but we will not assist with any issues you may
> encounter.
Does anyone el
day, October 02, 2006 6:46 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Call Quality / Echo / Problems
>
> Hi all
> I'm having a problem getting usable quality from my Asterisk setup.
>
> *SETUP*
> 2 Ghz PC with 1 GB Ram
Hi all
I'm having a problem getting usable quality from my Asterisk setup.
*SETUP*
2 Ghz PC with 1 GB Ram with TDM 400p 1 x FXS to route to analog phones
in the house and 2 x FXO to receive calls and in the future faxes.
Gentoo Linux
Here is what I've done so far
(1) Moved theTDM 400p (FXS,
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
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 for active calls.)
Suggestions appreciated!
- Mike
__
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
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.
The WAN comes in from the Cisco IAD and into a LAN switch (DLink
DGS-1005D w/ 802.1p) where the two public IPs are switched to different
devices.
Hi
all,Do you monitor call quality ?If positive, how do you proceed
?
How do you
estimate user's experience from rough lattency, MOS, throughput and so on
?
Which issues (echo ? call interruption ?) do you prevent with such
monitoring and which conter-measures do you engage when a problem
Hi
all,Do you monitor call quality ?If positive, how do you proceed
?
How do you
estimate user's experience from rough lattency, MOS, throughput and so on
?
Which
issues (echo ? call interruption ?) do you prevent with such monitoring
and which conter-measures do you engage when a problem
Hi all,
Do you monitor call quality ?
If positive, how do you proceed ?
Which issues (echo ? call interruption ?) do you prevent with such
monitoring and which conter-measures do you engage when a problem occurs ?
Cheers
Olivier
___
--Bandwidth a
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
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 making a call to
other extension (in
the same ne
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
HI
I JUST IMPLEMENTED ASTERISK AND IS SUCH A GRATE SOLUTION...I AM USING IT
POLYCOM 301 AND 501 PHONESON LAN A IAM USING G.711 AND I HAVE A 16 PORT
LINKSYS SWITCH...
THE PROBLEM IS WHEN SOMEBODY INSIDE THE NETWORK IS MAKING A CALL TO OTHER
EXTENSION (IN THE SAME NETWORK) AND FOR EXAMPLE IS S
I am having quality problems on SIP bound calls made over the Zap channels.
All Sip only calls (Cisco phone through Asterisk to another Sip device sound
fine).
Our setup looks like this:
User --> Executone PBX --> Asterisk Server --> Router --> Internet
The user is using a legacy handset that wo
From what I've searched in the archives, having Asterisk report call
quality statistics for each call has been discussed and mulled a little
bit, but it is not implemented in * and doesn't have a plan for it at
the moment. For the time being, such things are a pipe dream.
Is this correct? I'
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
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 their sound keeps
'cutting in and out'. I believe that the inc
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
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 info on this.
Thanks,
Calin.
_
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
I need to debug a call quality issue with remote users on the other
end of a satellite link. The symptoms are: we here on the Internet
side can hear them just fine. On their end, things work sorta OK most
times, but they often suffer from severe dropouts and digital
warbling, both of which I attr
Hello
All,
I have a system up
and running that will be used as a PBX lcaolly with SIP phones. Because I
am dumping all my calls into my X100Ps and have a very small number of clients
(15), I woudl like to set all my call quality variables to their highest
levels. I ahve a 100 meg networ
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,
D]
Subject: [Asterisk-Users] Call quality questions
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 hardw
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 at all with little over 20% proc
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