hi group ,
i am working on dahdi_monitor for measuring voice quality , so i want to
know that on which data i can tell that this PRI
lines are working properly, is there any measurement on basis of that i can
make MOS. i am working from last 2-3 days
but i only get idea about making .raw file and
) or just contact us directly.
Best Regards,
Sevana Oy
Finland
- Original Message -
From: DHAVAL INDRODIYA
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, February 04, 2011 12:53 PM
Subject: [asterisk-users] voice quality measurement using dahdi_monitor
Am 04.02.2011 10:53, schrieb DHAVAL INDRODIYA:
hi group ,
i am working on dahdi_monitor for measuring voice quality , so i want
to know that on which data i can tell that this PRI
lines are working properly, is there any measurement on basis of that
i can make MOS. i am working from last 2-3
On Fri, Oct 08, 2010 at 05:51:48PM +0400, Sevana Oy wrote:
One quick clarification please... With Fluke ACEs you measure MOS according
G.107, E-model, right?
Don't know if it is g.107 but it has to be something similar if it
isn't. I'm personally more interessted in the network statitics.
--
Hi,
How do you typically test voice quality in Asterisk? For example if you like to
do load testing, or monitor voice quality and get notified if certain calls had
bad quality for proactive maintenance?
Thank you!
Best Regards,
Sevana Oy
http://www.sevana.fi--
The professional way is to do a series of test calls, play a reference
file and record the audio at the incoming side. You then use both files
to calculate a MOS score. This method is used by telco's to do quality
checks.
https://secure.wikimedia.org/wikipedia/en/wiki/Mean_opinion_score
On Fri, Oct 08, 2010 at 02:24:11PM +0200, Bert Van Kets wrote:
The professional way is to do a series of test calls, play a reference
file and record the audio at the incoming side. You then use both files
to calculate a MOS score. This method is used by telco's to do quality
checks.
Take a
.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bert Van Kets
Sent: Friday, October 08, 2010 8:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voice quality assessment in Asterisk
@lists.digium.com
Sent: Friday, October 08, 2010 4:41 PM
Subject: Re: [asterisk-users] Voice quality assessment in Asterisk
On Fri, Oct 08, 2010 at 02:24:11PM +0200, Bert Van Kets wrote:
The professional way is to do a series of test calls, play a reference
file and record the audio at the incoming
On Thu, 23 Apr 2009, Rilawich Ango wrote:
Hi all,
I wonder who has the same voice quality problem as what we have.
Below is our configuration.
Company --- asterisk 1.4.22 (g729) --- CISCO --- T1 --- customer
Sometimes, customers told me that they heard our voice not very clear,
like a
Normally, there are 10 concurrent calls in peak. You are right that
usage g729 is due to bandwidth consideration.
On Thu, Apr 23, 2009 at 2:42 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Thu, 23 Apr 2009, Rilawich Ango wrote:
Hi all,
I wonder who has the same voice quality
Hi all,
I wonder who has the same voice quality problem as what we have.
Below is our configuration.
Company --- asterisk 1.4.22 (g729) --- CISCO --- T1 --- customer
Sometimes, customers told me that they heard our voice not very clear,
like a call from far far away. I heard the recording is
Hi all;
I have two asterisk boxes installed in two separated
sites, the internet bandwidth between them is very
good and I am using G729 codec to communicate with
them and IAX.
The problem that side A hears well side B while side B
does not hear well side A !!
I did one thing in side B that in
bilal ghayyad wrote:
Hi all;
I have two asterisk boxes installed in two separated
sites, the internet bandwidth between them is very
good and I am using G729 codec to communicate with
them and IAX.
Try playing around with the adaptive / fixed jitter buffer settings for
IAX2. Also, if
On Wed, Mar 5, 2008 at 6:09 PM, Alex Balashov [EMAIL PROTECTED] wrote:
bilal ghayyad wrote:
Hi all;
I have two asterisk boxes installed in two separated
sites, the internet bandwidth between them is very
good and I am using G729 codec to communicate with
them and IAX.
Try
On Wed, Mar 5, 2008 at 6:53 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Wed, Mar 5, 2008 at 6:09 PM, Alex Balashov [EMAIL PROTECTED] wrote:
bilal ghayyad wrote:
Hi all;
I have two asterisk boxes installed in two separated
sites, the internet bandwidth between them is
Steve Totaro wrote:
Try using SIP. Post back with results.
Or that. Certainly, my own experiences with IAX2 would support this
conclusion.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706)
Being new to Asterisk, I have set up a little test rig at home.
Asterisk itself is the Debian Etch package, running on a Celeron 1.7G
machine
I have two clients on a 100Mb LAN, a Windows XP machine with an Athlon
XP2200+ processor, and a Linux Core Duo 6300 machine.
On the windows XP machine
On Sun, Mar 04, 2007 at 11:03:21AM +, Alan Chandler wrote:
Being new to Asterisk, I have set up a little test rig at home.
Asterisk itself is the Debian Etch package, running on a Celeron 1.7G
machine
I have two clients on a 100Mb LAN, a Windows XP machine with an Athlon
XP2200+
Hi Folks,
Looking for feedback on the cordless phones with the Aastra 480i CT handset. Is
voice quality comparable to standard consumer residential 2.4GHz cordless
phones in the US$30 - $50 price range, or better/worse?
How about handset and speakerphone quality for the main phone?
Seems
a few users need basic in-office mobility.
Cory Andrews
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Keagy
Sent: Friday, November 17, 2006 2:09 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] voice quality of Aastra
not been
able to find another unit that has this same feature.
Curt
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Keagy
Sent: Friday, November 17, 2006 1:09 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] voice quality of Aastra 480i CT
Scott,
I've used the phone for 9 months. It's a truly outstanding phone. The cordless
handset sounds great. It is limited to two ongoing calls at one time, but that
has not been an issue for me.
The range on the cordless is comparable to the Panasonic KX-TG4000 KSU that I
used to use and a
Aastra is a great phone for sound quality and other features. I didn't have
any problems with it and didn't go back to Grandstream once installed
Aastra. My only concern was some problem with its web UI bugs, but that will
be eventually fixed.
___
Hi ,
Does anyone encounter this problem ? We have installed Asterisk at Site
A and have 128k Frame Relay over to Site B.
We are using Zyxel 2 port FXS at Site B and Linksys PAP2-NA at Site A.
We are using Ulaw at Site A and G729 at Site B.
When the calls are originated from Site A to Site B,
On Tue, 04 Oct 2005 14:28:47 +0800, Stephen wrote:
Hi ,
Does anyone encounter this problem ? We have installed Asterisk at Site
A and have 128k Frame Relay over to Site B.
We are using Zyxel 2 port FXS at Site B and Linksys PAP2-NA at Site A.
We are using Ulaw at Site A and G729 at Site B.
I've tried almost
any softphone available on the market with many different PC, soundcard,
headphones combinations.
None of them prooved
production reliable in a call center environment.
I've also tested
many IP Phones and Gateways. Even the cheapest one supplies much better quality.
Is
over.
MATT---
-Original Message-
From: Cenk Yabas [mailto:[EMAIL PROTECTED]
Sent: Saturday, June 11, 2005 10:03 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voice quality of Softphones vs. IP Phones and
Gateways.
I've tried almost any softphone available on the market
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voice Quality
Hello,
I have setup two * servers and they are communicating using IAX. I'm
passing calls from SRV A (internet connection T1) to SRV B (internet
connection: 512).
For some reasons I have an issue with the quality
What's your end device? if it's a voip device (eg SIP phone or a soft
phone) then you shouldn't need a jitter buffer.
Also, you don't need bandwidth=low if you specify the codecs (the
disallow=all will override the bandwidth=low) and maxjitterbuffer is the
param you're after with this line
Jonathan wrote:
Andrew Kohlsmith wrote:
BTW are you *really* saving any time by bastardizing your email so
much (ur, u, bcz)... jeez.
I think they teach that crap in school these days ... kids and their
sms cell phones..
I thought it was trying to simulate a high packet loss. :-)
Regards,
: Wednesday, May 04, 2005 2:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Voice Quality
I would use g.729, and if this is an issue, GSM.
Setup trunking between both IAX peers so that you can save a lot of
bandwidth.
-Original Message-
From
Hello,
I have setup two * servers and they are communicating using IAX. I'm
passing calls from SRV A (internet connection T1) to SRV B (internet
connection: 512).
For some reasons I have an issue with the quality. The voice is a bit
scratchy. I have tried iLBC and SPEEX, but it didn't make any
[EMAIL PROTECTED] wrote:
Hello,
I have setup two * servers and they are communicating using IAX. I'm
passing calls from SRV A (internet connection T1) to SRV B (internet
connection: 512).
For some reasons I have an issue with the quality. The voice is a bit
scratchy. I have tried iLBC and SPEEX,
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, May 03, 2005 2:51 PM
Subject: [Asterisk-Users] Voice Quality
Hello,
I have setup two * servers and they are communicating using IAX. I'm
passing calls from SRV A (internet connection T1) to SRV B
] On Behalf Of Sean Kennedy
Sent: Tuesday, May 03, 2005 11:33 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Voice Quality
[EMAIL PROTECTED] wrote:
Hello,
I have setup two * servers and they are communicating using IAX. I'm
passing calls from SRV A (internet connection T1
David:
Bandwidth may be an issue; however, do you have any timing devices
installed? Digium's hardware (or any generic knockoffs) will provide
this. There are also some other ways, such as ztdummy or a usb
controller (haven't used either of these, so I don't know any
specifics. Check the Wiki).
allow=ulaw or allow=gsm is all you need at both locations.
On 5/3/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello,
I have setup two * servers and they are communicating using IAX. I'm
passing calls from SRV A (internet connection T1) to SRV B (internet
connection: 512).
For some
:[EMAIL PROTECTED] On Behalf Of Andrew Latham
Sent: Tuesday, May 03, 2005 1:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voice Quality
allow=ulaw or allow=gsm is all you need at both locations.
On 5/3/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote
-Commercial Discussion
Subject: Re: [Asterisk-Users] Voice Quality
David:
Bandwidth may be an issue; however, do you have any timing devices
installed? Digium's hardware (or any generic knockoffs) will provide this.
There are also some other ways, such as ztdummy or a usb controller (haven't
used
On 16:14, Tue 03 May 05, David wrote:
Andrew,
Thanks for ur response. Hmmm, But I will have a bandwidth issue if I use
ulaw. 20 calls at the same time... 512 connection can't support that... What
do u think? The only reason I'm using iLBC is bcz of the Bandwidth. What
about the packet
On May 3, 2005 04:14 pm, David wrote:
Thanks for ur response. Hmmm, But I will have a bandwidth issue if I use
ulaw. 20 calls at the same time... 512 connection can't support that...
What do u think? The only reason I'm using iLBC is bcz of the Bandwidth.
What about the packet lost, I see some
Andrew Kohlsmith wrote:
BTW are you *really* saving any time by bastardizing your email so much (ur,
u, bcz)... jeez.
I think they teach that crap in school these days ... kids and their sms
cell phones..
Jonathan
___
Asterisk-Users mailing list
On 5/3/05, David [EMAIL PROTECTED] wrote:
Andrew,
Thanks for ur response. Hmmm, But I will have a bandwidth issue if I use
ulaw. 20 calls at the same time... 512 connection can't support that... What
do u think?
I would try ulaw just to get a basis to work from. Frankly I would
expect
I have 2 * server, each connects to a IPtel SER server.
i have another ATA which also connects to IPtel SER server.
when * call * thru IPtel server, i have jitter and voice quality problem,
but when i use the ATA to call both * server, there is no problem, the
quality is superb. the same thing
hello list ,
iam using a simple setup as shown below
ip device --- ser -- asterisk (astcc) ---pstn gatewsy
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hello list ,
my set up is like this
ip device --ser --- asterisk(astcc) -- pstn gatewsy
my asterisk version is 1.0.2
iam using the ser as registration and asterisk aa the
prepaid one with the help of the astcc.
now my problem is the destination people
i.e the pstn line s are listening
VoIP technocrat wrote:
hello list ,
iam using a simple setup as shown below
ip device --- ser -- asterisk (astcc) ---pstn gatewsy
:-)
Are you having some problems? Is the voice quality good/bad?
What is your network load?
What is your PC load?
What PSTN Gateway are you using?
What codecs?
Snippet
voip technocrat wrote:
hello list ,
Hello!
:-)
my set up is like this
ip device --ser --- asterisk(astcc) -- pstn gatewsy
my asterisk version is 1.0.2
Latest stable package is 1.03 with 1.04 to be released very shortly :-)
iam using the ser as registration and asterisk aa the
prepaid one with the
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