2013/1/5 joachim zoach...@securax.org
You are pretty much limited to measuring the delay and the jitter.
The delay you can somewhat estimate prior to the call (with qualify for
example).
The jitter / packetloss you can only figure out when the call is already
up for a while. (e.g. you might
2013/1/8 Lenz Emilitri lenz.lo...@gmail.com
2013/1/5 joachim zoach...@securax.org
You are pretty much limited to measuring the delay and the jitter.
The delay you can somewhat estimate prior to the call (with qualify for
example).
The jitter / packetloss you can only figure out when the
A few years ago I spoke to a Finnish company that had a commercial
solution for automated MOS estimation. So something exists though I
have not tested it first-hand.
l.
--
You need a lot of data to calculate a MOS score, you will need the
actual call.
The only solution i can think of is
Greetings.
I got two extensions on my asterisk that autenticates from outside our
network, via internet. Is there a way to monitor, in certain time periods,
if they are available (online) and send some sort of notification if they
don't?
There are two extensions to monitor, they belong to the
2013/1/8 Luis H. Forchesatto luisforchesa...@gmail.com
Greetings.
I got two extensions on my asterisk that autenticates from outside our
network, via internet. Is there a way to monitor, in certain time periods,
if they are available (online) and send some sort of notification if they
When i worked in an internet provider with asterisk telephony solution - we
used Aqua (http://www.sevana.fi) to measure voice quality. several nettops were
spread across our network. The nettop called to our asterisk, the asterisk
saved this voice file to the disk, then this file was sent to
Top and bottom post in the same email... don't open again the thread :-)
#!/bin/bash
res=`sudo /usr/sbin/asterisk -rx 'sip show peer $1' | grep Status | cut
-d\: -f 2 | cut -d\ -f 2`
if [ $res == OK ]
then
echo OK is registered
exit 0
else
echo WARNING peer not registered
exit 1
2013/1/8 Luis
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, January 07, 2013 6:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX2 support of video
According to this:
Hello Users,
I've been searching for a couple of hours now but I can't find the answers to
my questions, so here they go:
1) Is it possible to stream audio files from a webserver during a call by
configuring this in the dialplan? Something like
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant
Bagdasarian
Sent: Tuesday, January 08, 2013 9:24 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Streaming/Recording audio
Hello Users,
I've been searching for
Valer,
Thank you for the advice - I have support tickets open with Adtran and Digium
and we are tracking down the issue. Hopefully it doesn't come down to adding
more hardware, but I'll keep that in mind.
-Justin Killen
From:
Hey Satish,
I've worked this out. I'm sorry, you were completely right and the context is
fine. I was testing without answering the call, so the Dial was never
connected! Doh!
Thanks heaps for your help, it's all working perfectly.
Cheers,
Andrew
From:
Hi,
I am working on Asterisk-10.11.1,I tried to generating outbound call through
.call file and facing a issue that call retry was happening after call
Answered.Is it bug in that Version or i missed some thing.
Here is my call file is-
Channel: DAHDI/G1/09990212758
MaxRetries: 2
RetryTime: 60
Hello,
For some reason I did not receive any replies related to my question by mail,
but I found the topic back on the online mailing archives. I hope by supplying
the same subject this email will be logged in my previously created topic
instead of a new one. If it does not, I apologize.
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