Displaying 20 results from an estimated 1000 matches similar to: "voipmonitor.org"
2015 Mar 25
5
Call Quality Measuring
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? but can?t other than
measuring the packet loss I don?t really know what I?m supposed to be
looking for
2013 Jan 05
8
Detect Low Quality Calls - Realtime
Hi there,
I support a large number of enterprise users who contractually must connect to
our support center via a 4G VOIP connection.
I simply want to be able to auto detect all poor quality calls in realtme (as
they are being made), play a message and drop the call - without user
intervention. All decent call quality calls will be allowed through - to be
handled by support staff.
Its a
2017 Aug 14
2
VoIP monitor and multiple RTP streams
Hello.
Is someone here using VoIPmonitor?
I am using just the sniffer and I found some pcap files that contain some
odd streams.
For example, I have a file with 3 streams, but the weird stuff is that 2
streams are the same (e.g., have the same source address and port and same
destination address and port).
Example:
"Source Address","Source Port","Destination
2012 Jul 26
2
Call ID of the second call leg
Hello friends,
I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can
access the caller Call ID (fbasename field in voipmonitor cdr) looking at
the SIPCALLID variable in asterisk, but how can I access from within
asterisk the Call ID of the second leg of the call (the one originating
from asterisk to the destination peer)? is there a variable holding this
value?
Thank you
2007 Jun 05
3
Outlook dialing
The bar is getting raised yet again
http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack
et8+Virtual+Office.aspx
I personally use Snapanumber $30 or there abouts (after trialing a few
other TAPI solutions and finding them sub-par) and think it's a great
product but interesting to see how more people are expecting
desktop/phone integration applications.
Does anyone
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
examples of "interesting" information like ICE result and howto make
"minimal" configuration of pjproject.conf
i.e.
forĀ debugging app_queue.so
core set debug 5 app_queue.so
for debugging RTP
core set debug 10 rtp_engine
core set debug 10 res_rtp_asterisk
rtp set debug on
logger.conf
rtp => debug,verbose(5)
so i mean
in
2014 Feb 12
1
Strange incoming call issue.
Hi all,
I've got a customer who's reporting "ghost calls." Essentially, the phone
rings, they pick up, and there's no body there.
It is NOT one-way audio, and it doesn't happen all the time.
We use voipmonitor to watch calls, and this is what we saw for the call in
question:
| calldate | caller | called | duration | whohanged |
2013 Dec 02
1
Not able to get remote channel variables containing RTCP values
I am not sure if its just me, but i am able to get only local channel
variables containing RTCP QOS values.
The Version is 1.8.14.
I want to store values of bridged channel in CDR.
Phone is Cisco 7941 SIP and with sip show channelstats i see all the
relevant information (jitter,packet loss) i want to get. It even
calculates packet loss in %. But i am not able to store it to CDR.
Asterisk 1.4
2017 May 31
8
OT: Want to capture all SIP messages
I want to capture all SIP messages.
I have about 30 hosts in about 6 colos.
My first thought was dumpcap, but the output file name format bugs me.
What do you use for long term SIP capture?
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
2015 Apr 01
0
Call Quality Measuring
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 Tue, Mar 31, 2015 at 1:16 PM, Patrick Beaumont <
p.beaumont at hatsoffsoftware.co.uk>
2018 Dec 05
3
Capture SIP all the time
Is there a way to configure the old SIP channel to stay in sip set debug
all the time, without human intervention and also at boot time, by default?
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2007 Jul 23
12
GRUB, zfs-root + Xen: Error 16: Inconsistent filesystem structure
Hi Lin,
In addition to bug 6541114...
Bug ID 6541114
Synopsis GRUB/ZFS fails to load files from a default compressed (lzjb) root
... I found yet another way to get the "Error 16: Inconsistent filesystem
structure" from GRUB. This time when trying to boot a Xen Dom0 from a
zfs bootfs
Synopsis: grub/zfs-root: cannot boot xen from a zfs root
2006 Jun 16
2
MOS Scores and LCR
Is there any tool that can do LCR for Asterisk but also take into
account MOS scores?
Is it possible to automatically generate MOS scores on random "calls"
so as to keep an updated database on a per provider, per destination,
per time-of-day score? Hopefully, with that information we can create
a better LCR module or script?
Thanks,
Daniel
2013 Jan 09
3
PESQ calculated MoS-Values for Speex
Hello,
I just signed up to this mailing-list (note: my first mailing list at all),
because I'm having some problems related to speex.
Let me just introduce you to what I'm doing.
I am writing a short (really short) paper about VoIP techniques, especially
audio codecs for speech.
I pointed out basic technologies behind audio codecs; vector quantization,
lpc, long-term prediction and some
2014 Dec 15
3
[LLVMdev] LLVM 3.6 update
Hi all,
Just wanted to give a quick update on the plan for the 3.6 release.
I had hoped to have at least posted a schedule by now, but I also
don't want us to have two releases in flight at the same time, so the
plan is to wait until 3.5.1 is done.
My tentative plan is to branch early January.
Please let me know what you think.
Cheers,
Hans
2010 Mar 08
3
Calculating R Factor and MOS metrics for VoIP
Hello All,
MOS and R factor are the two QoS parameters used to estimate VoIP call
quality.
I have found that they are calculated from other metrics like jitter,
latency, packet loss,...etc. But, haven't found any formula or arithmetic
rule to calculate them.
Do you have an idea about their formulas or an open source that calculates
them. Is it possible to interpret them from wireshark.
2007 Nov 21
2
Testing Help
Hi,
I am wondering if you may be able to help me.
I have an application that has been developed by a third party and is being
used in some field locations. My challenge is that I have no way of
independently testing the product across the WAN at this point. All
testing they have proposed required manual intervention and subjective
analysis. I am not comfortable deploying the solution in this
2009 Apr 01
2
Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten => s,n,ResetCDR(vw)
exten => s,n,NoCDR()
So I retrieve
2007 Nov 25
1
Testing Help
Jean-Marc Valin wrote:
>> What commercial VOIP test products can I turn to in order to objectively
>> evaluate your CODEC; in order to analyze audio streams for MOS scores and
>> performance parameters?
>>
>
> The MOS evaluation is subjective, not objective and actually involves
> getting (many) real people to listen to the audio, not just buying a
>
2006 Jun 22
4
Quality monitoring
Does anyone out there have a recommendation for tools that will monitor the
quality of VoIP systems? I am looking for jitter and MOS monitoring. I have
a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms
but I am looking for a little more detail. I would not be against writing
something in Perl for Nagios to do but I don't really know where to start on
measuring jitter