Re: [asterisk-users] asterisk server stress test
Hi,
Curious why didn’t you try AQuA http://sevana.biz/products/aqua/ to score
the quality? Using voice files for tests has more representation to my
opinion.
Thanks,
vallu
On Thu, Aug 20, 2015 at 4:11 AM, Pete Mundy p...@fiberphone.co.nz wrote:
Markus
That's a fascinating concept!
Can you share any more about how you appraised the data and determined
your results?
ie once you had the recordings on the second host what did you do do
computationally score them? Do you look at the decoded (1khz?) waveform or
do you appraise in another way?
Pete
On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org
wrote:
Am 19.08.2015 um 19:07 schrieb Steve Edwards:
Please don't top post.
On Wed, 19 Aug 2015, James Cass wrote:
Steve, would you be willing to share that quick bash script?
There's no magic in the script, but here it is, embarrassing myself:
cp sample-call-file /tmp/
chmod +x /tmp/sample-call-file
for I in $(seq 1 $1)
do
sudo -u asterisk\
cp /tmp/sample-call-file\
/var/spool/asterisk/outgoing/${RANDOM}
done
sleep 10
Here's what's wrong with this snippet:
1) I don't know why I chmod the 'template.' No idea whatsoever. Alcohol
may have been involved.
2) I hate single character variable names. I love alcohol.
3) cp is ill advised. For a testing script, it was easy. For a production
application, use mv.
In use, I would execute it specifying how many call files to create, like
50. Then, take a look at top, iftop, and vmstat. Lather, rinse, repeat to
get to your goal.
We started the 500 calls and used milliwatt app on the first and record on
the second host to check the quality. Alternatively just start 500+ calls
and call yourself on top. So you can get a good idea how the quality is.
Call-Files are explained on
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Markus
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[asterisk-users] Connecting Samsung Galaxy to Asterisk for VoLTE
Hi, Has anybody tried to connect Samsung Galaxy to Asterisk PBX to be able to make calls over VoLTE? Thanks a lot in advance! Best regards, Sevana http://www.sevana.biz -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Samsung Galaxy to Asterisk for VoLTE
Thank you! Could you share links to information resources or manuals you used to connect these phones? Again, than you very much! On Fri, Apr 3, 2015 at 6:05 PM, Toufic Khreish (Gmail) toufic.khre...@gmail.com wrote: Hi, I have tried Groundwire on IOS , and Android Alcatel (voice and video calls with asterisk 13.3) Also tried Bria on both OS in video and voice. Regards Toufic *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sevana Oy *Sent:* Friday, April 03, 2015 12:28 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Connecting Samsung Galaxy to Asterisk for VoLTE Hi, Has anybody tried to connect Samsung Galaxy to Asterisk PBX to be able to make calls over VoLTE? Thanks a lot in advance! Best regards, Sevana http://www.sevana.biz -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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.beaum...@hatsoffsoftware.co.uk wrote: 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 oza.4...@gmail.com wrote: Some SIP hardphones (Polycom) or softphones (Counterpath) embed a module that metter MOS. Regards 2015-03-25 14:21 GMT+01:00 Patrick Beaumont p.beaum...@hatsoffsoftware.co.uk: 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 in order to say “ah ha! that’s the problem!”. I also don’t know what it’s limits are. Will the stats in “sip show channelstats” show a customer using a torrent client and saturating their own broadband connection? Regards, Patrick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What's the best average duration for a SIP test call?
Hi, What is your experience: if you plan to make a test SIP call to check voice quality overa connection what would be the best call duration? The point is that we should have a call long enough to be able to catch/hear impairments that the connection may have. This is partly a matter of curiosity, but I believe the roots of this question may be quite important. Thanks! Valeri on behalf of Sevana -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Figuring out gateway that degrades call quality
Hi, How do you figure out if one of gateways in your network leads to voice quality loss f.e. due to transcoding? The point is that all VoIP metrics in this case remain the same Thanks! Sevana http://www.sevana.fi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: AQuA Meter – waveform analysis to get continous MOS scores for your network
Hi, Although this is a repost from Asterisk biz, we would like to ask if somebody may help us to develop a native Asterisk module using AQuA technology for voice quality monitoring using the same web service AQuA Meter is using. Thanks, Sevana Finland/Estonia -- Forwarded message -- From: Sevana Oy sa...@sevana.fi Date: Mon, Jun 17, 2013 at 7:30 PM Subject: AQuA Meter – waveform analysis to get continous MOS scores for your network To: asterisk-...@lists.digium.com [image: AQuA Meter]http://blog.sevana.fi/wp-content/uploads/2013/03/screenshot.png Hi, We would like to offer you to learn about our new application that performs scheduled voice test calls to a predefined echo server and then uses our AQuA web service to evaluate the call quality. We developed it because several VoIP service providers have inquired us for a possibility to make test calls from local machines within their customers’ network. A typical example is when you provide VoIP communications to a company that rents its premises (including an Internet connection) in a business center. In this case it is quite important to monitor voice call quality from different computers in the office space to the service provider’s server. This is a cross platform (Windows, Linux, MAC) Java application and uses our latest developments in waveform analysis to evaluate voice call quality: http://www.sevana.fi/aquameter.zip The setup is simple: our application calls the echo server (apparently provided by the VoIP service provider), plays a reference audio and records the playback from the echo server and can thus provide overall (both ways) call quality analysis. We are very interested to receive your feedback and feature wishlist. The application is free. Best Regards, Sevana Oy/Oü Finland/Estonia http://blog.sevana.fi/aqua-meter-waveform-analysis-to-get-continous-mos-scores-for-your-network/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP call quality metrics: who cares?
Hi, How much do you care about call quality metrics to collect and analyze them? What metrics are of interest for you (of course packet loss, jitter, latency, but what else?). We have collected some for your review and would be happy to expand them with those you are using in your Asterisk systems. http://blog.sevana.fi/recommended-voip-call-quality-metrics/ Best Regards, Sevana http://www.sevana.fi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] server performance....
Hi, We have worked out another approach for load testing: - generate using sipp certain number of test calls and that go to PBX echo server playing and receiving back pre-defined audio - generate +1 test call, which also plays and receives back an audio file Then we test the audio we received from the +1 test call using AQuA (Audio Quality Analyzer) and obtain a MOS score (AQuA is doing perceptual audio quality assessment, it's not calculating MOS as in G.107, but more likely in P.862, although the algorithms are absolutely different). In this way we can always know how many calls can the PBX under test handle before actual call quality goes down. The whole test suit is put together with other testing (loop back call testing, conference bridge testing) capabilities into what we call Asterisk VQM. If my previous message goes through moderation you will be able to see screenshots as well :) Best Regards, Sevana Oy http://www.sevana.fi - Original Message - From: Andrew Latham lath...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2011 8:20 PM Subject: Re: [asterisk-users] server performance On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna viswavardhanre...@gmail.com wrote: Hi every one, I am doing some experiments on asterisk server performance.. How can we know server performance? can any one explain me plz I have 2 doubts regarding the asterisk server performance... 1. When can we know asterisk server performance? 1. when server is in idle state ? 2. when the server is in busy state? can any one please tell me when can the server performance is known i mean when server is busy or in idle state? Best Regards, viswavardhanreddy Many people test their servers with call-setups and call tear-downs. Using another tool like sipp you can send 100-1000s of call-setups and then do call tear-downs. You can also use transcoding loops to test the load. If you have 1 call that is sent to a context where it dials exten+1 and continues the loop until a target number, you can then set the codec for each dialed number. I know that there are many methods of testing and this is just a common one. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstriEurope coference
Are there other European Asterisk conferences? Thanks, Sevana Oy Vendor of Asterisk VQM - Original Message - From: randulo rand...@randulo.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 22, 2011 9:56 AM Subject: Re: [asterisk-users] AstriEurope coference On Mon, Feb 21, 2011 at 11:56 PM, Albert alber...@wp.pl wrote: does anyone know is AstriEurope coference is still on ? http://www.astrieurop.com/fr/cloture.php Cancelled. Hello, It is with regret that we announce you the cancellation of the AstriEurop exhibition on May, 3rd and 4th 2011 in Paris. We thank all the companies/partners having supported this project. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voice quality measurement using dahdi_monitor
Hi, The question is can you record the audio to evaluate its quality? There is intrusive approach when you have a reference file that you can test against the recorded audio, or non-intrusive approach, which allows you evaluate voice quality of any call recording (no reference needed). Both correspond to ITU-T standards: P.862 for intrusive and P.563 for non-intrusive, or to Sevana AQuA (for intrusive) and Sevana NIQA (for non-intrusive). The difference is that all ITU-T recommendations related to voice quality measurement are quite expensive and involve annual royalties, but they are recognized standards. Sevana products are not recognized standards, but are used by many happy customers doing call quality assessment in VoIP, PSTN and mobile networks. Welcome to our web site: http://www.sevana.fi for further information and customer references (many are Asterisk owners) 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 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 making .wav file and visulal mode of RX and TX of PRI line. what i want is measurement of voice quality so that i can talk with provider that i am getting % of voice quality.i am sure there is some better way to solve or debug .raw file and taking a decision. let me help please to solve and finding problem of voice quality. regards Dhaval -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sound quality issue
Hello, Can you record audio at different locations on its route? Our experience would suggest (of course) using intrusive or non-intrusive perceptual voice quality evaluation at different parts of the network to localize the one where it drops down. Best regards, Sevana Oy http://www.sevana.fi http://twitter.com/sevana - Original Message - From: Cédric Lemarchand cedric.lemarch...@ixcore.com To: asterisk-users@lists.digium.com Sent: Saturday, January 15, 2011 10:38 PM Subject: [asterisk-users] Sound quality issue Hello, Our Asterisk runs with multiple remote sites (12 over an MPLS network), everything works fine except for the last site we have juste installed. When VOIP flows comes/goes from/to this site, there are sound quality issues, persistent, 100% reproducible, on every call. This is not a bandwidth or latency or jitter problem, everything is fine on the network. Our MPLS provider does all check on his network equipments, everything is fine too, no packets loss recorded on routers's interfaces ect ... We have, on our side, check and replace all the VOIP equipments (spare rocks), an reduce the configuration to its simpliest (MPLS router = ethernet cable = VOIP equipment), quality problem still there. I am sure there are RTP packets losses somewhere, except RTP debug in the asterisk CLI, how can i determine where the problem come from ? Any help would be greatly appreciated, thx. Cédric -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Exchange - a waste of money?
Hi, This is it: http://www.asteriskexchange.com/ It's also good to know that people from such respectful community may not know it at all. Besides, the ones from Digium who read and moderate also don't reply to my post - good to know that too :) - Original Message - From: Goke M Aruna To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, December 09, 2010 6:39 PM Subject: Re: [asterisk-users] Asterisk Exchange - a waste of money? can someone give more eduaction to me about what the asterisk exchange is all about? thanks On Thu, Dec 9, 2010 at 5:43 AM, Sevana Oy sa...@sevana.fi wrote: Hi, A couple of months ago we registered our product AQuA at Asterisk Exchange. We were told that it collects like 14K visitors per month and knowing interest to our product from Asterisk community we have calculated a certain super-mini-minimal % of visitors coming from Asterisk Exchange to our web site... Here comes the funny thing - there was no traffic increase since then, there were no referral visist from Asterisk Exchange... We are getting something a bit less than 100 product inquiries a month and NONE has ever mentioned that learnt about us from Asterisk Exchange... My question is: have we just wasted $2500 for being listed there? Is Asterisk Exchange some kind of bubble? Unfortunately we never got response from the people who sold us this service :-) Thanks and cheers, Vallu Sevana Oy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [POTS/BRI] Neutral comparisons of PCI vs. box?
Hello, We would be happy to offer you Asterisk VQM for voice quality assessment, however, it's Asterisk based and works with every hardware that works with Asterisk: http://www.sevana.fi/aqua-powered-asterisk-voice-quality-monitoring-solution.php - Original Message - From: Gilles codecompl...@free.fr To: asterisk-users@lists.digium.com Sent: Wednesday, December 08, 2010 5:06 PM Subject: [asterisk-users] [POTS/BRI] Neutral comparisons of PCI vs. box? Hello I need to find a recent and neutral comparison of the major products available to connect an Asterisk server to the telephone network, whether ISDN (BRI) or PSTN, and through a PCI card or some external box. I'm told there are less issues (echo, stability) with external boxes compared to PCI cards. Apparently, the main brands are Digium, Sangoma, Rhino Equipment, Patton, and Audiocodes. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Exchange - a waste of money?
Hi, A couple of months ago we registered our product AQuA at Asterisk Exchange. We were told that it collects like 14K visitors per month and knowing interest to our product from Asterisk community we have calculated a certain super-mini-minimal % of visitors coming from Asterisk Exchange to our web site... Here comes the funny thing - there was no traffic increase since then, there were no referral visist from Asterisk Exchange... We are getting something a bit less than 100 product inquiries a month and NONE has ever mentioned that learnt about us from Asterisk Exchange... My question is: have we just wasted $2500 for being listed there? Is Asterisk Exchange some kind of bubble? Unfortunately we never got response from the people who sold us this service :-) Thanks and cheers, Vallu Sevana Oy-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Voice Quality Monitoring Framework
Hi, Please don't consider this as an advertizing, but since we received feedback from the community that the only voice quality assessment solutions available are worth 50K, we have started to develop an Asterisk based voice quality monitoring framework. The core technology for perceptual voice quality evaluation is based on our products AQuA and NIQA that are quite widely used already among some Asterisk owners and even some well-recognized brands. However, the point of this message is to ask the community to assist with the demanded features. So far our customers reported that they would like Asterisk VQM Framework having: - blast dialing to evaluate voice quality depending on the calls load sent to the server under test - reporting mechanism for obtained call statistics - implement alerts on test call results (quality going below 3 and below 2) - implement SaaS as demo and be ready to sell it as a service - sniffer + NIQA + G.107 with alerting for non-intrusive voice quality testing - Asterisk VQM to Asterisk VQM call simulation (to simulate how monitoring works when there are a couple of servers installed) - detection of reasons for failed test calls Our goal is to develop an affordable solution assisting VoIP community to monitor, evaluate, control and manage call quality in their networks; to have quantitative arguments towards PSTN gateway owners when voice quality drops down in PSTN network, to be able to route calls over the trunks with the best quality and in automated manner... And your support in presenting your wishes and ideas will help us a lot to achieve this. The first version of the framework is already available and even has customers already. Thanks a lot in advance! Best Regards, Vallu Sevana Oy Finland -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How many Asterisk PBX operating in the World?
Hi, Sorry for maybe not a very list related topic, but I have always been curious if there is information on how many Asterisk based PBXs are operating Worldwide? Thanks and hope the community will not reject my curiosity! :) Best Regards, Vallu Sevana Oy-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many Asterisk PBX operating in the World?
Thanks! What I managed to learn from one article is: Asterisk PBX has about 18% of the World market share of PBXs. Asterisk claims 75% of the World market share for Open Source PBXs. (source: http://www.asteriskexpert.co.uk/about-asterisk.php) This must be quite a lot I believe, and I do agree amount of downloads does not show the picture I tend to see. Thank you! - Original Message - From: Danny Nicholas To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, November 17, 2010 7:47 PM Subject: Re: [asterisk-users] How many Asterisk PBX operating in the World? -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sevana Oy Sent: Wednesday, November 17, 2010 10:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How many Asterisk PBX operating in the World? Hi, Sorry for maybe not a very list related topic, but I have always been curious if there is information on how many Asterisk based PBXs are operating Worldwide? Thanks and hope the community will not reject my curiosity! :) Best Regards, Vallu Sevana Oy There is probably not a reliable answer to this question since there are at least 4 major flavors of Asterisk out there (1.0/1.2, 1.4, 1.6, 1.8) and open and commercial source. It is reliably 10,000 and quite possibly over 100,000 or even over 1 million. The asterisk folks might be willing to tell you how many downloads have been done from www.asterisk.org, but that wouldn't tell you the real number. -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Test numbers Worldwide
Hi, We are searching for a pool of test numbers to call from Asterisk, record voice and test it with our non-intrusive voice quality testing software (NIQA). The problem is that we could find some test numbers, but our customer would like to have a global pool of test numbers, so that we can call them and test voice quality. Greatly appreciate any help! Thank you! Sevana Oy, Finland http://www.sevana.fi-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice quality assessment in Asterisk
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-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice quality assessment in Asterisk
One quick clarification please... With Fluke ACEs you measure MOS according G.107, E-model, right? Thanks a lot to all who replied and will reply! - Original Message - From: Daniel Tryba dan...@tryba.nl To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@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 side. You then use both files to calculate a MOS score. This method is used by telco's to do quality checks. Take a look at the website mentioned in GPs post. He/they already know this, I guess it is a fishing expedition for competitors :) We don't do the test calls method, but use inline probes (Fluke ACEs) that analyze all traffic and give a MOS score to SIP calls and save network statistics per call (can be retrieved from the RTCP reports in asterisk). These probes and the analyzer software aren't bug free and perfect but give a good indication of all historic calls. Once a problem is spotted we move to test calls to trace the problem. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
