similar to: VoIP call quality metrics: who cares?

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 -------------- next part -------------- An HTML attachment was scrubbed... URL:

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.

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>

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 -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 <sales at sevana.fi> Date: Mon, Jun 17, 2013 at 7:30 PM Subject: AQuA Meter

Hello, As I mentioned in the previous message we are developing solution for wholesale companies to analyze their sales transactions by associative rules. I would very much appreciated if the community could give us some hint of what is a typical PC configuration of a professional statist (processor, RAM, HDD...)? Thanks a lot in advance and I highly appreciate your feedback! Kind regards,

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

Unsubscribe Elvin G. Nodalo -----Original Message----- From: asterisk-users-request at lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit

Unsubscribe Elvin G. Nodalo -----Original Message----- From: asterisk-users-request at lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit

Overview Asterisk-powered dialer software Web Interface UNIX/Linux Cron-based Schedule Logic Open-Source Code Graphing Monitoring Stats MySQL Database for Call Records Current Features Dial by SIP or PSTN - Asterisk base capable of dialing via any medium Blast-Dialing - send multiple calls to 1 trunk for specified duration - No QoS/MOS scoring performed, designed for load testing

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 -------------- next part -------------- An HTML attachment was scrubbed... URL:

> On May 26, 2016, at 7:08 AM, Elena Lepilkina via llvm-dev <llvm-dev at lists.llvm.org> wrote: > > I understood your modules and I see as them can be used in LNT. But there are some question about old features. > 1. With Makefiles we can write pass and collect some statistics. There was an example of branch pass. Metrics can be collected by > @-$(LOPT) -load

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150403/ac9b4a31/attachment.html>

I understood your modules and I see as them can be used in LNT. But there are some question about old features. 1. With Makefiles we can write pass and collect some statistics. There was an example of branch pass. Metrics can be collected by @-$(LOPT) -load dcc888$(SHLIBEXT) -branch-counter -stats \ -time-passes -disable-output $< 2>>$@ in makefile. In report file we write how

Hi! metrics.dovecot.org will be removed by end of April. It's not really used that much and it's not compatible with new style metrics. --- Aki Tuomi Open-Xchange oy

Hi, We have worked out an approach to optimize audio compression for OGG files achieving best or pre-defined quality and best compression ratio. If there is interest in this please consider reading this blog post: http://blog.sevana.fi/optimize-bitrate-and-size-preserving-high-audio-quality-in-tracks-podcasts-tunes-with-aqua-wideband/ Thanks! Best Regards, Sevana Oy -------------- next part

> On May 25, 2016, at 1:54 AM, Elena Lepilkina via llvm-dev <llvm-dev at lists.llvm.org> wrote: > > Hi Matthias, > > Thank you for your answer. > But can you answer for some more questions? > First of all, now LNT uses make-style of running tests and parse results from result csv file. Are there any plans to go to cmake? As James already said "lnt runtest

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-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sevana Oy Sent: Friday, April 03, 2015 12:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

Hello, I am going to be on vacation and would like to temporary unsubscribe from the list. Sending "unsubscribe" didn't help. Could you please guide me on the appropriate step? Thanks a lot in advance! Best regards, Endre -- Endre Domiczi, CEO Sevana Oy, http://www.sevana.fi Email : ceo at sevana.fi GSM : +372 53485178 Skype : emddom

I have voip quality issues I would like to minimize. I have a ~= 3M/384k (Comcast) cable modem and a CentOS based Linux router (SME 7, 2.6.9 kernel) with 5 NAT''d devices (3 PCs "DHCP", 2 Vonage adapters "static 10.10.2.10-11"). The quality problems are audio cutting out and popping. I tried the following (see below) based on a Cookbook example, but I still have