similar to: What's the best average duration for a SIP test call?

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,

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

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, 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>

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:

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

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:

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

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, 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

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 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

Hi Everyone, There are situations when internet connection is lost, SIP provider fails, or even authentication to SIP provider fails, and we want to use the backup Dahdi channels (PSTN). As simple as it may sound but with the many different situations and error messages it seems like it's not so easy to predict all the errors. Is there any single parameter value that can be changed to send

Version 1.4.2 but to be honest I have no reason at all to suspect that this is a problem with the asterisk software. I've able to replicate this from a few different "client" net connections and a across a few different linksys ata's. Where when you call into the host and enter the extension to connect to you miss the last digit of the extension. Almost every time you

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

Hi. I'm Bill Cox, and I volunteer a bit for the Vinux project, which is Linux for people with vision impairments. Most blind users use a closed-source speech synthesis tool called voxin, as it's very easy to understand at high speed. I would like to make TTS synthesizers based on large recorded vocabularies of actual speech, but to make it useful for the blind, I need to be able to

================================================================= CALL FOR PAPERS 10th Workshop on Virtualization in High-Performance Cloud Computing (VHPC '15) held in conjunction with Euro-Par 2015, August 24-28, Vienna, Austria (Springer LNCS) ================================================================= Date: August 25, 2015 Workshop URL: http://vhpc.org Paper Submission