similar to: PC configuration you are using

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

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, 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, 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, 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, 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, I'm using mdbox now (Dovecot 2.2.4), with indexes on a separate disk. Works fine, but the directories for indexes are also present in the storage folder, and are empty. Userdb (sql) returns this: home /var/vmail/domains/somedomain/someuser mail mdbox:~/mdbox:INDEX=/mnt/spool/dovecot/index/somedomain/s/someuser And the created directory structure is:

Hi, I'm new in this list. As the title explains, I would like know which game that work in wine, with intel graphics. I've experinced that games like Heroes of Might and Magic III won't work, but a game like Soldier of Fortune, which is a newer game, works. If you know of a game that works in wine, please try to list them here, with additional tweaks as to how you got it to work,

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, 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

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 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

Hello, finale notepad 2008 is a free (as in beer) score writer program which is fairly user-friendly. It can save scores in its own format and as a midi file. The problem is that the file dialog doesn't show any of the file types, it's all blank. It automatically saves in its own format, so I'm not lost, but it's just that saving to midi makes sharing my music easier. Wine

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, 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, 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 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

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, I am getting the errors given below, when I am trying to add the prio to any table. What can be the problem and how can i resolve it.... Kindly help [root /root]# ip rule RTNETLINK answers: Invalid argument Dump terminated [root /root]# ip rule list RTNETLINK answers: Invalid argument Dump terminated [root /root]# ip rule list table main "ip rule show" need not eny arguments.