search for: sevana

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20101117/a6ef7254/attachment.htm

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>

...actions 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, endre -- Endre Domiczi, CEO Sevana Oy, http://www.sevana.fi Email : ceo at sevana.fi GSM : +372 53485178 Skype : emddom

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 ? waveform analysis to get continous MOS scores for your network To: asterisk-biz at lists.digium.com [image: AQuA Meter]<http://blog.sevan...

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: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140527/025f1c2e/attachment.html>

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20101008/47aee455/attachment.htm

...llen) 4. Re: analog phone digit delay (jg) 5. Re: analog phone digit delay (Steve Edwards) 6. Re: PCI Passthrough of T1 cards (Mauricio Tavares) 7. Re: PCI Passthrough of T1 cards (Nick Khamis) 8. Fwd: AQuA Meter ? waveform analysis to get continous MOS scores for your network (Sevana Oy) ---------------------------------------------------------------------- Message: 1 Date: Mon, 8 Jul 2013 10:14:31 -0700 From: Justin Killen <jkillen at allamericanasphalt.com> Subject: [asterisk-users] analog phone digit delay To: Asterisk Users Mailing List - Non-Commercial Discussion...

...llen) 4. Re: analog phone digit delay (jg) 5. Re: analog phone digit delay (Steve Edwards) 6. Re: PCI Passthrough of T1 cards (Mauricio Tavares) 7. Re: PCI Passthrough of T1 cards (Nick Khamis) 8. Fwd: AQuA Meter ? waveform analysis to get continous MOS scores for your network (Sevana Oy) ---------------------------------------------------------------------- Message: 1 Date: Mon, 8 Jul 2013 10:14:31 -0700 From: Justin Killen <jkillen at allamericanasphalt.com> Subject: [asterisk-users] analog phone digit delay To: Asterisk Users Mailing List - Non-Commercial Discussion...

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, 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 -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis/attachments/20110827/375a5cdf/attachment.h...

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 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 a...

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

...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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130617/9bb34eb1/attachment.htm>

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

...re source, reference WAVs Initiate Dials via Web Service - All calls are initiated by HTTP POST (even internally) Upcoming Features Roll-up Reporting Dashboard Scheduled email reports Email notifications - Threshold definitions per schedule, email notifications Read more at: http://www.sevana.fi/aqua-powered-asterisk-voice-quality-monitoring-solution.php -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20100526/8ca7099c/attachment.html>

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> 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 discussi...

...onnection 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150327/d83318cc/attachment.html>

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

Hi All, I have configured WebRTC according to the install document. The clients register correctly. I'm use SIPjs. The clients are able to send messages to the server. The SIP debug shows the messages being received. However I'm stumped for directions on how to route the messages between the clients. Asterisk 11.11.0 Here is my client sip config: [1060] type=friend username=1060 ; The