similar to: How to quickly move on to Dahdi channels when SIP provider fails?

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, 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>

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,

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 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, 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

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

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

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

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 >

Hi there, I support a large number of enterprise users who contractually must connect to our support center via a 4G VOIP connection. I simply want to be able to auto detect all poor quality calls in realtme (as they are being made), play a message and drop the call - without user intervention. All decent call quality calls will be allowed through - to be handled by support staff. Its a

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 I am using a 4FXO , TDM400P card I am able to call outside , after modifiying extensions.conf with exten => _9X.,1,Dial(ZAP/1/${EXTEN:1}) using this , I can only dial through one of the port , Actually I want to dial outside using round - robin search After reading the manuals , I have plans to modified the above line as exten =>

Hi, I have got one X100P telephone card in my Asterisk server and it's working well. I have two phones lines and would like to install a second card so I can use both lines. I have installed the card and tried to set it up, but all to no avail ! Could someone outline the changes that I need to make (and in which .conf files) in order to get the second card going ? Thanks in advance,

Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with

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