1) Does it happen every time at the 5 minute work? 2) Have you done a dump on the client side to see if the NAT device is dropping the packets? 3) Is the phone behind a load balance internet connection and is the RTP port changing?
On Thu, Aug 11, 2016 at 5:33 PM, Jonas Christoffersen <[email protected]> wrote: > Hi all, > > Just installed Asterisk 13 on CentOS 7 and have run into a problem. > > The Scenario is this: > > Asterisk is on the internet > the Phone, a D40, is behind NAT > > So someone calls the number and Asterisk routes the call to the D40 > Everything works fine and the call is established, but then after 5 min. > the caller stops getting audio from the D40 but there is still audio to the > D40. > > using both RTP and SIP debug on the Asterisk console does not reveal > anything. > Actually I can see from the RTP debug that RTP packages are send and > received even after lose of the audio. > > So does anyone have any ideas where to look for the problem or perhaps a > solution? > > > > Med venlig hilsen / Kind Regards, > > Jonas Christoffersen > > > Slotsbryggen 14 A-D > DK-4800 Nykøbing F. > > Tel. +45 3841 0960 > www.showitmedia.eu > [email protected] > > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
