Jonas Christoffersen
2016-Aug-11 21:33 UTC
[asterisk-users] loosing audio from one end after 5 min.
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 Nyk?bing F. >> >>Tel. +45 3841 0960 >>www.showitmedia.eu >>jonc at showitmedia.eu >> >> >> >>-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160811/c8f8f951/attachment.html>
Dovid Bender
2016-Aug-11 21:35 UTC
[asterisk-users] loosing audio from one end after 5 min.
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 <jonc at showitmedia.eu> 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 Nyk?bing F. > > Tel. +45 3841 0960 > www.showitmedia.eu > jonc at showitmedia.eu > > > > > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160811/131553b2/attachment.html>
Carlos Rojas
2016-Aug-12 02:16 UTC
[asterisk-users] loosing audio from one end after 5 min.
Hi Is the keep alive activated on the phone? On Thu, Aug 11, 2016, 5:36 PM Dovid Bender <dovid at telecurve.com> wrote:> 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 <jonc at showitmedia.eu > > 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 Nyk?bing F. >> >> Tel. +45 3841 0960 >> www.showitmedia.eu >> jonc at showitmedia.eu >> >> >> >> >> >> >> -- >> _____________________________________________________________________ >> -- 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160812/24a0c49b/attachment.html>