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>
Jonas Christoffersen
2016-Aug-12 16:44 UTC
[asterisk-users] loosing audio from one end after 5 min.
Just tested the connection in the other direction and when calling out there is no problem. only when calling in.>>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 >> >> >> >>------ Original Message ------ From: "Carlos Rojas" <crt.rojas at gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>; "Jonas Christoffersen" <jonc at showitmedia.eu> Sent: 12-08-2016 04:16:24 Subject: Re: [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 >>>>>http://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/e3972400/attachment.html>
D'Arcy J.M. Cain
2016-Aug-12 16:58 UTC
[asterisk-users] losing audio from one end after 5 min.
On Fri, 12 Aug 2016 16:44:32 +0000 "Jonas Christoffersen" <jonc at showitmedia.eu> wrote:> Just tested the connection in the other direction and when calling > out there is no problem. > only when calling in.It sure sounds like a NAT problem. Missing audio has a 90% or more probability of being NAT related. Maybe it's a problem on the gateway device. What is the modem/router? P.S. The subject was driving me nuts. I had to correct it. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net VoIP: sip:darcy at Vex.Net