VoIP Mailing List Archives

[jonc at showitmedia.eu]

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 [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbNlRIaldrdHhPeXM&export=download[/img]Slotsbryggen 14 A-DDK-4800 Nykøbing F. Tel. +45 3841 0960jonc@showitmedia.eu (jonc@showitmedia.eu) [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbUjVpeWptN3p6T3c&export=download[/img] [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbMDJqUnIwVDVYZzQ&export=download[/img] [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbNkt2azlFY1poOGs&export=download[/img][img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbTF9YdmppMmVqdDA&export=download[/img] [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbUENqREhQTG5hcnM&export=download[/img] [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbSDloSF9TbnVJWUk&export=download[/img]

[dovid at telecurve.com]

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@showitmedia.eu (jonc@showitmedia.eu)> wrote:

[jonc at showitmedia.eu]

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 [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbNlRIaldrdHhPeXM&export=download[/img]Slotsbryggen 14 A-DDK-4800 Nykøbing F. Tel. +45 3841 0960jonc@showitmedia.eu (jonc@showitmedia.eu) [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbUjVpeWptN3p6T3c&export=download[/img] [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbMDJqUnIwVDVYZzQ&export=download[/img] [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbNkt2azlFY1poOGs&export=download[/img][img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbTF9YdmppMmVqdDA&export=download[/img] [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbUENqREhQTG5hcnM&export=download[/img] [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbSDloSF9TbnVJWUk&export=download[/img] ------ Original Message ------ From: "Carlos Rojas" <crt.rojas@gmail.com (crt.rojas@gmail.com)> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)>; "Jonas Christoffersen" <jonc@showitmedia.eu (jonc@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@telecurve.com (dovid@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@showitmedia.eu (jonc@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 [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbNlRIaldrdHhPeXM&export=download[/img]Slotsbryggen 14 A-DDK-4800 Nykøbing F. Tel. jonc@showitmedia.eu (jonc@showitmedia.eu) [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbUjVpeWptN3p6T3c&export=download[/img] [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbMDJqUnIwVDVYZzQ&export=download[/img] [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbNkt2azlFY1poOGs&export=download[/img][img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbTF9YdmppMmVqdDA&export=download[/img] [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbUENqREhQTG5hcnM&export=download[/img] [img]https://docs.google.com/a/showitmedia.eu/uc?id=0B-rtu7Yjm4jbSDloSF9TbnVJWUk&export=download[/img] --_____________________________________________________________________-- 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/helloasterisk-users mailing listTo 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/helloasterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users