[asterisk-users] loosing audio from one end after 5 min.

[Jonas Christoffersen]

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 Nykøbing 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 Nykøbing F.
>>>>>
>>>>>Tel. +45 3841 0960
>>>>>http://www.showitmedia.eu/
>>>>>jonc at showitmedia.eu
>>>>>
>>>>>
>>>>>
>>>>>
>>>
>>>
>>>--
>>>_____________________________________________________________________
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>>
>>--
>>_____________________________________________________________________
>>-- 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
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