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

[Carlos Rojas]

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
>> 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
>
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