asterisk users - Oct 2011 - RTP ports used by Asterisk in dialplan

Dear all,

Do you know if there is a way to know the 2 RTP ports that Asterisk is using for
audio flow in a call in the dialplan?
I would like to launch a Linux shell command "tcpdump" to capture
audio flow in those 2 RTP ports before call starts and stop capturing at the end
of the call.

Regards,
Isabel

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You could temporarily change rtp.conf to use just 4 ports (say 10001-10004)
and monitor 10001 and 10002.  On a production system you would have to do
something with a tool like netstat to try and predict which ports in the
range would be used.

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of ISABEL ORDAS
ARNAL
Sent: Thursday, October 20, 2011 8:28 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] RTP ports used by Asterisk in dialplan

 

Dear all,  

 

Do you know if there is a way to know the 2 RTP ports that Asterisk is using
for audio flow in a call in the dialplan?

I would like to launch a Linux shell command ?tcpdump? to capture audio flow
in those 2 RTP ports before call starts and stop capturing at the end of the
call. 

 

Regards,

Isabel

 

  _____  

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receive email on the basis of the terms set out at.
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Hi Isabel,

Could you not just filter out after the fact using something like Wireshark?

Regards

On Thu, Oct 20, 2011 at 3:28 PM, ISABEL ORDAS ARNAL <ioa at tid.es> wrote:
>  Dear all,  ****
>
> ** **
>
> Do you know if there is a way to know the 2 RTP ports that Asterisk is
> using for audio flow in a call in the dialplan?****
>
> I would like to launch a Linux shell command ?tcpdump? to capture audio
> flow in those 2 RTP ports before call starts and stop capturing at the end
> of the call. ****
>
> ** **
>
> Regards,****
>
> Isabel****
>
> ------------------------------
> Este mensaje se dirige exclusivamente a su destinatario. Puede consultar
> nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace
> situado m?s abajo.
> This message is intended exclusively for its addressee. We only send and
> receive email on the basis of the terms set out at.
> http://www.tid.es/ES/PAGINAS/disclaimer.aspx
>
> --
> _____________________________________________________________________
> -- 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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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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You can use tcpdump portrange 10000-20000 udp

2011/10/20 Andrew Higgs <andrew.m.higgs at gmail.com>
> Hi Isabel,
>
> Could you not just filter out after the fact using something like
> Wireshark?
>
> Regards
>
> On Thu, Oct 20, 2011 at 3:28 PM, ISABEL ORDAS ARNAL <ioa at tid.es>
wrote:
>
>>  Dear all,  ****
>>
>> ** **
>>
>> Do you know if there is a way to know the 2 RTP ports that Asterisk is
>> using for audio flow in a call in the dialplan?****
>>
>> I would like to launch a Linux shell command ?tcpdump? to capture audio
>> flow in those 2 RTP ports before call starts and stop capturing at the
end
>> of the call. ****
>>
>> ** **
>>
>> Regards,****
>>
>> Isabel****
>>
>> ------------------------------
>> Este mensaje se dirige exclusivamente a su destinatario. Puede
consultar
>> nuestra pol?tica de env?o y recepci?n de correo electr?nico en el
enlace
>> situado m?s abajo.
>> This message is intended exclusively for its addressee. We only send
and
>> receive email on the basis of the terms set out at.
>> http://www.tid.es/ES/PAGINAS/disclaimer.aspx
>>
>> --
>> _____________________________________________________________________
>> -- 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
>
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