[asterisk-users] how to know RTP por of a SIP client in

[Bruce B]

Then you may use system() in dial-plan to run that shell command along with
what I suggested.

-Bruce

On Sun, Oct 23, 2011 at 5:22 AM, ISABEL ORDAS ARNAL <ioa at tid.es> wrote:

>
> Yes, I need to know to get in in dialplan because I want to capture traffic
> per call. I would like to launch $SHELL{tcpdump src port xxxx} in the
> dialplan or something like this. And I want RTP traffic only of a certain
> call.
> Thank you!
>
> =======================
> Date: Fri, 21 Oct 2011 09:41:39 -0400
> From: Bruce B <bruceb444 at gmail.com>
> Subject: Re: [asterisk-users] how to know RTP por of a SIP client in
>        the     dialplan
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>        <asterisk-users at lists.digium.com>
> Message-ID:
>        <CAJyE_uWLRXkhrWQ-6SvNW1ihN-nGA3HFwHt=PU-tfR6LYbi5mg at mail.gmail.com
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Do you need to know to get it in dialplan? If I not, from shell (not
> Asterisk CLI) I usually use:
>
> netstata -a | grep asterisk
>
> By default Asterisk settings it should be something between 10k-20k
>
> -Bruce
>
> On Fri, Oct 21, 2011 at 3:46 AM, ISABEL ORDAS ARNAL <ioa at tid.es> wrote:
>
> >  Hi all, ****
> >
> > How can I get the RTP port one SIP client is using for sending/receiving
> > RTP flow? Can I obtain it in from SIP_HEADER of something like that in
> the
> > dialplan?****
> >
> > Thank you!****
> >
> > ** **
> >
> > Isabel****
> >
>
>
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