ISABEL ORDAS ARNAL
2011-Oct-23 09:22 UTC
[asterisk-users] how to know RTP por of a SIP client in
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
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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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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**** > > > > > 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111023/969165c9/attachment.htm>