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 <[email protected]> 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 política de envío y recepción de correo electrónico en el enlace > situado más 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
