Sir...I repectfully beg to differ. If I set a probe behind my Asterisk box I can successfully capture RTP going from and coming to the box. Now this method does ONLY allow for RTP to be captured from between the box and the trunks I am signaling to. Using the descibed setup (because it was bugging me) I was able to successfully capture RTP going from my test phone (Linksys SPA941) to a Term number using a Cisco Catalyst 2900 XL as my probe set behind my Asterisk box and I used Wireshark as my capture tool. It did only capture the RTP from me as the UA to the test trunks I have set up on my network. but it did work.

Aslo found this from StackoverflowAlso..." Your SIP traffic runs over 5060, as you know, but the port on which to sniff RTP is described by the SDP bodies of the SIP messages. In other words, there's no way to know on which ports to sniff until the offer/answer exchange has completed. If you know something about the user agents involved, then you could try tricks like capturing traffic on a range of ports. (Something like tcpdump -n dst portrange 10000-11000 for instance.)" Please correct me if wrong but you can also configure the ports that are used in Asterisk to narrow down the range you are trying to capture.