Well, I enabled RTP debugging on siproxd and telnet'ed to the debug port. I tried an extension routed through siproxd. it was not routed to the qVOIP. But on the RTP debug, I noticed that the UDP port my Polycom 650 was originating with 2224. It hit me that the Polycom default UDP port for RDP was 2222 (from past experience). So, for grins, I rolled it up to 7070 (the starting port for siproxd on my end) and tried the call again. still nothing. But in further examination, I noticed that the destination UDP port that siproxd was using for my remote Asterisk server was 12478 (outside the siproxd specified range of 7070-7099). The originating port siproxd used was 7076 (within the range). Now, i have static ports set for outbound NAT. But, siproxd is side-stepping NAT, so I guess it negotiates with the remote, and the Asterisk server's range is 10000-20000. So, on a hunch, I expanded the floating qVOIP outbound rule to cover UDP 7070-20000. Damned if that didn't do the trick! Now, my SIP and RTP routed through through siproxd is being routed into qVOIP. I am going to keeping investigating it further, but this must be why it was not matching the qVOIP rule. FYI!