VOIP over VPN dropp RTP protocol

Question

We are installing a new 2911 ISR in our office and connecting with a Linksys (CISCO) RV016 VPN router.  These are two small doctors offices that need to have computer, and Voip traffic over a VPN.
Currently we connect an older RV082 and the RV016 together and have NO issues with VOIP traffic.  If we establish a connection with the 2911 router then we are having an issue with no voice or RTP traffic coming through.  Phones will connect, and dial out, but no voice can be heard.
The First office is on a Verizon Fios Network with a MTU of 1492. The Network and servers are as follows:
Remote Office                                                                            Main Office
Linksys Spa 942 phones
               |                                                                        
Netgear 10/100 POE Switch
               |                                                                        =================
Linksys (cisco) RV016 VPN                                            |          2911                 |
               |                                                                       |    POE Module Sw     |
Comcast Cable Modem                                                  -------------------------------       
               |                                                                                       |          |
             VPN                                                                                VPN       |
               +=======================================+     Asterisk
                                                                                                              (Call Man)
Basically we have the Internet coming in from Gig0/0 and routing traffic to multiple outside IP addresses so we are using 3 subs in our configuration.
192.168.1.X          192.168.2.X          192.168.3.X               192.168.0.X (Remote Group)
When we connect the old routers (RV016 and RV082) VPN VOIP and Data traffic go fine.   We are using a Term Server on one end, Web Server, and the Asterisk PBX for our VOIP Call Manager.
So far we connect up the 2911 and the RV016 and have no issues with data traffic.  But the VOIP is dead on the remote end.  No sound.  We did a Wireshark on traffic, and we are getting some 407 errors from the Astersick Host, and a unknown RTP version 1 error message.  THe only thing that we had to do on the RV082 router was port forward UDP 506 and 10001 - 20000 for the traffic, and setup a access rule, but nothing else.
We are getting traffic on the 2911, but nothing else.  We have excluded the 192.158.0.X traffic from the NAT so not to get into that issue, and have even tried forwarding ports but nothing seems to help.  Is there a good way to route this traffic?  Our bandwidth is pretty fast so I am not sure if QoS is needed, but if so it is not one of my strong areas.  What is the best way to route this traffic through the VPN without loosing the RTP part of the call.

Answer 1

I put this line in and still not getting audio on the other end.  I will be doing captures tonight from working and non working phones.  I need to get this resolved.  I have spent 3 weeks on this issue and I have run out of time.  Should I use the DEBUG VOIP SIP command for the capture on the router?  I believe this would be the best resolution to the service to see what is going on.  The phones work with a RV016 and RV082 router in place.  All data traffic works fine in sending and recieving calls.
I have read about all of the articles on Cisco and voip traffic.  We are going to be shutting off the natting on the router to see if I can just get the voip traffic to flow.  Once we get it flowing then I can work on building up the house on a stable foundation.
At this time, we are routing multiple IP addresses throught the 2911 and have IP NAT OUTSIDE on the G0/0 port and IP NAT INSIDE on the G1/0 Interface, which is a POE Switch Module in the 2911.
I know that the cisco router wants to act as a call manager, or terminate the SIP traffic on the 2911, but we have a working Asterisk box that handles all SIP traffic.  If there is a way to just forward the traffic there properly, without the 2911 trying to intercept the traffic, that would be wonderful.  I am looking at the possibility of creating dial-peer groups for all of the phones, but really is this needed?  What is so frustrating about the whole situation is that I put in a 5 year old sub $200 router and everything works.
Dale