[cisco-voip] Debugging RTP stream creation

[Joseph Mays]

I have an Allworx server that is using a Cisco AS5400 as a media gateway. 
There is an Allworx phone plugged into the LAN port of the allworx server. 
Calls out from the phone to the world work fine. Calls in work fine as long 
as the audio is going to the Allworx server. So there is two way audio when 
working with the audio attendant, two way audio while ringing the desk phone 
if you select the phones extension, two way audio for leaving messages if 
you don't answer the desk phone, but if you enter the phones extension and 
answer it when it rings, there is only one-way outgoing audio from the 
phone. A wan packet capture from the Allworx server shows that both incoming 
and outgoing RTP streams are created for everything else, but no incoming 
RTP stream appears after the INVITE from answering the phone, and ccsip 
debugging output shows that the Cisco AS5400 is receiving and OK'ing the 
INVITE, but is not transmitting outgoing packets.

Is there some debug option for the cisco that will show me if an error is 
occurring on the cisco side when it tries to create the stream, or something 
that will show why it is not creating the stream?