We have merged two small offices that are using different VoIP solutions. One office is using a 3825 running CME (8.1) and the other office is running FreePBX (Asterisk 2.9). I believe that I have correctly configured the SIP trunk/Dial Peers so the CME and FreePBX can talk to each other. I can initiate a call from the CME side to the FreePBX side with no issue. Calls from a Cisco Phone to a SIP Phone connects and sounds good (two way audio). However, if I initiate from the Asterisk side to the CME the call fails to connect. I debugged the CME and placed a packet sniffer and I am seeing a Q.850 Cause Code 65 disconnect error (SIP_UNACCEPTABLE_MEDIA_ERR) coming from the CME. I am matching an "incoming" dial-peer that I created and all dial-peers are built for g711ulaw, as are all the extensions/trunks on the Asterisk. Long term we will probably move all SIP phones onto the CME but for now I need to get these sites talking. Any tips/assistance would be appreciated. Here is a diagram and a debug capture for further clarification.