I am testing Asterisk Beta 2 in our lab and I have found a possible bug, the box is setup with a T410P. Call path looks like this: T1 PRI --> Asterisk Server(1.2.0beta2) --> SIP Interaction Proxy --> Asterisk Server (1.0.9) --> SIP Phone. This works perfectly. SIP Phone --> Asterisk Server (1.0.9) --> SIP Interaction Proxy --> Asterisk Server(1.2.0beta2) --> T1 PRI No Audio either direction, there are no firewalls or nat traversals between the any of the equipment. If I change the Asterisk(1.2.0beta2) server back to (1.0.9) everything works great. The only error I see is generated on my Interaction SIP Proxy aka "to retrieve next Via, don't know where to send responseSIP/2.0 200 OK" I have included the entire message bellow. Unfortunately I am not educated in SIP messaging to spot the problem right off. I would be willing to test with anybody that would like to tackle the problem. Chris SIPSession::proxyResponseImmediately(): Failed to retrieve next Via, don't know where to send responseSIP/2.0 200 OK From: "Veracity Communications" <sip:8017654321@192.168.201.18>;tag=as4177fb3e To: <sip:99918011234567@192.168.201.10>;tag=as4a7c573e Call-ID: 0ad778fe68521e5823395118731bb234@192.168.201.18 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: <sip:8011234567@192.168.201.14> Content-Type: application/sdp Content-Length: 214 v=0 o=root 1076 1077 IN IP4 192.168.201.14 s=session c=IN IP4 192.168.201.14 t=0 0 m=audio 17268 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051101/c3777617/attachment.htm