Hi, I am having trouble receiving calls from a sippy voip server. Freeswitch returns 400 for a normal INVITE I received from my vendor. Does anyone know what is wrong? Is there anyway to configure freeswitch to get this interoperability issue resolved? U x.x.x.x:5061 -> :5060 INVITE sip:13453334444@y.y.y.y:5060 SIP/2.0. Via: SIP/2.0/UDP x.x.x.x:5061;branch=z9hG4bK-cjuna7hd7oy4rvgl;rport. Max-Forwards: 70. From: ;tag=uyfnqeaupzqsi3az.o. To: . Call-ID: YTRhZTRjMzAyZTg0ZjgxNTg4MmI5ZjRjYjljNjY4YTY.~1o. CSeq: 621 INVITE. Contact: Anonymous . Expires: 300. User-Agent: Sippy. cisco-GUID: 1714481388-1998655968-2851059884-1870679645. h323-conf-id: 1714481388-1998655968-2851059884-1870679645. Content-disposition: session. Content-Length: 403. Content-Type: application/sdp. . v=0. o=Sippy 476627024 0 IN IP4 x.x.x.x. s=CounterPath X-Lite 4.0. t=0 0. a=ice-ufrag:e60b22. a=ice-pwd:3dc27abbae9db2484de85b0a2ac295fe. m=audio 34354 RTP/AVP 0 8 101. c=IN IP4 70.52.58.139. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.1.104 56716 typ host. a=candidate:1 2 UDP 659134 192.168.1.104 56717 typ host. a=oldmediaip:192.168.1.105. U y.y.y.y:5060 -> x.x.x.x:5061 SIP/2.0 400 Bad Session Description. Via: SIP/2.0/UDP x.x.x.x:5061;branch=z9hG4bK-cjuna7hd7oy4rvgl;rport=5061. From: ;tag=uyfnqeaupzqsi3az.o. To: ;tag=QggHUaHjv0SSr. Call-ID: YTRhZTRjMzAyZTg0ZjgxNTg4MmI5ZjRjYjljNjY4YTY.~1o. CSeq: 621 INVITE. User-Agent: freeswitch. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0. . Thanks, Mark