I am having a problem where I am not able to receive inbound calls. 3cx sees the call, you can see the trunk status showing IN and the caller ID but it will not ring an individual extension or a ring group. Here is my current config 3cx server is in the cloud - no nat, just windows firewall. Multiple ip addresses (static) for other websites and services 1 extension is 3cx softphone connected through tunnel 2 cisco spa504g remote extensions All phones register, I can make outbound calls an all 3 extensions (individually and simultaneously) no problem that I have noticed. 1) dial softphone extension from hardphone - Success 2) dial hardphone extension from softphone - fail 3) dial hardphone extension from hardphone - fail 4) dial in from external pots line - fail I tried to dial in from an outside pots line and checked the 3cx log. You'll notice my server has multiple static IP's but even if I remove all but 1 IP I still have this problem. Code: 10:04:01.380 Currently active calls [none] 10:03:53.944 [CM503003]: Call(2): Call to sip:111@server_primary_ip.126:5060 has failed; Cause: 408 Request Timeout; internal 10:03:46.691 [CM503008]: Call(2): Call is terminated 10:03:29.380 Currently active calls - 1:  10:03:21.896 [CM503025]: Call(2): Calling Ext:Ext.111@[Dev:sip:111@remote_extension_ip.241.22:12184;transport=UDP;rinstance=55252ef4a6b17ef0] 10:03:21.895 [MS210002] C:2.2:Offer provided. Connection(transcoding mode): server_primary_ip.126:9006(9007) 10:03:21.857 [MS210000] C:2.1:Offer received. RTP connection: 220.127.116.11:20220(20221) 10:03:21.856 [CM503004]: Call(2): Route 1: Ext:Ext.111@[Dev:sip:111@remote_extension_ip.241.22:12184;transport=UDP;rinstance=55252ef4a6b17ef0] 10:03:21.853 [CM503010]: Making route(s) to <sip:111@server_ALT_ip.84:5060> 10:03:21.852 Remote SDP is set for legC:2.1 10:03:21.852 [CM505003]: Provider:[NexVortex] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent:  PBX contact: [sip:dW8808jwzN@server_primary_ip.126:5060] 10:03:21.850 [CM503001]: Call(2): Incoming call from 1externalPOTSline_i_called_from@(Ln.10012@NexVortex) to <sip:111@server_ALT_ip.84:5060> 10:03:21.844 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10012 forwards to DN:111 10:03:21.842 Looking for inbound target: called=1DID5433; caller=1externalPOTSline_i_called_from 10:03:21.841 [CM500002]: Info on incoming INVITE: INVITE sip:biblesatcost@server_primary_ip.126;transport=udp SIP/2.0 Via: SIP/2.0/UDP 18.104.22.168:5060;branch=z9hG4bKfd23.d70e64f6.0 Via: SIP/2.0/UDP 22.214.171.124:5060;branch=z9hG4bK-847ea-4fa80057-bafcf200-527e5ce9 Max-Forwards: 16 Record-Route: <sip:1DID5433@126.96.36.199:5060;nat=yes;ftag=b9f5ed0-13c4-4fa80057-bafcf200-49fd3170;lr=on> Contact: <sip:1externalPOTSline_i_called_from@188.8.131.52:5060;maddr=184.108.40.206;transport=udp> To: <sip:1DID5433@220.127.116.11:5060> From: <sip:1externalPOTSline_i_called_from@18.104.22.168:5060>;tag=b9f5ed0-13c4-4fa80057-bafcf200-49fd3170 Call-ID: CXCfirstname.lastname@example.org CSeq: 1 INVITE P-Asserted-Identity: <sip:1externalPOTSline_i_called_from@cxc.dashcs.com:5060> Content-Length: 0 10:03:05.530 [CM504004]: Registration succeeded for: 10012@NexVortex 10:03:05.328 [CM504003]: Sent registration request for 10012@NexVortex 10:03:01.502 [CM504004]: Registration succeeded for: 10012@NexVortex 10:03:01.242 [CM504003]: Sent registration request for 10012@NexVortex Code: 10:55:59.483 IP(s) removed:[ALT_IP_51.84,anotherALT_IP_166.200] Now, if route my DID back to the server on my LAN, I can receive a call, but then I'm back to the problem with 1 way audio. So I thought by having a server in the cloud, static ip, no nat, I would be golden, but not so far Aside from shooting myself in the face, how can I solve my problems?