Hi All, just setup skype gateway and cannot get my newest skype account to work. Config: Avaya pbx analog trunks connected to grandstream fxs with ports registered as extensions in 3cx. Skype gateway is a pstn device using template with one port. The grandstream fxs (GXW4004) successfully register to 3cx, my skype gateway trunks successfully register, and I can use the skype account to call landlines directly from skype app. Out bound rule for skype says any extension that is registered on the grandstream fxs goes out skype pstn device, no digit strip/prepend. When the call fails I get a busy signal. Any thoughts? Skype accounts that I previously created (couple of weeks ago) work fine. following is a trace of a failed call: Code: 15:27:27.337 Currently active calls [none] 15:27:17.541 [CM503020]: Normal call termination. Reason: Not available 15:27:17.541 [CM503016]: Call(7): Attempt to reach <sip:email@example.com> failed. Reason: Temporarily Unavailable 15:27:17.539 [CM503003]: Call(7): Call to sip:firstname.lastname@example.org:6060 has failed; Cause: 480 Temporarily Unavailable; from IP:xx.x.x.153:6060 15:27:17.529 [CM505002]: Gateway:[SkypeOut] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXSkypeGateway 2.0.9919.0] PBX contact: [sip:email@example.com:5060] 15:27:17.528 [CM503002]: Call(7): Alerting sip:firstname.lastname@example.org:6060;rinstance=f14b0d15f3f39a3d 15:27:17.372 [CM503025]: Call(7): Calling PSTNline:xxxxxxxxxx@(Ln.10000@SkypeOut)@[Dev:sip:email@example.com:6060;rinstance=f14b0d15f3f39a3d] 15:27:17.368 [MS210002] C:7.2:Offer provided. Connection(transcoding mode): xx.x.x.153:7028(7029) 15:27:17.336 [CM503004]: Call(7): Route 1: PSTNline:xxxxxxxxxx@(Ln.10000@SkypeOut)@[Dev:sip:firstname.lastname@example.org:6060;rinstance=f14b0d15f3f39a3d] 15:27:17.328 [CM503010]: Making route(s) to <sip:email@example.com> 15:27:17.326 [MS210000] C:7.1:Offer received. RTP connection: xx.x.x.154:5004(5005) 15:27:17.323 Remote SDP is set for legC:7.1 15:27:17.322 [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW-4004 V1.5A 184.108.40.206] PBX contact: [sip:firstname.lastname@example.org:5060] 15:27:17.316 [CM503001]: Call(7): Incoming call from Ext.101 to <sip:email@example.com> 15:27:17.307 [CM500002]: Info on incoming INVITE: INVITE sip:firstname.lastname@example.org SIP/2.0 Via: SIP/2.0/UDP xx.x.x.154:5060;branch=z9hG4bK1663837088;rport=5060 Max-Forwards: 70 Contact: "101"<sip:email@example.com:5060> To: <sip:firstname.lastname@example.org> From: "101"<sip:email@example.com>;tag=870922287 Call-ID: firstname.lastname@example.org CSeq: 71 INVITE Accept: application/sdp, application/dtmf-relay Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Proxy-Authorization: Digest username="101",realm="3CXPhoneSystem",nonce="414d535c03d37d0553:ad6ec0abf4cfad92cd125f72351873a3",uri="sip:email@example.com",response="67ce122ab047d45e64b47c1bfbd194a4",algorithm=MD5 Supported: replaces, path, timer User-Agent: Grandstream GXW-4004 V1.5A 220.127.116.11 Privacy: none P-Asserted-Identity: "101" <sip:firstname.lastname@example.org> Content-Length: 0 Thanks, Cheers!