Hi to all 2 weeks ago I finisedh installed correcly the test pbx server, i'm using axtel provider for inbound and outbound, but some days ago i was not able to make calls from extensions, only i can recibe (inbound calls) when i call a number, the 3cx shows than the extension is trying to make call, (calling status) but in the trunk status it never shows calling or something, also in a polycom phone i recive the error 408 and in grand stream i recibe 480 in the lcd screen. so in the next log the info is: 89900000 = the number than I tried to call from the extension 192.168.0.100 = local ip of the server 10003815430ip = auth id of my voip provider mty13.axtel.net = host and out proxy of my voip provider NOTE: IN THE NEXT PARAGRAPH, AT THIRD LINE IT SAYS \\@mty1.axtel.// were it was exact copy/paste from log, but the server must be mty13.axtel.... as it shows in the other lines.... this is common? why ? or could be part of the problem? Code: 08:13:34.887 [CM503020]: Normal call termination. Reason: No answer 08:13:34.887 [CM503016]: Call(16): Attempt to reach <sip:firstname.lastname@example.org;user=phone> failed. Reason: No Answer 08:13:34.886 [CM503003]: Call(16): Call to sip:email@example.com:5060 has failed; Cause: 408 Request Timeout; from IP:207.***.***.154:5060 08:13:14.691 [CM503025]: Call(16): Calling VoIPline:89900000@(Ln.10005@AxtelHQ)@[Dev:sip:firstname.lastname@example.org:5060] 08:13:14.566 [CM503004]: Call(16): Route 1: VoIPline:89900000@(Ln.10008@AxtelHQ)@[Dev:sip:email@example.com:5060] 08:13:14.562 [CM503001]: Call(16): Incoming call from usrext1000 to <sip:firstname.lastname@example.org;user=phone> i believe than is a mismatch config of the outbound sip parameters... any suggestion ?!?