Just installed v9. And I'm trying to to setup a pots line and 1 softphone. Problem: When I call in (from my cell etc) it rings once or twice and then all I hear is a dial tone sound. I can't dial anything without a forbidden message and a couple of beeps. Doesn't matter if it is an extension or an outside number. I'm using a Grandstream GXW-4104 gateway. I have a single line connected into channel 1 on that device. I was trying to follow this tutorial http:/ /www.3cx.com/voip-gateways/Grandstream-GXW-41044108/ NO LONGER AVAILABLE Gateway: http://buggyonpurpose.com/random/SS-201 ... .31.20.png Status Tab http://buggyonpurpose.com/random/SS-201 ... .35.31.png Channels Tab http://buggyonpurpose.com/random/SS-201 ... .36.42.png Dial Plan http://buggyonpurpose.com/random/SS-201 ... .37.11.png Profile 1 3CX: http://buggyonpurpose.com/random/SS-201 ... .39.18.png Trunk Status http://buggyonpurpose.com/random/SS-201 ... .41.02.png Extension Status Softphone: http://buggyonpurpose.com/random/3CXPho ... .54.07.png Forbidden Server Activity Log: Code: 12:00:17.375 Requests rate from IP 127.0.0.1:1149 is too high! Blacklisted for 334 seconds 12:00:17.375 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification: INVITE sip:*999@localhost:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:1149;branch=z9hG4bK-d8754z-5a61e377cc480454-1---d8754z-;rport=1149 Max-Forwards: 70 Contact: <sip:email@example.com:1149;rinstance=157924ea8ffde435> To: <sip:*999@localhost:5060> From: <sip:100@localhost:5060>;tag=e71e1347 Call-ID: ODY3Y2Y0NGM4YzQ4NmI2M2Y2OGQ0MTg3ZGU5NzhlMmE. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Supported: replaces User-Agent: 3CXPhone 4.0.13679.0 Content-Length: 0 12:00:17.375 [CM302001]: Authorization system can not identify source of: SipReq: INVITE *999@localhost:5060 tid=5a61e377cc480454 cseq=INVITE firstname.lastname@example.org:1149 / 1 from(wire) 11:59:07.265 [CM306003]: SIP IP:port mapping (220.127.116.11:57100) resolved by STUN server 18.104.22.168:3478 differs from the one (22.214.171.124:64404 resolved by STUN server 126.96.36.199 11:59:07.015 [CM506003]: Resolved SIP external IP:port has changed to (188.8.131.52:57100) on Transport 192.168.13.163:5060 11:58:11.343 [CM302001]: Authorization system can not identify source of: SipReq: SUBSCRIBE 100@localhost:5060 tid=845f13688104353b cseq=SUBSCRIBE email@example.com:1149 / 1 from(wire) 11:54:06.234 Requests rate from IP 127.0.0.1:4972 is too high! Blacklisted for 334 seconds INVITE sip:*999@localhost:5060 SIP/2.0 Max-Forwards: 70 To: <sip:*999@localhost:5060> Call-ID: MjU2OGRhYmUwYjVjMGY0NTZiMDA4OTJkOTIxMmE5NDY. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE User-Agent: 3CXPhone 4.0.13679.0 11:54:03.125 [CM302001]: Authorization system can not identify source of: SipReq: SUBSCRIBE 100@localhost:5060 tid=6f780c45a6171502 cseq=SUBSCRIBE firstname.lastname@example.org:4972 / 1 from(wire) 11:52:07.218 Requests rate from IP 127.0.0.1:4045 is too high! Blacklisted for 334 seconds INVITE sip:2@localhost:5060 SIP/2.0 Max-Forwards: 70 To: <sip:2@localhost:5060> I only see server activity when trying to dial from the soft phone. If I try to call the line I don't see any activity. Thanks.