Busy VOIP Status 403

[jasonhayes]

Please can anyone help. I'm trying to setup 3CX with Siemens Gigaset A580 IP Phones. I can ring via 3CX My phone app on my PC the handsets, incoming calls via my GrandStream GX4104 ring the handsets, but if I try and ring another ext from the Handsets I get "Busy". The same happens if from the handsets I try ringing an external number via the GX4104. I have turned on VOIP Status on the handsets and I get "Status 403". My 3CX is installed on a Windows 2008 Server, can anyone help?

 

[mixig]

Check 3cx server activity log, there must be some clue for message 403 "Forbidden", and Busy

 

[jasonhayes]

What does this mean?
01-Jan-2013 13:15:16.010 [CM502001]: Source info: From: "10"<sip:[email protected]>;tag=946704651; To: <sip:[email protected];user=phone>
01-Jan-2013 13:15:16.010 [CM503013]: Call(C:21): Incoming call rejected, caller is unknown; msg=Ivite-IN Recv Req INVITE from 192.168.75.201:5060 tid=fcf356445aedb019ed92e673beca78ca Call-ID=363579809@192_168_75_201:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.75.201:5060;branch=z9hG4bKfcf356445aedb019ed92e673beca78ca;rport=5060
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>
From: "10"<sip:[email protected]>;tag=946704651
Call-ID: 363579809@192_168_75_201
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Proxy-Authorization: Digest username="10",realm="3CXPhoneSystem",algorithm=MD5,uri="sip:[email protected];user=phone",nonce="414d535c06f3726323:916529b81b0346e4645136f6fcfd4358",response="d788b4f58f950421fc13c2ad4761fb0f"
Supported: replaces
User-Agent: A580 IP/022270000000
Allow-Events: message-summary, refer
Content-Length: 370

v=0
o=20 5020 8 IN IP4 192.168.75.201
s=Mapping
c=IN IP4 192.168.75.201
t=0 0
m=audio 5020 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
01-Jan-2013 13:15:15.830 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
Invite-UNK Recv Req INVITE from 192.168.75.201:5060 tid=d8a951bf7a0e1504eb6c8b2655e95965 Call-ID=363579809@192_168_75_201:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.75.201:5060;branch=z9hG4bKd8a951bf7a0e1504eb6c8b2655e95965;rport=5060
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected];user=phone>
From: "10"<sip:[email protected]>;tag=946704651
Call-ID: 363579809@192_168_75_201
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Supported: replaces
User-Agent: A580 IP/022270000000
Allow-Events: message-summary, refer
Content-Length: 370

 

[sigma1]

First of all, please use an IP address an not a FQDN

 

[jasonhayes]

Fixed. The Siemens software had a field called domain in the handset web interface. My old Asterisk phone system it appears didn't matter that the field was completed, it didn't use it, with 3CX it was trying to use that to contact the 3CX software.

Thanks for the clue.

 

[leejor]

You can use FQDN in the remote extension settings, as a server contact info, if you put JASONHAYES.ME.UK in 3CX as your Domain Name (advanced settings). Otherwise the calls will be rejected, as you've experienced.