Ronald Hartmann
2004-Oct-21 08:37 UTC
[Asterisk-Users] Video Phone issues registering with asterisk
WookSung TelephoSee 2000 Help needed. Can not get the phone to register with asterisk. I am not sure what the problem is at this point. I have the setup of the phone as: Server1 192.168.3.1 Port1: 5060 Display: TelephoSee URI: <blank> Userid: 2205 Password: "password" Following is the debug. Any assistance would be helpful... pc-11*CLI> sip debug SIP Debugging Enabled pc-11*CLI> pc-11*CLI> pc-11*CLI> pc-11*CLI> pc-11*CLI> pc-11*CLI> pc-11*CLI> pc-11*CLI> Destroying call '043194642e7f2779c77bd47b885ca423@192.168.3.23' pc-11*CLI> Sip read: REGISTER sip:192.168.3.11:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.23:5060;branch=z9hG4bKaf0f84c42701849ef85c1812100b8137 To: <sip:@192.168.3.11:5060;user=phone> From: TelePhoSee <sip:@192.168.3.11:5060;user=phone>;tag=b7df9204 Call-ID: 579bb2735be96d498b77c70cf8c00706@192.168.3.23 CSeq: 1 REGISTER Max-Forwards: 70 Expires: 3600 Contact: <sip:@192.168.3.23:5060;user=phone> Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 192.168.3.23 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.3.23:5060;branch=z9hG4bKaf0f84c42701849ef85c1812100b8137 From: TelePhoSee <sip:@192.168.3.11:5060;user=phone>;tag=b7df9204 To: <sip:@192.168.3.11:5060;user=phone>;tag=as5975a76c Call-ID: 579bb2735be96d498b77c70cf8c00706@192.168.3.23 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:192.168.3.11> Content-Length: 0 to 192.168.3.23:5060 Scheduling destruction of call '579bb2735be96d498b77c70cf8c00706@192.168.3.23' in 15000 ms pc-11*CLI> Sip read: REGISTER sip:192.168.3.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.23:5060;branch=z9hG4bKaf0f84c42701849ef85c1812100b8137 To: <sip:@192.168.3.11:5060;user=phone> From: TelePhoSee <sip:@192.168.3.11:5060;user=phone>;tag=b7df9204 Call-ID: 579bb2735be96d498b77c70cf8c00706@192.168.3.23 CSeq: 2 REGISTER Max-Forwards: 70 Expires: 0 Contact: <sip:@192.168.3.23:5060;user=phone> Content-Length: 0 10 headers, 0 lines Using latest request as basis request Sending to 192.168.3.23 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.3.23:5060;branch=z9hG4bKaf0f84c42701849ef85c1812100b8137 From: TelePhoSee <sip:@192.168.3.11:5060;user=phone>;tag=b7df9204 To: <sip:@192.168.3.11:5060;user=phone>;tag=as5975a76c Call-ID: 579bb2735be96d498b77c70cf8c00706@192.168.3.23 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:192.168.3.11> Content-Length: 0 to 192.168.3.23:5060 Scheduling destruction of call '579bb2735be96d498b77c70cf8c00706@192.168.3.23' in 15000 ms Destroying call '579bb2735be96d498b77c70cf8c00706@192.168.3.23' -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041021/befc3b3e/attachment.htm