[Asterisk-Users] Video Phone issues registering with asterisk

Ronald Hartmann RHartmann at nnamtraH.com
Thu Oct 21 08:37:26 MST 2004


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 at 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 at 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 at 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 at 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 at 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 at 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 at 192.168.3.23' in 15000 ms
Destroying call '579bb2735be96d498b77c70cf8c00706 at 192.168.3.23'
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