[Asterisk-Users] SIP phone registering problem

[Rich Adamson]

download ethereal and take a peek at the packets on the wire. Without
something like that, no one is really going to be able to help you.

------------------------
> I am clearly doing something ridiculously wrong.
> 
> Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are
> unable to register. They keep trying and then time out.
> 
> With the sip debug on in Asterisk nothing is logged.
> Here is the trace from one of the phones (kphone):
> 
> (192.168.100.13 is kphone, 192.168.100.3 is Asterisk)
> 
> sipclient: sending: 21:47:45.454
> --------------------------------
> REGISTER sip:192.168.100.3 SIP/2.0
> Via: SIP/2.0/UDP 192.168.100.13;rport
> CSeq: 4399 REGISTER
> To: "sjphone2" <sip:sjphone2 at 192.168.100.3>
> Expires: 900
> From: "sjphone2" <sip:sjphone2 at 192.168.100.3>
> Call-ID: 797316263 at 192.168.100.13
> Content-Length: 0
> User-Agent: kphone/4.0
> Event: registration
> Allow-Events: presence
> Contact: "myusername" <sip:myusername at 192.168.100.13;transport=udp>;methods="INVITE, MESSAGE, 
INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
> 
> SipClient: Sending to '192.168.100.3:5060'
> SipClient: Receiving message...
> 
> SipClient: Received: 21:47:45.471
> ---------------------------------
> REGISTER sip:192.168.100.3 SIP/2.0
> Via: SIP/2.0/UDP 192.168.100.13;rport
> CSeq: 4399 REGISTER
> To: "sjphone2" <sip:sjphone2 at 192.168.100.3>
> Expires: 900
> From: "sjphone2" <sip:sjphone2 at 192.168.100.3>
> Call-ID: 797316263 at 192.168.100.13
> Content-Length: 0
> User-Agent: kphone/4.0
> Event: registration
> Allow-Events: presence
> Contact: "myusername" <sip:myusername at 192.168.100.13;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
> 
> SipCall: Incoming request
> SipCall: New transaction created
> SipTransaction: Incoming Request
> SipTransaction: Retransmit 1 (4000)
> 
> SipClient: Sending: 21:47:49.456
> --------------------------------
> REGISTER sip:192.168.100.3 SIP/2.0
> Via: SIP/2.0/UDP 192.168.100.13;rport
> CSeq: 4399 REGISTER
> To: "sjphone2" <sip:sjphone2 at 192.168.100.3>
> Expires: 900
> From: "sjphone2" <sip:sjphone2 at 192.168.100.3>
> Call-ID: 797316263 at 192.168.100.13
> Content-Length: 0
> User-Agent: kphone/4.0
> Event: registration
> Allow-Events: presence
> Contact: "myusername" <sip:myusername at 192.168.100.13;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
> 
> SipClient: Receiving message...
> 
> SipClient: Received: 21:47:49.466
> ---------------------------------
> REGISTER sip:192.168.100.3 SIP/2.0
> Via: SIP/2.0/UDP 192.168.100.13;rport
> CSeq: 4399 REGISTER
> To: "sjphone2" <sip:sjphone2 at 192.168.100.3>
> Expires: 900
> From: "sjphone2" <sip:sjphone2 at 192.168.100.3>
> Call-ID: 797316263 at 192.168.100.13
> Content-Length: 0
> User-Agent: kphone/4.0
> Event: registration
> Allow-Events: presence
> Contact: "myusername" <sip:myusername at 192.168.100.13;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
> 
> 
> SipCall: Incoming request
> SipTransaction: Incoming Request Retransmission
> SipTransaction: Response Retransmission
> SipTransaction: Retransmit 2 (4000)
> 
> (and so it continues)
> 
> Seems like the REGISTER messages are being recieved at Asterisk but
> then just echoed back to the SIP phone? What am I doing wrong?
> 
> thanks!
> 
> Richard.
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