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:[EMAIL PROTECTED]> > Expires: 900 > From: "sjphone2" <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > Content-Length: 0 > User-Agent: kphone/4.0 > Event: registration > Allow-Events: presence > Contact: "myusername" <sip:[EMAIL PROTECTED];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:[EMAIL PROTECTED]> > Expires: 900 > From: "sjphone2" <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > Content-Length: 0 > User-Agent: kphone/4.0 > Event: registration > Allow-Events: presence > Contact: "myusername" <sip:[EMAIL PROTECTED];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:[EMAIL PROTECTED]> > Expires: 900 > From: "sjphone2" <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > Content-Length: 0 > User-Agent: kphone/4.0 > Event: registration > Allow-Events: presence > Contact: "myusername" <sip:[EMAIL PROTECTED];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:[EMAIL PROTECTED]> > Expires: 900 > From: "sjphone2" <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > Content-Length: 0 > User-Agent: kphone/4.0 > Event: registration > Allow-Events: presence > Contact: "myusername" <sip:[EMAIL PROTECTED];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. > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ---------------End of Original Message----------------- _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
