Re: [Serusers] kphone registration problem

Nils: I'm afraid there is no more interesting dump to send you, kphone keeps sending the same register message and SER keeps answering the same. By the way yesterday I tried to register kphone without authentication configured in SER and it didn't work either. Kphone keeps trying again and again although it reiceives an OK answer from SER every time. Oddly I've tested kphone with other proxies and always worked fine. Here is a dump of this other situation (by the way I'm using kphone v. 3.12): SipClient: Listening UDP on port: 5060 SipClient: Our address: 192.168.2.175 SipRegister: Auth is '(null)' SipRegister: Proxy Auth is '(null)' SipClient: Sending: 09:00:38.025 -------------------------------- REGISTER sip:asterix SIP/2.0 Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" <sip:jorge@asterix> Expires: 900 From: "jorge" <sip:jorge@asterix> Call-ID: 123630670(a)192.168.2.175 Content-Length: 0 User-Agent: KPhone/3.12 Event: registration Allow-Events: presence Contact: "jorge" <sip:jorge@192.168.2.175;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK" SipClient: Sending to '192.168.2.176:5060' SipClient: Receiving message... SipClient: Received: 09:00:38.041 --------------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" <sip:jorge@asterix>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22 From: "jorge" <sip:jorge@asterix> Call-ID: 123630670(a)192.168.2.175 Contact: <sip:jorge@192.168.2.175:5062;transport=udp>;q=0.00;expires=337 Contact: <sip:jorge@192.168.2.175;transport=udp>;q=0.00;expires=900 Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0 SipTransaction: Retransmit 1 (4000) SipClient: Sending: 09:00:42.026 -------------------------------- REGISTER sip:asterix SIP/2.0 Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" <sip:jorge@asterix> Expires: 900 From: "jorge" <sip:jorge@asterix> Call-ID: 123630670(a)192.168.2.175 Content-Length: 0 User-Agent: KPhone/3.12 Event: registration Allow-Events: presence Contact: "jorge" <sip:jorge@192.168.2.175;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK" SipClient: Receiving message... SipClient: Received: 09:00:42.038 --------------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.175 CSeq: 4764 REGISTER To: "jorge" <sip:jorge@asterix>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7f22 From: "jorge" <sip:jorge@asterix> Call-ID: 123630670(a)192.168.2.175 Contact: <sip:jorge@192.168.2.175:5062;transport=udp>;q=0.00;expires=333 Contact: <sip:jorge@192.168.2.175;transport=udp>;q=0.00;expires=900 Server: Sip EXpress router (0.8.11 (i386/linux)) Content-Length: 0 ----------------------------------------------------------------------------------------------------- Here is an example of a successful connection with other proxy (asterisk in this case) SipClient: Listening UDP on port: 5062 SipClient: Our address: 192.168.2.176 SipRegister: Auth is '(null)' SipRegister: Proxy Auth is '(null)' SipClient: Sending: 08:47:08.026 -------------------------------- REGISTER sip:192.168.2.175 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.176:5062;rport CSeq: 1124 REGISTER To: "Luis Vazquez" <sip:101@192.168.2.175> Expires: 600 From: "Luis Vazquez" <sip:101@192.168.2.175> Call-ID: 546198833(a)192.168.2.176 Content-Length: 0 User-Agent: KPhone/3.14 Event: registration Allow-Events: presence Contact: "Luis Vazquez" <sip:luis@192.168.2.176:5062;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK" SipClient: Sending to '192.168.2.175:5060' SipClient: Receiving message... SipClient: Received: 08:47:08.048 --------------------------------- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.176:5062;rport From: "Luis Vazquez" <sip:101@192.168.2.175> To: "Luis Vazquez" <sip:101@192.168.2.175>;tag=as3f7be343 Call-ID: 546198833(a)192.168.2.176 CSeq: 1124 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:101@192.168.2.175> Content-Length: 0 SipCall: Incoming response SipTransaction: Incoming Response SipClient: Receiving message... SipClient: Received: 08:47:08.051 --------------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.176:5062;rport From: "Luis Vazquez" <sip:101@192.168.2.175> To: "Luis Vazquez" <sip:101@192.168.2.175>;tag=as3f7be343 Call-ID: 546198833(a)192.168.2.176 CSeq: 1124 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 600 Contact: <sip:101@192.168.2.175>;expires=600 Date: Tue, 02 Dec 2003 11:46:29 GMT Content-Length: 0 SipCall: Incoming response SipCall: Checking for Contact and Record-Route SipCall: Setting Contact for this Call Member SipTransaction: Incoming Response ReRegistrationTimer (ms): 540000 Regards Jorge