Unfortunately there's nothing in the log that gives us any info about this, so you should ask the other end why it is disconnecting the call. Cheers Benny 2010/1/3 Bo Song <songbo_highlight at hotmail.com>: > > Hi, > ? I have met a problem that run the PJSIP symbian_ua_gui example. > ? I bulid a sisx file with Carbide C/C++ and install in NOKIA N73. I make a > call with this application, it can register successfully. But, after the > confirmed by > callee, it's strange that the caller receive the BYE request and the > application have to exit this call. > The log message as below: > RX 373 bytes Response msg 200/REGISTER/cseq=25494 (rdata0x71da04) from UDP > 210.52.252.108:5060: > SIP/2.0 200 OK > CSeq: 25494 REGISTER > Via: SIP/2.0/UDP > 10.14.230.182:5060;rport;branch=z9hG4bKPjnghsLkjHbXoDxcByoZGf0jFBPBP4itTN > From: <sip:wsz@210.52.252.108>;tag=gk3RNJJp7Ne.Hqasm4FBJ5tg0NjnTCC9 > Call-ID: c66ZHx5fG5HFuROZqsKPq3P5Yy8J.J4e > To: <sip:wsz at 210.52.252.108>;tag=281242092106 > Contact: <sip:wsz at 10.14.230.182:5060>;expires=30 > Expires: 30 > Content-Length: 0 > > --end msg--sip:wsz at 210.52.252.108: registration success, status=200 (OK), > will re-register in 30 secondsKeep-alive timer started for acc 1, > destination:210.52.252.108:5060, interval:15sRegistration success!RX 633 > bytes Response msg 200/INVITE/cseq=14946 (rdata0x71da04) from UDP > 210.52.252.108:5060: > SIP/2.0 > 200 OK > CSeq: 14946 INVITE > Via: SIP/2.0/UDP > 10.14.230.182:5060;rport;branch=z9hG4bKPjUL89AEql5o50wLomdkDK5Tpp9TnyaIKw > From: sip:wsz@210.52.252.108;tag=oMiS2VsfsgIMlTWFdT0v7HcC9RZ7O9X- > Call-ID: MOTwWfT8guaDRb35fe-SNYEntSNolOtM > To: sip:013880767030 at 210.52.252.108;tag=2812410921537011883127973 > Contact: <sip:210.52.252.108:5060;transport=udp> > Content-Type: application/sdp > Content-Length: 224 > v=0 > o=VoipSwitch 8972 8972 IN IP4 210.52.252.108 > s=VoipSIP > i=Audio Session > c=IN IP4 210.52.252.108 > t=0 0 > m=audio 7972 RTP/AVP 3 101 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > --end msg--Call 0 state=CONNECTING > VAD temporarily disabledEncoder stream startedDecoder stream startedMedia > updates, stream #0: GSM (sendrecv)Port 1 (sip:013880767030 at 210.52.252.108) > transmitting to > port 0 (Symbian Audio)Port 0 (Symbian Audio) transmitting to port 1 > (sip:013880767030 at 210.52.252.108)TX 377 bytes Request msg ACK/cseq=14946 > (tdta0x74d540) to UDP > 210.52.252.108:5060: > ACK sip:210.52.252.108:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP > 10.14.230.182:5060;rport;branch=z9hG4bKPju9qRLRFifXPfgBJYle6hofuMLJFP09BC > Max-Forwards: 70 > From: sip:wsz@210.52.252.108;tag=oMiS2VsfsgIMlTWFdT0v7HcC9RZ7O9X- > To: sip:013880767030 at 210.52.252.108;tag=2812410921537011883127973 > Call-ID: MOTwWfT8guaDRb35fe-SNYEntSNolOtM > CSeq: 14946 ACK > Content-Length:? 0 > > --end msg--Call 0 state=CONFIRMED > RX 322 bytes Request msg BYE/cseq=2 (rdata0x71da04) from UDP > 210.52.252.108:5060: > BYE sip:wsz at 210.52.252.108;tag=oMiS2VsfsgIMlTWFdT0v7HcC9RZ7O9X- SIP/2.0 > CSeq: 2 BYE > Via: SIP/2.0/UDP 210.52.252.108:5060 > From: sip:013880767030@210.52.252.108;tag=2812410921537011883127973 > Call-ID: MOTwWfT8guaDRb35fe-SNYEntSNolOtM > To: sip:wsz at 210.52.252.108;tag=oMiS2VsfsgIMlTWFdT0v7HcC9RZ7O9X- > Content-Length: 0 > > --end msg--TX 294 bytes Response msg 200/BYE/cseq=2 (tdta0x751c50) to UDP > 210.52.252.108:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 210.52.252.108:5060;received=210.52.252.108 > Call-ID: MOTwWfT8guaDRb35fe-SNYEntSNolOtM > From: <sip:013880767030@210.52.252.108>;tag=2812410921537011883127973 > To: <sip:wsz at 210.52.252.108>;tag=oMiS2VsfsgIMlTWFdT0v7HcC9RZ7O9X- > CSeq: 2 BYE > Content-Length:? 0 > > Can you help me to explain the reason why the server or the callee send the > BYE request to the caller, result that the call be stopped? > Thanks very much! > > Best regards. > > Bob Soong >
