Hi, I really think it is too old. But the point is it's in a almost completed project. At the last test.I encounter this issue. If i change to V 0.9.0 or some other version.It will take to much time. If any body has a idea.Please try to help me. Thanks! Best Regards Michael Liu > Date: Fri, 27 Jun 2008 10:49:31 +0100> From: bennylp@xxxxxxxxx> To: pjsip at lists.pjsip.org> Subject: Re: Pjsip segmentation fault issue when received a INVITE.> > 2008/6/27 Michael Liu <skysoshy at msn.com>:> >> > Hi all,> >> >> > I get a problem.> > I use pjlib 0.7.0 in my project as sip stack.> > I think all I can say now is version 0.7 is very old. It's more than a> year old, and in pjsip time scale that is light years behind :).> Please update to the latest SVN version, or the 0.9 which will be> released tomorrow.> > Cheers> Benny> > > And set up a Asterisk 1.0.2.1 as Sip Proxy Server.> >> >> > When i use a softphone(sjphone) call pjsip phone.> > At the first time.> >> > Pjsip will reply a 500 "internel server " response.> >> > The second time call pjsip phone will encounter a segmentation fault after> > received the INVITE message.> >> > I have no idea about it.Please help me,if you get any clues.> >> > Thank a lot!> >> > Pjsip Log:> >> > 22:07:25.948 pjsua_core.c RX 857 bytes Request msg INVITE/cseq=102> > (rdata0x103dde64) from UDP 192.168.1.11:5060:> > INVITE sip:5001 at 192.168.1.200:5060;transport=UDP SIP/2.0> > Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK74a60ee5;rport> > From: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as2858a32c> > To: <sip:5001 at 192.168.1.200:5060;transport=UDP>> > Contact: <sip:66660000 at 192.168.1.11>> > Call-ID: 0bc7612c665e875a4a46411442b930a6 at 192.168.1.11> > CSeq: 102 INVITE> > User-Agent: Asterisk PBX> > Max-Forwards: 70> > Date: Fri, 27 Jun 2008 08:46:47 GMT> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY> > Supported: replaces> > Content-Type: application/sdp> > Content-Length: 285> > v=0> > o=root 4236 4236 IN IP4 192.168.1.11> > s=session> > c=IN IP4 192.168.1.11> > t=0 0> > m=audio 14390 RTP/AVP 0 3 8 101> > a=rtpmap:0 PCMU/8000> > a=rtpmap:3 GSM/8000> > a=rtpmap:8 PCMA/8000> > a=rtpmap:101 telephone-event/8000> > a=fmtp:101 0-16> > a=silenceSupp:off - - - -> > a=ptime:20> > a=sendrecv> > --end msg--> > 22:07:25.949 pjsua_core.c TX 317 bytes Response msg 500/INVITE/cseq=102> > (tdta0x10475304) to UDP 192.168.1.11:5060:> > SIP/2.0 500 Internal Server Error> > Via: SIP/2.0/UDP> > 192.168.1.11:5060;rport=5060;received=192.168.1.11;branch=z9hG4bK74a60ee5> > Call-ID: 0bc7612c665e875a4a46411442b930a6 at 192.168.1.11> > From: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as2858a32c> > To: <sip:5001 at 192.168.1.200>> > CSeq: 102 INVITE> > Content-Length: 0> >> > --end msg--> > 22:07:25.952 pjsua_core.c RX 407 bytes Request msg ACK/cseq=102> > (rdata0x103dde64) from UDP 192.168.1.11:5060:> > ACK sip:5001 at 192.168.1.200:5060;transport=UDP SIP/2.0> > Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK74a60ee5;rport> > From: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as2858a32c> > To: <sip:5001 at 192.168.1.200:5060;transport=UDP>> > Contact: <sip:66660000 at 192.168.1.11>> > Call-ID: 0bc7612c665e875a4a46411442b930a6 at 192.168.1.11> > CSeq: 102 ACK> > User-Agent: Asterisk PBX> > Max-Forwards: 70> > Content-Length: 0> >> > --end msg--> > 22:07:25.952 sip_endpoint.c Message Request msg ACK/cseq=102> > (rdata0x103dde64) from 192.168.1.11:5060 was dropped/unhandled by any> > modules> >> >> > 22:07:29.177 pjsua_core.c RX 857 bytes Request msg INVITE/cseq=102> > (rdata0x103dde64) from UDP 192.168.1.11:5060:> > INVITE sip:5001 at 192.168.1.200:5060;transport=UDP SIP/2.0> > Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK1f760b73;rport> > From: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as6378f8ca> > To: <sip:5001 at 192.168.1.200:5060;transport=UDP>> > Contact: <sip:66660000 at 192.168.1.11>> > Call-ID: 2689f1ea79179f9d0f21486d37ada856 at 192.168.1.11> > CSeq: 102 INVITE> > User-Agent: Asterisk PBX> > Max-Forwards: 70> > Date: Fri, 27 Jun 2008 08:46:50 GMT> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY> > Supported: replaces> > Content-Type: application/sdp> > Content-Length: 285> > v=0> > o=root 4236 4236 IN IP4 192.168.1.11> > s=session> > c=IN IP4 192.168.1.11> > t=0 0> > m=audio 11852 RTP/AVP 0 3 8 101> > a=rtpmap:0 PCMU/8000> > a=rtpmap:3 GSM/8000> > a=rtpmap:8 PCMA/8000> > a=rtpmap:101 telephone-event/8000> > a=fmtp:101 0-16> > a=silenceSupp:off - - - -> > a=ptime:20> > a=sendrecv> > --end msg--> > Segmentation fault> > #> >> >> > Best Regards> > Michael Liu> >> > Chengdu,> > Sichuan,> > China> > Email:skysoshy at msn.com> >> > ________________________________> > ? 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