Hi all, I get a problem. I use pjlib 0.7.0 in my project as sip stack. 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@192.168.1.200:5060;transport=UDP SIP/2.0Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK74a60ee5;rportFrom: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as2858a32cTo: <sip:5001 at 192.168.1.200:5060;transport=UDP>Contact: <sip:66660000 at 192.168.1.11>Call-ID: 0bc7612c665e875a4a46411442b930a6 at 192.168.1.11CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 27 Jun 2008 08:46:47 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 285 v=0o=root 4236 4236 IN IP4 192.168.1.11s=sessionc=IN IP4 192.168.1.11t=0 0m=audio 14390 RTP/AVP 0 3 8 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=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 ErrorVia: SIP/2.0/UDP 192.168.1.11:5060;rport=5060;received=192.168.1.11;branch=z9hG4bK74a60ee5Call-ID: 0bc7612c665e875a4a46411442b930a6@192.168.1.11From: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as2858a32cTo: <sip:5001 at 192.168.1.200>CSeq: 102 INVITEContent-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@192.168.1.200:5060;transport=UDP SIP/2.0Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK74a60ee5;rportFrom: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as2858a32cTo: <sip:5001 at 192.168.1.200:5060;transport=UDP>Contact: <sip:66660000 at 192.168.1.11>Call-ID: 0bc7612c665e875a4a46411442b930a6 at 192.168.1.11CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-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@192.168.1.200:5060;transport=UDP SIP/2.0Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK1f760b73;rportFrom: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as6378f8caTo: <sip:5001 at 192.168.1.200:5060;transport=UDP>Contact: <sip:66660000 at 192.168.1.11>Call-ID: 2689f1ea79179f9d0f21486d37ada856 at 192.168.1.11CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 27 Jun 2008 08:46:50 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 285 v=0o=root 4236 4236 IN IP4 192.168.1.11s=sessionc=IN IP4 192.168.1.11t=0 0m=audio 11852 RTP/AVP 0 3 8 101a=rtpmap:0 PCMU/8000a=rtpmap:3 GSM/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv --end msg--Segmentation fault# Best Regards Michael Liu Chengdu, Sichuan, China Email:skysoshy at msn.com _________________________________________________________________ ?????????live mail???????? http://get.live.cn/product/mail.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080627/54ba61a1/attachment.html
