Hi all, I got a problem in my project. There is a pjsip 0.7.0 that used in it. I set up a AsteriskNow 1.0.2 as Sip Proxy Server. And install two SjPhones,One on my PC,the other one on another PC. I do some simple configuration on Asterisk Sever: Add four accout for two Pjsip phone and my SjPhones. Like this: [5001]context=occ_siptype=friendhost=dynamicusername=5001secret=sparc10nat=nocanreinvite=yesinsecure=port,invite Add four dial rule: [occ_sip]exten => 66660000,1,dial(SIP/66660000)exten => 66661111,1,dial(SIP/66661111)exten => 5001,1,dial(SIP/5001)exten => 5002,1,dial(SIP/5002) Any body has a clue or suggestion.Please let me know.Thanks! Following are problem description: 1. When SjPhone call Pjsip Phone. Pjsip will response a 500 "Internal Server Error" for INVITE Message. If you call again.There will encounter a "Segmentation fault" Following is debug log of Pjsip: ###################################################################################################### First Call: ###################################################################################################### 19:03:06.346 pjsua_core.c RX 857 bytes Request msg INVITE/cseq=102 (rdata0x104793dc) 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=z9hG4bK6c1d9bb1;rportFrom: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as62628929To: <sip:5001 at 192.168.1.200:5060;transport=UDP>Contact: <sip:66660000 at 192.168.1.11>Call-ID: 1797e7c14ead3bed679382095579af28 at 192.168.1.11CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 27 Jun 2008 05:42:27 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 10786 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-- 19:03:06.347 pjsua_core.c TX 317 bytes Response msg 500/INVITE/cseq=102 (tdta0x1047eb64) 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=z9hG4bK6c1d9bb1Call-ID: 1797e7c14ead3bed679382095579af28@192.168.1.11From: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as62628929To: <sip:5001 at 192.168.1.200>CSeq: 102 INVITEContent-Length: 0 --end msg-- 19:03:06.349 pjsua_core.c RX 407 bytes Request msg ACK/cseq=102 (rdata0x104793dc) 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=z9hG4bK6c1d9bb1;rportFrom: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as62628929To: <sip:5001 at 192.168.1.200:5060;transport=UDP>Contact: <sip:66660000 at 192.168.1.11>Call-ID: 1797e7c14ead3bed679382095579af28 at 192.168.1.11CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0 --end msg-- 19:03:06.349 sip_endpoint.c Message Request msg ACK/cseq=102 (rdata0x104793dc) from 192.168.1.11:5060 was dropped/unhandled by any modules ###################################################################################################### Second Call: ###################################################################################################### 19:20:22.633 pjsua_core.c RX 857 bytes Request msg INVITE/cseq=102 (rdata0x104793dc) 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=z9hG4bK6feef7f9;rportFrom: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as54c98a19To: <sip:5001 at 192.168.1.200:5060;transport=UDP>Contact: <sip:66660000 at 192.168.1.11>Call-ID: 3b15421d5405cf87366858b0007f618e at 192.168.1.11CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 27 Jun 2008 05:59:43 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 16936 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 ###################################################################################################### 2. When Pjsip Phone call SjPhone.Asterisk will prompt some error: ###################################################################################################### [Jun 27 01:22:02] NOTICE[4271]: chan_sip.c:13885 handle_request_invite: Call from '5001' to extension '192.168.1.11:5060' rejected because extension not found.AsteriskServer*CLI> ###################################################################################################### Pjsip receive a 404 "Not found" Message. ###################################################################################################### 19:42:41.370 pjsua_core.c TX 1059 bytes Request msg INVITE/cseq=846930887 (tdta0x1047e614) to UDP 192.168.1.11:5060:INVITE sip:192.168.1.11:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.1.200:5060;rport;branch=z9hG4bKPj00ee0000001f6b8b4567Max-Forwards: 70From: "Freescale1" <sip:5001@192.168.1.11>;tag=00ee0000001c6b8b4567To: sip:5002 at 192.168.1.11Contact: "Freescale1" <sip:5001 at 192.168.1.200:5060;transport=UDP>Call-ID: 00ee0000001d6b8b4567CSeq: 846930887 INVITERoute: <sip:5002 at 192.168.1.11:5060>Route: <sip:192.168.1.11:5060>Allow: INVITE, ACK, BYE, CANCEL, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONSSupported: norefersubUser-Agent: PJSUA v0.7.0/linuxProxy-Authorization: Digest username="5001", realm="asterisk", nonce="1643cd23", uri="sip:192.168.1.11:5060", response="14f189611f61006e1d9539241fb111d1", algorithm=md5Content-Type: application/sdpContent-Length: 269 v=0o=- 2215885361 2215885361 IN IP4 192.168.1.200s=pjmediac=IN IP4 192.168.1.200t=0 0m=audio 11000 RTP/AVP 0 8 101a=rtcp:11001 IN IP4 192.168.1.200a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=sendrecva=rtpmap:101 telephone-event/8000a=fmtp:101 0-15 --end msg-- 19:42:41.373 pjsua_core.c RX 433 bytes Response msg 404/INVITE/cseq=846930887 (rdata0x1047941c) from UDP 192.168.1.11:5060:SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bKPj00ee0000001f6b8b4567;received=192.168.1.200;rport=5060From: "Freescale1" <sip:5001@192.168.1.11>;tag=00ee0000001c6b8b4567To: sip:5002 at 192.168.1.11;tag=as4b230045Call-ID: 00ee0000001d6b8b4567CSeq: 846930887 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 0 ###################################################################################################### Please get me some suggestion. Thank you so much! Best Regards Michael Liu Email: skysoshy at msn.com dongdong27 at 163.com _________________________________________________________________ Windows Live Photo gallery ????????????????????????????? http://get.live.cn/product/photo.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080627/3aafec12/attachment-0001.html
