Hi, 1. The users register to Asterisk successfully.Since I can call Pjsip Phoen by SjPhone(On my PC).Pjsip receive the sip packet. 2. The dst_uri that used to invoke pjsua_call_make_call is "sip:5002@192.168.1.11:5060".Should i add "<>" to include them? 3. Following is Sip Log:3.1 5001 call 5002 (Both on Pjsip) Asterisk Sip Log:#############################################################################################################################################Really destroying SIP dialog '2FECF925-4515-4FE0-886D-F592C3845965 at 192.168.1.69' Method: OPTIONSAsteriskServer*CLI> <--- SIP read from 192.168.1.200:5060 --->INVITE sip:192.168.1.11:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.1.200:5060;rport;branch=z9hG4bKPj00ee000000226b8b4567Max-Forwards: 70From: "Freescale1" <sip:5001 at 192.168.1.11>;tag=00ee000000206b8b4567To: sip:5002 at 192.168.1.11Contact: "Freescale1" <sip:5001 at 192.168.1.200:5060;transport=UDP>Call-ID: 00ee000000216b8b4567CSeq: 846930886 INVITERoute: <sip:192.168.1.11:5060>Route: <sip:5002 at 192.168.1.11:5060>Allow: INVITE, ACK, BYE, CANCEL, SUBSCRIBE, NOTIFY, PUBLISH, REFER, MESSAGE, OPTIONSSupported: norefersubUser-Agent: PJSUA v0.7.0/linuxContent-Type: application/sdpContent-Length: 269v=0o=- 2215886540 2215886540 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<------------->--- (15 headers 12 lines) ---Sending to 192.168.1.200 : 5060 (NAT)Using INVITE request as basis request - 00ee000000216b8b4567<--- Reliably Transmitting (no NAT) to 192.168.1.200:5060 --->SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bKPj00ee000000226b8b4567;received=192.168.1.200;rport=5060From: "Freescale1" <sip:5001 at 192.168.1.11>;tag=00ee000000206b8b4567To: sip:5002 at 192.168.1.11;tag=as12bc66f2Call-ID: 00ee000000216b8b4567CSeq: 846930886 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesProxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="325d69e2"Content-Length: 0<------------>Scheduling destruction of SIP dialog '00ee000000216b8b4567' in 32000 ms (Method: INVITE)Found user '5001'AsteriskServer*CLI> <--- SIP read from 192.168.1.200:5060 --->ACK sip:192.168.1.11:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.1.200:5060;rport;branch=z9hG4bKPj00ee000000226b8b4567Max-Forwards: 70From: "Freescale1" <sip:5001 at 192.168.1.11>;tag=00ee000000206b8b4567To: sip:5002 at 192.168.1.11;tag=as12bc66f2Call-ID: 00ee000000216b8b4567CSeq: 846930886 ACKRoute: <sip:192.168.1.11:5060>Route: <sip:5002 at 192.168.1.11:5060>Content-Length: 0<------------->--- (10 headers 0 lines) ---AsteriskServer*CLI> <--- SIP read from 192.168.1.200:5060 --->INVITE sip:192.168.1.11:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.1.200:5060;rport;branch=z9hG4bKPj00ee000000236b8b4567Max-Forwards: 70From: "Freescale1" <sip:5001 at 192.168.1.11>;tag=00ee000000206b8b4567To: sip:5002 at 192.168.1.11Contact: "Freescale1" <sip:5001 at 192.168.1.200:5060;transport=UDP>Call-ID: 00ee000000216b8b4567CSeq: 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="325d69e2", uri="sip:192.168.1.11:5060", response="24d7cd7b6970bcd1501e951b2d16a9e0", algorithm=md5Content-Type: application/sdpContent-Length: 269v=0o=- 2215886540 2215886540 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<------------->--- (16 headers 12 lines) ---Sending to 192.168.1.200 : 5060 (NAT)Using INVITE request as basis request - 00ee000000216b8b4567Found user '5001'Found RTP audio format 0Found RTP audio format 8Found RTP audio format 101Peer audio RTP is at port 192.168.1.200:11000Found audio description format PCMU for ID 0Found audio description format PCMA for ID 8Found audio description format telephone-event for ID 101Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)Peer audio RTP is at port 192.168.1.200:11000Looking for s in occ_sip (domain 192.168.1.11)<--- Reliably Transmitting (no NAT) to 192.168.1.200:5060 --->SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bKPj00ee000000236b8b4567;received=192.168.1.200;rport=5060From: "Freescale1" <sip:5001 at 192.168.1.11>;tag=00ee000000206b8b4567To: sip:5002 at 192.168.1.11;tag=as12bc66f2Call-ID: 00ee000000216b8b4567CSeq: 846930887 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Length: 0<------------>[Jun 27 01:41:41] NOTICE[4271]: chan_sip.c:13885 handle_request_invite: Call from '5001' to extension '192.168.1.11:5060' rejected because extension not found.Scheduling destruction of SIP dialog '00ee000000216b8b4567' in 32000 ms (Method: INVITE)AsteriskServer*CLI> <--- SIP read from 192.168.1.200:5060 --->ACK sip:192.168.1.11:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.1.200:5060;rport;branch=z9hG4bKPj00ee000000236b8b4567Max-Forwards: 70From: "Freescale1" <sip:5001 at 192.168.1.11>;tag=00ee000000206b8b4567To: sip:5002 at 192.168.1.11;tag=as12bc66f2Call-ID: 00ee000000216b8b4567CSeq: 846930887 ACKRoute: <sip:5002 at 192.168.1.11:5060>Route: <sip:192.168.1.11:5060>Content-Length: 0<------------->--- (10 headers 0 lines) ---AsteriskServer*CLI> #############################################################################################################################################3.2 66660000 call 5002 (SjPhone call Pjsip phone) Asterisk Sip Log:#############################################################################################################################################AsteriskServer*CLI> <--- SIP read from 192.168.1.69:5060 --->INVITE sip:5002 at 192.168.1.11:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.1.69;rport;branch=z9hG4bKc0a801450000008648648d0a00005b4200000321Content-Length: 337Contact: <sip:66660000 at 192.168.1.69:5060>Call-ID: 073C10DF-E494-46D2-9D16-30B2E543DCED at 192.168.1.69Content-Type: application/sdpCSeq: 1 INVITEFrom: "Michael Liu"<sip:66660000 at 192.168.1.11:5060>;tag=569600020446Max-Forwards: 70To: <sip:5002 at 192.168.1.11:5060>User-Agent: SJphone/1.60.289a (SJ Labs)v=0o=- 3423538058 3423538058 IN IP4 192.168.1.69s=SJphonec=IN IP4 192.168.1.69t=0 0a=direction:activem=audio 49168 RTP/AVP 3 97 98 8 0 101a=rtpmap:3 GSM/8000a=rtpmap:97 iLBC/8000a=rtpmap:98 iLBC/8000a=fmtp:98 mode=20a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-11,16<------------->--- (11 headers 15 lines) ---Sending to 192.168.1.69 : 5060 (NAT)Using INVITE request as basis request - 073C10DF-E494-46D2-9D16-30B2E543DCED at 192.168.1.69<--- Reliably Transmitting (no NAT) to 192.168.1.69:5060 --->SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 192.168.1.69;branch=z9hG4bKc0a801450000008648648d0a00005b4200000321;received=192.168.1.69;rport=5060From: "Michael Liu"<sip:66660000 at 192.168.1.11:5060>;tag=569600020446To: <sip:5002 at 192.168.1.11:5060>;tag=as0747ca93Call-ID: 073C10DF-E494-46D2-9D16-30B2E543DCED at 192.168.1.69CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesProxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a64d56e"Content-Length: 0<------------>Scheduling destruction of SIP dialog '073C10DF-E494-46D2-9D16-30B2E543DCED at 192.168.1.69' in 32000 ms (Method: INVITE)Found user '66660000'AsteriskServer*CLI> <--- SIP read from 192.168.1.69:5060 --->ACK sip:5002 at 192.168.1.11:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.1.69;rport;branch=z9hG4bKc0a801450000008648648d0a00005b4200000321Content-Length: 0Call-ID: 073C10DF-E494-46D2-9D16-30B2E543DCED at 192.168.1.69CSeq: 1 ACKFrom: "Michael Liu"<sip:66660000 at 192.168.1.11:5060>;tag=569600020446Max-Forwards: 70To: <sip:5002 at 192.168.1.11:5060>;tag=as0747ca93User-Agent: SJphone/1.60.289a (SJ Labs)<------------->--- (9 headers 0 lines) ---AsteriskServer*CLI> <--- SIP read from 192.168.1.69:5060 --->INVITE sip:5002 at 192.168.1.11:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.1.69;rport;branch=z9hG4bKc0a801450000008648648d0a0000295e00000322Content-Length: 337Contact: <sip:66660000 at 192.168.1.69:5060>Call-ID: 073C10DF-E494-46D2-9D16-30B2E543DCED at 192.168.1.69Content-Type: application/sdpCSeq: 2 INVITEFrom: "Michael Liu"<sip:66660000 at 192.168.1.11:5060>;tag=569600020446Max-Forwards: 70To: <sip:5002 at 192.168.1.11:5060>User-Agent: SJphone/1.60.289a (SJ Labs)Proxy-Authorization: Digest username="66660000",realm="asterisk",nonce="0a64d56e",uri="sip:5002 at 192.168.1.11:5060",response="d35b3b82f618c6b563246d3af1eb458b",algorithm="MD5"v=0o=- 3423538058 3423538058 IN IP4 192.168.1.69s=SJphonec=IN IP4 192.168.1.69t=0 0a=direction:activem=audio 49168 RTP/AVP 3 97 98 8 0 101a=rtpmap:3 GSM/8000a=rtpmap:97 iLBC/8000a=rtpmap:98 iLBC/8000a=fmtp:98 mode=20a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-11,16<------------->--- (12 headers 15 lines) ---Sending to 192.168.1.69 : 5060 (NAT)Using INVITE request as basis request - 073C10DF-E494-46D2-9D16-30B2E543DCED at 192.168.1.69Found user '66660000'Found RTP audio format 3Found RTP audio format 97Found RTP audio format 98Found RTP audio format 8Found RTP audio format 0Found RTP audio format 101Peer audio RTP is at port 192.168.1.69:49168Found audio description format GSM for ID 3Found audio description format iLBC for ID 97Found audio description format iLBC for ID 98Found audio description format PCMA for ID 8Found audio description format PCMU for ID 0Found audio description format telephone-event for ID 101Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)Peer audio RTP is at port 192.168.1.69:49168Looking for 5002 in occ_sip (domain 192.168.1.11)list_route: hop: <sip:66660000 at 192.168.1.69:5060><--- Transmitting (no NAT) to 192.168.1.69:5060 --->SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.1.69;branch=z9hG4bKc0a801450000008648648d0a0000295e00000322;received=192.168.1.69;rport=5060From: "Michael Liu"<sip:66660000 at 192.168.1.11:5060>;tag=569600020446To: <sip:5002 at 192.168.1.11:5060>Call-ID: 073C10DF-E494-46D2-9D16-30B2E543DCED at 192.168.1.69CSeq: 2 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip:5002 at 192.168.1.11>Content-Length: 0<------------>Audio is at 192.168.1.11 port 15496Adding codec 0x4 (ulaw) to SDPAdding codec 0x2 (gsm) to SDPAdding codec 0x8 (alaw) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 192.168.1.200:5060:INVITE sip:5002 at 192.168.1.200:5060;transport=UDP SIP/2.0Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK42d1da69;rportFrom: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as4a4ccddaTo: <sip:5002 at 192.168.1.200:5060;transport=UDP>Contact: <sip:66660000 at 192.168.1.11>Call-ID: 7c7db2a975d0f0550fc43c354624754d at 192.168.1.11CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 27 Jun 2008 06:47:36 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 285v=0o=root 4236 4236 IN IP4 192.168.1.11s=sessionc=IN IP4 192.168.1.11t=0 0m=audio 15496 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---AsteriskServer*CLI> <--- SIP read from 192.168.1.200:5060 --->SIP/2.0 500 Internal Server ErrorVia: SIP/2.0/UDP 192.168.1.11:5060;rport=5060;received=192.168.1.11;branch=z9hG4bK42d1da69Call-ID: 7c7db2a975d0f0550fc43c354624754d at 192.168.1.11From: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as4a4ccddaTo: <sip:5002 at 192.168.1.200>CSeq: 102 INVITEContent-Length: 0<------------->--- (7 headers 0 lines) ---Transmitting (no NAT) to 192.168.1.200:5060:ACK sip:5002 at 192.168.1.200:5060;transport=UDP SIP/2.0Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK42d1da69;rportFrom: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as4a4ccddaTo: <sip:5002 at 192.168.1.200:5060;transport=UDP>Contact: <sip:66660000 at 192.168.1.11>Call-ID: 7c7db2a975d0f0550fc43c354624754d at 192.168.1.11CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0---AsteriskServer*CLI> <--- Transmitting (no NAT) to 192.168.1.69:5060 --->SIP/2.0 503 Service UnavailableVia: SIP/2.0/UDP 192.168.1.69;branch=z9hG4bKc0a801450000008648648d0a0000295e00000322;received=192.168.1.69;rport=5060From: "Michael Liu"<sip:66660000 at 192.168.1.11:5060>;tag=569600020446To: <sip:5002 at 192.168.1.11:5060>;tag=as17dfab5eCall-ID: 073C10DF-E494-46D2-9D16-30B2E543DCED at 192.168.1.69CSeq: 2 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip:5002 at 192.168.1.11>Content-Length: 0X-Asterisk-HangupCause: Network out of orderX-Asterisk-HangupCauseCode: 38<------------>AsteriskServer*CLI> <--- SIP read from 192.168.1.69:5060 --->ACK sip:5002 at 192.168.1.11:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.1.69;rport;branch=z9hG4bKc0a801450000008648648d0a0000295e00000322Content-Length: 0Call-ID: 073C10DF-E494-46D2-9D16-30B2E543DCED at 192.168.1.69CSeq: 2 ACKFrom: "Michael Liu"<sip:66660000 at 192.168.1.11:5060>;tag=569600020446Max-Forwards: 70To: <sip:5002 at 192.168.1.11:5060>;tag=as17dfab5eUser-Agent: SJphone/1.60.289a (SJ Labs)<------------->--- (9 headers 0 lines) ---Really destroying SIP dialog '7c7db2a975d0f0550fc43c354624754d at 192.168.1.11' Method: INVITEReally destroying SIP dialog '073C10DF-E494-46D2-9D16-30B2E543DCED at 192.168.1.69' Method: ACKAsteriskServer*CLI> ############################################################################################################################################# 3.3 66660000 call 5002 (SjPhone call Pjsip phone) PjSip Log: ############################################################################################################################################# 20:08:15.338 pjsua_core.c RX 857 bytes Request msg INVITE/cseq=102 (rdata0x1047941c) from UDP 192.168.1.11:5060:INVITE sip:5002 at 192.168.1.200:5060;transport=UDP SIP/2.0Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK42d1da69;rportFrom: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as4a4ccddaTo: <sip:5002 at 192.168.1.200:5060;transport=UDP>Contact: <sip:66660000 at 192.168.1.11>Call-ID: 7c7db2a975d0f0550fc43c354624754d at 192.168.1.11CSeq: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 27 Jun 2008 06:47:36 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContent-Type: application/sdpContent-Length: 285v=0o=root 4236 4236 IN IP4 192.168.1.11s=sessionc=IN IP4 192.168.1.11t=0 0m=audio 15496 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-- 20:08:15.338 pjsua_core.c TX 317 bytes Response msg 500/INVITE/cseq=102 (tdta0x1047e004) 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=z9hG4bK42d1da69Call-ID: 7c7db2a975d0f0550fc43c354624754d at 192.168.1.11From: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as4a4ccddaTo: <sip:5002 at 192.168.1.200>CSeq: 102 INVITEContent-Length: 0--end msg-- 20:08:15.341 pjsua_core.c RX 407 bytes Request msg ACK/cseq=102 (rdata0x1047941c) from UDP 192.168.1.11:5060:ACK sip:5002 at 192.168.1.200:5060;transport=UDP SIP/2.0Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK42d1da69;rportFrom: "Michael Liu" <sip:66660000 at 192.168.1.11>;tag=as4a4ccddaTo: <sip:5002 at 192.168.1.200:5060;transport=UDP>Contact: <sip:66660000 at 192.168.1.11>Call-ID: 7c7db2a975d0f0550fc43c354624754d at 192.168.1.11CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0--end msg-- 20:08:15.341 sip_endpoint.c Message Request msg ACK/cseq=102 (rdata0x1047941c) from 192.168.1.11:5060 was dropped/unhandled by any modules############################################################################################################################################ Thanks and Regards Michael Date: Thu, 26 Jun 2008 23:38:40 -0700From: muki_champs@yahoo.comTo: pjsip at lists.pjsip.orgCC: liuzhidong at opencon3322.orgSubject: Re: Help: PjSip INVITE Message problem Hi, Please check the sip format to dialing the users.or can you plesae send us the sip Registrar format to me. a. have u registrar the users in Asterisk server along iwth the ip. check the invite request in the "Asterisk server"... and send me the log.Have a look @ my small web-page:http://www.geocities.com/muki_champsRegards, Mukesh Kumar, Sr.Software Engineer, Mobile Application Developer.Hyderabad. India. +91-9397845485 (M) ----- Original Message ----From: ?? <skysoshy@xxxxxxx>To: pjsip at lists.pjsip.orgCc: liuzhidong at opencon3322.orgSent: Friday, June 27, 2008 11:56:10 AMSubject: Help: PjSip INVITE Message problem 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,inviteAdd 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: 285v=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 at 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 at 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 at 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: 285v=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 at 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: 269v=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 at 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 RegardsMichael Liu Email:skysoshy at msn.comdongdong27@xxxxxxx ????? Windows Live Messenger ???????? ????? _________________________________________________________________ ?????????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/20080627/72db4007/attachment-0001.html
