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. Following is Sip Log: 2.1 5001 call 5002 (Both on Pjsip) Asterisk Sip Log: ############################################################################################################################################# Really destroying SIP dialog '2FECF925-4515-4FE0-886D-F592C3845965@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: 269 v=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@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@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@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: 269 v=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@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@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> ############################################################################################################################################# 2.2 66660000 call 5002 (SjPhone call Pjsip phone) Asterisk Sip Log: ############################################################################################################################################# AsteriskServer*CLI> <--- SIP read from 192.168.1.69:5060 --->INVITE sip:5002@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@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@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@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@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@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: 285 v=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@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@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@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@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> ############################################################################################################################################# 2.3 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@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: 285 v=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@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@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 _________________________________________________________________ ?????????live mail???????? http://get.live.cn/product/mail.html -------------- next part -------------- An HTML attachment was scrubbed... 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