User-Agent: SJphone/1.60.289a (SJ Labs) Proxy-Authorization: Digest username="106",realm="asterisk",nonce="67d05f14",uri="sip:97418592@pt3",response="ee61bf0f7210eb0508e02400487d99af",algorithm="MD5" v=0 o=- 3384830620 3384830620 IN IP4 218.212.116.89 s=SJphone c=IN IP4 218.212.116.89 t=0 0 a=direction:active m=audio 49164 RTP/AVP 3 97 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (12 headers 15 lines) --- Using INVITE request as basis request - 8CCD023D-A4FA-4887-9A8F-AAAD9CDC9B93@192.168.0.145 Sending to 192.168.0.145 : 5060 (NAT) Found user '106' Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 218.212.116.89:49164 Peer video RTP is at port 218.212.116.89:65535 Found description format GSM Found description format iLBC Found description format iLBC Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 0x1c050e (gsm|ulaw|alaw|g729|ilbc|h261|h263|h263p), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x40e (gsm|ulaw|alaw|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 97418592 in from-internal (domain pt3) list_route: hop: Transmitting (NAT) to 218.212.116.89:60919: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.145;branch=z9hG4bKc0a800910000004d4615ec1c00006fba0000009d;received=218.212.116.89;rport=60919 From: "unknown";tag=5205131213109 To: Call-ID: 8CCD023D-A4FA-4887-9A8F-AAAD9CDC9B93@192.168.0.145 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Macro("SIP/106-082435b0", "dialout-trunk|2|97418592||") in new stack -- Executing Set("SIP/106-082435b0", "DIAL_TRUNK=2") in new stack -- Executing Set("SIP/106-082435b0", "_NODEST=") in new stack -- Executing Set("SIP/106-082435b0", "DIAL_NUMBER=97418592") in new stack -- Executing Set("SIP/106-082435b0", "ROUTE_PASSWD=") in new stack -- Executing Set("SIP/106-082435b0", "DIAL_TRUNK_OPTIONS=trwW") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?noauth") in new stack -- Goto (macro-dialout-trunk,s,8) -- Executing Set("SIP/106-082435b0", "GROUP()=OUT_2") in new stack -- Executing Macro("SIP/106-082435b0", "user-callerid|SKIPTTL") in new stack -- Executing NoOp("SIP/106-082435b0", "user-callerid: device 106") in new stack -- Executing GotoIf("SIP/106-082435b0", "0?report") in new stack -- Executing GotoIf("SIP/106-082435b0", "0?start") in new stack -- Executing Set("SIP/106-082435b0", "REALCALLERIDNUM=106") in new stack -- Executing NoOp("SIP/106-082435b0", "REALCALLERIDNUM is 106") in new stack -- Executing Set("SIP/106-082435b0", "AMPUSER=106") in new stack -- Executing Set("SIP/106-082435b0", "AMPUSERCIDNAME=Remote1") in new stack -- Executing GotoIf("SIP/106-082435b0", "0?report") in new stack -- Executing Set("SIP/106-082435b0", "CALLERID(all)=Remote1 <106>") in new stack -- Executing Set("SIP/106-082435b0", "REALCALLERIDNUM=106") in new stack -- Executing NoOp("SIP/106-082435b0", "TTL: ARG1: SKIPTTL") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/106-082435b0", "Using CallerID "Remote1" <106>") in new stack -- Executing Macro("SIP/106-082435b0", "record-enable|106|OUT") in new stack -- Executing GotoIf("SIP/106-082435b0", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/106-082435b0", "recordingcheck|20070406-144312|1175841791.22") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20070406-144312|1175841791.22: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/106-082435b0", "No recording needed") in new stack -- Executing GotoIf("SIP/106-082435b0", "0?skipoutcid") in new stack -- Executing Set("SIP/106-082435b0", "DIAL_TRUNK_OPTIONS=TrwW") in new stack -- Executing Macro("SIP/106-082435b0", "outbound-callerid|2") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?start") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp("SIP/106-082435b0", "REALCALLERIDNUM is 106") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,9) -- Executing Set("SIP/106-082435b0", "USEROUTCID=") in new stack -- Executing Set("SIP/106-082435b0", "EMERGENCYCID=") in new stack -- Executing Set("SIP/106-082435b0", "TRUNKOUTCID="Totalhearing" <6568845638>") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,16) -- Executing GotoIf("SIP/106-082435b0", "0?usercid") in new stack -- Executing Set("SIP/106-082435b0", "CALLERID(all)="Totalhearing" <6568845638>") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?report") in new stack -- Goto (macro-outbound-callerid,s,22) -- Executing NoOp("SIP/106-082435b0", "CallerID set to "Totalhearing" <6568845638>") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,16) -- Executing DeadAGI("SIP/106-082435b0", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix > fixlocalprefix: Using pattern 6XXXXXXX > fixlocalprefix: Using pattern 9XXXXXXX == fixlocalprefix: Dialpattern 9XXXXXXX matched. 97418592 -> 97418592 -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/106-082435b0", "OUTNUM=97418592") in new stack -- Executing Set("SIP/106-082435b0", "custom=ZAP/4") in new stack -- Executing GotoIf("SIP/106-082435b0", "0?customtrunk") in new stack -- Executing Dial("SIP/106-082435b0", "ZAP/4/97418592|300|TrwW") in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto("SIP/106-082435b0", "s-CHANUNAVAIL|1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp("SIP/106-082435b0", "Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack -- Executing Macro("SIP/106-082435b0", "dialout-trunk|1|97418592||") in new stack -- Executing Set("SIP/106-082435b0", "DIAL_TRUNK=1") in new stack -- Executing Set("SIP/106-082435b0", "_NODEST=") in new stack -- Executing Set("SIP/106-082435b0", "DIAL_NUMBER=97418592") in new stack -- Executing Set("SIP/106-082435b0", "ROUTE_PASSWD=") in new stack -- Executing Set("SIP/106-082435b0", "DIAL_TRUNK_OPTIONS=trwW") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?noauth") in new stack -- Goto (macro-dialout-trunk,s,8) -- Executing Set("SIP/106-082435b0", "GROUP()=OUT_1") in new stack -- Executing Macro("SIP/106-082435b0", "user-callerid|SKIPTTL") in new stack -- Executing NoOp("SIP/106-082435b0", "user-callerid: Totalhearing 6568845638") in new stack -- Executing GotoIf("SIP/106-082435b0", "0?report") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?start") in new stack -- Goto (macro-user-callerid,s,5) -- Executing NoOp("SIP/106-082435b0", "REALCALLERIDNUM is 106") in new stack -- Executing Set("SIP/106-082435b0", "AMPUSER=106") in new stack -- Executing Set("SIP/106-082435b0", "AMPUSERCIDNAME=Remote1") in new stack -- Executing GotoIf("SIP/106-082435b0", "0?report") in new stack -- Executing Set("SIP/106-082435b0", "CALLERID(all)=Remote1 <106>") in new stack -- Executing Set("SIP/106-082435b0", "REALCALLERIDNUM=106") in new stack -- Executing NoOp("SIP/106-082435b0", "TTL: ARG1: SKIPTTL") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/106-082435b0", "Using CallerID "Remote1" <106>") in new stack -- Executing Macro("SIP/106-082435b0", "record-enable|106|OUT") in new stack -- Executing GotoIf("SIP/106-082435b0", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/106-082435b0", "recordingcheck|20070406-144312|1175841791.22") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20070406-144312|1175841791.22: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/106-082435b0", "No recording needed") in new stack -- Executing GotoIf("SIP/106-082435b0", "0?skipoutcid") in new stack -- Executing Set("SIP/106-082435b0", "DIAL_TRUNK_OPTIONS=TrwW") in new stack -- Executing Macro("SIP/106-082435b0", "outbound-callerid|1") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?start") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp("SIP/106-082435b0", "REALCALLERIDNUM is 106") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,9) -- Executing Set("SIP/106-082435b0", "USEROUTCID=") in new stack -- Executing Set("SIP/106-082435b0", "EMERGENCYCID=") in new stack -- Executing Set("SIP/106-082435b0", "TRUNKOUTCID="Totalhearing" <6568845638>") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,16) -- Executing GotoIf("SIP/106-082435b0", "0?usercid") in new stack -- Executing Set("SIP/106-082435b0", "CALLERID(all)="Totalhearing" <6568845638>") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?report") in new stack -- Goto (macro-outbound-callerid,s,22) -- Executing NoOp("SIP/106-082435b0", "CallerID set to "Totalhearing" <6568845638>") in new stack -- Executing GotoIf("SIP/106-082435b0", "0?nomax") in new stack -- Executing GotoIf("SIP/106-082435b0", "0?chanfull") in new stack -- Executing DeadAGI("SIP/106-082435b0", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix > fixlocalprefix: Using pattern 6XXXXXXX > fixlocalprefix: Using pattern 9XXXXXXX == fixlocalprefix: Dialpattern 9XXXXXXX matched. 97418592 -> 97418592 -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/106-082435b0", "OUTNUM=97418592") in new stack -- Executing Set("SIP/106-082435b0", "custom=ZAP/3") in new stack -- Executing GotoIf("SIP/106-082435b0", "0?customtrunk") in new stack -- Executing Dial("SIP/106-082435b0", "ZAP/3/97418592|300|TrwW") in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto("SIP/106-082435b0", "s-CHANUNAVAIL|1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing NoOp("SIP/106-082435b0", "Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack -- Executing Macro("SIP/106-082435b0", "outisbusy|") in new stack -- Executing Playback("SIP/106-082435b0", "all-circuits-busy-now|noanswer") in new stack We're at 192.168.0.102 port 10042 Video is at 192.168.0.102 port 10014 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x400 (ilbc) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (NAT) to 218.212.116.89:60919: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.145;branch=z9hG4bKc0a800910000004d4615ec1c00006fba0000009d;received=218.212.116.89;rport=60919 From: "unknown";tag=5205131213109 To: ;tag=as2afdcbc1 Call-ID: 8CCD023D-A4FA-4887-9A8F-AAAD9CDC9B93@192.168.0.145 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 289 v=0 o=root 3990 3990 IN IP4 192.168.0.102 s=session c=IN IP4 192.168.0.102 t=0 0 m=audio 10042 RTP/AVP 0 8 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Playing 'all-circuits-busy-now' (language 'en') -- Executing Playback("SIP/106-082435b0", "pls-try-call-later|noanswer") in new stack -- Playing 'pls-try-call-later' (language 'en') pt3*CLI> <-- SIP read from 192.168.0.9:5060: REGISTER sip:PT3 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bKt0RQhwEmSCeedK7s Max-Forwards: 70 User-Agent: PA168S From: "101" ;tag=vp3ryfGbJOXpILO4 To: "101" Call-ID: cMVIZ3m56skNefSo@192.168.0.9 CSeq: 19512 REGISTER Contact: Expires: 60 Content-Length: 0 --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.9 : 5060 (NAT) Transmitting (no NAT) to 192.168.0.9:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bKt0RQhwEmSCeedK7s;received=192.168.0.9 From: "101" ;tag=vp3ryfGbJOXpILO4 To: "101" Call-ID: cMVIZ3m56skNefSo@192.168.0.9 CSeq: 19512 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.9:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bKt0RQhwEmSCeedK7s;received=192.168.0.9 From: "101" ;tag=vp3ryfGbJOXpILO4 To: "101" ;tag=as19b83778 Call-ID: cMVIZ3m56skNefSo@192.168.0.9 CSeq: 19512 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40b1fd32" Content-Length: 0 --3*CLI> Scheduling destruction of call 'cMVIZ3m56skNefSo@192.168.0.9' in 15000 ms pt3*CLI> <-- SIP read from 218.212.116.89:60919: OPTIONS sip:banshing.dyndns.org:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.145;rport;branch=z9hG4bKc0a80091000000104615eba0000058c400000084 Content-Length: 0 Call-ID: B2ADAA9F-7436-4D5C-97CC-6346281B3645@192.168.0.145 CSeq: 22 OPTIONS From: ;tag=5192726517924 Max-Forwards: 70 To: --- (8 headers 0 lines) --- Looking for s in from-sip-external (domain banshing.dyndns.org) Transmitting (NAT) to 218.212.116.89:60919: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.145;branch=z9hG4bKc0a80091000000104615eba0000058c400000084;received=218.212.116.89;rport=60919 From: ;tag=5192726517924 To: ;tag=as596812f0 Call-ID: B2ADAA9F-7436-4D5C-97CC-6346281B3645@192.168.0.145 CSeq: 22 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'B2ADAA9F-7436-4D5C-97CC-6346281B3645@192.168.0.145' -- Executing Macro("SIP/106-082435b0", "hangupcall") in new stack -- Executing ResetCDR("SIP/106-082435b0", "w") in new stack -- Executing NoCDR("SIP/106-082435b0", "") in new stack -- Executing GotoIf("SIP/106-082435b0", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing GotoIf("SIP/106-082435b0", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing Wait("SIP/106-082435b0", "5") in new stack pt3*CLI> <-- SIP read from 192.168.0.9:5060: REGISTER sip:PT3 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bKpLWU0Wgovz2JtlIl Max-Forwards: 70 User-Agent: PA168S From: "101" ;tag=4QSjH2RsTV3WIFKS To: "101" Call-ID: cMVIZ3m56skNefSo@192.168.0.9 CSeq: 19513 REGISTER Contact: Expires: 60 Authorization: Digest username="101", realm="asterisk", nonce="40b1fd32", uri="sip:PT3", response="1765a937ced98e75d4bedefb7585e04f", algorithm=MD5 Content-Length: 0 --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.9 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.0.9:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bKpLWU0Wgovz2JtlIl;received=192.168.0.9 From: "101" ;tag=4QSjH2RsTV3WIFKS To: "101" Call-ID: cMVIZ3m56skNefSo@192.168.0.9 CSeq: 19513 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY ontact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.9:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bKpLWU0Wgovz2JtlIl;received=192.168.0.9 From: "101" ;tag=4QSjH2RsTV3WIFKS To: "101" ;tag=as19b83778 Call-ID: cMVIZ3m56skNefSo@192.168.0.9 CSeq: 19513 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 60 Contact: ;expires=60 Date: Fri, 06 Apr 2007 06:41:08 GMT Content-Length: 0 --- Scheduling destruction of call 'cMVIZ3m56skNefSo@192.168.0.9' in 15000 ms -- Executing Hangup("SIP/106-082435b0", "") in new stack == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/106-082435b0' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/106-082435b0' in macro 'outisbusy' == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/106-082435b0' Scheduling destruction of call '69AE3CC2-1D8B-48B2-9B79-12B7900E3C90@192.168.0.145' in 32000 ms Reliably Transmitting (NAT) to 218.212.116.89:60919: SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.0.145;branch=z9hG4bKc0a80091000000454615eb9c0000127000000081;received=218.212.116.89;rport=60919 From: "unknown";tag=5192321815974 To: ;tag=as60f2a70c Call-ID: 69AE3CC2-1D8B-48B2-9B79-12B7900E3C90@192.168.0.145 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- pt3*CLI> <-- SIP read from 218.212.116.89:60919: ACK sip:97418592@pt3 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.145;rport;branch=z9hG4bKc0a80091000000454615eb9c0000127000000081 Content-Length: 0 Call-ID: 69AE3CC2-1D8B-48B2-9B79-12B7900E3C90@192.168.0.145 CSeq: 2 ACK From: "unknown";tag=5192321815974 Max-Forwards: 70 To: ;tag=as60f2a70c User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines) --- 12 headers, 3 lines Reliably Transmitting (no NAT) to 192.168.0.9:5060: NOTIFY sip:101@192.168.0.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK475a80db;rport From: "asterisk" ;tag=as65b628ed To: Contact: Call-ID: 346ac58824c1e7b9381e8d8408f309a8@192.168.0.102 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 95 Messages-Waiting: yes Message-Account: sip:asterisk@192.168.0.102 Voice-Message: 13/0 (0/0) --- Scheduling destruction of call '346ac58824c1e7b9381e8d8408f309a8@192.168.0.102' in 15000 ms pt3*CLI> <-- SIP read from 192.168.0.9:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK475a80db;rport Call-ID: 346ac58824c1e7b9381e8d8408f309a8@192.168.0.102 CSeq: 102 NOTIFY From: "asterisk" ;tag=as65b628ed To: Contact: Content-Length: 0