Really destroying SIP dialog 'a4a58edf-8fe659b8@192.168.1.4' Method: REGISTER asx*CLI> -- Attempting call on Local/4483706@client/n for s@faxtx:1 (Retry 1) asx*CLI> -- Executing Macro("Local/4483706@client-2533;2", "client,4483706,fax,1,KamailioSPB,1,3364079,1") -- Executing [s@macro-client:1] Set("Local/4483706@client-2533;2", "__CALLEE=4483706") in new stack -- Executing [s@macro-client:2] Set("Local/4483706@client-2533;2", "__TALKNAME=fax") in new stack -- Executing [s@macro-client:3] Set("Local/4483706@client-2533;2", "__TALKCOUNT=1") in new stack -- Executing [s@macro-client:4] Set("Local/4483706@client-2533;2", "__CALLEDCHANNEL=KamailioSPB") in new stack -- Executing [s@macro-client:5] Set("Local/4483706@client-2533;2", "__CDRGENERATE=1") in new stack -- Executing [s@macro-client:6] Set("Local/4483706@client-2533;2", "__CALLER=3364079") in new stack -- Executing [s@macro-client:7] Set("Local/4483706@client-2533;2", "__LOCALCHANNEL=1") in new stack [Feb 2 09:27:01] WARNING[24344]: ast_expr2.fl:433 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '>', expecting $end; Input: > 0 ^ [Feb 2 09:27:01] WARNING[24344]: ast_expr2.fl:437 ast_yyerror: If you have questions, please refer to doc/tex/channelvariables.tex in the asterisk source. -- Executing [s@macro-client:8] GotoIf("Local/4483706@client-2533;2", "?calleridcheck,s,1") in new stack -- Executing [s@macro-client:9] Goto("Local/4483706@client-2533;2", "clientout,s,1") in new stack -- Goto (clientout,s,1) == Channel 'Local/4483706@client-2533;2' jumping out of macro 'client' -- Executing [s@clientout:1] Set("Local/4483706@client-2533;2", "GROUP()=fax") in new stack -- Executing [s@clientout:2] GotoIf("Local/4483706@client-2533;2", "0?endc") in new stack -- Executing [s@clientout:3] Set("Local/4483706@client-2533;2", "CALLERID(num)=3364079") in new stack -- Executing [s@clientout:4] GotoIf("Local/4483706@client-2533;2", "0?:cdr") in new stack -- Goto (clientout,s,6) -- Executing [s@clientout:6] GotoIf("Local/4483706@client-2533;2", "1?:phone") in new stack -- Executing [s@clientout:7] Set("Local/4483706@client-2533;2", "DB(CALLID/123456)=1265092021.1") in new stack asx*CLI> -- Executing [s@clientout:8] Dial("Local/4483706@client-2533;2", "SIP/KamailioSPB/4483706") in new stack == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Audio is at 1.1.1.1 port 14814 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 2.2.2.2:5060: INVITE sip:4483706@2.2.2.2 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6a13871d;rport Max-Forwards: 70 From: "3364079" ;tag=as06933882 To: Contact: Call-ID: 33f0da563fb8be3a71553bea55ed6c87@1.1.1.1 CSeq: 102 INVITE User-Agent: GW Date: Tue, 02 Feb 2010 06:27:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 642954861 642954861 IN IP4 1.1.1.1 s=Asterisk PBX 1.6.2.1 c=IN IP4 1.1.1.1 t=0 0 m=audio 14814 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called KamailioSPB/4483706 asx*CLI> <--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6a13871d;rport=5060 From: "3364079" ;tag=as06933882 To: Call-ID: 33f0da563fb8be3a71553bea55ed6c87@1.1.1.1 CSeq: 102 INVITE Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 <-------------> asx*CLI> --- (8 headers 0 lines) --- asx*CLI> <--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK6a13871d;rport=5060 From: "3364079" ;tag=as06933882 To: ;tag=556FB62C-1220 Date: Wed, 08 Nov 2000 16:20:18 GMT Call-ID: 33f0da563fb8be3a71553bea55ed6c87@1.1.1.1 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Contact: Record-Route: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 7748 1528 IN IP4 213.170.74.62 s=SIP Call c=IN IP4 213.170.74.62 t=0 0 m=audio 19100 RTP/AVP 8 101 c=IN IP4 213.170.74.62 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> asx*CLI> --- (14 headers 11 lines) --- asx*CLI> Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 asx*CLI> Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.170.74.62:19100 asx*CLI> -- SIP/KamailioSPB-00000000 is making progress passing it to Local/4483706@client-2533;2 asx*CLI> asx*CLI> <--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK6a13871d;rport=5060 From: "3364079" ;tag=as06933882 To: ;tag=556FB62C-1220 Date: Wed, 08 Nov 2000 16:20:18 GMT Call-ID: 33f0da563fb8be3a71553bea55ed6c87@1.1.1.1 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Contact: Record-Route: Content-Type: application/sdp Content-Length: 250 v=0 o=CiscoSystemsSIP-GW-UserAgent 7748 1528 IN IP4 213.170.74.62 s=SIP Call c=IN IP4 213.170.74.62 t=0 0 m=audio 19100 RTP/AVP 8 101 c=IN IP4 213.170.74.62 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> asx*CLI> --- (14 headers 11 lines) --- list_route: hop: asx*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 2.2.2.2, port 5060 Transmitting (no NAT) to 2.2.2.2:5060: ACK sip:4483706@213.170.74.62:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3a512ebb;rport Route: Max-Forwards: 70 From: "3364079" ;tag=as06933882 To: ;tag=556FB62C-1220 Contact: Call-ID: 33f0da563fb8be3a71553bea55ed6c87@1.1.1.1 CSeq: 102 ACK User-Agent: GW Content-Length: 0 --- asx*CLI> -- SIP/KamailioSPB-00000000 answered Local/4483706@client-2533;2 > Channel Local/4483706@client-2533;1 was answered. -- Executing [s@faxtx:1] NoOp("Local/4483706@client-2533;1", "**** SENDING FAX ****") in new stack -- Executing [s@faxtx:2] Wait("Local/4483706@client-2533;1", "3") in new stack asx*CLI> -- Executing [s@faxtx:3] Playback("Local/4483706@client-2533;1", "pls-try-manually") in new stack -- Playing 'pls-try-manually.slin' (language 'en') asx*CLI> -- Executing [s@faxtx:4] Wait("Local/4483706@client-2533;1", "2") in new stack asx*CLI> -- Executing [s@faxtx:5] Set("Local/4483706@client-2533;1", "FAXUNIQNAME=/asterisk/demand/phoneptl/faxdemand.tif") in new stack -- Executing [s@faxtx:6] Set("Local/4483706@client-2533;1", "CALLID=1265092021.1") in new stack asx*CLI> -- Executing [s@faxtx:7] Set("Local/4483706@client-2533;1", "1265092021.1") in new stack -- Executing [s@faxtx:8] SendFAX("Local/4483706@client-2533;1", "/asterisk/demand/phoneptl/faxdemand.tif") in new stack asx*CLI> <--- SIP read from UDP:2.2.2.2:5060 ---> INVITE sip:3364079@1.1.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK5486.464c4271.0 Via: SIP/2.0/UDP 213.170.74.62:5060 From: ;tag=556FB62C-1220 To: "3364079" ;tag=as06933882 Date: Wed, 08 Nov 2000 16:20:31 GMT Call-ID: 33f0da563fb8be3a71553bea55ed6c87@1.1.1.1 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 3706518479-3033076180-3194068891-3892023434 User-Agent: Cisco-SIPGateway/IOS-12.x Accept-Language: ru Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: ;party=calling;screen=no;privacy=off Timestamp: 973700431 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 398 v=0 o=CiscoSystemsSIP-GW-UserAgent 7748 1529 IN IP4 213.170.74.62 s=SIP Call c=IN IP4 213.170.74.62 t=0 0 m=image 19100 udptl t38 c=IN IP4 213.170.74.62 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> asx*CLI> --- (22 headers 16 lines) --- Sending to 2.2.2.2 : 5060 (no NAT) Got T.38 offer in SDP in dialog 33f0da563fb8be3a71553bea55ed6c87@1.1.1.1 Capabilities: us - 0x8 (alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. <--- Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK5486.464c4271.0;received=2.2.2.2 Via: SIP/2.0/UDP 213.170.74.62:5060 From: ;tag=556FB62C-1220 To: "3364079" ;tag=as06933882 Call-ID: 33f0da563fb8be3a71553bea55ed6c87@1.1.1.1 CSeq: 101 INVITE Server: GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> asx*CLI> <--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK5486.464c4271.0;received=2.2.2.2 Via: SIP/2.0/UDP 213.170.74.62:5060 From: ;tag=556FB62C-1220 To: "3364079" ;tag=as06933882 Call-ID: 33f0da563fb8be3a71553bea55ed6c87@1.1.1.1 CSeq: 101 INVITE Server: GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 273 v=0 o=root 642954861 642954862 IN IP4 1.1.1.1 s=Asterisk PBX 1.6.2.1 c=IN IP4 1.1.1.1 t=0 0 m=image 4035 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy <------------> asx*CLI> [Feb 2 09:27:14] ERROR[24343]: app_fax.c:795 transmit: Audio loop reports T38 switchover but t38state != T38_STATE_NEGOTIATED [Feb 2 09:27:14] WARNING[24343]: app_fax.c:806 transmit: Transmission error == Spawn extension (faxtx, s, 8) exited non-zero on 'Local/4483706@client-2533;1' -- Executing [h@faxtx:1] Wait("Local/4483706@client-2533;1", "50") in new stack == Spawn extension (faxtx, h, 1) exited non-zero on 'Local/4483706@client-2533;1' asx*CLI> [Feb 2 09:27:14] NOTICE[24343]: pbx_spool.c:349 attempt_thread: Call completed to Local/4483706@client/n -- Executing [h@clientout:1] GotoIf("Local/4483706@client-2533;2", "0?fax:endc") in new stack -- Goto (clientout,h,3) -- Executing [h@clientout:3] Hangup("Local/4483706@client-2533;2", "") in new stack asx*CLI> Scheduling destruction of SIP dialog '33f0da563fb8be3a71553bea55ed6c87@1.1.1.1' in 32000 ms (Method: INVITE) asx*CLI> == Spawn extension (clientout, s, 8) exited non-zero on 'Local/4483706@client-2533;2' asx*CLI> <--- SIP read from UDP:2.2.2.2:5060 ---> ACK sip:3364079@1.1.1.1:5060 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK5486.464c4271.2 Via: SIP/2.0/UDP 213.170.74.62:5060 From: ;tag=556FB62C-1220 To: "3364079" ;tag=as06933882 Date: Wed, 08 Nov 2000 16:20:31 GMT Call-ID: 33f0da563fb8be3a71553bea55ed6c87@1.1.1.1 Max-Forwards: 6 Content-Length: 0 CSeq: 101 ACK <-------------> --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 2.2.2.2, port 5060 Reliably Transmitting (no NAT) to 2.2.2.2:5060: BYE sip:4483706@213.170.74.62:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bfef70b;rport Route: Max-Forwards: 70 From: "3364079" ;tag=as06933882 To: ;tag=556FB62C-1220 C asx*CLI> all-ID: 33f0da563fb8be3a71553bea55ed6c87@1.1.1.1 CSeq: 103 BYE User-Agent: GW X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '33f0da563fb8be3a71553bea55ed6c87@1.1.1.1' in 32000 ms (Method: ACK) asx*CLI> <--- SIP read from UDP:2.2.2.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK4bfef70b;rport=5060 From: "3364079" ;tag=as06933882 To: ;tag=556FB62C-1220 Date: Wed, 08 Nov 2000 16:20:31 GMT Call-ID: 33f0da563fb8be3a71553bea55ed6c87@1.1.1.1 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 103 BYE