-- Attempting call on SIP/07112529826@qsc for application AGI(routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif) (Retry 1) == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Audio is at 93.189.169.91 port 17360 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.148.136.2:5060: INVITE sip:07112529826@sip.qsc.de SIP/2.0 Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK6b1e750f;rport Max-Forwards: 70 From: "0711123456" ;tag=as75239362 To: Contact: Call-ID: 3d1be2e92264d347650ee5855a454c09@qsc.de CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208502 Date: Fri, 24 Jul 2009 15:08:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 282 v=0 o=root 1391638937 1391638937 IN IP4 93.189.169.91 s=Asterisk PBX SVN-branch-1.6.0-r208502 c=IN IP4 93.189.169.91 t=0 0 m=audio 17360 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK6b1e750f;rport=5060 Call-ID: 3d1be2e92264d347650ee5855a454c09@qsc.de From: "0711123456";tag=as75239362 To: CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK6b1e750f;rport=5060 Call-ID: 3d1be2e92264d347650ee5855a454c09@qsc.de From: "0711123456";tag=as75239362 To: ;tag=a52b8d31 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="qsc.de",nonce="4a69cfa683d605dff5c80ed01cda5ebdc3fbd3c3",qop="auth" Server: QSC SIP Router Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (no NAT) to 213.148.136.2:5060: ACK sip:07112529826@sip.qsc.de SIP/2.0 Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK6b1e750f;rport Max-Forwards: 70 From: "0711123456" ;tag=as75239362 To: ;tag=a52b8d31 Contact: Call-ID: 3d1be2e92264d347650ee5855a454c09@qsc.de CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.6.0-r208502 Content-Length: 0 --- Audio is at 93.189.169.91 port 17360 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 213.148.136.2:5060: INVITE sip:07112529826@sip.qsc.de SIP/2.0 Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK0a44c5cf;rport Max-Forwards: 70 From: "0711123456" ;tag=as75239362 To: Contact: Call-ID: 3d1be2e92264d347650ee5855a454c09@qsc.de CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208502 Proxy-Authorization: Digest username="02152994430", realm="qsc.de", algorithm=MD5, uri="sip:07112529826@sip.qsc.de", nonce="4a69cfa683d605dff5c80ed01cda5ebdc3fbd3c3", response="ee18c3729ce703659de3bc2ca531bc62", qop=auth, cnonce="7f214517", nc=00000001 Date: Fri, 24 Jul 2009 15:08:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 282 v=0 o=root 1391638937 1391638938 IN IP4 93.189.169.91 s=Asterisk PBX SVN-branch-1.6.0-r208502 c=IN IP4 93.189.169.91 t=0 0 m=audio 17360 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK0a44c5cf;rport=5060 Call-ID: 3d1be2e92264d347650ee5855a454c09@qsc.de From: "0711123456";tag=as75239362 To: CSeq: 103 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK0a44c5cf;rport=5060 Call-ID: 3d1be2e92264d347650ee5855a454c09@qsc.de From: "0711123456";tag=as75239362 To: ;tag=c8312f1d CSeq: 103 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Supported: 100rel,replaces,timer,precondition,histinfo Contact: Content-Length: 207 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 9385832 9385832 IN IP4 213.148.136.2 s=Sip Call c=IN IP4 213.148.136.2 t=0 0 m=audio 22644 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 213.148.136.2:22644 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.148.136.2:22644 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 213.148.136.2, port 5060 Transmitting (no NAT) to 213.148.136.2:5060: ACK sip:213.148.136.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK4ff70643;rport Max-Forwards: 70 From: "0711123456" ;tag=as75239362 To: ;tag=c8312f1d Contact: Call-ID: 3d1be2e92264d347650ee5855a454c09@qsc.de CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.6.0-r208502 Content-Length: 0 --- > Channel SIP/qsc-0822f7a0 was answered. > Launching AGI(routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif) on SIP/qsc-0822f7a0 -- Launched AGI Script /var/lib/asterisk/agi-bin/routing.php routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_request' => 'routing.php' routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_channel' => 'SIP/qsc-0822f7a0' routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_language' => 'en' routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_type' => 'SIP' routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_uniqueid' => '1248448122.10' routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_version' => 'SVN-branch-1.6.0-r208502' routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_callerid' => routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_calleridname routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_callingpres' routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_callingani2' routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_callington' routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_callingtns' routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_dnid' => 'un routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_rdnis' => 'u routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_context' => routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_extension' = routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_priority' => routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_enhanced' => routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_accountcode' routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_threadid' => routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: 'agi_arg_1' => '/ routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: string(58) "unkno routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: >> EXEC SendFAX / -- AGI Script Executing Application: (SendFAX) Options: (/tmp/email2fax/1248 stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> hello <-------------> [Jul 24 15:13:43] WARNING[11081]: app_fax.c:479 transmit_audio: It looks like we hung. Aborting. [Jul 24 15:13:43] WARNING[11081]: app_fax.c:178 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. [Jul 24 15:13:43] WARNING[11081]: app_fax.c:677 transmit: Transmission error routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: >> GET VARIABLE ANSWEREDTIME routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: >> GET VARIABLE DIALSTATUS routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: >> GET VARIABLE FAXSTATUS routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: >> GET VARIABLE FAXERROR routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: >> GET VARIABLE FAXMODE routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: >> GET VARIABLE FAXPAGES routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: >> GET VARIABLE FAXBITRATE routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: >> GET VARIABLE FAXRESOLUTION routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: >> GET VARIABLE REMOTESTATIONID routing.php,/tmp/email2fax/1248448121-921117000/test.pdf.tif: string(48) " FAILED The call dropped prematurely audio "n -- AGI Script routing.php completed, returning 0 Scheduling destruction of SIP dialog '3d1be2e92264d347650ee5855a454c09@qsc.de' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 213.148.136.2, port 5060 Reliably Transmitting (no NAT) to 213.148.136.2:5060: BYE sip:213.148.136.2:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK6f8888b5;rport Max-Forwards: 70 From: "0711123456" ;tag=as75239362 To: ;tag=c8312f1d Call-ID: 3d1be2e92264d347650ee5855a454c09@qsc.de CSeq: 104 BYE User-Agent: Asterisk PBX SVN-branch-1.6.0-r208502 Proxy-Authorization: Digest username="02152994430", realm="qsc.de", algorithm=MD5, uri="sip:213.148.136.2:5060", nonce="4a69cfa683d605dff5c80ed01cda5ebdc3fbd3c3", response="631c2081fec2c2e6664a9c09fc3c4cf5", qop=auth, cnonce="1aaf74a6", nc=00000002 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jul 24 15:13:43] NOTICE[11081]: pbx_spool.c:357 attempt_thread: Call completed to SIP/07112529826@qsc stars24*CLI> <--- SIP read from UDP://213.148.136.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 93.189.169.91:5060;branch=z9hG4bK6f8888b5;rport=5060 Call-ID: 3d1be2e92264d347650ee5855a454c09@qsc.de From: "0711123456";tag=as75239362 To: ;tag=c8312f1d CSeq: 104 BYE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '3d1be2e92264d347650ee5855a454c09@qsc.de' Method: INVITE