<-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '121eed2872e6b94669c9f6075a75bbf7@212.121.135.135' Method: ACK -- Attempting call on SIP/01234567@colt for application AGI(routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif) (Retry 1) == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Audio is at 192.168.0.39 port 12226 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 212.121.135.135:5060: INVITE sip:01234567@212.121.135.135 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.39:5060;branch=z9hG4bK470fa7b2;rport Max-Forwards: 70 From: "asterisk" ;tag=as619e9991 To: Contact: Call-ID: 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r209394 Date: Thu, 20 Aug 2009 06:34:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 280 v=0 o=root 1293770576 1293770576 IN IP4 192.168.0.39 s=Asterisk PBX SVN-branch-1.6.0-r209394 c=IN IP4 192.168.0.39 t=0 0 m=audio 12226 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 --- faxserver*CLI> <--- SIP read from UDP://212.121.135.135:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.39:5060;branch=z9hG4bK470fa7b2 From: "asterisk" ;tag=as619e9991 To: Call-ID: 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- faxserver*CLI> <--- SIP read from UDP://212.121.135.135:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.39:5060;branch=z9hG4bK470fa7b2 From: "asterisk" ;tag=as619e9991 To: ;tag=SDqmig299-566319325 Call-ID: 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 CSeq: 102 INVITE Contact: Content-Type: application/sdp Content-Length: 371 v=0 o=- 1294238 0 IN IP4 212.121.135.135 s=Cisco SDP 0 c=IN IP4 212.121.135.135 t=0 0 m=audio 30786 RTP/AVP 8 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 192-194,200-202 a=X-cap: 2 image udptl t38 <-------------> --- (9 headers 15 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found RTP audio format 100 Peer audio RTP is at port 212.121.135.135:30786 Found audio description format telephone-event for ID 101 Found unknown media description format X-NSE for ID 100 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 212.121.135.135:30786 faxserver*CLI> <--- SIP read from UDP://212.121.135.135:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.39:5060;branch=z9hG4bK470fa7b2 From: "asterisk" ;tag=as619e9991 To: ;tag=SDqmig299-566319325 Call-ID: 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 CSeq: 102 INVITE Contact: Content-Type: application/sdp Content-Length: 371 v=0 o=- 1294238 0 IN IP4 212.121.135.135 s=Cisco SDP 0 c=IN IP4 212.121.135.135 t=0 0 m=audio 30786 RTP/AVP 8 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 192-194,200-202 a=X-cap: 2 image udptl t38 <-------------> --- (9 headers 15 lines) --- faxserver*CLI> <--- SIP read from UDP://212.121.135.135:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.39:5060;branch=z9hG4bK470fa7b2 From: "asterisk" ;tag=as619e9991 To: ;tag=SDqmig299-566319325 Call-ID: 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 CSeq: 102 INVITE Contact: Content-Type: application/sdp Content-Length: 371 v=0 o=- 1294238 0 IN IP4 212.121.135.135 s=Cisco SDP 0 c=IN IP4 212.121.135.135 t=0 0 m=audio 30786 RTP/AVP 8 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 192-194,200-202 a=X-cap: 2 image udptl t38 <-------------> --- (9 headers 15 lines) --- faxserver*CLI> <--- SIP read from UDP://212.121.135.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.39:5060;branch=z9hG4bK470fa7b2 From: "asterisk" ;tag=as619e9991 To: ;tag=SDqmig299-566319325 Call-ID: 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, PRACK, SUBSCRIBE, BYE, CANCEL, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Content-Length: 371 v=0 o=- 1294238 0 IN IP4 212.121.135.135 s=Cisco SDP 0 c=IN IP4 212.121.135.135 t=0 0 m=audio 30786 RTP/AVP 8 101 100 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194,200-202 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 192-194,200-202 a=X-cap: 2 image udptl t38 <-------------> --- (10 headers 15 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 212.121.135.135, port 5060 Transmitting (NAT) to 212.121.135.135:5060: ACK sip:01234567@212.121.135.135:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.39:5060;branch=z9hG4bK1262ac24;rport Max-Forwards: 70 From: "asterisk" ;tag=as619e9991 To: ;tag=SDqmig299-566319325 Contact: Call-ID: 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.6.0-r209394 Content-Length: 0 --- -- Launched AGI Script /var/lib/asterisk/agi-bin/routing.php routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_request' => 'routing.php' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_channel' => 'SIP/colt-084834d8' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_language' => 'en' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_type' => 'SIP' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_uniqueid' => '1250750090.32' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_version' => 'SVN-branch-1.6.0-r209394' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_callerid' => 'unknown' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_calleridname' => 'unknown' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_callingpres' => '0' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_callingani2' => '0' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_callington' => '0' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_callingtns' => '0' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_dnid' => 'unknown' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_rdnis' => 'unknown' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_context' => 'default' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_extension' => '' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_priority' => '1' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_enhanced' => '0.0' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_accountcode' => '' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_threadid' => '-1225544816' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: 'agi_arg_1' => '/tmp/email2fax/1250750089-743863495/Womaco.tif.tif' routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: string(59) "unknown ; SIP/colt-084834d8 ; 1250750090.32 ; ; ; unknown"n routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: >> EXEC SendFAX /tmp/email2fax/1250750089-743863495/Womaco.tif.tif -- AGI Script Executing Application: (SendFAX) Options: (/tmp/email2fax/1250750089-743863495/Womaco.tif.tif) faxserver*CLI> <--- SIP read from UDP://212.121.135.135:5060 ---> INVITE sip:asterisk@192.168.0.34:1428 SIP/2.0 Via: SIP/2.0/UDP 212.121.135.135:5060;branch=z9hG4bKcvh9ob30ag1g6bojf2k1sb0000g00.1 From: ;tag=SDqmig299-566319325 To: "asterisk" ;tag=as619e9991 Call-ID: 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 CSeq: 1 INVITE Max-Forwards: 69 Supported: timer Session-Expires: 1800 Contact: Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE Content-Type: application/sdp Content-Length: 258 v=0 o=- 1294238 1 IN IP4 212.121.135.135 s=Cisco SDP 0 c=IN IP4 212.121.135.135 t=0 0 m=image 30786 udptl t38 a=X-sqn:0 a=X-cap: 1 audio RTP/AVP 100 a=X-cpar: a=rtpmap:100 X-NSE/8000 a=X-cpar: a=fmtp:100 192-194,200-202 a=X-cap: 2 image udptl t38 <-------------> --- (13 headers 11 lines) --- Sending to 212.121.135.135 : 5060 (NAT) Got T.38 offer in SDP in dialog 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 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) faxserver*CLI> <--- Transmitting (NAT) to 212.121.135.135:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.121.135.135:5060;branch=z9hG4bKcvh9ob30ag1g6bojf2k1sb0000g00.1;received=212.121.135.135 From: ;tag=SDqmig299-566319325 To: "asterisk" ;tag=as619e9991 Call-ID: 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r209394 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> faxserver*CLI> <--- Reliably Transmitting (NAT) to 212.121.135.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 212.121.135.135:5060;branch=z9hG4bKcvh9ob30ag1g6bojf2k1sb0000g00.1;received=212.121.135.135 From: ;tag=SDqmig299-566319325 To: "asterisk" ;tag=as619e9991 Call-ID: 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 CSeq: 1 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r209394 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 281 v=0 o=root 1293770576 1293770577 IN IP4 192.168.0.39 s=Asterisk PBX SVN-branch-1.6.0-r209394 c=IN IP4 192.168.0.39 t=0 0 m=image 4694 udptl t38 a=T38Faxversion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPFEC <------------> faxserver*CLI> <--- SIP read from UDP://212.121.135.135:5060 ---> ACK sip:asterisk@192.168.0.34:1428 SIP/2.0 Via: SIP/2.0/UDP 212.121.135.135:5060;branch=z9hG4bKcvh9ob30ag1g6bojf2k1sc0000g00.1 From: ;tag=SDqmig299-566319325 To: "asterisk" ;tag=as619e9991 Call-ID: 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 CSeq: 1 ACK Max-Forwards: 69 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- [Aug 20 06:35:32] WARNING[9620]: app_fax.c:178 phase_e_handler: Error transmitting fax. result=20: Received no response to DCS or TCF. [Aug 20 06:35:32] WARNING[9620]: app_fax.c:704 transmit: Transmission failed routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: >> GET VARIABLE ANSWEREDTIME routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: >> GET VARIABLE DIALSTATUS routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: >> GET VARIABLE FAXSTATUS routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: >> GET VARIABLE FAXERROR routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: >> GET VARIABLE FAXMODE routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: >> GET VARIABLE FAXPAGES routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: >> GET VARIABLE FAXBITRATE routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: >> GET VARIABLE FAXRESOLUTION routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: >> GET VARIABLE REMOTESTATIONID routing.php,/tmp/email2fax/1250750089-743863495/Womaco.tif.tif: string(52) " FAILED Received no response to DCS or TCF T38 "n [Aug 20 06:35:32] ERROR[9620]: utils.c:1019 ast_carefulwrite: write() returned error: Broken pipe -- AGI Script routing.php completed, returning 0 Scheduling destruction of SIP dialog '4bcfafa45830c068321c5edf5b5cc875@212.121.135.135' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 212.121.135.135, port 5060 Reliably Transmitting (NAT) to 212.121.135.135:5060: BYE sip:01234567@212.121.135.135:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.39:5060;branch=z9hG4bK3a9bc977;rport Max-Forwards: 70 From: "asterisk" ;tag=as619e9991 To: ;tag=SDqmig299-566319325 Call-ID: 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 CSeq: 103 BYE User-Agent: Asterisk PBX SVN-branch-1.6.0-r209394 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Aug 20 06:35:32] NOTICE[9620]: pbx_spool.c:357 attempt_thread: Call completed to SIP/01234567@colt faxserver*CLI> <--- SIP read from UDP://212.121.135.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.39:5060;branch=z9hG4bK3a9bc977 From: "asterisk" ;tag=as619e9991 To: ;tag=SDqmig299-566319325 Call-ID: 4bcfafa45830c068321c5edf5b5cc875@212.121.135.135 CSeq: 103 BYE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '4bcfafa45830c068321c5edf5b5cc875@212.121.135.135' Method: ACK faxserver*CLI>