[2009-10-20 02:56:27.7388] DEBUG[6797] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 67.221.172.46:5060 [2009-10-20 02:56:28.1710] VERBOSE[6797] chan_sip.c: <--- SIP read from UDP://192.168.200.152:5060 ---> INVITE sip:5551234567@192.168.135.103;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.200.152:5060;branch=z9hG4bK2g54gs204gag6jgda4h1.1 From: ;tag=20c013a8+1+5b6c0004+71ff275f To: CSeq: 825934346 INVITE Expires: 45 Call-ID: 457753420c013a8 Remote-Party-ID: ;party=calling;id-type=subscriber;privacy=off;screen=no Max-Forwards: 15 Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=PVG 1256031719920 1256031719920 IN IP4 192.168.200.152 s=- p=+1 6135555555 c=IN IP4 192.168.200.152 t=0 0 m=audio 10658 RTP/AVP 18 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=fmtp:18 annexb=no <-------------> [2009-10-20 02:56:28.1711] VERBOSE[6797] chan_sip.c: --- (12 headers 11 lines) --- <--- Transmitting (no NAT) to 192.168.200.152:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.200.152:5060;branch=z9hG4bK2g54gs204gag6jgda4h1.1;received=192.168.200.152 From: ;tag=20c013a8+1+5b6c0004+71ff275f To: Call-ID: 457753420c013a8 CSeq: 825934346 INVITE Server: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [2009-10-20 02:56:28.1765] DEBUG[6797] chan_sip.c: Trying to put 'SIP/2.0 10' onto UDP socket destined for 192.168.200.152:5060 [2009-10-20 02:56:28.4790] DEBUG[29097] pbx.c: Launching 'ReceiveFAX' [2009-10-20 02:56:28.4790] VERBOSE[29097] pbx.c: -- Executing [21906@fax_receive:9] ReceiveFAX("SIP/provider-ada1c0f0", "/var/spool/hylafax/recvq/fax_in_1256032588.42.tiff") in new stack [2009-10-20 02:56:28.4801] DEBUG[29097] chan_sip.c: SIP answering channel: SIP/provider-ada1c0f0 [2009-10-20 02:56:28.4802] DEBUG[29097] chan_sip.c: Setting framing from config on incoming call [2009-10-20 02:56:28.4803] DEBUG[29097] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [2009-10-20 02:56:28.4803] DEBUG[29097] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [2009-10-20 02:56:28.4804] VERBOSE[29097] chan_sip.c: Audio is at 192.168.135.103 port 42834 [2009-10-20 02:56:28.4805] VERBOSE[29097] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-10-20 02:56:28.4805] VERBOSE[29097] chan_sip.c: Adding codec 0x8 (alaw) to SDP [2009-10-20 02:56:28.4807] VERBOSE[29097] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.200.152:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.200.152:5060;branch=z9hG4bK2g54gs204gag6jgda4h1.1;received=192.168.200.152 From: ;tag=20c013a8+1+5b6c0004+71ff275f To: ;tag=as12ab5a38 Call-ID: 457753420c013a8 CSeq: 825934346 INVITE Server: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 233 v=0 o=root 1453031403 1453031403 IN IP4 192.168.135.103 s=Asterisk PBX 1.6.1.6 c=IN IP4 192.168.135.103 t=0 0 m=audio 42834 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2009-10-20 02:56:28.4809] DEBUG[29097] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.200.152:5060 [2009-10-20 02:56:28.5067] VERBOSE[6797] chan_sip.c: <--- SIP read from UDP://192.168.200.152:5060 ---> ACK sip:5551234567@192.168.135.103 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.152:5060;branch=z9hG4bK20c7i5309gj0dl4b02b1.1 Call-ID: 457753420c013a8 From: ;tag=20c013a8+1+5b6c0004+71ff275f To: ;tag=as12ab5a38 CSeq: 825934346 ACK Contact: Content-Length: 0 Max-Forwards: 15 <-------------> [2009-10-20 02:56:28.5068] VERBOSE[6797] chan_sip.c: --- (9 headers 0 lines) --- [2009-10-20 02:56:28.5069] DEBUG[6797] chan_sip.c: Stopping retransmission on '457753420c013a8' of Response 825934346: Match Found [2009-10-20 02:56:28.7553] DEBUG[29097] chan_sip.c: Strict routing enforced for session 457753420c013a8 [2009-10-20 02:56:28.7554] VERBOSE[29097] chan_sip.c: set_destination: Parsing for address/port to send to [2009-10-20 02:56:28.7555] VERBOSE[29097] chan_sip.c: set_destination: set destination to 192.168.200.152, port 5060 [2009-10-20 02:56:28.7556] DEBUG[29097] chan_sip.c: T.38 UDPTL is at 192.168.135.103 port 4356 [2009-10-20 02:56:28.7556] DEBUG[29097] chan_sip.c: Initializing already initialized SIP dialog 457753420c013a8 (presumably reinvite) [2009-10-20 02:56:28.7557] VERBOSE[29097] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.200.152:5060: INVITE sip:192.168.200.152:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.135.103:5060;branch=z9hG4bK629ca788;rport Max-Forwards: 70 From: ;tag=as12ab5a38 To: ;tag=20c013a8+1+5b6c0004+71ff275f Contact: Call-ID: 457753420c013a8 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 339 v=0 o=root 1453031403 1453031404 IN IP4 192.168.135.103 s=Asterisk PBX 1.6.1.6 c=IN IP4 192.168.135.103 t=0 0 m=image 4356 udptl t38 a=T38Faxversion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPFEC --- [2009-10-20 02:56:28.7558] DEBUG[29097] chan_sip.c: Trying to put 'INVITE sip' onto UDP socket destined for 192.168.200.152:5060 [2009-10-20 02:56:28.7559] DEBUG[29097] app_fax.c: Negotiating T.38 for receive on SIP/provider-ada1c0f0 [2009-10-20 02:56:28.7765] VERBOSE[6797] chan_sip.c: <--- SIP read from UDP://192.168.200.152:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.135.103:5060;branch=z9hG4bK629ca788;rport=5060 From: ;tag=as12ab5a38 To: ;tag=20c013a8+1+5b6c0004+71ff275f Call-ID: 457753420c013a8 CSeq: 102 INVITE <-------------> [2009-10-20 02:56:28.7766] VERBOSE[6797] chan_sip.c: --- (6 headers 0 lines) --- [2009-10-20 02:56:28.7767] DEBUG[6797] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '457753420c013a8' Request 102: Found [2009-10-20 02:56:28.7800] VERBOSE[6797] chan_sip.c: <--- SIP read from UDP://192.168.200.152:5060 ---> SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 192.168.135.103:5060;branch=z9hG4bK629ca788;rport=5060 From: ;tag=as12ab5a38 To: ;tag=20c013a8+1+5b6c0004+71ff275f Call-ID: 457753420c013a8 CSeq: 102 INVITE Server: DC-SIP/2.0 Content-Length: 0 <-------------> [2009-10-20 02:56:28.7801] VERBOSE[6797] chan_sip.c: --- (8 headers 0 lines) --- [2009-10-20 02:56:28.7801] DEBUG[6797] chan_sip.c: Acked pending invite 102 [2009-10-20 02:56:28.7802] DEBUG[6797] chan_sip.c: Stopping retransmission on '457753420c013a8' of Request 102: Match Found [2009-10-20 02:56:28.7803] VERBOSE[6797] chan_sip.c: -- Got SIP response 415 "Unsupported Media Type" back from 192.168.200.152 [2009-10-20 02:56:28.7803] DEBUG[6797] chan_sip.c: Strict routing enforced for session 457753420c013a8 [2009-10-20 02:56:28.7804] VERBOSE[6797] chan_sip.c: set_destination: Parsing for address/port to send to [2009-10-20 02:56:28.7805] VERBOSE[6797] chan_sip.c: set_destination: set destination to 192.168.200.152, port 5060 [2009-10-20 02:56:28.7806] VERBOSE[6797] chan_sip.c: Transmitting (no NAT) to 192.168.200.152:5060: ACK sip:192.168.200.152:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.135.103:5060;branch=z9hG4bK629ca788;rport Max-Forwards: 70 From: ;tag=as12ab5a38 To: ;tag=20c013a8+1+5b6c0004+71ff275f Contact: Call-ID: 457753420c013a8 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.6 Content-Length: 0 --- [2009-10-20 02:56:28.7806] DEBUG[6797] chan_sip.c: Trying to put 'ACK sip:20' onto UDP socket destined for 192.168.200.152:5060 [2009-10-20 02:56:28.7826] WARNING[29097] app_fax.c: Transmission error [2009-10-20 02:56:28.7826] DEBUG[29097] pbx.c: Spawn extension (fax_receive,21906,9) exited non-zero on 'SIP/provider-ada1c0f0' [2009-10-20 02:56:28.7827] VERBOSE[29097] pbx.c: == Spawn extension (fax_receive, 21906, 9) exited non-zero on 'SIP/provider-ada1c0f0' [2009-10-20 02:56:28.7828] DEBUG[29097] channel.c: Soft-Hanging up channel 'SIP/provider-ada1c0f0'