Asterisk SVN-branch-1.6.0-r207783, Copyright (C) 1999 - 2009 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk SVN-branch-1.6.0-r207783 currently running on proxy07 (pid = 27402) proxy07*CLI> score set debug 4 proxy07*CLI> Core debug was 0 and is now 4 proxy07*CLI> core set verbose 4 proxy07*CLI> Verbosity was 0 and is now 4 proxy07*CLI> sp sip set debug on proxy07*CLI> SIP Debugging enabled proxy07*CLI> proxy07*CLI> proxy07*CLI> <--- SIP read from UDP://74.13.233.144:5060 ---> INVITE sip:1905@209.167.0.151 SIP/2.0 Via: SIP/2.0/UDP 74.13.233.144:5060;branch=z9hG4bK7e8e9869;rport Max-Forwards: 70 From: "123456" ;tag=as3550d235 To: Contact: Call-ID: 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r207783 Date: Tue, 21 Jul 2009 20:19:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 490 v=0 o=root 699930702 699930702 IN IP4 74.13.233.144 s=Asterisk PBX SVN-branch-1.6.0-r207783 c=IN IP4 74.13.233.144 t=0 0 m=audio 17674 RTP/AVP 10 3 0 8 112 5 7 111 9 101 a=rtpmap:10 L16/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 0 [ 37]: INVITE sip:1905@209.167.0.151 SIP/2.0 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 1 [ 64]: Via: SIP/2.0/UDP 74.13.233.144:5060;branch=z9hG4bK7e8e9869;rport [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 2 [ 16]: Max-Forwards: 70 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 3 [ 56]: From: "123456" ;tag=as3550d235 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 4 [ 28]: To: [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 5 [ 35]: Contact: [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 6 [ 55]: Call-ID: 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 7 [ 16]: CSeq: 102 INVITE [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 8 [ 49]: User-Agent: Asterisk PBX SVN-branch-1.6.0-r207783 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 9 [ 35]: Date: Tue, 21 Jul 2009 20:19:59 GMT [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 11 [ 26]: Supported: replaces, timer [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 12 [ 29]: Content-Type: application/sdp [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 13 [ 19]: Content-Length: 490 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 14 [ 0]: [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 0 [ 3]: v=0 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 1 [ 47]: o=root 699930702 699930702 IN IP4 74.13.233.144 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 2 [ 39]: s=Asterisk PBX SVN-branch-1.6.0-r207783 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 3 [ 22]: c=IN IP4 74.13.233.144 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 4 [ 5]: t=0 0 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 5 [ 48]: m=audio 17674 RTP/AVP 10 3 0 8 112 5 7 111 9 101 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 6 [ 20]: a=rtpmap:10 L16/8000 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 7 [ 19]: a=rtpmap:3 GSM/8000 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 10 [ 30]: a=rtpmap:112 AAL2-G726-32/8000 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 11 [ 20]: a=rtpmap:5 DVI4/8000 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 12 [ 19]: a=rtpmap:7 LPC/8000 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 13 [ 25]: a=rtpmap:111 G726-32/8000 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 14 [ 20]: a=rtpmap:9 G722/8000 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 15 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 16 [ 15]: a=fmtp:101 0-16 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 17 [ 25]: a=silenceSupp:off - - - - [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 18 [ 10]: a=ptime:20 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Body 19 [ 10]: a=sendrecv --- (14 headers 20 lines) --- [Jul 21 16:01:55] DEBUG[27419]: acl.c:490 ast_ouraddrfor: Found IP address for this socket proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:2847 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 209.167.0.151:5060 proxy07*CLI> == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:4108 do_setnat: Setting NAT on RTP to Off [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:4116 do_setnat: Setting NAT on UDPTL to Off [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6351 sip_alloc: Allocating new SIP dialog for 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 - INVITE (With RTP) proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:19277 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:2566 parse_sip_options: Begin: parsing SIP "Supported: replaces, timer" [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:2574 parse_sip_options: Found SIP option: -replaces- [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:2580 parse_sip_options: Matched SIP option: replaces [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:2574 parse_sip_options: Found SIP option: -timer- proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:2580 parse_sip_options: Matched SIP option: timer Sending to 74.13.233.144 : 5060 (no NAT) [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:17716 handle_request_invite: Initializing initreq for method INVITE - callid 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 Using INVITE request as basis request - 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 proxy07*CLI> No user '123456' in SIP users list No matching peer for '123456' from '74.13.233.144:5060' proxy07*CLI> Found RTP audio format 10 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 112 Found RTP audio format 5 Found RTP audio format 7 Found RTP audio format 111 Found RTP audio format 9 Found RTP audio format 101 proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:7228 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 74.13.233.144:17674 Found audio description format L16 for ID 10 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format AAL2-G726-32 for ID 112 Found audio description format DVI4 for ID 5 Found audio description format LPC for ID 7 Found audio description format G726-32 for ID 111 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x18fe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g726aal2|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 74.13.233.144:17674 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:7591 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:17801 handle_request_invite: Checking SIP call limits for device [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:4729 update_call_counter: Updating call counter for incoming call Looking for 1905 in default (domain 209.167.0.151) proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: frame.c:1240 ast_codec_choose: Could not find preferred codec - Going for the best codec proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:5759 sip_new: *** Our native formats are 0x4 (ulaw) [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:5760 sip_new: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:5761 sip_new: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Jul 21 16:01:55] DEBUG[27419]: frame.c:1240 ast_codec_choose: Could not find preferred codec - Going for the best codec proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:5762 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:5790 sip_new: This channel will not be able to handle video. proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:10815 build_route: build_route: Contact hop: list_route: hop: proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:17874 handle_request_invite: Incoming INVITE with 'timer' option enabled [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:20115 start_session_timer: Session timer started: 1 - 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:18030 handle_request_invite: SIP/74.13.233.144-09baab78: New call is still down.... Trying... proxy07*CLI> <--- Transmitting (NAT) to 74.13.233.144:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 74.13.233.144:5060;branch=z9hG4bK7e8e9869;received=74.13.233.144;rport=5060 From: "123456" ;tag=as3550d235 To: Call-ID: 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r207783 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Length: 0 <------------> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:2716 __sip_xmit: Trying to put 'SIP/2.0 10' onto UDP socket destined for 74.13.233.144:5060 [Jul 21 16:01:55] DEBUG[27419]: devicestate.c:452 ast_devstate_changed_literal: Notification of state change to be queued on device/channel SIP/74.13.233.144 proxy07*CLI> [Jul 21 16:01:55] DEBUG[27465]: pbx.c:3101 pbx_extension_helper: Launching 'Answer' -- Executing [1905@default:1] Answer("SIP/74.13.233.144-09baab78", "") in new stack [Jul 21 16:01:55] DEBUG[27465]: devicestate.c:452 ast_devstate_changed_literal: Notification of state change to be queued on device/channel SIP/74.13.233.144 [Jul 21 16:01:55] DEBUG[27465]: chan_sip.c:5274 sip_answer: SIP answering channel: SIP/74.13.233.144-09baab78 [Jul 21 16:01:55] DEBUG[27465]: chan_sip.c:9050 transmit_response_with_sdp: Setting framing from config on incoming call [Jul 21 16:01:55] DEBUG[27465]: chan_sip.c:8719 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True Text flag: True [Jul 21 16:01:55] DEBUG[27465]: chan_sip.c:8720 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 209.167.0.151 port 17770 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Jul 21 16:01:55] DEBUG[27465]: chan_sip.c:8861 add_sdp: -- Done with adding codecs to SDP [Jul 21 16:01:55] DEBUG[27465]: chan_sip.c:8993 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) <--- Reliably Transmitting (NAT) to 74.13.233.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 74.13.233.144:5060;branch=z9hG4bK7e8e9869;received=74.13.233.144;rport=5060 From: "123456" ;tag=as3550d235 To: ;tag=as0019cd13 Call-ID: 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.0-r207783 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 329 v=0 o=root 1318238108 1318238108 IN IP4 209.167.0.151 s=Asterisk PBX SVN-branch-1.6.0-r207783 c=IN IP4 209.167.0.151 t=0 0 m=audio 17770 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jul 21 16:01:55] DEBUG[27465]: chan_sip.c:3069 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #2 [Jul 21 16:01:55] DEBUG[27465]: chan_sip.c:2716 __sip_xmit: Trying to put 'SIP/2.0 20' onto UDP socket destined for 74.13.233.144:5060 [Jul 21 16:01:55] DEBUG[27407]: devicestate.c:325 _ast_device_state: No provider found, checking channel drivers for SIP - 74.13.233.144 [Jul 21 16:01:55] DEBUG[27407]: chan_sip.c:20545 sip_devicestate: Checking device state for peer 74.13.233.144 [Jul 21 16:01:55] DEBUG[27407]: devicestate.c:443 do_state_change: Changing state for SIP/74.13.233.144 - state 2 (In use) [Jul 21 16:01:55] DEBUG[27407]: devicestate.c:325 _ast_device_state: No provider found, checking channel drivers for SIP - 74.13.233.144 [Jul 21 16:01:55] DEBUG[27407]: chan_sip.c:20545 sip_devicestate: Checking device state for peer 74.13.233.144 [Jul 21 16:01:55] DEBUG[27407]: devicestate.c:443 do_state_change: Changing state for SIP/74.13.233.144 - state 2 (In use) proxy07*CLI> [Jul 21 16:01:55] DEBUG[27415]: app_queue.c:766 handle_statechange: Device 'SIP/74.13.233.144' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jul 21 16:01:55] DEBUG[27415]: app_queue.c:766 handle_statechange: Device 'SIP/74.13.233.144' changed to state '2' (In use) but we don't care because they're not a member of any queue. proxy07*CLI> <--- SIP read from UDP://74.13.233.144:5060 ---> ACK sip:1905@209.167.0.151 SIP/2.0 Via: SIP/2.0/UDP 74.13.233.144:5060;branch=z9hG4bK05657e56;rport Max-Forwards: 70 From: "123456" ;tag=as3550d235 To: ;tag=as0019cd13 Contact: Call-ID: 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.6.0-r207783 Content-Length: 0 <-------------> proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 0 [ 34]: ACK sip:1905@209.167.0.151 SIP/2.0 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 1 [ 64]: Via: SIP/2.0/UDP 74.13.233.144:5060;branch=z9hG4bK05657e56;rport [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 2 [ 16]: Max-Forwards: 70 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 3 [ 56]: From: "123456" ;tag=as3550d235 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 4 [ 43]: To: ;tag=as0019cd13 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 5 [ 35]: Contact: [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 6 [ 55]: Call-ID: 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 7 [ 13]: CSeq: 102 ACK [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 8 [ 49]: User-Agent: Asterisk PBX SVN-branch-1.6.0-r207783 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 9 [ 17]: Content-Length: 0 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 10 [ 0]: --- (10 headers 0 lines) --- proxy07*CLI> [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:19277 handle_incoming: **** Received ACK (6) - Command in SIP ACK [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:3208 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #2 [Jul 21 16:01:55] DEBUG[27419]: chan_sip.c:3237 __sip_ack: Stopping retransmission on '1bc9a92843077da96d7d0eb866fd644f@74.13.233.144' of Response 102: Match Found proxy07*CLI> [Jul 21 16:01:55] DEBUG[27465]: channel.c:1823 __ast_answer: Didn't receive a media frame from SIP/74.13.233.144-09baab78 within 500 ms of answering. Continuing anyway [Jul 21 16:01:55] DEBUG[27465]: pbx.c:3101 pbx_extension_helper: Launching 'Wait' -- Executing [1905@default:2] Wait("SIP/74.13.233.144-09baab78", "3") in new stack proxy07*CLI> [Jul 21 16:01:58] DEBUG[27465]: pbx.c:3101 pbx_extension_helper: Launching 'Set' -- Executing [1905@default:3] Set("SIP/74.13.233.144-09baab78", "FAXFILE=test_rec.tif") in new stack proxy07*CLI> [Jul 21 16:01:58] DEBUG[27465]: pbx.c:3101 pbx_extension_helper: Launching 'ReceiveFAX' -- Executing [1905@default:4] ReceiveFAX("SIP/74.13.233.144-09baab78", "/home/sip/fax/test_rec.tif") in new stack proxy07*CLI> [Jul 21 16:01:58] DEBUG[27465]: channel.c:3612 set_format: Set channel SIP/74.13.233.144-09baab78 to read format slin [Jul 21 16:01:58] DEBUG[27465]: channel.c:3612 set_format: Set channel SIP/74.13.233.144-09baab78 to write format slin [Jul 21 16:01:58] DEBUG[27465]: app_fax.c:378 transmit_audio: Setting up CNG detection on SIP/74.13.233.144-09baab78 [Jul 21 16:01:58] DEBUG[27465]: dsp.c:408 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jul 21 16:01:58] DEBUG[27465]: dsp.c:408 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 proxy07*CLI> <--- SIP read from UDP://74.13.233.144:5060 ---> BYE sip:1905@209.167.0.151 SIP/2.0 Via: SIP/2.0/UDP 74.13.233.144:5060;branch=z9hG4bK35fcafee;rport Max-Forwards: 70 From: "123456" ;tag=as3550d235 To: ;tag=as0019cd13 Call-ID: 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 CSeq: 103 BYE User-Agent: Asterisk PBX SVN-branch-1.6.0-r207783 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 0 [ 34]: BYE sip:1905@209.167.0.151 SIP/2.0 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 1 [ 64]: Via: SIP/2.0/UDP 74.13.233.144:5060;branch=z9hG4bK35fcafee;rport proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 2 [ 16]: Max-Forwards: 70 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 3 [ 56]: From: "123456" ;tag=as3550d235 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 4 [ 43]: To: ;tag=as0019cd13 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 5 [ 55]: Call-ID: 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 6 [ 13]: CSeq: 103 BYE proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 7 [ 49]: User-Agent: Asterisk PBX SVN-branch-1.6.0-r207783 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 8 [ 39]: X-Asterisk-HangupCause: Normal Clearing proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 9 [ 30]: X-Asterisk-HangupCauseCode: 16 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 10 [ 17]: Content-Length: 0 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:6667 parse_request: Header 11 [ 0]: proxy07*CLI> --- (11 headers 0 lines) --- proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:19277 handle_incoming: **** Received BYE (8) - Command in SIP BYE proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:18742 handle_request_bye: Initializing initreq for method BYE - callid 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 proxy07*CLI> Sending to 74.13.233.144 : 5060 (NAT) proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:2464 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:20096 stop_session_timer: Session timer stopped: -1 - 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:18799 handle_request_bye: Received bye, issuing owner hangup proxy07*CLI> <--- Transmitting (NAT) to 74.13.233.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 74.13.233.144:5060;branch=z9hG4bK35fcafee;received=74.13.233.144;rport=5060 From: "123456" ;tag=as3550d235 To: ;tag=as0019cd13 Call-ID: 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 CSeq: 103 BYE User-Agent: Asterisk PBX SVN-branch-1.6.0-r207783 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 proxy07*CLI> <------------> proxy07*CLI> [Jul 21 16:06:56] DEBUG[27419]: chan_sip.c:2716 __sip_xmit: Trying to put 'SIP/2.0 20' onto UDP socket destined for 74.13.233.144:5060 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: app_fax.c:398 transmit_audio: Channel hangup [Jul 21 16:06:56] DEBUG[27465]: app_fax.c:486 transmit_audio: Loop finished, res=-1 [Jul 21 16:06:56] DEBUG[27465]: app_fax.c:166 phase_e_handler: Fax phase E handler. result=49 [Jul 21 16:06:56] WARNING[27465]: app_fax.c:178 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. [Jul 21 16:06:56] DEBUG[27465]: channel.c:3612 set_format: Set channel SIP/74.13.233.144-09baab78 to write format ulaw [Jul 21 16:06:56] DEBUG[27465]: channel.c:3612 set_format: Set channel SIP/74.13.233.144-09baab78 to read format ulaw [Jul 21 16:06:56] WARNING[27465]: app_fax.c:678 transmit: Transmission error [Jul 21 16:06:56] DEBUG[27465]: pbx.c:3701 __ast_pbx_run: Spawn extension (default,1905,4) exited non-zero on 'SIP/74.13.233.144-09baab78' == Spawn extension (default, 1905, 4) exited non-zero on 'SIP/74.13.233.144-09baab78' [Jul 21 16:06:56] DEBUG[27465]: channel.c:1610 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/74.13.233.144-09baab78' [Jul 21 16:06:56] DEBUG[27465]: channel.c:1703 ast_hangup: Hanging up channel 'SIP/74.13.233.144-09baab78' [Jul 21 16:06:56] DEBUG[27465]: chan_sip.c:5098 sip_hangup: Hangup call SIP/74.13.233.144-09baab78, SIP callid 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is '2009-07-21 16:01:55' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is '"123456" <123456>' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is 'default' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is 'SIP/74.13.233.144-09baab78' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is '' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is 'ReceiveFAX' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is '/home/sip/fax/test_rec.tif' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is '301' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is '301' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is '' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is '1248206515.0' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is '' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: pbx.c:2935 pbx_substitute_variables_helper_full: Function result is '' proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: cdr_sqlite3_custom.c:261 sqlite3_log: About to log: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,test) VALUES ('2009-07-21 16:01:55','"123456" <123456>','default','SIP/74.13.233.144-09baab78','','ReceiveFAX','/home/sip/fax/test_rec.tif','301','301','ANSWERED','DOCUMENTATION','','1248206515.0','','') proxy07*CLI> [Jul 21 16:06:56] DEBUG[27465]: devicestate.c:452 ast_devstate_changed_literal: Notification of state change to be queued on device/channel SIP/74.13.233.144 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27407]: devicestate.c:325 _ast_device_state: No provider found, checking channel drivers for SIP - 74.13.233.144 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27407]: chan_sip.c:20545 sip_devicestate: Checking device state for peer 74.13.233.144 proxy07*CLI> [Jul 21 16:06:56] DEBUG[27407]: devicestate.c:443 do_state_change: Changing state for SIP/74.13.233.144 - state 1 (Not in use) proxy07*CLI> [Jul 21 16:06:56] DEBUG[27415]: app_queue.c:766 handle_statechange: Device 'SIP/74.13.233.144' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. proxy07*CLI> [Jul 21 16:06:57] DEBUG[27419]: chan_sip.c:4886 sip_destroy: Destroying SIP dialog 1bc9a92843077da96d7d0eb866fd644f@74.13.233.144 Really destroying SIP dialog '1bc9a92843077da96d7d0eb866fd644f@74.13.233.144' Method: BYE proxy07*CLI>