Asterisk SVN-branch-1.6.1-r207784, 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-1.6.1-r207784 currently running on proxy07 (pid = 10528) proxy07*CLI> core stet debug 4 proxy07*CLI> Core debug was 0 and is now 4 proxy07*CLI> core set averbose 4 proxy07*CLI> Verbosity was 0 and is now 4 proxy07*CLI> sip set debug on proxy07*CLI> SIP Debugging enabled 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=z9hG4bK48c02858;rport Max-Forwards: 70 From: "123456" ;tag=as2184c44d To: Contact: Call-ID: 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.1-r207784 Date: Tue, 21 Jul 2009 20:08:17 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 550744479 550744479 IN IP4 74.13.233.144 s=Asterisk PBX SVN-branch-1.6.1-r207784 c=IN IP4 74.13.233.144 t=0 0 m=audio 11044 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 <-------------> proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 0 [ 37]: INVITE sip:1905@209.167.0.151 SIP/2.0 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 1 [ 64]: Via: SIP/2.0/UDP 74.13.233.144:5060;branch=z9hG4bK48c02858;rport proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 2 [ 16]: Max-Forwards: 70 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 3 [ 56]: From: "123456" ;tag=as2184c44d proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 4 [ 28]: To: proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 5 [ 35]: Contact: proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 6 [ 55]: Call-ID: 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 7 [ 16]: CSeq: 102 INVITE proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 8 [ 49]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r207784 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 9 [ 35]: Date: Tue, 21 Jul 2009 20:08:17 GMT proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 11 [ 26]: Supported: replaces, timer proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 12 [ 29]: Content-Type: application/sdp proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 13 [ 19]: Content-Length: 490 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 14 [ 0]: proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 0 [ 3]: v=0 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 1 [ 47]: o=root 550744479 550744479 IN IP4 74.13.233.144 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 2 [ 39]: s=Asterisk PBX SVN-branch-1.6.1-r207784 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 3 [ 22]: c=IN IP4 74.13.233.144 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 4 [ 5]: t=0 0 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 5 [ 48]: m=audio 11044 RTP/AVP 10 3 0 8 112 5 7 111 9 101 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 6 [ 20]: a=rtpmap:10 L16/8000 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 7 [ 19]: a=rtpmap:3 GSM/8000 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 8 [ 20]: a=rtpmap:0 PCMU/8000 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 9 [ 20]: a=rtpmap:8 PCMA/8000 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 10 [ 30]: a=rtpmap:112 AAL2-G726-32/8000 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 11 [ 20]: a=rtpmap:5 DVI4/8000 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 12 [ 19]: a=rtpmap:7 LPC/8000 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 13 [ 25]: a=rtpmap:111 G726-32/8000 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 14 [ 20]: a=rtpmap:9 G722/8000 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 15 [ 33]: a=rtpmap:101 telephone-event/8000 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 16 [ 15]: a=fmtp:101 0-16 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 17 [ 25]: a=silenceSupp:off - - - - proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 18 [ 10]: a=ptime:20 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 19 [ 10]: a=sendrecv proxy07*CLI> --- (14 headers 20 lines) --- proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: acl.c:490 ast_ouraddrfor: Found IP address for this socket proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:3049 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 209.167.0.151:5060 proxy07*CLI> == Using SIP RTP CoS mark 5 proxy07*CLI> == Using UDPTL CoS mark 5 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:4325 do_setnat: Setting NAT on RTP to Off proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:4333 do_setnat: Setting NAT on UDPTL to Off proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6602 sip_alloc: Allocating new SIP dialog for 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 - INVITE (With RTP) proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:20208 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:2768 parse_sip_options: Begin: parsing SIP "Supported: replaces, timer" proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:2776 parse_sip_options: Found SIP option: -replaces- proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:2782 parse_sip_options: Matched SIP option: replaces proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:2776 parse_sip_options: Found SIP option: -timer- proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:2782 parse_sip_options: Matched SIP option: timer proxy07*CLI> Sending to 74.13.233.144 : 5060 (no NAT) proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:18562 handle_request_invite: Initializing initreq for method INVITE - callid 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 proxy07*CLI> Using INVITE request as basis request - 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 proxy07*CLI> No matching peer for '123456' from '74.13.233.144:5060' proxy07*CLI> Found RTP audio format 10 proxy07*CLI> Found RTP audio format 3 proxy07*CLI> Found RTP audio format 0 proxy07*CLI> Found RTP audio format 8 proxy07*CLI> Found RTP audio format 112 proxy07*CLI> Found RTP audio format 5 proxy07*CLI> Found RTP audio format 7 proxy07*CLI> Found RTP audio format 111 proxy07*CLI> Found RTP audio format 9 proxy07*CLI> Found RTP audio format 101 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:7526 process_sdp: Peer doesn't provide T.38 UDPTL proxy07*CLI> Peer audio RTP is at port 74.13.233.144:11044 proxy07*CLI> Found audio description format L16 for ID 10 proxy07*CLI> Found audio description format GSM for ID 3 proxy07*CLI> Found audio description format PCMU for ID 0 proxy07*CLI> Found audio description format PCMA for ID 8 proxy07*CLI> Found audio description format AAL2-G726-32 for ID 112 proxy07*CLI> Found audio description format DVI4 for ID 5 proxy07*CLI> Found audio description format LPC for ID 7 proxy07*CLI> Found audio description format G726-32 for ID 111 proxy07*CLI> Found audio description format G722 for ID 9 proxy07*CLI> Found audio description format telephone-event for ID 101 proxy07*CLI> Got unsupported a:fmtp in SDP offer proxy07*CLI> 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) proxy07*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) proxy07*CLI> Peer audio RTP is at port 74.13.233.144:11044 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:7919 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:18647 handle_request_invite: Checking SIP call limits for device proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:4968 update_call_counter: Updating call counter for incoming call proxy07*CLI> Looking for 1905 in default (domain 209.167.0.151) proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: frame.c:1240 ast_codec_choose: Could not find preferred codec - Going for the best codec proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6020 sip_new: *** Our native formats are 0x4 (ulaw) proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6021 sip_new: *** Joint capabilities are 0xe (gsm|ulaw|alaw) proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6022 sip_new: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: frame.c:1240 ast_codec_choose: Could not find preferred codec - Going for the best codec proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6023 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6053 sip_new: This channel will not be able to handle video. proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:11306 build_route: build_route: Contact hop: proxy07*CLI> list_route: hop: proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:18720 handle_request_invite: Incoming INVITE with 'timer' option enabled proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:21060 start_session_timer: Session timer started: 1 - 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:18876 handle_request_invite: SIP/74.13.233.144-084215c8: 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=z9hG4bK48c02858;received=74.13.233.144;rport=5060 From: "123456" ;tag=as2184c44d To: Call-ID: 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 CSeq: 102 INVITE Server: Asterisk PBX SVN-branch-1.6.1-r207784 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Length: 0 <------------> proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:2918 __sip_xmit: Trying to put 'SIP/2.0 10' onto UDP socket destined for 74.13.233.144:5060 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10533]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 74.13.233.144 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10533]: chan_sip.c:21488 sip_devicestate: Checking device state for peer 74.13.233.144 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10533]: devicestate.c:486 do_state_change: Changing state for SIP/74.13.233.144 - state 2 (In use) proxy07*CLI> [Jul 21 15:50:14] DEBUG[10533]: devicestate.c:466 devstate_event: device 'SIP/74.13.233.144' state '2' proxy07*CLI> [Jul 21 15:50:14] DEBUG[10587]: pbx.c:3182 pbx_extension_helper: Launching 'Answer' proxy07*CLI> -- Executing [1905@default:1] Answer("SIP/74.13.233.144-084215c8", "") in new stack proxy07*CLI> [Jul 21 15:50:14] DEBUG[10533]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 74.13.233.144 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10533]: chan_sip.c:21488 sip_devicestate: Checking device state for peer 74.13.233.144 [Jul 21 15:50:14] DEBUG[10533]: channel.c:1231 channel_find_locked: Avoiding initial deadlock for channel '0x842ae38' [Jul 21 15:50:14] DEBUG[10555]: app_queue.c:787 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> [Jul 21 15:50:14] DEBUG[10587]: chan_sip.c:5531 sip_answer: SIP answering channel: SIP/74.13.233.144-084215c8 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10587]: chan_sip.c:9384 transmit_response_with_sdp: Setting framing from config on incoming call proxy07*CLI> [Jul 21 15:50:14] DEBUG[10587]: chan_sip.c:9046 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True Text flag: True proxy07*CLI> [Jul 21 15:50:14] DEBUG[10587]: chan_sip.c:9047 add_sdp: ** Our prefcodec: 0x0 (nothing) proxy07*CLI> Audio is at 209.167.0.151 port 12112 proxy07*CLI> Adding codec 0x2 (gsm) to SDP proxy07*CLI> Adding codec 0x4 (ulaw) to SDP proxy07*CLI> Adding codec 0x8 (alaw) to SDP proxy07*CLI> Adding non-codec 0x1 (telephone-event) to SDP proxy07*CLI> [Jul 21 15:50:14] DEBUG[10587]: chan_sip.c:9188 add_sdp: -- Done with adding codecs to SDP proxy07*CLI> [Jul 21 15:50:14] DEBUG[10587]: chan_sip.c:9320 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) proxy07*CLI> <--- 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=z9hG4bK48c02858;received=74.13.233.144;rport=5060 From: "123456" ;tag=as2184c44d To: ;tag=as5ef848b2 Call-ID: 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 CSeq: 102 INVITE Server: Asterisk PBX SVN-branch-1.6.1-r207784 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 1255310944 1255310944 IN IP4 209.167.0.151 s=Asterisk PBX SVN-branch-1.6.1-r207784 c=IN IP4 209.167.0.151 t=0 0 m=audio 12112 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 <------------> proxy07*CLI> [Jul 21 15:50:14] DEBUG[10587]: chan_sip.c:3279 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #2 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10587]: chan_sip.c:2918 __sip_xmit: Trying to put 'SIP/2.0 20' onto UDP socket destined for 74.13.233.144:5060 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10533]: devicestate.c:486 do_state_change: Changing state for SIP/74.13.233.144 - state 2 (In use) proxy07*CLI> [Jul 21 15:50:14] DEBUG[10533]: devicestate.c:466 devstate_event: device 'SIP/74.13.233.144' state '2' proxy07*CLI> [Jul 21 15:50:14] DEBUG[10555]: app_queue.c:787 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=z9hG4bK7ccf5447;rport Max-Forwards: 70 From: "123456" ;tag=as2184c44d To: ;tag=as5ef848b2 Contact: Call-ID: 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.6.1-r207784 Content-Length: 0 <-------------> proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 0 [ 34]: ACK sip:1905@209.167.0.151 SIP/2.0 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 1 [ 64]: Via: SIP/2.0/UDP 74.13.233.144:5060;branch=z9hG4bK7ccf5447;rport proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 2 [ 16]: Max-Forwards: 70 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 3 [ 56]: From: "123456" ;tag=as2184c44d proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 4 [ 43]: To: ;tag=as5ef848b2 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 5 [ 35]: Contact: proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 6 [ 55]: Call-ID: 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 7 [ 13]: CSeq: 102 ACK proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 8 [ 49]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r207784 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 9 [ 17]: Content-Length: 0 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 10 [ 0]: --- (10 headers 0 lines) --- proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:20208 handle_incoming: **** Received ACK (6) - Command in SIP ACK proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:3431 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #2 proxy07*CLI> [Jul 21 15:50:14] DEBUG[10551]: chan_sip.c:3463 __sip_ack: Stopping retransmission on '75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144' of Response 102: Match Found proxy07*CLI> [Jul 21 15:50:14] DEBUG[10587]: pbx.c:3182 pbx_extension_helper: Launching 'Wait' proxy07*CLI> -- Executing [1905@default:2] Wait("SIP/74.13.233.144-084215c8", "3") in new stack proxy07*CLI> [Jul 21 15:50:17] DEBUG[10587]: pbx.c:3182 pbx_extension_helper: Launching 'Set' -- Executing [1905@default:3] Set("SIP/74.13.233.144-084215c8", "FAXFILE=test_rec.tif") in new stack proxy07*CLI> [Jul 21 15:50:17] DEBUG[10587]: pbx.c:3182 pbx_extension_helper: Launching 'ReceiveFAX' -- Executing [1905@default:4] ReceiveFAX("SIP/74.13.233.144-084215c8", "/home/sip/fax/test_rec.tif") in new stack [Jul 21 15:50:17] DEBUG[10587]: channel.c:3606 set_format: Set channel SIP/74.13.233.144-084215c8 to read format slin proxy07*CLI> [Jul 21 15:50:17] DEBUG[10587]: channel.c:3606 set_format: Set channel SIP/74.13.233.144-084215c8 to write format slin [Jul 21 15:50:17] DEBUG[10587]: app_fax.c:378 transmit_audio: Setting up CNG detection on SIP/74.13.233.144-084215c8 [Jul 21 15:50:17] DEBUG[10587]: dsp.c:468 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jul 21 15:50:17] DEBUG[10587]: dsp.c:468 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jul 21 15:50:17] DEBUG[10587]: channel.c:2358 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second proxy07*CLI> [Jul 21 15:50:17] DEBUG[10587]: channel.c:2471 ast_read_generator_actions: Generator got voice, switching to phase locked mode [Jul 21 15:50:17] DEBUG[10587]: channel.c:2358 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 21 15:50:17] DEBUG[10587]: rtp.c:3788 ast_rtp_write: Ooh, format changed from unknown to ulaw [Jul 21 15:50:17] DEBUG[10587]: rtp.c:3804 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: dsp.c:584 tone_detect: 1100 Hz done detected [Jul 21 15:50:18] DEBUG[10587]: app_fax.c:421 transmit_audio: Fax tone detected. Requesting T38 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:4399 change_t38_state: T38 state changed to 1 on channel SIP/74.13.233.144-084215c8 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:8429 reqprep: Strict routing enforced for session 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 set_destination: Parsing for address/port to send to proxy07*CLI> set_destination: set destination to 74.13.233.144, port 5060 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:9218 add_sdp: T.38 UDPTL is at 209.167.0.151 port 4225 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:9227 add_sdp: Our T38 capability (3859), peer T38 capability (0), joint capability (3859) proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:8886 t38_get_rate: T38MaxBitRate 9600 found proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:9320 add_sdp: Done building SDP. Settling with this capability: 0x0 (nothing) proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:2653 initialize_initreq: Initializing already initialized SIP dialog 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 (presumably reinvite) [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 0 [ 39]: INVITE sip:123456@74.13.233.144 SIP/2.0 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 1 [ 64]: Via: SIP/2.0/UDP 209.167.0.151:5060;branch=z9hG4bK1ba2b95c;rport proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 2 [ 16]: Max-Forwards: 70 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 3 [ 45]: From: ;tag=as5ef848b2 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 4 [ 54]: To: "123456" ;tag=as2184c44d proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 5 [ 33]: Contact: [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 6 [ 55]: Call-ID: 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 7 [ 16]: CSeq: 102 INVITE proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 8 [ 49]: User-Agent: Asterisk PBX SVN-branch-1.6.1-r207784 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 9 [ 14]: Require: timer proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 10 [ 33]: Session-Expires: -1;refresher=uas [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 11 [ 10]: Min-SE: 90 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 12 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 13 [ 26]: Supported: replaces, timer [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 14 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 15 [ 29]: Content-Type: application/sdp proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 16 [ 19]: Content-Length: 353 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Header 17 [ 0]: [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 0 [ 3]: v=0 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 1 [ 49]: o=root 1255310944 1255310945 IN IP4 209.167.0.151 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 2 [ 39]: s=Asterisk PBX SVN-branch-1.6.1-r207784 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 3 [ 22]: c=IN IP4 209.167.0.151 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 4 [ 5]: t=0 0 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 5 [ 22]: m=image 4225 udptl t38 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 6 [ 17]: a=T38FaxVersion:0 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 7 [ 20]: a=T38MaxBitRate:9600 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 8 [ 22]: a=T38FaxFillBitRemoval [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 9 [ 22]: a=T38FaxTranscodingMMR [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 10 [ 37]: a=T38FaxRateManagement:transferredTCF proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 11 [ 21]: a=T38FaxMaxBuffer:400 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 12 [ 23]: a=T38FaxMaxDatagram:400 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:6955 parse_request: Body 13 [ 23]: a=T38FaxUdpEC:t38UDPFEC Reliably Transmitting (NAT) to 74.13.233.144:5060: INVITE sip:123456@74.13.233.144 SIP/2.0 Via: SIP/2.0/UDP 209.167.0.151:5060;branch=z9hG4bK1ba2b95c;rport Max-Forwards: 70 From: ;tag=as5ef848b2 To: "123456" ;tag=as2184c44d Contact: Call-ID: 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.6.1-r207784 Require: timer Session-Expires: -1;refresher=uas Min-SE: 90 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: 353 v=0 o=root 1255310944 1255310945 IN IP4 209.167.0.151 s=Asterisk PBX SVN-branch-1.6.1-r207784 c=IN IP4 209.167.0.151 t=0 0 m=image 4225 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPFEC --- proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:3279 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #4 [Jul 21 15:50:18] DEBUG[10587]: chan_sip.c:2918 __sip_xmit: Trying to put 'INVITE sip' onto UDP socket destined for 74.13.233.144:5060 proxy07*CLI> <--- SIP read from UDP://74.13.233.144:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.167.0.151:5060;branch=z9hG4bK1ba2b95c;received=209.167.0.151;rport=5060 From: ;tag=as5ef848b2 To: "123456" ;tag=as2184c44d Call-ID: 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 CSeq: 102 INVITE Server: Asterisk PBX SVN-branch-1.6.1-r207784 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 90;refresher=uas Contact: Content-Length: 0 <-------------> proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 1 [ 92]: Via: SIP/2.0/UDP 209.167.0.151:5060;branch=z9hG4bK1ba2b95c;received=209.167.0.151;rport=5060 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 2 [ 45]: From: ;tag=as5ef848b2 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 3 [ 54]: To: "123456" ;tag=as2184c44d proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 4 [ 55]: Call-ID: 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 5 [ 16]: CSeq: 102 INVITE proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 6 [ 45]: Server: Asterisk PBX SVN-branch-1.6.1-r207784 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 8 [ 26]: Supported: replaces, timer proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 9 [ 14]: Require: timer [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 10 [ 33]: Session-Expires: 90;refresher=uas [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 11 [ 35]: Contact: proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 12 [ 17]: Content-Length: 0 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 13 [ 0]: --- (13 headers 0 lines) --- proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:3497 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #4 - INVITE (got response) [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:3504 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144' Request 102: Found proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:16366 handle_response_invite: SIP response 100 to RE-invite on outgoing call 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 proxy07*CLI> <--- SIP read from UDP://74.13.233.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.167.0.151:5060;branch=z9hG4bK1ba2b95c;received=209.167.0.151;rport=5060 From: ;tag=as5ef848b2 To: "123456" ;tag=as2184c44d Call-ID: 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 CSeq: 102 INVITE Server: Asterisk PBX SVN-branch-1.6.1-r207784 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: 90;refresher=uas Contact: Content-Type: application/sdp Content-Length: 351 v=0 o=root 550744479 550744480 IN IP4 74.13.233.144 s=Asterisk PBX SVN-branch-1.6.1-r207784 c=IN IP4 74.13.233.144 t=0 0 m=image 4482 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPFEC <-------------> proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 0 [ 14]: SIP/2.0 200 OK [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 1 [ 92]: Via: SIP/2.0/UDP 209.167.0.151:5060;branch=z9hG4bK1ba2b95c;received=209.167.0.151;rport=5060 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 2 [ 45]: From: ;tag=as5ef848b2 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 3 [ 54]: To: "123456" ;tag=as2184c44d proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 4 [ 55]: Call-ID: 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 5 [ 16]: CSeq: 102 INVITE [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 6 [ 45]: Server: Asterisk PBX SVN-branch-1.6.1-r207784 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 7 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 8 [ 26]: Supported: replaces, timer [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 9 [ 14]: Require: timer [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 10 [ 33]: Session-Expires: 90;refresher=uas proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 11 [ 35]: Contact: [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 12 [ 29]: Content-Type: application/sdp [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 13 [ 19]: Content-Length: 351 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Header 14 [ 0]: [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 0 [ 3]: v=0 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 1 [ 47]: o=root 550744479 550744480 IN IP4 74.13.233.144 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 2 [ 39]: s=Asterisk PBX SVN-branch-1.6.1-r207784 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 3 [ 22]: c=IN IP4 74.13.233.144 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 4 [ 5]: t=0 0 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 5 [ 22]: m=image 4482 udptl t38 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 6 [ 17]: a=T38FaxVersion:0 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 7 [ 20]: a=T38MaxBitRate:9600 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 8 [ 22]: a=T38FaxFillBitRemoval proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 9 [ 22]: a=T38FaxTranscodingMMR [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 10 [ 37]: a=T38FaxRateManagement:transferredTCF proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 11 [ 21]: a=T38FaxMaxBuffer:400 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 12 [ 23]: a=T38FaxMaxDatagram:400 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:6955 parse_request: Body 13 [ 23]: a=T38FaxUdpEC:t38UDPFEC --- (14 headers 14 lines) --- proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:3426 __sip_ack: Acked pending invite 102 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:3463 __sip_ack: Stopping retransmission on '75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144' of Request 102: Match Found proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:16366 handle_response_invite: SIP response 200 to RE-invite on outgoing call 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 proxy07*CLI> Got T.38 offer in SDP in dialog 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:7522 process_sdp: Peer T.38 UDPTL is at port 74.13.233.144:4482 Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:7738 process_sdp: FaxVersion: 0 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:7715 process_sdp: T38MaxBitRate: 9600 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:7757 process_sdp: FillBitRemoval [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:7767 process_sdp: Transcoding MMR proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:7782 process_sdp: RateManagement: transferredTCF [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:7712 process_sdp: MaxBufferSize:400 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:7745 process_sdp: FaxMaxDatagram: 400 [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:7789 process_sdp: UDP EC: t38UDPFEC [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:7812 process_sdp: Our T38 capability = (3859), peer T38 capability (3859), joint T38 capability (3859) proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:4399 change_t38_state: T38 state changed to 3 on channel SIP/74.13.233.144-084215c8 proxy07*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), 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) [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:7860 process_sdp: Have T.38 but no audio codecs, accepting offer anyway proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:4968 update_call_counter: Updating call counter for incoming call proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:8429 reqprep: Strict routing enforced for session 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 set_destination: Parsing for address/port to send to set_destination: set destination to 74.13.233.144, port 5060 Transmitting (NAT) to 74.13.233.144:5060: ACK sip:123456@74.13.233.144 SIP/2.0 Via: SIP/2.0/UDP 209.167.0.151:5060;branch=z9hG4bK1d1b2119;rport Max-Forwards: 70 From: ;tag=as5ef848b2 To: "123456" ;tag=as2184c44d Contact: Call-ID: 75d0e3f762b31d2e2f7b4e3a6d7f4712@74.13.233.144 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.6.1-r207784 Content-Length: 0 --- proxy07*CLI> [Jul 21 15:50:18] DEBUG[10551]: chan_sip.c:2918 __sip_xmit: Trying to put 'ACK sip:12' onto UDP socket destined for 74.13.233.144:5060 proxy07*CLI> [Jul 21 15:50:18] DEBUG[10587]: app_fax.c:451 transmit_audio: T38 negotiated, finishing audio loop [Jul 21 15:50:18] DEBUG[10587]: app_fax.c:486 transmit_audio: Loop finished, res=1 [Jul 21 15:50:18] DEBUG[10587]: channel.c:2358 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 21 15:50:18] DEBUG[10587]: channel.c:3606 set_format: Set channel SIP/74.13.233.144-084215c8 to write format ulaw [Jul 21 15:50:18] DEBUG[10587]: channel.c:3606 set_format: Set channel SIP/74.13.233.144-084215c8 to read format ulaw proxy07*CLI>