=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.11.20 14:18:54 =~=~=~=~=~=~=~=~=~=~=~= [Nov 20 14:21:39] DEBUG[30653]: acl.c:490 ast_ouraddrfor: Found IP address for this socket asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:3295 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 10.9.1.121:5060 asterisk*CLI> == Using SIP RTP CoS mark 5 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:4626 do_setnat: Setting NAT on RTP to Off asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:6818 sip_alloc: Allocating new SIP dialog for M2YxYzY3YmFiZTc3MjYwOTZiYTZkNTQ3OGNkMjU5YjQ. - INVITE (With RTP) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:20568 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:4626 do_setnat: Setting NAT on RTP to Off asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:3174 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.9.5.63:63152 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:20568 handle_incoming: **** Received ACK (6) - Command in SIP ACK asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:3709 __sip_ack: Stopping retransmission on 'M2YxYzY3YmFiZTc3MjYwOTZiYTZkNTQ3OGNkMjU5YjQ.' of Response 1: Match Found asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:20568 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:4626 do_setnat: Setting NAT on RTP to Off asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:7548 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:7548 process_sdp: Processing session-level SDP o=- 3 2 IN IP4 10.9.5.63... UNSUPPORTED. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:7548 process_sdp: Processing session-level SDP s=CounterPath eyeBeam 1.5... UNSUPPORTED. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:7548 process_sdp: Processing session-level SDP c=IN IP4 10.9.5.63... OK. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:7548 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:7712 process_sdp: Processing media-level (audio) SDP a=alt:1 1 : RZbLgXTa p3G3F0wD 10.9.5.63 61036... UNSUPPORTED. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:7712 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=yes... UNSUPPORTED. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:7712 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:7712 process_sdp: Processing media-level (audio) SDP a=rtpmap:107 BV32/16000... UNSUPPORTED. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:7712 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:7712 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:7891 process_sdp: We're settling with these formats: 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:19006 handle_request_invite: Checking SIP call limits for device 38678 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:5222 update_call_counter: Updating call counter for incoming call asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:6218 sip_new: *** Our native formats are 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:6219 sip_new: *** Joint capabilities are 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:6220 sip_new: *** Our capabilities are 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:6221 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:6251 sip_new: This channel will not be able to handle video. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:11640 build_route: build_route: Contact hop: asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:19236 handle_request_invite: SIP/38678-00000000: New call is still down.... Trying... asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:3174 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.9.5.63:63152 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: pbx.c:3200 pbx_extension_helper: Launching 'Answer' asterisk*CLI> -- Executing [3300@default:1] Answer("SIP/38678-00000000", "") in new stack asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:5772 sip_answer: SIP answering channel: SIP/38678-00000000 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9721 transmit_response_with_sdp: Setting framing from config on incoming call asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9390 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9391 add_sdp: ** Our prefcodec: 0x0 (nothing) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9523 add_sdp: -- Done with adding codecs to SDP asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3025 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=29) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9658 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:3174 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.9.5.63:63152 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 38678 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: chan_sip.c:21907 sip_devicestate: Checking device state for peer 38678 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:486 do_state_change: Changing state for SIP/38678 - state 1 (Not in use) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:466 devstate_event: device 'SIP/38678' state '1' asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 38678 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: chan_sip.c:21907 sip_devicestate: Checking device state for peer 38678 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:486 do_state_change: Changing state for SIP/38678 - state 1 (Not in use) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:466 devstate_event: device 'SIP/38678' state '1' asterisk*CLI> [Nov 20 14:21:39] DEBUG[30656]: app_queue.c:788 handle_statechange: Device 'SIP/38678' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30656]: app_queue.c:788 handle_statechange: Device 'SIP/38678' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 132 bytes asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: pbx.c:3200 pbx_extension_helper: Launching 'Dial' asterisk*CLI> -- Executing [3300@default:2] Dial("SIP/38678-00000000", "SIP/SIP_VM/3300,,TTr") in new stack asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:22001 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) asterisk*CLI> == Using SIP RTP CoS mark 5 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6818 sip_alloc: Allocating new SIP dialog for 6ca6d4c66c93f3bd21eea59378fefd2d@127.0.0.1 - INVITE (With RTP) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:4626 do_setnat: Setting NAT on RTP to Off asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: acl.c:490 ast_ouraddrfor: Found IP address for this socket asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:3295 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_TCP with address 10.9.1.121:5060 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: frame.c:1211 ast_codec_choose: Could not find preferred codec - Going for the best codec asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6218 sip_new: *** Our native formats are 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6219 sip_new: *** Joint capabilities are 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6220 sip_new: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: frame.c:1211 ast_codec_choose: Could not find preferred codec - Going for the best codec asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6221 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6223 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6251 sip_new: This channel will not be able to handle video. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable DIALEDTIME. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable DIALSTATUS. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable SIPCALLID. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable SIPURI. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:5030 sip_call: Outgoing Call for 3300 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:5222 update_call_counter: Updating call counter for outgoing call asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9377 add_sdp: This call needs video offers, but there's no video support enabled! asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9390 add_sdp: ** Our capability: 0x8000e (gsm|ulaw|alaw|h263) Video flag: False Text flag: False asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9391 add_sdp: ** Our prefcodec: 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9523 add_sdp: -- Done with adding codecs to SDP asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3025 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=33) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9658 add_sdp: Done building SDP. Settling with this capability: 0x8000e (gsm|ulaw|alaw|h263) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:2897 initialize_initreq: Initializing initreq for method INVITE - callid 101678025b30ffdd7450a9f03b4c41bd@10.9.1.121 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:3174 __sip_xmit: Trying to put 'INVITE sip:' onto TCP socket destined for 10.9.1.13:5060 asterisk*CLI> -- Called SIP_VM/3300 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3178 ast_indicate_data: Driver for channel 'SIP/38678-00000000' does not support indication 3, emulating it asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3655 set_format: Set channel SIP/38678-00000000 to write format slin asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:2377 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3025 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=29) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:2490 ast_read_generator_actions: Generator got voice, switching to phase locked mode asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:2377 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: rtp.c:3804 ast_rtp_write: Ooh, format changed from unknown to ulaw asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: rtp.c:3820 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30673]: chan_sip.c:16721 handle_response_invite: SIP response 100 to standard invite asterisk*CLI> -- Got SIP response 302 "Moved Temporarily" back from 10.9.1.13 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30673]: chan_sip.c:16621 parse_moved_contact: Found promiscuous redirection to 'SIP/3300::::TCP@10.9.1.13:5067' asterisk*CLI> -- Now forwarding SIP/38678-00000000 to 'SIP/3300::::TCP@10.9.1.13:5067' (thanks to SIP/SIP_VM-00000001) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:22001 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) asterisk*CLI> == Using SIP RTP CoS mark 5 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6818 sip_alloc: Allocating new SIP dialog for 408a843944ff350f33fbfe816138dc22@127.0.0.1 - INVITE (With RTP) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:4626 do_setnat: Setting NAT on RTP to Off asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: acl.c:490 ast_ouraddrfor: Found IP address for this socket asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:3295 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_TCP with address 10.9.1.121:5060 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: frame.c:1211 ast_codec_choose: Could not find preferred codec - Going for the best codec asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6218 sip_new: *** Our native formats are 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6219 sip_new: *** Joint capabilities are 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6220 sip_new: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: frame.c:1211 ast_codec_choose: Could not find preferred codec - Going for the best codec asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6221 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6223 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:6251 sip_new: This channel will not be able to handle video. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable DIALEDTIME. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable DIALSTATUS. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable SIPCALLID. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:4262 ast_channel_inherit_variables: Not copying variable SIPURI. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:5030 sip_call: Outgoing Call for 3300 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:5222 update_call_counter: Updating call counter for outgoing call asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9377 add_sdp: This call needs video offers, but there's no video support enabled! asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9390 add_sdp: ** Our capability: 0x8000e (gsm|ulaw|alaw|h263) Video flag: False Text flag: False asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9391 add_sdp: ** Our prefcodec: 0x4 (ulaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9523 add_sdp: -- Done with adding codecs to SDP asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3025 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=37) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:9658 add_sdp: Done building SDP. Settling with this capability: 0x8000e (gsm|ulaw|alaw|h263) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:2897 initialize_initreq: Initializing initreq for method INVITE - callid 0ff2ea170a538880219d54db673ffdf8@10.9.1.121 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:3174 __sip_xmit: Trying to put 'INVITE sip:' onto TCP socket destined for 10.9.1.13:5067 [Nov 20 14:21:39] DEBUG[30673]: chan_sip.c:3174 __sip_xmit: Trying to put 'ACK sip:330' onto TCP socket destined for 10.9.1.13:5060 [Nov 20 14:21:39] DEBUG[30673]: chan_sip.c:2908 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 101678025b30ffdd7450a9f03b4c41bd@10.9.1.121 [Nov 20 14:21:39] DEBUG[30676]: channel.c:1711 ast_hangup: Hanging up channel 'SIP/SIP_VM-00000001' [Nov 20 14:21:39] DEBUG[30676]: chan_sip.c:5583 sip_hangup: Hangup call SIP/SIP_VM-00000001, SIP callid 101678025b30ffdd7450a9f03b4c41bd@10.9.1.121 [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - SIP_VM [Nov 20 14:21:39] DEBUG[30645]: chan_sip.c:21907 sip_devicestate: Checking device state for peer SIP_VM [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:486 do_state_change: Changing state for SIP/SIP_VM - state 1 (Not in use) [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:466 devstate_event: device 'SIP/SIP_VM' state '1' [Nov 20 14:21:39] DEBUG[30656]: app_queue.c:788 handle_statechange: Device 'SIP/SIP_VM' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:2581 _sip_tcp_helper_thread: Starting thread for TCP server [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:16721 handle_response_invite: SIP response 100 to standard invite asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:16721 handle_response_invite: SIP response 180 to standard invite asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 10.9.1.13:5067 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: chan_sip.c:21907 sip_devicestate: Checking device state for peer 10.9.1.13:5067 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:486 do_state_change: Changing state for SIP/10.9.1.13:5067 - state 6 (Ringing) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:466 devstate_event: device 'SIP/10.9.1.13:5067' state '6' asterisk*CLI> [Nov 20 14:21:39] DEBUG[30656]: app_queue.c:788 handle_statechange: Device 'SIP/10.9.1.13:5067' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. asterisk*CLI> -- SIP/10.9.1.13:5067-00000002 is ringing asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3178 ast_indicate_data: Driver for channel 'SIP/38678-00000000' does not support indication 3, emulating it asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3655 set_format: Set channel SIP/38678-00000000 to write format ulaw asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3655 set_format: Set channel SIP/38678-00000000 to write format slin asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:2377 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3025 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=29) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:2490 ast_read_generator_actions: Generator got voice, switching to phase locked mode asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:2377 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3025 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=29) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3025 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=29) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:20568 handle_incoming: **** Received ACK (6) - Command in SIP ACK asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:3709 __sip_ack: Stopping retransmission on 'M2YxYzY3YmFiZTc3MjYwOTZiYTZkNTQ3OGNkMjU5YjQ.' of Response 2: Match Found asterisk*CLI> [Nov 20 14:21:39] DEBUG[30653]: chan_sip.c:5369 sip_destroy: Destroying SIP dialog 101678025b30ffdd7450a9f03b4c41bd@10.9.1.121 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3025 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=29) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3025 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=29) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3025 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=29) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:16721 handle_response_invite: SIP response 200 to standard invite asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:7548 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:7548 process_sdp: Processing session-level SDP o=- 0 0 IN IP4 10.9.1.13... UNSUPPORTED. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:7548 process_sdp: Processing session-level SDP s=Microsoft Exchange Speech Engine... UNSUPPORTED. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:7548 process_sdp: Processing session-level SDP c=IN IP4 10.9.1.13... OK. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:7548 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:7712 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:7712 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:7891 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:7896 process_sdp: We have an owner, now see if we need to change this call asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:5222 update_call_counter: Updating call counter for outgoing call asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:11640 build_route: build_route: Contact hop: ;automata asterisk*CLI> [Nov 20 14:21:39] DEBUG[30677]: chan_sip.c:3174 __sip_xmit: Trying to put 'ACK sip:dia' onto TCP socket destined for 10.9.1.13:5067 asterisk*CLI> -- SIP/10.9.1.13:5067-00000002 answered SIP/38678-00000000 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:3655 set_format: Set channel SIP/38678-00000000 to write format ulaw asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: channel.c:2377 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: features.c:2503 ast_bridge_call: bridge answer set, chan answer set asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 10.9.1.13:5067 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: chan_sip.c:21907 sip_devicestate: Checking device state for peer 10.9.1.13:5067 asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:486 do_state_change: Changing state for SIP/10.9.1.13:5067 - state 2 (In use) asterisk*CLI> [Nov 20 14:21:39] DEBUG[30645]: devicestate.c:466 devstate_event: device 'SIP/10.9.1.13:5067' state '2' asterisk*CLI> [Nov 20 14:21:39] DEBUG[30656]: app_queue.c:788 handle_statechange: Device 'SIP/10.9.1.13:5067' changed to state '2' (In use) but we don't care because they're not a member of any queue. asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: rtp.c:3804 ast_rtp_write: Ooh, format changed from unknown to ulaw asterisk*CLI> [Nov 20 14:21:39] DEBUG[30676]: rtp.c:3820 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 asterisk*CLI> [Nov 20 14:21:41] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:41] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:42] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 200 bytes asterisk*CLI> [Nov 20 14:21:44] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:44] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:45] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:45] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:45] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:45] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:45] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:45] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:46] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:46] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:46] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 200 bytes asterisk*CLI> [Nov 20 14:21:46] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:46] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:46] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:46] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [Nov 20 14:21:47] DEBUG[30676]: rtp.c:886 send_dtmf: Sending dtmf: 56 (8), at 10.9.5.63 [Nov 20 14:21:47] DEBUG[30676]: channel.c:4854 ast_generic_bridge: Got DTMF begin on channel (SIP/38678-00000000) [Nov 20 14:21:47] DEBUG[30676]: channel.c:5227 ast_channel_bridge: Bridge stops bridging channels SIP/38678-00000000 and SIP/10.9.1.13:5067-00000002 [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [Nov 20 14:21:47] DEBUG[30676]: rtp.c:886 send_dtmf: Sending dtmf: 56 (8), at 10.9.5.63 [Nov 20 14:21:47] DEBUG[30676]: channel.c:4854 ast_generic_bridge: Got DTMF end on channel (SIP/38678-00000000) [Nov 20 14:21:47] DEBUG[30676]: channel.c:5227 ast_channel_bridge: Bridge stops bridging channels SIP/38678-00000000 and SIP/10.9.1.13:5067-00000002 [Nov 20 14:21:47] DEBUG[30676]: features.c:2046 ast_feature_interpret: Feature interpret: chan=SIP/38678-00000000, peer=SIP/10.9.1.13:5067-00000002, code=8, sense=1, features=2, dynamic=# asterisk*CLI> [Nov 20 14:21:47] DEBUG[30678]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30678]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 128 bytes asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) [Nov 20 14:21:47] DEBUG[30676]: rtp.c:886 send_dtmf: Sending dtmf: 53 (5), at 10.9.5.63 [Nov 20 14:21:47] DEBUG[30676]: channel.c:4854 ast_generic_bridge: Got DTMF begin on channel (SIP/38678-00000000) [Nov 20 14:21:47] DEBUG[30676]: channel.c:5227 ast_channel_bridge: Bridge stops bridging channels SIP/38678-00000000 and SIP/10.9.1.13:5067-00000002 [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) [Nov 20 14:21:47] DEBUG[30676]: rtp.c:886 send_dtmf: Sending dtmf: 53 (5), at 10.9.5.63 [Nov 20 14:21:47] DEBUG[30676]: channel.c:4854 ast_generic_bridge: Got DTMF end on channel (SIP/38678-00000000) [Nov 20 14:21:47] DEBUG[30676]: channel.c:5227 ast_channel_bridge: Bridge stops bridging channels SIP/38678-00000000 and SIP/10.9.1.13:5067-00000002 [Nov 20 14:21:47] DEBUG[30676]: features.c:2046 ast_feature_interpret: Feature interpret: chan=SIP/38678-00000000, peer=SIP/10.9.1.13:5067-00000002, code=5, sense=1, features=2, dynamic=# asterisk*CLI> [Nov 20 14:21:47] DEBUG[30678]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) [Nov 20 14:21:47] DEBUG[30678]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) asterisk*CLI> [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:47] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:48] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:48] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 80 bytes asterisk*CLI> [Nov 20 14:21:48] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 20 14:21:48] DEBUG[30676]: rtp.c:886 send_dtmf: Sending dtmf: 51 (3), at 10.9.5.63 [Nov 20 14:21:48] DEBUG[30676]: channel.c:4854 ast_generic_bridge: Got DTMF begin on channel (SIP/38678-00000000) [Nov 20 14:21:48] DEBUG[30676]: channel.c:5227 ast_channel_bridge: Bridge stops bridging channels SIP/38678-00000000 and SIP/10.9.1.13:5067-00000002 asterisk*CLI> [Nov 20 14:21:48] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 20 14:21:48] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:21:48] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:21:48] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:21:48] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:21:48] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:21:48] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:21:48] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 20 14:21:48] DEBUG[30676]: rtp.c:886 send_dtmf: Sending dtmf: 51 (3), at 10.9.5.63 [Nov 20 14:21:48] DEBUG[30676]: channel.c:4854 ast_generic_bridge: Got DTMF end on channel (SIP/38678-00000000) [Nov 20 14:21:48] DEBUG[30676]: channel.c:5227 ast_channel_bridge: Bridge stops bridging channels SIP/38678-00000000 and SIP/10.9.1.13:5067-00000002 [Nov 20 14:21:48] DEBUG[30676]: features.c:2046 ast_feature_interpret: Feature interpret: chan=SIP/38678-00000000, peer=SIP/10.9.1.13:5067-00000002, code=3, sense=1, features=2, dynamic=# asterisk*CLI> [Nov 20 14:21:48] DEBUG[30678]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 20 14:21:48] DEBUG[30678]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:21:48] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:48] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 80 bytes asterisk*CLI> [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 80 bytes asterisk*CLI> [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 20 14:21:49] DEBUG[30676]: rtp.c:886 send_dtmf: Sending dtmf: 50 (2), at 10.9.5.63 [Nov 20 14:21:49] DEBUG[30676]: channel.c:4854 ast_generic_bridge: Got DTMF begin on channel (SIP/38678-00000000) [Nov 20 14:21:49] DEBUG[30676]: channel.c:5227 ast_channel_bridge: Bridge stops bridging channels SIP/38678-00000000 and SIP/10.9.1.13:5067-00000002 [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) asterisk*CLI> [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) asterisk*CLI> [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) asterisk*CLI> [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) asterisk*CLI> [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 20 14:21:49] DEBUG[30676]: rtp.c:886 send_dtmf: Sending dtmf: 50 (2), at 10.9.5.63 [Nov 20 14:21:49] DEBUG[30676]: channel.c:4854 ast_generic_bridge: Got DTMF end on channel (SIP/38678-00000000) [Nov 20 14:21:49] DEBUG[30676]: channel.c:5227 ast_channel_bridge: Bridge stops bridging channels SIP/38678-00000000 and SIP/10.9.1.13:5067-00000002 [Nov 20 14:21:49] DEBUG[30676]: features.c:2046 ast_feature_interpret: Feature interpret: chan=SIP/38678-00000000, peer=SIP/10.9.1.13:5067-00000002, code=2, sense=1, features=2, dynamic=# asterisk*CLI> [Nov 20 14:21:49] DEBUG[30678]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 20 14:21:49] DEBUG[30678]: rtp.c:1038 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) asterisk*CLI> [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 80 bytes asterisk*CLI> [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 224 bytes asterisk*CLI> [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:49] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 80 bytes asterisk*CLI> [Nov 20 14:21:50] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:50] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 80 bytes asterisk*CLI> [Nov 20 14:21:50] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:50] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 80 bytes asterisk*CLI> [Nov 20 14:21:50] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:50] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 80 bytes asterisk*CLI> [Nov 20 14:21:51] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:51] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 80 bytes asterisk*CLI> [Nov 20 14:21:51] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:51] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:51] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:51] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:52] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:52] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:52] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:52] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:52] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:21:52] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:21:53] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 248 bytes asterisk*CLI> [Nov 20 14:21:53] DEBUG[30676]: rtp.c:1234 ast_rtcp_read: Got RTCP report of 160 bytes asterisk*CLI> [Nov 20 14:21:53] DEBUG[30653]: chan_sip.c:20568 handle_incoming: **** Received BYE (8) - Command in SIP BYE [Nov 20 14:21:53] DEBUG[30653]: chan_sip.c:2908 sip_alreadygone: Setting SIP_ALREADYGONE on dialog M2YxYzY3YmFiZTc3MjYwOTZiYTZkNTQ3OGNkMjU5YjQ. [Nov 20 14:21:53] DEBUG[30653]: chan_sip.c:20072 handle_request_bye: Received bye, issuing owner hangup [Nov 20 14:21:53] DEBUG[30653]: chan_sip.c:3174 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.9.5.63:63152 asterisk*CLI> [Nov 20 14:21:53] DEBUG[30676]: channel.c:4803 ast_generic_bridge: Didn't get a frame from channel: SIP/38678-00000000 [Nov 20 14:21:53] DEBUG[30676]: channel.c:5227 ast_channel_bridge: Bridge stops bridging channels SIP/38678-00000000 and SIP/10.9.1.13:5067-00000002 [Nov 20 14:21:53] DEBUG[30676]: channel.c:1711 ast_hangup: Hanging up channel 'SIP/10.9.1.13:5067-00000002' [Nov 20 14:21:53] DEBUG[30676]: chan_sip.c:5583 sip_hangup: Hangup call SIP/10.9.1.13:5067-00000002, SIP callid 0ff2ea170a538880219d54db673ffdf8@10.9.1.121 [Nov 20 14:21:53] DEBUG[30676]: chan_sip.c:3174 __sip_xmit: Trying to put 'BYE sip:dia' onto TCP socket destined for 10.9.1.13:5067 [Nov 20 14:21:53] DEBUG[30676]: rtp.c:2114 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Nov 20 14:21:53] DEBUG[30676]: app_dial.c:2113 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Nov 20 14:21:53] DEBUG[30676]: pbx.c:3806 __ast_pbx_run: Spawn extension (default,3300,2) exited non-zero on 'SIP/38678-00000000' == Spawn extension (default, 3300, 2) exited non-zero on 'SIP/38678-00000000' [Nov 20 14:21:53] DEBUG[30676]: channel.c:1606 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/38678-00000000' [Nov 20 14:21:53] DEBUG[30676]: channel.c:1711 ast_hangup: Hanging up channel 'SIP/38678-00000000' [Nov 20 14:21:53] DEBUG[30676]: chan_sip.c:5583 sip_hangup: Hangup call SIP/38678-00000000, SIP callid M2YxYzY3YmFiZTc3MjYwOTZiYTZkNTQ3OGNkMjU5YjQ. asterisk*CLI> [Nov 20 14:21:53] DEBUG[30645]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 10.9.1.13:5067 [Nov 20 14:21:53] DEBUG[30645]: chan_sip.c:21907 sip_devicestate: Checking device state for peer 10.9.1.13:5067 [Nov 20 14:21:53] DEBUG[30645]: devicestate.c:486 do_state_change: Changing state for SIP/10.9.1.13:5067 - state 0 (Unknown) [Nov 20 14:21:53] DEBUG[30645]: devicestate.c:466 devstate_event: device 'SIP/10.9.1.13:5067' state '0' [Nov 20 14:21:53] DEBUG[30645]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 38678 [Nov 20 14:21:53] DEBUG[30645]: chan_sip.c:21907 sip_devicestate: Checking device state for peer 38678 [Nov 20 14:21:53] DEBUG[30645]: devicestate.c:486 do_state_change: Changing state for SIP/38678 - state 1 (Not in use) [Nov 20 14:21:53] DEBUG[30645]: devicestate.c:466 devstate_event: device 'SIP/38678' state '1' asterisk*CLI> [Nov 20 14:21:53] DEBUG[30656]: app_queue.c:788 handle_statechange: Device 'SIP/10.9.1.13:5067' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. [Nov 20 14:21:53] DEBUG[30656]: app_queue.c:788 handle_statechange: Device 'SIP/38678' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. asterisk*CLI> [Nov 20 14:21:54] DEBUG[30653]: chan_sip.c:5369 sip_destroy: Destroying SIP dialog 0ff2ea170a538880219d54db673ffdf8@10.9.1.121 [Nov 20 14:21:54] DEBUG[30653]: chan_sip.c:5369 sip_destroy: Destroying SIP dialog M2YxYzY3YmFiZTc3MjYwOTZiYTZkNTQ3OGNkMjU5YjQ. asterisk*CLI>