=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.11.20 14:17:38 =~=~=~=~=~=~=~=~=~=~=~= [Nov 20 14:20:25] DEBUG[29852]: acl.c:490 ast_ouraddrfor: Found IP address for this socket asterisk*CLI> [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:3059 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 14:20:25] DEBUG[29852]: chan_sip.c:4390 do_setnat: Setting NAT on RTP to Off asterisk*CLI> [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:6570 sip_alloc: Allocating new SIP dialog for NzZlNzIyMWJmZDg3MDdjOGViNTgzOGYwODk3OWRjMDY. - INVITE (With RTP) asterisk*CLI> [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:20147 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE asterisk*CLI> [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:4390 do_setnat: Setting NAT on RTP to Off asterisk*CLI> [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:2927 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.9.5.63:63152 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:20147 handle_incoming: **** Received ACK (6) - Command in SIP ACK asterisk*CLI> [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:3473 __sip_ack: Stopping retransmission on 'NzZlNzIyMWJmZDg3MDdjOGViNTgzOGYwODk3OWRjMDY.' of Response 1: Match Found asterisk*CLI> [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:20147 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:4390 do_setnat: Setting NAT on RTP to Off [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:7872 process_sdp: We're settling with these formats: 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:18585 handle_request_invite: Checking SIP call limits for device 38678 [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:4975 update_call_counter: Updating call counter for incoming call [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:5987 sip_new: *** Our native formats are 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:5988 sip_new: *** Joint capabilities are 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:5989 sip_new: *** Our capabilities are 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:5990 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:6020 sip_new: This channel will not be able to handle video. [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:11252 build_route: build_route: Contact hop: [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:18815 handle_request_invite: SIP/38678-084f4b50: New call is still down.... Trying... [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:2927 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.9.5.63:63152 [Nov 20 14:20:25] DEBUG[29887]: pbx.c:3182 pbx_extension_helper: Launching 'Answer' -- Executing [3300@default:1] Answer("SIP/38678-084f4b50", "") in new stack [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:5540 sip_answer: SIP answering channel: SIP/38678-084f4b50 [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9344 transmit_response_with_sdp: Setting framing from config on incoming call [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9013 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9014 add_sdp: ** Our prefcodec: 0x0 (nothing) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9146 add_sdp: -- Done with adding codecs to SDP [Nov 20 14:20:25] DEBUG[29887]: channel.c:3028 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=27) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9281 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:2927 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.9.5.63:63152 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 38678 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29844]: chan_sip.c:21427 sip_devicestate: Checking device state for peer 38678 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:486 do_state_change: Changing state for SIP/38678 - state 1 (Not in use) asterisk*CLI> [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:466 devstate_event: device 'SIP/38678' state '1' asterisk*CLI> [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 38678 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29844]: chan_sip.c:21427 sip_devicestate: Checking device state for peer 38678 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:486 do_state_change: Changing state for SIP/38678 - state 1 (Not in use) asterisk*CLI> [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:466 devstate_event: device 'SIP/38678' state '1' asterisk*CLI> [Nov 20 14:20:25] DEBUG[29855]: app_queue.c:787 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 14:20:25] DEBUG[29855]: app_queue.c:787 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 14:20:25] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 132 bytes asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: pbx.c:3182 pbx_extension_helper: Launching 'Dial' -- Executing [3300@default:2] Dial("SIP/38678-084f4b50", "SIP/SIP_VM/3300,,TTr") in new stack [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:21521 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) == Using SIP RTP CoS mark 5 [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:6570 sip_alloc: Allocating new SIP dialog for 5feac3a36b2c217724e8c0132b61c231@127.0.0.1 - INVITE (With RTP) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:4390 do_setnat: Setting NAT on RTP to Off [Nov 20 14:20:25] DEBUG[29887]: acl.c:490 ast_ouraddrfor: Found IP address for this socket [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:3059 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_TCP with address 10.9.1.121:5060 [Nov 20 14:20:25] DEBUG[29887]: frame.c:1222 ast_codec_choose: Could not find preferred codec - Going for the best codec [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:5987 sip_new: *** Our native formats are 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:5988 sip_new: *** Joint capabilities are 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:5989 sip_new: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Nov 20 14:20:25] DEBUG[29887]: frame.c:1222 ast_codec_choose: Could not find preferred codec - Going for the best codec [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:5990 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:5992 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:6020 sip_new: This channel will not be able to handle video. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable DIALEDTIME. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable DIALSTATUS. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable SIPURI. [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:4783 sip_call: Outgoing Call for 3300 [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:4975 update_call_counter: Updating call counter for outgoing call [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9000 add_sdp: This call needs video offers, but there's no video support enabled! [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9013 add_sdp: ** Our capability: 0x8000e (gsm|ulaw|alaw|h263) Video flag: False Text flag: False [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9014 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9146 add_sdp: -- Done with adding codecs to SDP [Nov 20 14:20:25] DEBUG[29887]: channel.c:3028 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=31) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9281 add_sdp: Done building SDP. Settling with this capability: 0x8000e (gsm|ulaw|alaw|h263) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:2650 initialize_initreq: Initializing initreq for method INVITE - callid 1ffeb7fa1d5f66d65a43f24b34e34e75@10.9.1.121 [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:2927 __sip_xmit: Trying to put 'INVITE sip:' onto TCP socket destined for 10.9.1.13:5060 -- Called SIP_VM/3300 [Nov 20 14:20:25] DEBUG[29887]: channel.c:3181 ast_indicate_data: Driver for channel 'SIP/38678-084f4b50' does not support indication 3, emulating it [Nov 20 14:20:25] DEBUG[29887]: channel.c:3650 set_format: Set channel SIP/38678-084f4b50 to write format slin [Nov 20 14:20:25] DEBUG[29887]: channel.c:2377 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Nov 20 14:20:25] DEBUG[29887]: channel.c:3028 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=27) [Nov 20 14:20:25] DEBUG[29887]: channel.c:2490 ast_read_generator_actions: Generator got voice, switching to phase locked mode [Nov 20 14:20:25] DEBUG[29887]: channel.c:2377 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Nov 20 14:20:25] DEBUG[29887]: rtp.c:3791 ast_rtp_write: Ooh, format changed from unknown to ulaw [Nov 20 14:20:25] DEBUG[29887]: rtp.c:3807 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29871]: chan_sip.c:16296 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 14:20:25] DEBUG[29871]: chan_sip.c:16196 parse_moved_contact: Found promiscuous redirection to 'SIP/3300::::TCP@10.9.1.13:5067' asterisk*CLI> -- Now forwarding SIP/38678-084f4b50 to 'SIP/3300::::TCP@10.9.1.13:5067' (thanks to SIP/SIP_VM-084fb3b8) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:21521 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) == Using SIP RTP CoS mark 5 [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:6570 sip_alloc: Allocating new SIP dialog for 3258ded056c2b09b1df87dcb36d29413@127.0.0.1 - INVITE (With RTP) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:4390 do_setnat: Setting NAT on RTP to Off [Nov 20 14:20:25] DEBUG[29887]: acl.c:490 ast_ouraddrfor: Found IP address for this socket [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:3059 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_TCP with address 10.9.1.121:5060 [Nov 20 14:20:25] DEBUG[29887]: frame.c:1222 ast_codec_choose: Could not find preferred codec - Going for the best codec [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:5987 sip_new: *** Our native formats are 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:5988 sip_new: *** Joint capabilities are 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:5989 sip_new: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Nov 20 14:20:25] DEBUG[29887]: frame.c:1222 ast_codec_choose: Could not find preferred codec - Going for the best codec [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:5990 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:5992 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:6020 sip_new: This channel will not be able to handle video. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable DIALEDTIME. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable DIALSTATUS. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Nov 20 14:20:25] DEBUG[29887]: channel.c:4253 ast_channel_inherit_variables: Not copying variable SIPURI. [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:4783 sip_call: Outgoing Call for 3300 [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:4975 update_call_counter: Updating call counter for outgoing call [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9000 add_sdp: This call needs video offers, but there's no video support enabled! [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9013 add_sdp: ** Our capability: 0x8000e (gsm|ulaw|alaw|h263) Video flag: False Text flag: False asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9014 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9146 add_sdp: -- Done with adding codecs to SDP [Nov 20 14:20:25] DEBUG[29887]: channel.c:3028 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=35) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:9281 add_sdp: Done building SDP. Settling with this capability: 0x8000e (gsm|ulaw|alaw|h263) [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:2650 initialize_initreq: Initializing initreq for method INVITE - callid 1c0adba905eed9a0407a0a4a478c6eed@10.9.1.121 [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:2927 __sip_xmit: Trying to put 'INVITE sip:' onto TCP socket destined for 10.9.1.13:5067 [Nov 20 14:20:25] DEBUG[29888]: chan_sip.c:2394 _sip_tcp_helper_thread: Starting thread for TCP server asterisk*CLI> [Nov 20 14:20:25] DEBUG[29871]: chan_sip.c:2927 __sip_xmit: Trying to put 'ACK sip:330' onto TCP socket destined for 10.9.1.13:5060 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29871]: chan_sip.c:2945 __sip_xmit: Socket type is TCP but no tcptls_session is present to write to asterisk*CLI> [Nov 20 14:20:25] DEBUG[29871]: chan_sip.c:2661 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 1ffeb7fa1d5f66d65a43f24b34e34e75@10.9.1.121 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: channel.c:1711 ast_hangup: Hanging up channel 'SIP/SIP_VM-084fb3b8' [Nov 20 14:20:25] DEBUG[29887]: chan_sip.c:5336 sip_hangup: Hangup call SIP/SIP_VM-084fb3b8, SIP callid 1ffeb7fa1d5f66d65a43f24b34e34e75@10.9.1.121 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - SIP_VM [Nov 20 14:20:25] DEBUG[29844]: chan_sip.c:21427 sip_devicestate: Checking device state for peer SIP_VM [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:486 do_state_change: Changing state for SIP/SIP_VM - state 1 (Not in use) [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:466 devstate_event: device 'SIP/SIP_VM' state '1' asterisk*CLI> [Nov 20 14:20:25] DEBUG[29855]: app_queue.c:787 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. asterisk*CLI> [Nov 20 14:20:25] DEBUG[29888]: chan_sip.c:16296 handle_response_invite: SIP response 100 to standard invite asterisk*CLI> [Nov 20 14:20:25] DEBUG[29888]: chan_sip.c:16296 handle_response_invite: SIP response 180 to standard invite [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 10.9.1.13:5067 [Nov 20 14:20:25] DEBUG[29844]: chan_sip.c:21427 sip_devicestate: Checking device state for peer 10.9.1.13:5067 [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:486 do_state_change: Changing state for SIP/10.9.1.13:5067 - state 6 (Ringing) [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:466 devstate_event: device 'SIP/10.9.1.13:5067' state '6' [Nov 20 14:20:25] DEBUG[29855]: app_queue.c:787 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. -- SIP/10.9.1.13:5067-08507a78 is ringing [Nov 20 14:20:25] DEBUG[29887]: channel.c:3181 ast_indicate_data: Driver for channel 'SIP/38678-084f4b50' does not support indication 3, emulating it asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: channel.c:3650 set_format: Set channel SIP/38678-084f4b50 to write format ulaw [Nov 20 14:20:25] DEBUG[29887]: channel.c:3650 set_format: Set channel SIP/38678-084f4b50 to write format slin [Nov 20 14:20:25] DEBUG[29887]: channel.c:2377 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: channel.c:3028 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=27) asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: channel.c:2490 ast_read_generator_actions: Generator got voice, switching to phase locked mode asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: channel.c:2377 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: channel.c:3028 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=27) asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: channel.c:3028 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=27) asterisk*CLI> [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:20147 handle_incoming: **** Received ACK (6) - Command in SIP ACK [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:3473 __sip_ack: Stopping retransmission on 'NzZlNzIyMWJmZDg3MDdjOGViNTgzOGYwODk3OWRjMDY.' of Response 2: Match Found [Nov 20 14:20:25] DEBUG[29852]: chan_sip.c:5122 sip_destroy: Destroying SIP dialog 1ffeb7fa1d5f66d65a43f24b34e34e75@10.9.1.121 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: channel.c:3028 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=27) asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: channel.c:3028 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=27) asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: channel.c:3028 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=27) asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: channel.c:3028 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=27) asterisk*CLI> [Nov 20 14:20:25] DEBUG[29888]: chan_sip.c:16296 handle_response_invite: SIP response 200 to standard invite asterisk*CLI> [Nov 20 14:20:25] DEBUG[29888]: chan_sip.c:7872 process_sdp: We're settling with these formats: 0xc (ulaw|alaw) asterisk*CLI> [Nov 20 14:20:25] DEBUG[29888]: chan_sip.c:7877 process_sdp: We have an owner, now see if we need to change this call asterisk*CLI> [Nov 20 14:20:25] DEBUG[29888]: chan_sip.c:4975 update_call_counter: Updating call counter for outgoing call asterisk*CLI> [Nov 20 14:20:25] DEBUG[29888]: chan_sip.c:11252 build_route: build_route: Contact hop: ;automata asterisk*CLI> [Nov 20 14:20:25] DEBUG[29888]: chan_sip.c:2927 __sip_xmit: Trying to put 'ACK sip:dia' onto TCP socket destined for 10.9.1.13:5067 asterisk*CLI> -- SIP/10.9.1.13:5067-08507a78 answered SIP/38678-084f4b50 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: channel.c:3650 set_format: Set channel SIP/38678-084f4b50 to write format ulaw asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: channel.c:2377 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: features.c:2499 ast_bridge_call: bridge answer set, chan answer set asterisk*CLI> [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 10.9.1.13:5067 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29844]: chan_sip.c:21427 sip_devicestate: Checking device state for peer 10.9.1.13:5067 asterisk*CLI> [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:486 do_state_change: Changing state for SIP/10.9.1.13:5067 - state 2 (In use) asterisk*CLI> [Nov 20 14:20:25] DEBUG[29844]: devicestate.c:466 devstate_event: device 'SIP/10.9.1.13:5067' state '2' asterisk*CLI> [Nov 20 14:20:25] DEBUG[29855]: app_queue.c:787 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 14:20:25] DEBUG[29887]: rtp.c:3791 ast_rtp_write: Ooh, format changed from unknown to ulaw asterisk*CLI> [Nov 20 14:20:25] DEBUG[29887]: rtp.c:3807 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 asterisk*CLI> [Nov 20 14:20:27] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:27] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:28] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 176 bytes asterisk*CLI> [Nov 20 14:20:30] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:30] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:30] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:30] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 176 bytes asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [Nov 20 14:20:31] DEBUG[29887]: rtp.c:885 send_dtmf: Sending dtmf: 56 (8), at 10.9.5.63 [Nov 20 14:20:31] DEBUG[29887]: channel.c:4845 ast_generic_bridge: Got DTMF begin on channel (SIP/38678-084f4b50) [Nov 20 14:20:31] DEBUG[29887]: channel.c:5218 ast_channel_bridge: Bridge stops bridging channels SIP/38678-084f4b50 and SIP/10.9.1.13:5067-08507a78 [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [Nov 20 14:20:31] DEBUG[29887]: rtp.c:885 send_dtmf: Sending dtmf: 56 (8), at 10.9.5.63 [Nov 20 14:20:31] DEBUG[29887]: channel.c:4845 ast_generic_bridge: Got DTMF end on channel (SIP/38678-084f4b50) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: channel.c:5218 ast_channel_bridge: Bridge stops bridging channels SIP/38678-084f4b50 and SIP/10.9.1.13:5067-08507a78 [Nov 20 14:20:31] DEBUG[29887]: features.c:2046 ast_feature_interpret: Feature interpret: chan=SIP/38678-084f4b50, peer=SIP/10.9.1.13:5067-08507a78, code=8, sense=1, features=2, dynamic=# asterisk*CLI> [Nov 20 14:20:31] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) [Nov 20 14:20:31] DEBUG[29887]: rtp.c:885 send_dtmf: Sending dtmf: 53 (5), at 10.9.5.63 [Nov 20 14:20:31] DEBUG[29887]: channel.c:4845 ast_generic_bridge: Got DTMF begin on channel (SIP/38678-084f4b50) [Nov 20 14:20:31] DEBUG[29887]: channel.c:5218 ast_channel_bridge: Bridge stops bridging channels SIP/38678-084f4b50 and SIP/10.9.1.13:5067-08507a78 [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) [Nov 20 14:20:31] DEBUG[29887]: rtp.c:885 send_dtmf: Sending dtmf: 53 (5), at 10.9.5.63 [Nov 20 14:20:31] DEBUG[29887]: channel.c:4845 ast_generic_bridge: Got DTMF end on channel (SIP/38678-084f4b50) [Nov 20 14:20:31] DEBUG[29887]: channel.c:5218 ast_channel_bridge: Bridge stops bridging channels SIP/38678-084f4b50 and SIP/10.9.1.13:5067-08507a78 [Nov 20 14:20:31] DEBUG[29887]: features.c:2046 ast_feature_interpret: Feature interpret: chan=SIP/38678-084f4b50, peer=SIP/10.9.1.13:5067-08507a78, code=5, sense=1, features=2, dynamic=# asterisk*CLI> [Nov 20 14:20:31] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) [Nov 20 14:20:31] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4) asterisk*CLI> [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:31] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 20 14:20:32] DEBUG[29889]: rtp.c:885 send_dtmf: Sending dtmf: 51 (3), at 10.9.5.63 asterisk*CLI> [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 20 14:20:32] DEBUG[29887]: rtp.c:885 send_dtmf: Sending dtmf: 51 (3), at 10.9.5.63 [Nov 20 14:20:32] DEBUG[29887]: channel.c:4845 ast_generic_bridge: Got DTMF begin on channel (SIP/38678-084f4b50) [Nov 20 14:20:32] DEBUG[29887]: channel.c:5218 ast_channel_bridge: Bridge stops bridging channels SIP/38678-084f4b50 and SIP/10.9.1.13:5067-08507a78 [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000003 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:32] DEBUG[29887]: channel.c:4845 ast_generic_bridge: Got DTMF end on channel (SIP/38678-084f4b50) [Nov 20 14:20:32] DEBUG[29887]: channel.c:5218 ast_channel_bridge: Bridge stops bridging channels SIP/38678-084f4b50 and SIP/10.9.1.13:5067-08507a78 [Nov 20 14:20:32] DEBUG[29887]: features.c:2046 ast_feature_interpret: Feature interpret: chan=SIP/38678-084f4b50, peer=SIP/10.9.1.13:5067-08507a78, code=3, sense=1, features=2, dynamic=# asterisk*CLI> [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 20 14:20:32] DEBUG[29889]: rtp.c:885 send_dtmf: Sending dtmf: 50 (2), at 10.9.5.63 [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) [Nov 20 14:20:32] DEBUG[29889]: rtp.c:885 send_dtmf: Sending dtmf: 50 (2), at 10.9.5.63 [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29889]: rtp.c:1037 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4) asterisk*CLI> [Nov 20 14:20:32] DEBUG[29887]: channel.c:4845 ast_generic_bridge: Got DTMF begin on channel (SIP/38678-084f4b50) [Nov 20 14:20:32] DEBUG[29887]: channel.c:5218 ast_channel_bridge: Bridge stops bridging channels SIP/38678-084f4b50 and SIP/10.9.1.13:5067-08507a78 asterisk*CLI> [Nov 20 14:20:32] DEBUG[29887]: channel.c:4845 ast_generic_bridge: Got DTMF end on channel (SIP/38678-084f4b50) [Nov 20 14:20:32] DEBUG[29887]: channel.c:5218 ast_channel_bridge: Bridge stops bridging channels SIP/38678-084f4b50 and SIP/10.9.1.13:5067-08507a78 [Nov 20 14:20:32] DEBUG[29887]: features.c:2046 ast_feature_interpret: Feature interpret: chan=SIP/38678-084f4b50, peer=SIP/10.9.1.13:5067-08507a78, code=2, sense=1, features=2, dynamic=# asterisk*CLI> [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:32] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:33] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:33] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:33] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:33] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:33] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:33] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:34] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 176 bytes asterisk*CLI> [Nov 20 14:20:34] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:34] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:34] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:34] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:35] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 28 bytes [Nov 20 14:20:35] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 104 bytes asterisk*CLI> [Nov 20 14:20:35] DEBUG[29887]: rtp.c:1233 ast_rtcp_read: Got RTCP report of 160 bytes asterisk*CLI> [Nov 20 14:20:35] DEBUG[29852]: chan_sip.c:20147 handle_incoming: **** Received BYE (8) - Command in SIP BYE [Nov 20 14:20:35] DEBUG[29852]: chan_sip.c:2661 sip_alreadygone: Setting SIP_ALREADYGONE on dialog NzZlNzIyMWJmZDg3MDdjOGViNTgzOGYwODk3OWRjMDY. [Nov 20 14:20:35] DEBUG[29852]: chan_sip.c:19651 handle_request_bye: Received bye, issuing owner hangup [Nov 20 14:20:35] DEBUG[29852]: chan_sip.c:2927 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.9.5.63:63152 asterisk*CLI> [Nov 20 14:20:35] DEBUG[29887]: channel.c:4794 ast_generic_bridge: Didn't get a frame from channel: SIP/38678-084f4b50 [Nov 20 14:20:35] DEBUG[29887]: channel.c:5218 ast_channel_bridge: Bridge stops bridging channels SIP/38678-084f4b50 and SIP/10.9.1.13:5067-08507a78 [Nov 20 14:20:35] DEBUG[29887]: channel.c:1711 ast_hangup: Hanging up channel 'SIP/10.9.1.13:5067-08507a78' [Nov 20 14:20:35] DEBUG[29887]: chan_sip.c:5336 sip_hangup: Hangup call SIP/10.9.1.13:5067-08507a78, SIP callid 1c0adba905eed9a0407a0a4a478c6eed@10.9.1.121 [Nov 20 14:20:35] DEBUG[29887]: chan_sip.c:2927 __sip_xmit: Trying to put 'BYE sip:dia' onto TCP socket destined for 10.9.1.13:5067 [Nov 20 14:20:35] DEBUG[29887]: rtp.c:2108 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Nov 20 14:20:35] DEBUG[29887]: app_dial.c:2033 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Nov 20 14:20:35] DEBUG[29887]: pbx.c:3787 __ast_pbx_run: Spawn extension (default,3300,2) exited non-zero on 'SIP/38678-084f4b50' == Spawn extension (default, 3300, 2) exited non-zero on 'SIP/38678-084f4b50' [Nov 20 14:20:35] DEBUG[29887]: channel.c:1606 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/38678-084f4b50' [Nov 20 14:20:35] DEBUG[29887]: channel.c:1711 ast_hangup: Hanging up channel 'SIP/38678-084f4b50' [Nov 20 14:20:35] DEBUG[29887]: chan_sip.c:5336 sip_hangup: Hangup call SIP/38678-084f4b50, SIP callid NzZlNzIyMWJmZDg3MDdjOGViNTgzOGYwODk3OWRjMDY. asterisk*CLI> [Nov 20 14:20:35] DEBUG[29844]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 10.9.1.13:5067 [Nov 20 14:20:35] DEBUG[29844]: chan_sip.c:21427 sip_devicestate: Checking device state for peer 10.9.1.13:5067 [Nov 20 14:20:35] DEBUG[29844]: devicestate.c:486 do_state_change: Changing state for SIP/10.9.1.13:5067 - state 0 (Unknown) [Nov 20 14:20:35] DEBUG[29844]: devicestate.c:466 devstate_event: device 'SIP/10.9.1.13:5067' state '0' [Nov 20 14:20:35] DEBUG[29844]: devicestate.c:368 _ast_device_state: No provider found, checking channel drivers for SIP - 38678 [Nov 20 14:20:35] DEBUG[29844]: chan_sip.c:21427 sip_devicestate: Checking device state for peer 38678 [Nov 20 14:20:35] DEBUG[29844]: devicestate.c:486 do_state_change: Changing state for SIP/38678 - state 1 (Not in use) [Nov 20 14:20:35] DEBUG[29844]: devicestate.c:466 devstate_event: device 'SIP/38678' state '1' [Nov 20 14:20:35] DEBUG[29855]: app_queue.c:787 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 14:20:35] DEBUG[29855]: app_queue.c:787 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 14:20:36] DEBUG[29852]: chan_sip.c:5122 sip_destroy: Destroying SIP dialog NzZlNzIyMWJmZDg3MDdjOGViNTgzOGYwODk3OWRjMDY. [Nov 20 14:20:36] DEBUG[29852]: chan_sip.c:5122 sip_destroy: Destroying SIP dialog 1c0adba905eed9a0407a0a4a478c6eed@10.9.1.121 asterisk*CLI>