[Jan 12 11:37:06] DEBUG[5768] acl.c: For destination '192.168.6.149', our source address is '192.168.6.105'.
[Jan 12 11:37:06] DEBUG[5768] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.6.105:5060
[Jan 12 11:37:06] DEBUG[5768] chan_sip.c: Allocating new SIP dialog for 6d4dc7040b15da91367612bc056df48f@192.168.6.149:5060 - INVITE (No RTP)
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Jan 12 11:37:06] DEBUG[5768][C-00000003] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, timer"
[Jan 12 11:37:06] DEBUG[5768][C-00000003] sip/reqresp_parser.c: Found SIP option: -replaces-
[Jan 12 11:37:06] DEBUG[5768][C-00000003] sip/reqresp_parser.c: Matched SIP option: replaces
[Jan 12 11:37:06] DEBUG[5768][C-00000003] sip/reqresp_parser.c: Found SIP option: -timer-
[Jan 12 11:37:06] DEBUG[5768][C-00000003] sip/reqresp_parser.c: Matched SIP option: timer
[Jan 12 11:37:06] DEBUG[5768][C-00000003] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7ff15801afe8'
[Jan 12 11:37:06] DEBUG[5768][C-00000003] res_rtp_asterisk.c: Allocated port 35334 for RTP instance '0x7ff15801afe8'
[Jan 12 11:37:06] DEBUG[5768][C-00000003] res_rtp_asterisk.c: Creating ICE session 192.168.6.105:35334 (35334) for RTP instance '0x7ff15801afe8'
[Jan 12 11:37:06] DEBUG[5768][C-00000003] rtp_engine.c: RTP instance '0x7ff15801afe8' is setup and ready to go
[Jan 12 11:37:06] DEBUG[5768][C-00000003] acl.c: Attached to given IP address
[Jan 12 11:37:06] DEBUG[5768][C-00000003] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7ff15801afe8'
[Jan 12 11:37:06] VERBOSE[5768][C-00000003] netsock2.c: Using SIP RTP TOS bits 184
[Jan 12 11:37:06] VERBOSE[5768][C-00000003] netsock2.c: Using SIP RTP CoS mark 5
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Setting NAT on RTP to Off
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Processing session-level SDP o=root 1704477879 1704477879 IN IP4 192.168.6.149... OK.
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Processing session-level SDP s=Asterisk PBX 13.7.2... UNSUPPORTED OR FAILED.
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.6.149... OK.
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Jan 12 11:37:06] DEBUG[5768][C-00000003] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7ff0eeb1d260
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED.
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Jan 12 11:37:06] DEBUG[5768][C-00000003] rtp_engine.c: Crossover copying tx to rx payload mapping 8 (0x7ff158004fe0) from 0x7ff0eeb1d260 to 0x7ff0eeb1d260
[Jan 12 11:37:06] DEBUG[5768][C-00000003] acl.c: For destination '192.168.6.149', our source address is '192.168.6.105'.
[Jan 12 11:37:06] DEBUG[5768][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7ff15801afe8'
[Jan 12 11:37:06] VERBOSE[5768][C-00000003] res_rtp_asterisk.c: 0x7ff1580114e0 -- Strict RTP learning after remote address set to: 192.168.6.149:12574
[Jan 12 11:37:06] DEBUG[5768][C-00000003] rtp_engine.c: Copying rx payload mapping 8 (0x7ff158004fe0) from 0x7ff0eeb1d260 to 0x7ff15801b1b0
[Jan 12 11:37:06] DEBUG[5768][C-00000003] rtp_engine.c: Copying tx payload mapping 8 (0x7ff158004fe0) from 0x7ff0eeb1d260 to 0x7ff15801b1b0
[Jan 12 11:37:06] DEBUG[5768][C-00000003] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7ff15801afe8'
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: We're settling with these formats: (alaw)
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Checking SIP call limits for device XXXX
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Updating call counter for incoming call
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Incoming INVITE with 'timer' option supported
[Jan 12 11:37:06] DEBUG[5768][C-00000003] channel.c: Channel 0x7ff1580155f8 'SIP/XXXX-00000002' allocated
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: *** Our native formats are (alaw) 
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: *** Joint capabilities are (alaw) 
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: *** Our capabilities are (alaw) 
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw 
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: This channel will not be able to handle video.
[Jan 12 11:37:06] DEBUG[5768][C-00000003] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Jan 12 11:37:06] DEBUG[5768][C-00000003] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: SIP/XXXX-00000002: New call is still down.... Trying... 
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.6.149:5060
[Jan 12 11:37:06] DEBUG[5729] devicestate.c: No provider found, checking channel drivers for SIP - XXXX
[Jan 12 11:37:06] DEBUG[5729] chan_sip.c: Checking device state for peer XXXX
[Jan 12 11:37:06] DEBUG[5729] devicestate.c: Changing state for SIP/XXXX - state 1 (Not in use)
[Jan 12 11:37:06] DEBUG[6166][C-00000003] pbx.c: Launching 'Answer'
[Jan 12 11:37:06] VERBOSE[6166][C-00000003] pbx.c: Executing [302@incoming_call:1] Answer("SIP/XXXX-00000002", "") in new stack
[Jan 12 11:37:06] DEBUG[6166][C-00000003] chan_sip.c: SIP answering channel: SIP/XXXX-00000002
[Jan 12 11:37:06] DEBUG[5729] devicestate.c: No provider found, checking channel drivers for SIP - XXXX
[Jan 12 11:37:06] DEBUG[6166][C-00000003] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Jan 12 11:37:06] DEBUG[5729] chan_sip.c: Checking device state for peer XXXX
[Jan 12 11:37:06] DEBUG[5729] devicestate.c: Changing state for SIP/XXXX - state 1 (Not in use)
[Jan 12 11:37:06] DEBUG[6166][C-00000003] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True
[Jan 12 11:37:06] DEBUG[6166][C-00000003] chan_sip.c: ** Our prefcodec: (nothing) 
[Jan 12 11:37:06] DEBUG[6166][C-00000003] chan_sip.c: -- Done with adding codecs to SDP
[Jan 12 11:37:06] DEBUG[6166][C-00000003] chan_sip.c: Setting framing on incoming call: 0
[Jan 12 11:37:06] DEBUG[6166][C-00000003] chan_sip.c: Done building SDP. Settling with this capability: (alaw)
[Jan 12 11:37:06] DEBUG[6166][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.6.149:5060
[Jan 12 11:37:06] DEBUG[5768] chan_sip.c: Session timer started: 2 - 6d4dc7040b15da91367612bc056df48f@192.168.6.149:5060 60000ms
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Jan 12 11:37:06] DEBUG[5768][C-00000003] chan_sip.c: Stopping retransmission on '6d4dc7040b15da91367612bc056df48f@192.168.6.149:5060' of Response 102: Match Found
[Jan 12 11:37:06] VERBOSE[6166][C-00000003] res_rtp_asterisk.c: 0x7ff1580114e0 -- Strict RTP switching to RTP target address 192.168.6.149:12574 as source
[Jan 12 11:37:06] DEBUG[6166][C-00000003] pbx.c: Launching 'Ringing'
[Jan 12 11:37:06] VERBOSE[6166][C-00000003] pbx.c: Executing [302@incoming_call:2] Ringing("SIP/XXXX-00000002", "") in new stack
[Jan 12 11:37:06] DEBUG[6166][C-00000003] channel.c: Driver for channel 'SIP/XXXX-00000002' does not support indication 3, emulating it
[Jan 12 11:37:06] DEBUG[6166][C-00000003] channel.c: Channel SIP/XXXX-00000002 setting write format path: slin -> alaw
[Jan 12 11:37:06] DEBUG[6166][C-00000003] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Jan 12 11:37:06] DEBUG[6166][C-00000003] pbx.c: Launching 'Set'
[Jan 12 11:37:06] VERBOSE[6166][C-00000003] pbx.c: Executing [302@incoming_call:3] Set("SIP/XXXX-00000002", "DB(junk/mychan)=SIP/XXXX-00000002") in new stack
[Jan 12 11:37:06] DEBUG[6166][C-00000003] pbx.c: Launching 'BridgeWait'
[Jan 12 11:37:06] VERBOSE[6166][C-00000003] pbx.c: Executing [302@incoming_call:4] BridgeWait("SIP/XXXX-00000002", ",,e(n)") in new stack
[Jan 12 11:37:06] DEBUG[6166][C-00000003] bridge_roles.c: Set role 'holding_participant'
[Jan 12 11:37:06] DEBUG[6166][C-00000003] bridge.c: Bridge technology native_rtp does not have any capabilities we want.
[Jan 12 11:37:06] DEBUG[6166][C-00000003] bridge.c: Bridge technology softmix does not have any capabilities we want.
[Jan 12 11:37:06] DEBUG[6166][C-00000003] bridge.c: Bridge technology simple_bridge does not have any capabilities we want.
[Jan 12 11:37:06] DEBUG[6166][C-00000003] bridge.c: Chose bridge technology holding_bridge
[Jan 12 11:37:06] DEBUG[6166][C-00000003] bridge.c: Bridge 42bd3c5b-f4cd-45a0-9e97-c89cf895378d: calling holding_bridge technology constructor
[Jan 12 11:37:06] DEBUG[6166][C-00000003] bridge.c: Bridge 42bd3c5b-f4cd-45a0-9e97-c89cf895378d: calling holding_bridge technology start
[Jan 12 11:37:06] VERBOSE[6166][C-00000003] app_bridgewait.c: SIP/XXXX-00000002 is entering waiting bridge default:42bd3c5b-f4cd-45a0-9e97-c89cf895378d
[Jan 12 11:37:06] DEBUG[6166][C-00000003] bridge_channel.c: Bridge 42bd3c5b-f4cd-45a0-9e97-c89cf895378d: 0x7ff16c0021a8(SIP/XXXX-00000002) is joining
[Jan 12 11:37:06] DEBUG[6166][C-00000003] bridge_channel.c: Bridge 42bd3c5b-f4cd-45a0-9e97-c89cf895378d: pushing 0x7ff16c0021a8(SIP/XXXX-00000002)
[Jan 12 11:37:06] DEBUG[6166][C-00000003] bridge_roles.c: Set role 'holding_participant'
[Jan 12 11:37:06] VERBOSE[6166][C-00000003] bridge_channel.c: Channel SIP/XXXX-00000002 joined 'holding_bridge' base-bridge <42bd3c5b-f4cd-45a0-9e97-c89cf895378d>
[Jan 12 11:37:06] DEBUG[6166][C-00000003] bridge.c: Bridge 42bd3c5b-f4cd-45a0-9e97-c89cf895378d: 0x7ff16c0021a8(SIP/XXXX-00000002) is joining holding_bridge technology
[Jan 12 11:37:06] DEBUG[6166][C-00000003] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet
[Jan 12 11:37:06] DEBUG[6166][C-00000003] res_rtp_asterisk.c: Ooh, format changed from none to alaw
[Jan 12 11:37:07] VERBOSE[6166][C-00000003] res_rtp_asterisk.c: 0x7ff1580114e0 -- Strict RTP learning complete - Locking on source address 192.168.6.149:12574
[Jan 12 11:37:09] DEBUG[5768] chan_sip.c: Auto destroying SIP dialog '4b5381fe6e4dda69425fac3d25682632@192.168.6.149:5060'
[Jan 12 11:37:09] DEBUG[5768] chan_sip.c: Destroying SIP dialog 4b5381fe6e4dda69425fac3d25682632@192.168.6.149:5060
[Jan 12 11:37:09] DEBUG[5768] rtp_engine.c: Destroyed RTP instance '0x7ff1580058f8'
[Jan 12 11:37:12] DEBUG[5768] acl.c: For destination '192.168.6.149', our source address is '192.168.6.105'.
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.6.105:5060
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: Allocating new SIP dialog for 6dda749551b9eaa5757be8e4088cf65e@192.168.6.149:5060 - INVITE (No RTP)
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Jan 12 11:37:12] DEBUG[5768][C-00000004] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, timer"
[Jan 12 11:37:12] DEBUG[5768][C-00000004] sip/reqresp_parser.c: Found SIP option: -replaces-
[Jan 12 11:37:12] DEBUG[5768][C-00000004] sip/reqresp_parser.c: Matched SIP option: replaces
[Jan 12 11:37:12] DEBUG[5768][C-00000004] sip/reqresp_parser.c: Found SIP option: -timer-
[Jan 12 11:37:12] DEBUG[5768][C-00000004] sip/reqresp_parser.c: Matched SIP option: timer
[Jan 12 11:37:12] DEBUG[5768][C-00000004] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7ff158019c08'
[Jan 12 11:37:12] DEBUG[5768][C-00000004] res_rtp_asterisk.c: Allocated port 37788 for RTP instance '0x7ff158019c08'
[Jan 12 11:37:12] DEBUG[5768][C-00000004] res_rtp_asterisk.c: Creating ICE session 192.168.6.105:37788 (37788) for RTP instance '0x7ff158019c08'
[Jan 12 11:37:12] DEBUG[5768][C-00000004] rtp_engine.c: RTP instance '0x7ff158019c08' is setup and ready to go
[Jan 12 11:37:12] DEBUG[5768][C-00000004] acl.c: Attached to given IP address
[Jan 12 11:37:12] DEBUG[5768][C-00000004] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7ff158019c08'
[Jan 12 11:37:12] VERBOSE[5768][C-00000004] netsock2.c: Using SIP RTP TOS bits 184
[Jan 12 11:37:12] VERBOSE[5768][C-00000004] netsock2.c: Using SIP RTP CoS mark 5
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Setting NAT on RTP to Off
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing session-level SDP o=root 1229840277 1229840277 IN IP4 192.168.6.149... OK.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing session-level SDP s=Asterisk PBX 13.7.2... UNSUPPORTED OR FAILED.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.6.149... OK.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7ff0eeb1d260
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] rtp_engine.c: Crossover copying tx to rx payload mapping 8 (0x7ff158006350) from 0x7ff0eeb1d260 to 0x7ff0eeb1d260
[Jan 12 11:37:12] DEBUG[5768][C-00000004] acl.c: For destination '192.168.6.149', our source address is '192.168.6.105'.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7ff158019c08'
[Jan 12 11:37:12] VERBOSE[5768][C-00000004] res_rtp_asterisk.c: 0x7ff1580079a0 -- Strict RTP learning after remote address set to: 192.168.6.149:13048
[Jan 12 11:37:12] DEBUG[5768][C-00000004] rtp_engine.c: Copying rx payload mapping 8 (0x7ff158006350) from 0x7ff0eeb1d260 to 0x7ff158019dd0
[Jan 12 11:37:12] DEBUG[5768][C-00000004] rtp_engine.c: Copying tx payload mapping 8 (0x7ff158006350) from 0x7ff0eeb1d260 to 0x7ff158019dd0
[Jan 12 11:37:12] DEBUG[5768][C-00000004] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7ff158019c08'
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: We're settling with these formats: (alaw)
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Checking SIP call limits for device XXXX
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Updating call counter for incoming call
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Incoming INVITE with 'timer' option supported
[Jan 12 11:37:12] DEBUG[5768][C-00000004] channel.c: Channel 0x7ff158016a78 'SIP/XXXX-00000003' allocated
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: *** Our native formats are (alaw) 
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: *** Joint capabilities are (alaw) 
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: *** Our capabilities are (alaw) 
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw 
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: This channel will not be able to handle video.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Jan 12 11:37:12] DEBUG[5768][C-00000004] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: SIP/XXXX-00000003: New call is still down.... Trying... 
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.6.149:5060
[Jan 12 11:37:12] DEBUG[5729] devicestate.c: No provider found, checking channel drivers for SIP - XXXX
[Jan 12 11:37:12] DEBUG[5729] chan_sip.c: Checking device state for peer XXXX
[Jan 12 11:37:12] DEBUG[5729] devicestate.c: Changing state for SIP/XXXX - state 1 (Not in use)
[Jan 12 11:37:12] DEBUG[6172][C-00000004] pbx_variables.c: Function DB(junk/mychan) result is 'SIP/XXXX-00000002'
[Jan 12 11:37:12] DEBUG[6172][C-00000004] pbx.c: Launching 'Bridge'
[Jan 12 11:37:12] VERBOSE[6172][C-00000004] pbx.c: Executing [303@incoming_call:1] Bridge("SIP/XXXX-00000003", "SIP/XXXX-00000002") in new stack
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: SIP answering channel: SIP/XXXX-00000003
[Jan 12 11:37:12] DEBUG[5729] devicestate.c: No provider found, checking channel drivers for SIP - XXXX
[Jan 12 11:37:12] DEBUG[5729] chan_sip.c: Checking device state for peer XXXX
[Jan 12 11:37:12] DEBUG[6172][C-00000004] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Jan 12 11:37:12] DEBUG[5729] devicestate.c: Changing state for SIP/XXXX - state 1 (Not in use)
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: ** Our prefcodec: (nothing) 
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: -- Done with adding codecs to SDP
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: Setting framing on incoming call: 0
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: Done building SDP. Settling with this capability: (alaw)
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.6.149:5060
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: Session timer started: 9 - 6dda749551b9eaa5757be8e4088cf65e@192.168.6.149:5060 60000ms
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985' can not use native RTP bridge as two channels are required
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge.
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge technology holding_bridge does not have any capabilities we want.
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Chose bridge technology simple_bridge
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: calling simple_bridge technology constructor
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: calling simple_bridge technology start
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Moving 0x7ff16c0021a8(SIP/XXXX-00000002) into bridge 5e0aadfc-76d9-4abd-84b6-821d44894985
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_channel.c: Bridge 42bd3c5b-f4cd-45a0-9e97-c89cf895378d: pulling 0x7ff16c0021a8(SIP/XXXX-00000002)
[Jan 12 11:37:12] VERBOSE[6172][C-00000004] bridge_channel.c: Channel SIP/XXXX-00000002 left 'holding_bridge' base-bridge <42bd3c5b-f4cd-45a0-9e97-c89cf895378d>
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_channel.c: Bridge 42bd3c5b-f4cd-45a0-9e97-c89cf895378d: 0x7ff16c0021a8(SIP/XXXX-00000002) is leaving holding_bridge technology
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_channel.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: pushing 0x7ff16c0021a8(SIP/XXXX-00000002)
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_roles.c: Set role 'holding_participant'
[Jan 12 11:37:12] VERBOSE[6172][C-00000004] bridge_channel.c: Channel SIP/XXXX-00000002 joined 'simple_bridge' basic-bridge <5e0aadfc-76d9-4abd-84b6-821d44894985>
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985' can not use native RTP bridge as two channels are required
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge.
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge technology softmix does not have any capabilities we want.
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge technology holding_bridge does not have any capabilities we want.
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Chose bridge technology simple_bridge
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985 is already using the new technology.
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: 0x7ff16c0021a8(SIP/XXXX-00000002) is joining simple_bridge technology
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_channel.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: 0x7ff1740033f8(SIP/XXXX-00000003) is joining
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_channel.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: pushing 0x7ff1740033f8(SIP/XXXX-00000003)
[Jan 12 11:37:12] VERBOSE[6172][C-00000004] bridge_channel.c: Channel SIP/XXXX-00000003 joined 'simple_bridge' basic-bridge <5e0aadfc-76d9-4abd-84b6-821d44894985>
[Jan 12 11:37:12] DEBUG[5733] cdr.c: Finalized CDR for SIP/XXXX-00000003 - start 1515753432.065504 answer 1515753432.065824 end 1515753432.066318 dispo ANSWERED
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985'.  Checking compatability for channels 'SIP/XXXX-00000002' and 'SIP/XXXX-00000003'
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge technology softmix does not have any capabilities we want.
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge technology holding_bridge does not have any capabilities we want.
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping.
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Chose bridge technology native_rtp
[Jan 12 11:37:12] VERBOSE[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: switching from simple_bridge technology to native_rtp
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: calling native_rtp technology constructor
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: moving 0x7ff16c0021a8(SIP/XXXX-00000002) to dummy bridge temporarily
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: 0x7ff16c0021a8(SIP/XXXX-00000002) is leaving simple_bridge technology (dummy)
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: calling simple_bridge technology stop
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: 0x7ff1740033f8(SIP/XXXX-00000003) is joining native_rtp technology
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985'.  Channel 'SIP/XXXX-00000003' is joining bridge tech
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985'.  Attaching hook data 0x7ff174001b30 to 'SIP/XXXX-00000003'
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: 0x7ff16c0021a8(SIP/XXXX-00000002) is joining native_rtp technology
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985'.  Channel 'SIP/XXXX-00000002' is joining bridge tech
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985'.  Attaching hook data 0x7ff1740064f0 to 'SIP/XXXX-00000002'
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985'.  Tech starting 'SIP/XXXX-00000003' and 'SIP/XXXX-00000002' with target 'none'
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: Deferring reinvite on SIP '6dda749551b9eaa5757be8e4088cf65e@192.168.6.149:5060' - It's audio will be redirected to IP 192.168.6.149:12574
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: Sending reinvite on SIP '6d4dc7040b15da91367612bc056df48f@192.168.6.149:5060' - It's audio soon redirected to IP 192.168.6.149:13048
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: ** Our native-bridge filtered capablity: (alaw)
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: ** Our prefcodec: (nothing) 
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: -- Done with adding codecs to SDP
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: Setting framing on incoming call: 0
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: Done building SDP. Settling with this capability: (alaw)
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: Initializing already initialized SIP dialog 6d4dc7040b15da91367612bc056df48f@192.168.6.149:5060 (presumably reinvite)
[Jan 12 11:37:12] DEBUG[6172][C-00000004] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.6.149:5060
[Jan 12 11:37:12] VERBOSE[6172][C-00000004] bridge_native_rtp.c: Remotely bridged 'SIP/XXXX-00000003' and 'SIP/XXXX-00000002' - media will flow directly between them
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: calling native_rtp technology start
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985: calling simple_bridge technology destructor
[Jan 12 11:37:12] DEBUG[6172][C-00000004] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet
[Jan 12 11:37:12] DEBUG[6166][C-00000003] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985'.  Checking compatability for channels 'SIP/XXXX-00000003' and 'SIP/XXXX-00000002'
[Jan 12 11:37:12] DEBUG[6166][C-00000003] bridge.c: Bridge technology softmix does not have any capabilities we want.
[Jan 12 11:37:12] DEBUG[6166][C-00000003] bridge.c: Bridge technology holding_bridge does not have any capabilities we want.
[Jan 12 11:37:12] DEBUG[6166][C-00000003] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping.
[Jan 12 11:37:12] DEBUG[6166][C-00000003] bridge.c: Chose bridge technology native_rtp
[Jan 12 11:37:12] DEBUG[6166][C-00000003] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985 is already using the new technology.
[Jan 12 11:37:12] DEBUG[5718] threadpool.c: Increasing threadpool stasis-core's size by 1
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985'.  Checking compatability for channels 'SIP/XXXX-00000003' and 'SIP/XXXX-00000002'
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge technology softmix does not have any capabilities we want.
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge technology holding_bridge does not have any capabilities we want.
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping.
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Chose bridge technology native_rtp
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge.c: Bridge 5e0aadfc-76d9-4abd-84b6-821d44894985 is already using the new technology.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Stopping retransmission on '6dda749551b9eaa5757be8e4088cf65e@192.168.6.149:5060' of Response 102: Match Found
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: Sending pending reinvite on '6dda749551b9eaa5757be8e4088cf65e@192.168.6.149:5060'
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: ** Our native-bridge filtered capablity: (alaw)
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: ** Our capability: (alaw) Video flag: True Text flag: True
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: ** Our prefcodec: (nothing) 
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: -- Done with adding codecs to SDP
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: Setting framing on incoming call: 0
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: Done building SDP. Settling with this capability: (alaw)
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: Initializing already initialized SIP dialog 6dda749551b9eaa5757be8e4088cf65e@192.168.6.149:5060 (presumably reinvite)
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.6.149:5060
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6d4dc7040b15da91367612bc056df48f@192.168.6.149:5060' Request 102: Found
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: SIP response 100 to RE-invite on outgoing call 6d4dc7040b15da91367612bc056df48f@192.168.6.149:5060
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Acked pending invite 102
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Stopping retransmission on '6d4dc7040b15da91367612bc056df48f@192.168.6.149:5060' of Request 102: Match Found
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: SIP response 200 to RE-invite on outgoing call 6d4dc7040b15da91367612bc056df48f@192.168.6.149:5060
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Processing session-level SDP o=root 1704477879 1704477880 IN IP4 192.168.6.149... OK.
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Processing session-level SDP s=Asterisk PBX 13.7.2... UNSUPPORTED OR FAILED.
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.6.149... OK.
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Jan 12 11:37:12] DEBUG[5768][C-00000003] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7ff0eeb1c4c0
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED.
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Jan 12 11:37:12] DEBUG[5768][C-00000003] res_rtp_asterisk.c: Set role to CONTROLLING (0x7ff15801afe8)
[Jan 12 11:37:12] DEBUG[5768][C-00000003] res_rtp_asterisk.c: Set role failed; no ice instance (0x7ff15801afe8)
[Jan 12 11:37:12] DEBUG[5768][C-00000003] acl.c: For destination '192.168.6.149', our source address is '192.168.6.105'.
[Jan 12 11:37:12] DEBUG[5768][C-00000003] rtp_engine.c: Copying tx payload mapping 8 (0x7ff158015530) from 0x7ff0eeb1c4c0 to 0x7ff15801b1b0
[Jan 12 11:37:12] DEBUG[5768][C-00000003] acl.c: Attached to given IP address
[Jan 12 11:37:12] DEBUG[5768][C-00000003] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7ff15801afe8'
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: We're settling with these formats: (alaw)
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: We have an owner, now see if we need to change this call
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Updating call counter for incoming call
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Session-Expires: 120
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Refresher: UAC
[Jan 12 11:37:12] DEBUG[5768][C-00000003] chan_sip.c: Trying to put 'ACK sip:102' onto UDP socket destined for 192.168.6.149:5060
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: Session timer stopped: 2 - 6d4dc7040b15da91367612bc056df48f@192.168.6.149:5060
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: Session timer started: 30 - 6d4dc7040b15da91367612bc056df48f@192.168.6.149:5060 60000ms
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985'.  Tech starting 'SIP/XXXX-00000003' and 'SIP/XXXX-00000002' with target 'SIP/XXXX-00000003'
[Jan 12 11:37:12] DEBUG[6172][C-00000004] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985'.  Sending 'SIP/XXXX-00000003' back to remote
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '6dda749551b9eaa5757be8e4088cf65e@192.168.6.149:5060' Request 102: Found
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: SIP response 100 to RE-invite on outgoing call 6dda749551b9eaa5757be8e4088cf65e@192.168.6.149:5060
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Acked pending invite 102
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Stopping retransmission on '6dda749551b9eaa5757be8e4088cf65e@192.168.6.149:5060' of Request 102: Match Found
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: SIP response 200 to RE-invite on outgoing call 6dda749551b9eaa5757be8e4088cf65e@192.168.6.149:5060
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing session-level SDP o=root 1229840277 1229840278 IN IP4 192.168.6.149... OK.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing session-level SDP s=Asterisk PBX 13.7.2... UNSUPPORTED OR FAILED.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.6.149... OK.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7ff0eeb1c4c0
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing media-level (audio) SDP a=maxptime:150... UNSUPPORTED OR FAILED.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Jan 12 11:37:12] DEBUG[5768][C-00000004] res_rtp_asterisk.c: Set role to CONTROLLING (0x7ff158019c08)
[Jan 12 11:37:12] DEBUG[5768][C-00000004] res_rtp_asterisk.c: Set role failed; no ice instance (0x7ff158019c08)
[Jan 12 11:37:12] DEBUG[5768][C-00000004] acl.c: For destination '192.168.6.149', our source address is '192.168.6.105'.
[Jan 12 11:37:12] VERBOSE[5768][C-00000004] res_rtp_asterisk.c: 0x7ff1580079a0 -- Strict RTP learning after remote address set to: 192.168.6.149:13048
[Jan 12 11:37:12] DEBUG[5768][C-00000004] rtp_engine.c: Copying tx payload mapping 8 (0x7ff158001af0) from 0x7ff0eeb1c4c0 to 0x7ff158019dd0
[Jan 12 11:37:12] DEBUG[5768][C-00000004] acl.c: Attached to given IP address
[Jan 12 11:37:12] DEBUG[5768][C-00000004] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7ff158019c08'
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: We're settling with these formats: (alaw)
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: We have an owner, now see if we need to change this call
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Updating call counter for incoming call
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Session-Expires: 120
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Refresher: UAC
[Jan 12 11:37:12] DEBUG[5768][C-00000004] chan_sip.c: Trying to put 'ACK sip:101' onto UDP socket destined for 192.168.6.149:5060
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: Session timer stopped: 9 - 6dda749551b9eaa5757be8e4088cf65e@192.168.6.149:5060
[Jan 12 11:37:12] DEBUG[5768] chan_sip.c: Session timer started: 16 - 6dda749551b9eaa5757be8e4088cf65e@192.168.6.149:5060 60000ms
[Jan 12 11:37:12] DEBUG[6166][C-00000003] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985'.  Tech starting 'SIP/XXXX-00000003' and 'SIP/XXXX-00000002' with target 'SIP/XXXX-00000002'
[Jan 12 11:37:12] DEBUG[6166][C-00000003] bridge_native_rtp.c: Bridge '5e0aadfc-76d9-4abd-84b6-821d44894985'.  Sending 'SIP/XXXX-00000002' back to remote
[Jan 12 11:37:15] DEBUG[5772] res_pjsip_registrar_expire.c: Woke up at 1515753435  Interval: 30
[Jan 12 11:37:15] DEBUG[5772] res_pjsip_registrar_expire.c: Expiring 0 contacts