[Nov 27 08:14:24] DEBUG[8867] chan_sip.c: = Looking for  Call ID: 7194eaa250baa46b394a213d5fd00b81@sip.dev.avecezar.org (Checking To) --From tag as1b0cba04 --To-tag as7dcefc74  
[Nov 27 08:14:24] DEBUG[8867] chan_sip.c: Stopping retransmission on '7194eaa250baa46b394a213d5fd00b81@sip.dev.avecezar.org' of Request 103: Match Found
[Nov 27 08:14:24] DEBUG[8867] chan_sip.c: Destroying SIP dialog 7194eaa250baa46b394a213d5fd00b81@sip.dev.avecezar.org
[Nov 27 08:14:24] DEBUG[8867] rtp_engine.c: Destroyed RTP instance '0x9f191e0'
[Nov 27 08:14:45] DEBUG[8867] chan_sip.c: Auto destroying SIP dialog '4c294681-d2ef5e0f@192.168.2.24'
[Nov 27 08:14:45] DEBUG[8867] chan_sip.c: Destroying SIP dialog 4c294681-d2ef5e0f@192.168.2.24
[Nov 27 08:14:45] DEBUG[8867] rtp_engine.c: Destroyed RTP instance '0x9f056d0'
[Nov 27 08:14:56] DEBUG[8867] chan_sip.c: Auto destroying SIP dialog '12dc3d15-6f1cc026@192.168.2.24'
[Nov 27 08:14:56] DEBUG[8867] chan_sip.c: Destroying SIP dialog 12dc3d15-6f1cc026@192.168.2.24
[Nov 27 08:14:56] DEBUG[8867] rtp_engine.c: Destroyed RTP instance '0xb44946d8'
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: = Looking for  Call ID: f1ae6dc1-723aa231@192.168.2.24 (Checking From) --From tag 810e78819efc1671o1 --To-tag   
[Nov 27 08:16:50] DEBUG[8867] acl.c: For destination '10.0.6.23', our source address is '10.0.6.23'.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Target address 10.0.6.23:5060 is not local, substituting externaddr
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.6.23:5061
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Allocating new SIP dialog for f1ae6dc1-723aa231@192.168.2.24 - INVITE (No RTP)
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: Splitting '10.0.6.23:5060' into...
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: ...host '10.0.6.23' and port '5060'.
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: Splitting 'sip.dev.avecezar.org' into...
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: ...host 'sip.dev.avecezar.org' and port ''.
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x9f05988'
[Nov 27 08:16:50] DEBUG[8867] res_rtp_asterisk.c: Allocated port 17430 for RTP instance '0x9f05988'
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: RTP instance '0x9f05988' is setup and ready to go
[Nov 27 08:16:50] DEBUG[8867] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x9f05988'
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting NAT on RTP to On
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing session-level SDP o=- 8088191 8088191 IN IP4 192.168.2.24... UNSUPPORTED.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED.
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: Splitting '192.168.2.24' into...
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: ...host '192.168.2.24' and port ''.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.2.24... OK.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Setting payload 8 based on m type on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Setting payload 0 based on m type on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Setting payload 4 based on m type on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Setting payload 18 based on m type on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Setting payload 97 based on m type on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Setting payload 98 based on m type on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Setting payload 101 based on m type on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:4 G723/8000... OK.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729a/8000... OK.
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Unsetting payload 96 on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... UNSUPPORTED.
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Unsetting payload 97 on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... UNSUPPORTED.
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Unsetting payload 98 on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... UNSUPPORTED.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for ulaw to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for gsm to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for g723 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for adpcm to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for adpcm to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for lpc10 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for alaw to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for g722 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for slin to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for slin to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for adpcm to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for adpcm to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for g729 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for jpeg to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for h261 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for h263 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for ilbc to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for h263p to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for h264 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for siren7 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for h263p to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for mpeg4 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for red to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for t140 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for speex to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for g726 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for g726aal2 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for siren14 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for g719 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for speex16 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for slin16 to 30
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Incorporating payload 0 on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Incorporating payload 2 on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Incorporating payload 4 on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Incorporating payload 8 on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Incorporating payload 18 on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Incorporating payload 101 on 0xb46dbe78
[Nov 27 08:16:50] DEBUG[8867] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9f05988'
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Copying payload 0 from 0xb46dbe78 to 0x9f05b34
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Copying payload 2 from 0xb46dbe78 to 0x9f05b34
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Copying payload 4 from 0xb46dbe78 to 0x9f05b34
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Copying payload 8 from 0xb46dbe78 to 0x9f05b34
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Copying payload 18 from 0xb46dbe78 to 0x9f05b34
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Copying payload 101 from 0xb46dbe78 to 0x9f05b34
[Nov 27 08:16:50] DEBUG[8867] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x9f05988'
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: We're settling with these formats: 0x11c (ulaw|alaw|g729|g726aal2)
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Checking SIP call limits for device 
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Updating call counter for incoming call
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: Splitting '10.0.6.23:5061' into...
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: ...host '10.0.6.23' and port ''.
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: Splitting 'sip.dev.avecezar.org' into...
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: ...host 'sip.dev.avecezar.org' and port ''.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: *** Our native formats are 0x100 (g729) 
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: *** Joint capabilities are 0x11c (ulaw|alaw|g729|g726aal2) 
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: *** Our capabilities are 0x91e (gsm|ulaw|alaw|g726|g729|g726aal2) 
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x100 (g729) 
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: This channel will not be able to handle video.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: build_route: Record-Route hop: <sip:10.0.6.23;lr;ftag=810e78819efc1671o1>
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: build_route: Record-Route hop: <sip:10.0.6.40;lr;did=732.43>
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: SIP/int_1-00000004: New call is still down.... Trying... 
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.0.6.23:5060
[Nov 27 08:16:50] DEBUG[8859] devicestate.c: No provider found, checking channel drivers for SIP - int_1
[Nov 27 08:16:50] DEBUG[8859] chan_sip.c: Checking device state for peer int_1
[Nov 27 08:16:50] DEBUG[8859] devicestate.c: Changing state for SIP/int_1 - state 1 (Not in use)
[Nov 27 08:16:50] DEBUG[8859] devicestate.c: device 'SIP/int_1' state '1'
[Nov 27 08:16:50] DEBUG[8891] app_queue.c: Device 'SIP/int_1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Function result is 'int_1'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'NoOp'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Function result is 'int_1'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Expression result is '0'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'GotoIf'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Function result is 'int_1'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Set'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Function result is 'int_1'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Expression result is '1'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Set'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Function result is 'sip:+48613994801@sip.dev.avecezar.org'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Set'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Result of 'Var_X-PAI' is 'sip:+48613994801@sip.dev.avecezar.org'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Expression result is '1'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'GotoIf'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Result of 'Var_X-PAI' is 'sip:+48613994801@sip.dev.avecezar.org'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'SIPAddHeader'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Function result is '(null)'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Set'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Result of 'Var_X-PRIVACY' is ''
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Expression result is '0'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'GotoIf'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Goto'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'NoOp'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Result of 'SIPTRANSFER' is NULL
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Expression result is '0'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'GotoIf'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Result of 'VAR_PEERIN' is ''
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Expression result is '0'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Expression result is '0'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'GotoIf'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Result of 'VAR_PEEROUT' is '1'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Expression result is '1'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Expression result is '1'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'GotoIf'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Result of 'EXTEN' is '+48600600612'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Goto'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Macro'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Function result is 'false'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Set'
[Nov 27 08:16:50] DEBUG[9392] app_macro.c: Executed application: Set
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Result of 'ANNF' is 'false'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Expression result is '0'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'GotoIf'
[Nov 27 08:16:50] DEBUG[9392] app_macro.c: Executed application: GotoIf
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'NoOp'
[Nov 27 08:16:50] DEBUG[9392] app_macro.c: Executed application: NoOp
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Function result is 'f1ae6dc1-723aa231@192.168.2.24'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'SIPAddHeader'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Set'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Set'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Set'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Set'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Goto'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Function result is '+48613994801'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Function result is '+48613994801'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Set'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Function result is '9534'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Expression result is '1'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'GotoIf'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Result of 'EXTEN' is '+48600600612'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Result of 'VAR_PEEROUT' is '1'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Function result is '9534'
[Nov 27 08:16:50] DEBUG[9392] pbx.c: Launching 'Dial'
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: Asked to create a SIP channel with formats: 0x100 (g729)
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: Allocating new SIP dialog for 343c4e3c38fbf58f003ac8bd57319b78@10.0.6.23:5061 - INVITE (No RTP)
[Nov 27 08:16:50] DEBUG[9392] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb44930d8'
[Nov 27 08:16:50] DEBUG[9392] res_rtp_asterisk.c: Allocated port 15354 for RTP instance '0xb44930d8'
[Nov 27 08:16:50] DEBUG[9392] rtp_engine.c: RTP instance '0xb44930d8' is setup and ready to go
[Nov 27 08:16:50] DEBUG[9392] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb44930d8'
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: Setting NAT on RTP to On
[Nov 27 08:16:50] DEBUG[9392] acl.c: For destination '10.0.6.29', our source address is '10.0.6.23'.
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: Target address 10.0.6.29:5060 is not local, substituting externaddr
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.0.6.23:5061
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: *** Our native formats are 0x8 (alaw) 
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: *** Joint capabilities are 0x0 (nothing) 
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: *** Our capabilities are 0xc (ulaw|alaw) 
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) 
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: *** Our preferred formats from the incoming channel are 0x100 (g729) 
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: This channel will not be able to handle video.
[Nov 27 08:16:50] DEBUG[9392] rtp_engine.c: Seeded SDP of 'SIP/ext_1-00000005' with that of 'SIP/int_1-00000004'
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable DIALEDTIME.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable ANSWEREDTIME.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable DIALEDPEERNAME.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable DIALEDPEERNUMBER.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable DIALSTATUS.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable LIMIT_TIMEOUT_FILE.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable LIMIT_PLAYAUDIO_CALLEE.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable LIMIT_PLAYAUDIO_CALLER.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable LIMIT_WARNING_FILE.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Copying hard-transferable variable SIPADDHEADER02.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable MACRO_DEPTH.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable ANNF.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable Var_X-PRIVACY.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Copying hard-transferable variable SIPADDHEADER01.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable Var_X-PAI.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable VAR_PEEROUT.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable VAR_PEERIN.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable SIPCALLID.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable SIPDOMAIN.
[Nov 27 08:16:50] DEBUG[9392] channel.c: Not copying variable SIPURI.
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: Outgoing Call for 0048600600612
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: Updating call counter for outgoing call
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: ** Our prefcodec: 0x100 (g729) 
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: -- Done with adding codecs to SDP
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw)
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: Initializing initreq for method INVITE - callid 4fec010938b4fb961abbe8c32728b5c5@sip.dev.avecezar.org
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.0.6.23:5060
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: = Looking for  Call ID: 4fec010938b4fb961abbe8c32728b5c5@sip.dev.avecezar.org (Checking To) --From tag as20b4c700 --To-tag   
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4fec010938b4fb961abbe8c32728b5c5@sip.dev.avecezar.org' Request 102: Found
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: SIP response 100 to standard invite
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: = Looking for  Call ID: 4fec010938b4fb961abbe8c32728b5c5@sip.dev.avecezar.org (Checking To) --From tag as20b4c700 --To-tag as1b950a80  
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Acked pending invite 102
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Stopping retransmission on '4fec010938b4fb961abbe8c32728b5c5@sip.dev.avecezar.org' of Request 102: Match Found
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: SIP response 200 to standard invite
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing session-level SDP o=root 1173675399 1173675399 IN IP4 10.0.6.29... UNSUPPORTED.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing session-level SDP s=Asterisk PBX 1.6.2.9-2+squeeze6... UNSUPPORTED.
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: Splitting '10.0.6.29' into...
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: ...host '10.0.6.29' and port ''.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing session-level SDP c=IN IP4 10.0.6.29... OK.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Setting payload 8 based on m type on 0xb46dc438
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Setting payload 0 based on m type on 0xb46dc438
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Setting payload 101 based on m type on 0xb46dc438
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for ulaw to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for gsm to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for g723 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for adpcm to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for adpcm to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for lpc10 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for alaw to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for g722 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for slin to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for slin to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for adpcm to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for adpcm to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for g729 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for jpeg to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for h261 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for h263 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for ilbc to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for h263p to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for h264 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for siren7 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for h263p to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for mpeg4 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for red to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for t140 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for speex to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for g726 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for g726aal2 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for siren14 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for g719 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for speex16 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Setting framing for slin16 to 10
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=ptime:10... OK.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Incorporating payload 0 on 0xb46dc438
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Incorporating payload 8 on 0xb46dc438
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Incorporating payload 101 on 0xb46dc438
[Nov 27 08:16:50] DEBUG[8867] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb44930d8'
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Copying payload 0 from 0xb46dc438 to 0xb4493284
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Copying payload 8 from 0xb46dc438 to 0xb4493284
[Nov 27 08:16:50] DEBUG[8867] rtp_engine.c: Copying payload 101 from 0xb46dc438 to 0xb4493284
[Nov 27 08:16:50] DEBUG[8867] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb44930d8'
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw)
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: We have an owner, now see if we need to change this call
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Updating call counter for outgoing call
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: build_route: Record-Route hop: <sip:10.0.6.23;lr;ftag=as20b4c700>
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Session-Expires: 1800
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Refresher: UAS
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Session timer started: 30 - 4fec010938b4fb961abbe8c32728b5c5@sip.dev.avecezar.org
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: Splitting '10.0.6.23' into...
[Nov 27 08:16:50] DEBUG[8867] netsock2.c: ...host '10.0.6.23' and port ''.
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Trying to put 'ACK sip:004' onto UDP socket destined for 10.0.6.23:5060
[Nov 27 08:16:50] DEBUG[9392] channel.c: Set channel SIP/ext_1-00000005 to read format slin
[Nov 27 08:16:50] DEBUG[9392] channel.c: Set channel SIP/int_1-00000004 to write format slin
[Nov 27 08:16:50] DEBUG[9392] channel.c: Set channel SIP/int_1-00000004 to read format slin
[Nov 27 08:16:50] DEBUG[9392] channel.c: Set channel SIP/ext_1-00000005 to write format slin
[Nov 27 08:16:50] DEBUG[9392] rtp_engine.c: Setting early bridge SDP of 'SIP/int_1-00000004' with that of 'SIP/ext_1-00000005'
[Nov 27 08:16:50] DEBUG[8859] devicestate.c: No provider found, checking channel drivers for SIP - ext_1
[Nov 27 08:16:50] DEBUG[8859] chan_sip.c: Checking device state for peer ext_1
[Nov 27 08:16:50] DEBUG[8859] devicestate.c: Changing state for SIP/ext_1 - state 1 (Not in use)
[Nov 27 08:16:50] DEBUG[8859] devicestate.c: device 'SIP/ext_1' state '1'
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: SIP answering channel: SIP/int_1-00000004
[Nov 27 08:16:50] DEBUG[9392] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Nov 27 08:16:50] DEBUG[8859] devicestate.c: No provider found, checking channel drivers for SIP - int_1
[Nov 27 08:16:50] DEBUG[8859] chan_sip.c: Checking device state for peer int_1
[Nov 27 08:16:50] DEBUG[8859] devicestate.c: Changing state for SIP/int_1 - state 1 (Not in use)
[Nov 27 08:16:50] DEBUG[8859] devicestate.c: device 'SIP/int_1' state '1'
[Nov 27 08:16:50] DEBUG[8891] app_queue.c: Device 'SIP/ext_1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Nov 27 08:16:50] DEBUG[8891] app_queue.c: Device 'SIP/int_1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: ** Our capability: 0x11c (ulaw|alaw|g729|g726aal2) Video flag: True Text flag: True
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: ** Our prefcodec: 0x0 (nothing) 
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: -- Done with adding codecs to SDP
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: Done building SDP. Settling with this capability: 0x11c (ulaw|alaw|g729|g726aal2)
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.0.6.23:5060
[Nov 27 08:16:50] DEBUG[9392] features.c: bridge answer set, chan answer set
[Nov 27 08:16:50] DEBUG[9392] features.c: Removing dialed interfaces datastore on SIP/ext_1-00000005 since we're bridging
[Nov 27 08:16:50] DEBUG[9392] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Nov 27 08:16:50] DEBUG[9392] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Nov 27 08:16:50] DEBUG[9392] rtp_engine.c: rtp-engine-local-bridge: Oooh, formats changed, backing out
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: = Looking for  Call ID: f1ae6dc1-723aa231@192.168.2.24 (Checking From) --From tag 810e78819efc1671o1 --To-tag as42d4d69f  
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Nov 27 08:16:50] DEBUG[8867] chan_sip.c: Stopping retransmission on 'f1ae6dc1-723aa231@192.168.2.24' of Response 102: Match Found
[Nov 27 08:16:50] DEBUG[9392] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x9f05988'
[Nov 27 08:16:50] DEBUG[9392] res_rtp_asterisk.c: RTP NAT: Got audio from other end. Now sending to address 10.0.2.3:16442
[Nov 27 08:16:50] DEBUG[9392] chan_sip.c: Oooh, format changed to alaw
[Nov 27 08:16:50] DEBUG[9392] channel.c: Set channel SIP/int_1-00000004 to read format slin
[Nov 27 08:16:50] DEBUG[9392] channel.c: Set channel SIP/int_1-00000004 to write format slin
[Nov 27 08:16:50] DEBUG[9392] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw
[Nov 27 08:16:50] DEBUG[9392] res_rtp_asterisk.c: Created smoother: format: alaw ms: 10 len: 80
[Nov 27 08:16:50] DEBUG[9392] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xb44930d8'