[Feb 27 23:13:18] VERBOSE[12208] config.c:   == Parsing '/etc/asterisk/logger.conf': [Feb 27 23:13:18] DEBUG[12208] config.c: Parsing /etc/asterisk/logger.conf
[Feb 27 23:13:18] VERBOSE[12208] config.c:   == Found
[Feb 27 23:13:18] VERBOSE[12208] logger.c:  Asterisk Queue Logger restarted
[Feb 27 23:13:21] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 711500805@10.78.65.151 (Checking From) --From tag 983168428 --To-tag   
[Feb 27 23:13:21] DEBUG[11384] acl.c: For destination '10.78.65.151', our source address is '10.78.65.80'.
[Feb 27 23:13:21] DEBUG[11384] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:21] VERBOSE[11384] netsock.c:   == Using UDPTL CoS mark 5
[Feb 27 23:13:21] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:21] DEBUG[11384] chan_sip.c: Allocating new SIP dialog for 711500805@10.78.65.151 - INVITE (No RTP)
[Feb 27 23:13:21] DEBUG[11384] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Feb 27 23:13:21] DEBUG[11384] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces"
[Feb 27 23:13:21] DEBUG[11384] sip/reqresp_parser.c: Found SIP option: -replaces-
[Feb 27 23:13:21] DEBUG[11384] sip/reqresp_parser.c: Matched SIP option: replaces
[Feb 27 23:13:21] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:21] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:21] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:21] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:21] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 711500805@10.78.65.151 (Checking From) --From tag 983168428 --To-tag as70661dda  
[Feb 27 23:13:21] DEBUG[11384] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Feb 27 23:13:21] DEBUG[11384] chan_sip.c: Stopping retransmission on '711500805@10.78.65.151' of Response 1: Match Found
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 711500805@10.78.65.151 (Checking From) --From tag 983168428 --To-tag   
[Feb 27 23:13:22] DEBUG[11384] netsock2.c: Splitting '10.78.65.80' into...
[Feb 27 23:13:22] DEBUG[11384] netsock2.c: ...host '10.78.65.80' and port ''.
[Feb 27 23:13:22] DEBUG[11384] netsock2.c: Splitting '10.78.65.80' into...
[Feb 27 23:13:22] DEBUG[11384] netsock2.c: ...host '10.78.65.80' and port ''.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Feb 27 23:13:22] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:22] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xaf162880'
[Feb 27 23:13:22] DEBUG[11384] res_rtp_asterisk.c: Allocated port 16076 for RTP instance '0xaf162880'
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: RTP instance '0xaf162880' is setup and ready to go
[Feb 27 23:13:22] DEBUG[11384] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xaf162880'
[Feb 27 23:13:22] VERBOSE[11384] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Setting NAT on RTP to Off
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing session-level SDP o=- 20080 20080 IN IP4 10.78.65.151... UNSUPPORTED.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing session-level SDP s=SDP data... UNSUPPORTED.
[Feb 27 23:13:22] DEBUG[11384] netsock2.c: Splitting '10.78.65.151' into...
[Feb 27 23:13:22] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port ''.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing session-level SDP c=IN IP4 10.78.65.151... OK.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Setting payload 8 based on m type on 0xaf668964
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Setting payload 0 based on m type on 0xaf668964
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Setting payload 18 based on m type on 0xaf668964
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Setting payload 9 based on m type on 0xaf668964
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Setting payload 102 based on m type on 0xaf668964
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Setting payload 101 based on m type on 0xaf668964
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:102 iLBC/8000... OK.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Incorporating payload 0 on 0xaf668964
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Incorporating payload 8 on 0xaf668964
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Incorporating payload 9 on 0xaf668964
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Incorporating payload 18 on 0xaf668964
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Incorporating payload 101 on 0xaf668964
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Incorporating payload 102 on 0xaf668964
[Feb 27 23:13:22] DEBUG[11384] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaf162880'
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Copying payload 0 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Copying payload 8 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Copying payload 9 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Copying payload 18 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Copying payload 101 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Copying payload 102 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: We're settling with these formats: 0x110c (ulaw|alaw|g729|g722)
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Checking SIP call limits for device 21
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Updating call counter for incoming call
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:22] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 2 (In use)
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: device 'SIP/21' state '2'
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: *** Our native formats are 0x1000 (g722) 
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: *** Joint capabilities are 0x110c (ulaw|alaw|g729|g722) 
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: *** Our capabilities are 0x110e (gsm|ulaw|alaw|g729|g722) 
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) 
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: This channel will not be able to handle video.
[Feb 27 23:13:22] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: build_route: Contact hop: <sip:21@10.78.65.151:5062>
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: SIP/21-000000ba: New call is still down.... Trying... 
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:22] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:22] DEBUG[12209] pbx.c: Launching 'NoOp'
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 2 (In use)
[Feb 27 23:13:22] VERBOSE[12209] pbx.c:     -- Executing [20@internal_default:1] NoOp("SIP/21-000000ba", "Internal SIP") in new stack
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: device 'SIP/21' state '2'
[Feb 27 23:13:22] DEBUG[12209] pbx.c: Result of 'EXTEN' is '20'
[Feb 27 23:13:22] DEBUG[11359] app_queue.c: Extension '21@autohint' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:22] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:22] DEBUG[12209] pbx.c: Function result is '21'
[Feb 27 23:13:22] DEBUG[12209] pbx.c: Expression result is '0'
[Feb 27 23:13:22] DEBUG[12209] pbx.c: Launching 'ExecIf'
[Feb 27 23:13:22] VERBOSE[12209] pbx.c:     -- Executing [20@internal_default:2] ExecIf("SIP/21-000000ba", "0?Hangup(3)") in new stack
[Feb 27 23:13:22] DEBUG[12209] pbx.c: Result of 'EXTEN' is '20'
[Feb 27 23:13:22] DEBUG[12209] pbx.c: Launching 'Dial'
[Feb 27 23:13:22] VERBOSE[12209] pbx.c:     -- Executing [20@internal_default:3] Dial("SIP/21-000000ba", "SIP/20,60,j") in new stack
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Asked to create a SIP channel with formats: 0x1000 (g722)
[Feb 27 23:13:22] VERBOSE[12209] netsock.c:   == Using UDPTL CoS mark 5
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Allocating new SIP dialog for 21a7431c5c66dfb167bb39dd65e19db2@10.78.65.80:5060 - INVITE (No RTP)
[Feb 27 23:13:22] DEBUG[12209] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xa3e3380'
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: Allocated port 14534 for RTP instance '0xa3e3380'
[Feb 27 23:13:22] DEBUG[12209] rtp_engine.c: RTP instance '0xa3e3380' is setup and ready to go
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xa3e3380'
[Feb 27 23:13:22] VERBOSE[12209] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Setting NAT on RTP to Off
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:22] DEBUG[12209] acl.c: For destination '10.78.65.152', our source address is '10.78.65.80'.
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: *** Our native formats are 0x1000 (g722) 
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: *** Joint capabilities are 0x1000 (g722) 
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: *** Our capabilities are 0x110e (gsm|ulaw|alaw|g729|g722) 
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) 
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: *** Our preferred formats from the incoming channel are 0x1000 (g722) 
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: This channel will not be able to handle video.
[Feb 27 23:13:22] DEBUG[12209] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Feb 27 23:13:22] DEBUG[12209] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Feb 27 23:13:22] DEBUG[12209] rtp_engine.c: Seeded SDP of 'SIP/20-000000bb' with that of 'SIP/21-000000ba'
[Feb 27 23:13:22] DEBUG[12209] channel.c: Not copying variable DIALEDTIME.
[Feb 27 23:13:22] DEBUG[12209] channel.c: Not copying variable ANSWEREDTIME.
[Feb 27 23:13:22] DEBUG[12209] channel.c: Not copying variable DIALEDPEERNAME.
[Feb 27 23:13:22] DEBUG[12209] channel.c: Not copying variable DIALEDPEERNUMBER.
[Feb 27 23:13:22] DEBUG[12209] channel.c: Not copying variable DIALSTATUS.
[Feb 27 23:13:22] DEBUG[12209] channel.c: Not copying variable SIPCALLID.
[Feb 27 23:13:22] DEBUG[12209] channel.c: Not copying variable SIPDOMAIN.
[Feb 27 23:13:22] DEBUG[12209] channel.c: Not copying variable SIPURI.
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Outgoing Call for 20
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:22] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 6 (Ringing)
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: ** Our capability: 0x100e (gsm|ulaw|alaw|g722) Video flag: False Text flag: False
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: device 'SIP/20' state '6'
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: ** Our prefcodec: 0x1000 (g722) 
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: -- Done with adding codecs to SDP
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Done building SDP. Settling with this capability: 0x100e (gsm|ulaw|alaw|g722)
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Initializing initreq for method INVITE - callid 2b6bc11c3b53c95b79c609935e4c64d9@10.78.65.80:5060
[Feb 27 23:13:22] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:22] VERBOSE[12209] app_dial.c:     -- Called SIP/20
[Feb 27 23:13:22] DEBUG[11359] app_queue.c: Extension '20@autohint' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 2b6bc11c3b53c95b79c609935e4c64d9@10.78.65.80:5060 (Checking To) --From tag as3cfc9190 --To-tag   
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2b6bc11c3b53c95b79c609935e4c64d9@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: SIP response 100 to standard invite
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 2b6bc11c3b53c95b79c609935e4c64d9@10.78.65.80:5060 (Checking To) --From tag as3cfc9190 --To-tag 1116036964  
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2b6bc11c3b53c95b79c609935e4c64d9@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: SIP response 180 to standard invite
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:22] VERBOSE[12209] app_dial.c:     -- SIP/20-000000bb is ringing
[Feb 27 23:13:22] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 6 (Ringing)
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: device 'SIP/20' state '6'
[Feb 27 23:13:22] DEBUG[12209] rtp_engine.c: Setting early bridge SDP of 'SIP/21-000000ba' with that of 'SIP/20-000000bb'
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:22] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:22] DEBUG[12209] channel.c: Driver for channel 'SIP/21-000000ba' does not support indication 3, emulating it
[Feb 27 23:13:22] DEBUG[12209] channel.c: Set channel SIP/21-000000ba to write format slin
[Feb 27 23:13:22] DEBUG[12209] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Feb 27 23:13:22] DEBUG[12209] channel.c: Prodding channel 'SIP/21-000000ba'
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Setting framing from config on incoming call
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: ** Our capability: 0x110c (ulaw|alaw|g729|g722) Video flag: True Text flag: True
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: ** Our prefcodec: 0x0 (nothing) 
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: -- Done with adding codecs to SDP
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Done building SDP. Settling with this capability: 0x110c (ulaw|alaw|g729|g722)
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: Received frame with no data for RTP instance '0xaf162880' so dropping frame
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: Ooh, format changed from unknown to g722
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: Created smoother: format: g722 ms: 20 len: 160
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xaf162880'
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 711500805@10.78.65.151 (Checking From) --From tag 983168428 --To-tag   
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: **** Received CANCEL (14) - Command in SIP CANCEL
[Feb 27 23:13:22] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:22] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Setting SIP_ALREADYGONE on dialog 711500805@10.78.65.151
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Updating call counter for incoming call
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:22] DEBUG[11384] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaf162880'
[Feb 27 23:13:22] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 1 (Not in use)
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: device 'SIP/21' state '1'
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 487' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:22] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:22] DEBUG[12209] channel.c: Set channel SIP/21-000000ba to write format g722
[Feb 27 23:13:22] DEBUG[12209] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Feb 27 23:13:22] DEBUG[12209] channel.c: Hanging up channel 'SIP/20-000000bb'
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Hanging up zombie call. Be scared.
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Hanging up channel in state Ringing (not UP)
[Feb 27 23:13:22] DEBUG[11359] app_queue.c: Extension '21@autohint' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:22] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa3e3380'
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2b6bc11c3b53c95b79c609935e4c64d9@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:22] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:22] DEBUG[12209] app_dial.c: Exiting with DIALSTATUS=CANCEL.
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:22] DEBUG[12209] pbx.c: Spawn extension (internal_default,20,3) exited non-zero on 'SIP/21-000000ba'
[Feb 27 23:13:22] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:22] VERBOSE[12209] pbx.c:   == Spawn extension (internal_default, 20, 3) exited non-zero on 'SIP/21-000000ba'
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:13:22] DEBUG[12209] channel.c: Soft-Hanging up channel 'SIP/21-000000ba'
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:13:22] DEBUG[11359] app_queue.c: Extension '20@autohint' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:22] DEBUG[12209] pbx.c: Launching 'Hangup'
[Feb 27 23:13:22] VERBOSE[12209] pbx.c:     -- Executing [h@internal_default:1] Hangup("SIP/21-000000ba", "") in new stack
[Feb 27 23:13:22] DEBUG[12209] pbx.c: Spawn extension (internal_default,h,1) exited non-zero on 'SIP/21-000000ba'
[Feb 27 23:13:22] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:22] VERBOSE[12209] pbx.c:   == Spawn extension (internal_default, h, 1) exited non-zero on 'SIP/21-000000ba'
[Feb 27 23:13:22] DEBUG[12209] channel.c: Hanging up channel 'SIP/21-000000ba'
[Feb 27 23:13:22] DEBUG[12209] chan_sip.c: Hanging up zombie call. Be scared.
[Feb 27 23:13:22] DEBUG[12209] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaf162880'
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:22] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 1 (Not in use)
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: device 'SIP/21' state '1'
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 711500805@10.78.65.151 (Checking From) --From tag 983168428 --To-tag as239276c1  
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Feb 27 23:13:22] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Stopping retransmission on '711500805@10.78.65.151' of Response 2: Match Found
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Destroying SIP dialog 711500805@10.78.65.151
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Destroyed RTP instance '0xaf162880'
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 2b6bc11c3b53c95b79c609935e4c64d9@10.78.65.80:5060 (Checking To) --From tag as3cfc9190 --To-tag 1116036964  
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Acked pending invite 102
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Stopping retransmission on '2b6bc11c3b53c95b79c609935e4c64d9@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 2b6bc11c3b53c95b79c609935e4c64d9@10.78.65.80:5060 (Checking To) --From tag as3cfc9190 --To-tag 1116036964  
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Stopping retransmission on '2b6bc11c3b53c95b79c609935e4c64d9@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: SIP response 487 to standard invite
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Trying to put 'ACK sip:20@' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2b6bc11c3b53c95b79c609935e4c64d9@10.78.65.80:5060
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:22] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:13:22] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:13:22] DEBUG[11384] chan_sip.c: Destroying SIP dialog 2b6bc11c3b53c95b79c609935e4c64d9@10.78.65.80:5060
[Feb 27 23:13:22] DEBUG[11384] rtp_engine.c: Destroyed RTP instance '0xa3e3380'
[Feb 27 23:13:22] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 349803596@10.78.65.151 (Checking From) --From tag 1188250542 --To-tag   
[Feb 27 23:13:25] DEBUG[11384] acl.c: For destination '10.78.65.151', our source address is '10.78.65.80'.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:25] VERBOSE[11384] netsock.c:   == Using UDPTL CoS mark 5
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Allocating new SIP dialog for 349803596@10.78.65.151 - INVITE (No RTP)
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Feb 27 23:13:25] DEBUG[11384] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces"
[Feb 27 23:13:25] DEBUG[11384] sip/reqresp_parser.c: Found SIP option: -replaces-
[Feb 27 23:13:25] DEBUG[11384] sip/reqresp_parser.c: Matched SIP option: replaces
[Feb 27 23:13:25] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:25] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 349803596@10.78.65.151 (Checking From) --From tag 1188250542 --To-tag as2c7feed1  
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Stopping retransmission on '349803596@10.78.65.151' of Response 1: Match Found
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 349803596@10.78.65.151 (Checking From) --From tag 1188250542 --To-tag   
[Feb 27 23:13:25] DEBUG[11384] netsock2.c: Splitting '10.78.65.80' into...
[Feb 27 23:13:25] DEBUG[11384] netsock2.c: ...host '10.78.65.80' and port ''.
[Feb 27 23:13:25] DEBUG[11384] netsock2.c: Splitting '10.78.65.80' into...
[Feb 27 23:13:25] DEBUG[11384] netsock2.c: ...host '10.78.65.80' and port ''.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Feb 27 23:13:25] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:25] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xaf162880'
[Feb 27 23:13:25] DEBUG[11384] res_rtp_asterisk.c: Allocated port 15572 for RTP instance '0xaf162880'
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: RTP instance '0xaf162880' is setup and ready to go
[Feb 27 23:13:25] DEBUG[11384] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xaf162880'
[Feb 27 23:13:25] VERBOSE[11384] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Setting NAT on RTP to Off
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing session-level SDP o=- 20081 20081 IN IP4 10.78.65.151... UNSUPPORTED.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing session-level SDP s=SDP data... UNSUPPORTED.
[Feb 27 23:13:25] DEBUG[11384] netsock2.c: Splitting '10.78.65.151' into...
[Feb 27 23:13:25] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port ''.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing session-level SDP c=IN IP4 10.78.65.151... OK.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Setting payload 8 based on m type on 0xaf668964
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Setting payload 0 based on m type on 0xaf668964
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Setting payload 18 based on m type on 0xaf668964
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Setting payload 9 based on m type on 0xaf668964
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Setting payload 102 based on m type on 0xaf668964
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Setting payload 101 based on m type on 0xaf668964
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:102 iLBC/8000... OK.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Incorporating payload 0 on 0xaf668964
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Incorporating payload 8 on 0xaf668964
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Incorporating payload 9 on 0xaf668964
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Incorporating payload 18 on 0xaf668964
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Incorporating payload 101 on 0xaf668964
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Incorporating payload 102 on 0xaf668964
[Feb 27 23:13:25] DEBUG[11384] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaf162880'
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Copying payload 0 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Copying payload 8 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Copying payload 9 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Copying payload 18 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Copying payload 101 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:25] DEBUG[11384] rtp_engine.c: Copying payload 102 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: We're settling with these formats: 0x110c (ulaw|alaw|g729|g722)
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Checking SIP call limits for device 21
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Updating call counter for incoming call
[Feb 27 23:13:25] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:25] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:25] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 2 (In use)
[Feb 27 23:13:25] DEBUG[11358] devicestate.c: device 'SIP/21' state '2'
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: *** Our native formats are 0x1000 (g722) 
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: *** Joint capabilities are 0x110c (ulaw|alaw|g729|g722) 
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: *** Our capabilities are 0x110e (gsm|ulaw|alaw|g729|g722) 
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) 
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: This channel will not be able to handle video.
[Feb 27 23:13:25] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: build_route: Contact hop: <sip:21@10.78.65.151:5062>
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: SIP/21-000000bc: New call is still down.... Trying... 
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:25] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:25] DEBUG[12210] pbx.c: Launching 'NoOp'
[Feb 27 23:13:25] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:25] VERBOSE[12210] pbx.c:     -- Executing [20@internal_default:1] NoOp("SIP/21-000000bc", "Internal SIP") in new stack
[Feb 27 23:13:25] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 2 (In use)
[Feb 27 23:13:25] DEBUG[11358] devicestate.c: device 'SIP/21' state '2'
[Feb 27 23:13:25] DEBUG[11359] app_queue.c: Extension '21@autohint' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:25] DEBUG[12210] pbx.c: Result of 'EXTEN' is '20'
[Feb 27 23:13:25] DEBUG[12210] pbx.c: Function result is '21'
[Feb 27 23:13:25] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:25] DEBUG[12210] pbx.c: Expression result is '0'
[Feb 27 23:13:25] DEBUG[12210] pbx.c: Launching 'ExecIf'
[Feb 27 23:13:25] VERBOSE[12210] pbx.c:     -- Executing [20@internal_default:2] ExecIf("SIP/21-000000bc", "0?Hangup(3)") in new stack
[Feb 27 23:13:25] DEBUG[12210] pbx.c: Result of 'EXTEN' is '20'
[Feb 27 23:13:25] DEBUG[12210] pbx.c: Launching 'Dial'
[Feb 27 23:13:25] VERBOSE[12210] pbx.c:     -- Executing [20@internal_default:3] Dial("SIP/21-000000bc", "SIP/20,60,j") in new stack
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Asked to create a SIP channel with formats: 0x1000 (g722)
[Feb 27 23:13:25] VERBOSE[12210] netsock.c:   == Using UDPTL CoS mark 5
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Allocating new SIP dialog for 6a43e444271249137887bc1842fc0288@10.78.65.80:5060 - INVITE (No RTP)
[Feb 27 23:13:25] DEBUG[12210] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xa3e3380'
[Feb 27 23:13:25] DEBUG[12210] res_rtp_asterisk.c: Allocated port 14652 for RTP instance '0xa3e3380'
[Feb 27 23:13:25] DEBUG[12210] rtp_engine.c: RTP instance '0xa3e3380' is setup and ready to go
[Feb 27 23:13:25] DEBUG[12210] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xa3e3380'
[Feb 27 23:13:25] VERBOSE[12210] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Setting NAT on RTP to Off
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:25] DEBUG[12210] acl.c: For destination '10.78.65.152', our source address is '10.78.65.80'.
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: *** Our native formats are 0x1000 (g722) 
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: *** Joint capabilities are 0x1000 (g722) 
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: *** Our capabilities are 0x110e (gsm|ulaw|alaw|g729|g722) 
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) 
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: *** Our preferred formats from the incoming channel are 0x1000 (g722) 
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: This channel will not be able to handle video.
[Feb 27 23:13:25] DEBUG[12210] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Feb 27 23:13:25] DEBUG[12210] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Feb 27 23:13:25] DEBUG[12210] rtp_engine.c: Seeded SDP of 'SIP/20-000000bd' with that of 'SIP/21-000000bc'
[Feb 27 23:13:25] DEBUG[12210] channel.c: Not copying variable DIALEDTIME.
[Feb 27 23:13:25] DEBUG[12210] channel.c: Not copying variable ANSWEREDTIME.
[Feb 27 23:13:25] DEBUG[12210] channel.c: Not copying variable DIALEDPEERNAME.
[Feb 27 23:13:25] DEBUG[12210] channel.c: Not copying variable DIALEDPEERNUMBER.
[Feb 27 23:13:25] DEBUG[12210] channel.c: Not copying variable DIALSTATUS.
[Feb 27 23:13:25] DEBUG[12210] channel.c: Not copying variable SIPCALLID.
[Feb 27 23:13:25] DEBUG[12210] channel.c: Not copying variable SIPDOMAIN.
[Feb 27 23:13:25] DEBUG[12210] channel.c: Not copying variable SIPURI.
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Outgoing Call for 20
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:25] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:25] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:25] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 6 (Ringing)
[Feb 27 23:13:25] DEBUG[11358] devicestate.c: device 'SIP/20' state '6'
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: ** Our capability: 0x100e (gsm|ulaw|alaw|g722) Video flag: False Text flag: False
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: ** Our prefcodec: 0x1000 (g722) 
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: -- Done with adding codecs to SDP
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Done building SDP. Settling with this capability: 0x100e (gsm|ulaw|alaw|g722)
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Initializing initreq for method INVITE - callid 7f2386ee4db587821464a123565e3636@10.78.65.80:5060
[Feb 27 23:13:25] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:25] VERBOSE[12210] app_dial.c:     -- Called SIP/20
[Feb 27 23:13:25] DEBUG[11359] app_queue.c: Extension '20@autohint' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 7f2386ee4db587821464a123565e3636@10.78.65.80:5060 (Checking To) --From tag as5069aff6 --To-tag   
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7f2386ee4db587821464a123565e3636@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: SIP response 100 to standard invite
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 7f2386ee4db587821464a123565e3636@10.78.65.80:5060 (Checking To) --From tag as5069aff6 --To-tag 1661771114  
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7f2386ee4db587821464a123565e3636@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:25] DEBUG[11384] chan_sip.c: SIP response 180 to standard invite
[Feb 27 23:13:25] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:25] VERBOSE[12210] app_dial.c:     -- SIP/20-000000bd is ringing
[Feb 27 23:13:25] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:25] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 6 (Ringing)
[Feb 27 23:13:25] DEBUG[11358] devicestate.c: device 'SIP/20' state '6'
[Feb 27 23:13:25] DEBUG[12210] rtp_engine.c: Setting early bridge SDP of 'SIP/21-000000bc' with that of 'SIP/20-000000bd'
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:25] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:25] DEBUG[12210] channel.c: Driver for channel 'SIP/21-000000bc' does not support indication 3, emulating it
[Feb 27 23:13:25] DEBUG[12210] channel.c: Set channel SIP/21-000000bc to write format slin
[Feb 27 23:13:25] DEBUG[12210] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Feb 27 23:13:25] DEBUG[12210] channel.c: Prodding channel 'SIP/21-000000bc'
[Feb 27 23:13:25] DEBUG[12210] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Setting framing from config on incoming call
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: ** Our capability: 0x110c (ulaw|alaw|g729|g722) Video flag: True Text flag: True
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: ** Our prefcodec: 0x0 (nothing) 
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: -- Done with adding codecs to SDP
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Done building SDP. Settling with this capability: 0x110c (ulaw|alaw|g729|g722)
[Feb 27 23:13:25] DEBUG[12210] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:25] DEBUG[12210] res_rtp_asterisk.c: Received frame with no data for RTP instance '0xaf162880' so dropping frame
[Feb 27 23:13:25] DEBUG[12210] res_rtp_asterisk.c: Ooh, format changed from unknown to g722
[Feb 27 23:13:25] DEBUG[12210] res_rtp_asterisk.c: Created smoother: format: g722 ms: 20 len: 160
[Feb 27 23:13:25] DEBUG[12210] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xaf162880'
[Feb 27 23:13:26] DEBUG[12210] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 349803596@10.78.65.151 (Checking From) --From tag 1188250542 --To-tag   
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: **** Received CANCEL (14) - Command in SIP CANCEL
[Feb 27 23:13:26] DEBUG[12210] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:26] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:26] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: Setting SIP_ALREADYGONE on dialog 349803596@10.78.65.151
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: Updating call counter for incoming call
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:26] DEBUG[11384] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaf162880'
[Feb 27 23:13:26] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 1 (Not in use)
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: device 'SIP/21' state '1'
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 487' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:26] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:26] DEBUG[12210] channel.c: Set channel SIP/21-000000bc to write format g722
[Feb 27 23:13:26] DEBUG[12210] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Feb 27 23:13:26] DEBUG[12210] channel.c: Hanging up channel 'SIP/20-000000bd'
[Feb 27 23:13:26] DEBUG[12210] chan_sip.c: Hanging up zombie call. Be scared.
[Feb 27 23:13:26] DEBUG[11359] app_queue.c: Extension '21@autohint' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:26] DEBUG[12210] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:26] DEBUG[12210] chan_sip.c: Hanging up channel in state Ringing (not UP)
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:26] DEBUG[12210] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa3e3380'
[Feb 27 23:13:26] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:26] DEBUG[12210] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7f2386ee4db587821464a123565e3636@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:13:26] DEBUG[12210] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:26] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:26] DEBUG[12210] app_dial.c: Exiting with DIALSTATUS=CANCEL.
[Feb 27 23:13:26] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:13:26] DEBUG[12210] pbx.c: Spawn extension (internal_default,20,3) exited non-zero on 'SIP/21-000000bc'
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:13:26] VERBOSE[12210] pbx.c:   == Spawn extension (internal_default, 20, 3) exited non-zero on 'SIP/21-000000bc'
[Feb 27 23:13:26] DEBUG[12210] channel.c: Soft-Hanging up channel 'SIP/21-000000bc'
[Feb 27 23:13:26] DEBUG[12210] pbx.c: Launching 'Hangup'
[Feb 27 23:13:26] VERBOSE[12210] pbx.c:     -- Executing [h@internal_default:1] Hangup("SIP/21-000000bc", "") in new stack
[Feb 27 23:13:26] DEBUG[12210] pbx.c: Spawn extension (internal_default,h,1) exited non-zero on 'SIP/21-000000bc'
[Feb 27 23:13:26] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:26] DEBUG[11359] app_queue.c: Extension '20@autohint' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:26] VERBOSE[12210] pbx.c:   == Spawn extension (internal_default, h, 1) exited non-zero on 'SIP/21-000000bc'
[Feb 27 23:13:26] DEBUG[12210] channel.c: Hanging up channel 'SIP/21-000000bc'
[Feb 27 23:13:26] DEBUG[12210] chan_sip.c: Hanging up zombie call. Be scared.
[Feb 27 23:13:26] DEBUG[12210] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaf162880'
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 7f2386ee4db587821464a123565e3636@10.78.65.80:5060 (Checking To) --From tag as5069aff6 --To-tag 1661771114  
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:26] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 1 (Not in use)
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: Acked pending invite 102
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: device 'SIP/21' state '1'
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: Stopping retransmission on '7f2386ee4db587821464a123565e3636@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:13:26] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 349803596@10.78.65.151 (Checking From) --From tag 1188250542 --To-tag as71e94c3b  
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: Stopping retransmission on '349803596@10.78.65.151' of Response 2: Match Found
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: Destroying SIP dialog 349803596@10.78.65.151
[Feb 27 23:13:26] DEBUG[11384] rtp_engine.c: Destroyed RTP instance '0xaf162880'
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 7f2386ee4db587821464a123565e3636@10.78.65.80:5060 (Checking To) --From tag as5069aff6 --To-tag 1661771114  
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: Stopping retransmission on '7f2386ee4db587821464a123565e3636@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: SIP response 487 to standard invite
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: Trying to put 'ACK sip:20@' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7f2386ee4db587821464a123565e3636@10.78.65.80:5060
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:26] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:13:26] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:13:26] DEBUG[11384] chan_sip.c: Destroying SIP dialog 7f2386ee4db587821464a123565e3636@10.78.65.80:5060
[Feb 27 23:13:26] DEBUG[11384] rtp_engine.c: Destroyed RTP instance '0xa3e3380'
[Feb 27 23:13:26] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 1698705372@10.78.65.151 (Checking From) --From tag 1073600252 --To-tag   
[Feb 27 23:13:28] DEBUG[11384] acl.c: For destination '10.78.65.151', our source address is '10.78.65.80'.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:28] VERBOSE[11384] netsock.c:   == Using UDPTL CoS mark 5
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Allocating new SIP dialog for 1698705372@10.78.65.151 - INVITE (No RTP)
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Feb 27 23:13:28] DEBUG[11384] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces"
[Feb 27 23:13:28] DEBUG[11384] sip/reqresp_parser.c: Found SIP option: -replaces-
[Feb 27 23:13:28] DEBUG[11384] sip/reqresp_parser.c: Matched SIP option: replaces
[Feb 27 23:13:28] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:28] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 1698705372@10.78.65.151 (Checking From) --From tag 1073600252 --To-tag as5a377e63  
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Stopping retransmission on '1698705372@10.78.65.151' of Response 1: Match Found
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 1698705372@10.78.65.151 (Checking From) --From tag 1073600252 --To-tag   
[Feb 27 23:13:28] DEBUG[11384] netsock2.c: Splitting '10.78.65.80' into...
[Feb 27 23:13:28] DEBUG[11384] netsock2.c: ...host '10.78.65.80' and port ''.
[Feb 27 23:13:28] DEBUG[11384] netsock2.c: Splitting '10.78.65.80' into...
[Feb 27 23:13:28] DEBUG[11384] netsock2.c: ...host '10.78.65.80' and port ''.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Feb 27 23:13:28] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:28] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xa3e3380'
[Feb 27 23:13:28] DEBUG[11384] res_rtp_asterisk.c: Allocated port 15442 for RTP instance '0xa3e3380'
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: RTP instance '0xa3e3380' is setup and ready to go
[Feb 27 23:13:28] DEBUG[11384] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xa3e3380'
[Feb 27 23:13:28] VERBOSE[11384] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Setting NAT on RTP to Off
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing session-level SDP o=- 20082 20082 IN IP4 10.78.65.151... UNSUPPORTED.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing session-level SDP s=SDP data... UNSUPPORTED.
[Feb 27 23:13:28] DEBUG[11384] netsock2.c: Splitting '10.78.65.151' into...
[Feb 27 23:13:28] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port ''.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing session-level SDP c=IN IP4 10.78.65.151... OK.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Setting payload 8 based on m type on 0xaf668964
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Setting payload 0 based on m type on 0xaf668964
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Setting payload 18 based on m type on 0xaf668964
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Setting payload 9 based on m type on 0xaf668964
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Setting payload 102 based on m type on 0xaf668964
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Setting payload 101 based on m type on 0xaf668964
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:102 iLBC/8000... OK.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Incorporating payload 0 on 0xaf668964
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Incorporating payload 8 on 0xaf668964
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Incorporating payload 9 on 0xaf668964
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Incorporating payload 18 on 0xaf668964
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Incorporating payload 101 on 0xaf668964
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Incorporating payload 102 on 0xaf668964
[Feb 27 23:13:28] DEBUG[11384] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa3e3380'
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Copying payload 0 from 0xaf668964 to 0xa3e352c
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Copying payload 8 from 0xaf668964 to 0xa3e352c
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Copying payload 9 from 0xaf668964 to 0xa3e352c
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Copying payload 18 from 0xaf668964 to 0xa3e352c
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Copying payload 101 from 0xaf668964 to 0xa3e352c
[Feb 27 23:13:28] DEBUG[11384] rtp_engine.c: Copying payload 102 from 0xaf668964 to 0xa3e352c
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: We're settling with these formats: 0x110c (ulaw|alaw|g729|g722)
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Checking SIP call limits for device 21
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Updating call counter for incoming call
[Feb 27 23:13:28] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:28] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:28] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 2 (In use)
[Feb 27 23:13:28] DEBUG[11358] devicestate.c: device 'SIP/21' state '2'
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: *** Our native formats are 0x1000 (g722) 
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: *** Joint capabilities are 0x110c (ulaw|alaw|g729|g722) 
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: *** Our capabilities are 0x110e (gsm|ulaw|alaw|g729|g722) 
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) 
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: This channel will not be able to handle video.
[Feb 27 23:13:28] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: build_route: Contact hop: <sip:21@10.78.65.151:5062>
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: SIP/21-000000be: New call is still down.... Trying... 
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:28] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:28] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:28] DEBUG[12211] pbx.c: Launching 'NoOp'
[Feb 27 23:13:28] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 2 (In use)
[Feb 27 23:13:28] VERBOSE[12211] pbx.c:     -- Executing [20@internal_default:1] NoOp("SIP/21-000000be", "Internal SIP") in new stack
[Feb 27 23:13:28] DEBUG[11358] devicestate.c: device 'SIP/21' state '2'
[Feb 27 23:13:28] DEBUG[11359] app_queue.c: Extension '21@autohint' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:28] DEBUG[12211] pbx.c: Result of 'EXTEN' is '20'
[Feb 27 23:13:28] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:28] DEBUG[12211] pbx.c: Function result is '21'
[Feb 27 23:13:28] DEBUG[12211] pbx.c: Expression result is '0'
[Feb 27 23:13:28] DEBUG[12211] pbx.c: Launching 'ExecIf'
[Feb 27 23:13:28] VERBOSE[12211] pbx.c:     -- Executing [20@internal_default:2] ExecIf("SIP/21-000000be", "0?Hangup(3)") in new stack
[Feb 27 23:13:28] DEBUG[12211] pbx.c: Result of 'EXTEN' is '20'
[Feb 27 23:13:28] DEBUG[12211] pbx.c: Launching 'Dial'
[Feb 27 23:13:28] VERBOSE[12211] pbx.c:     -- Executing [20@internal_default:3] Dial("SIP/21-000000be", "SIP/20,60,j") in new stack
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Asked to create a SIP channel with formats: 0x1000 (g722)
[Feb 27 23:13:28] VERBOSE[12211] netsock.c:   == Using UDPTL CoS mark 5
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Allocating new SIP dialog for 3572ca272f9d3f47374eeac12e343f4b@10.78.65.80:5060 - INVITE (No RTP)
[Feb 27 23:13:28] DEBUG[12211] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xa460738'
[Feb 27 23:13:28] DEBUG[12211] res_rtp_asterisk.c: Allocated port 15040 for RTP instance '0xa460738'
[Feb 27 23:13:28] DEBUG[12211] rtp_engine.c: RTP instance '0xa460738' is setup and ready to go
[Feb 27 23:13:28] DEBUG[12211] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xa460738'
[Feb 27 23:13:28] VERBOSE[12211] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Setting NAT on RTP to Off
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:28] DEBUG[12211] acl.c: For destination '10.78.65.152', our source address is '10.78.65.80'.
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: *** Our native formats are 0x1000 (g722) 
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: *** Joint capabilities are 0x1000 (g722) 
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: *** Our capabilities are 0x110e (gsm|ulaw|alaw|g729|g722) 
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) 
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: *** Our preferred formats from the incoming channel are 0x1000 (g722) 
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: This channel will not be able to handle video.
[Feb 27 23:13:28] DEBUG[12211] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Feb 27 23:13:28] DEBUG[12211] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Feb 27 23:13:28] DEBUG[12211] rtp_engine.c: Seeded SDP of 'SIP/20-000000bf' with that of 'SIP/21-000000be'
[Feb 27 23:13:28] DEBUG[12211] channel.c: Not copying variable DIALEDTIME.
[Feb 27 23:13:28] DEBUG[12211] channel.c: Not copying variable ANSWEREDTIME.
[Feb 27 23:13:28] DEBUG[12211] channel.c: Not copying variable DIALEDPEERNAME.
[Feb 27 23:13:28] DEBUG[12211] channel.c: Not copying variable DIALEDPEERNUMBER.
[Feb 27 23:13:28] DEBUG[12211] channel.c: Not copying variable DIALSTATUS.
[Feb 27 23:13:28] DEBUG[12211] channel.c: Not copying variable SIPCALLID.
[Feb 27 23:13:28] DEBUG[12211] channel.c: Not copying variable SIPDOMAIN.
[Feb 27 23:13:28] DEBUG[12211] channel.c: Not copying variable SIPURI.
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Outgoing Call for 20
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:28] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:28] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:28] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 6 (Ringing)
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: ** Our capability: 0x100e (gsm|ulaw|alaw|g722) Video flag: False Text flag: False
[Feb 27 23:13:28] DEBUG[11358] devicestate.c: device 'SIP/20' state '6'
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: ** Our prefcodec: 0x1000 (g722) 
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: -- Done with adding codecs to SDP
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Done building SDP. Settling with this capability: 0x100e (gsm|ulaw|alaw|g722)
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Initializing initreq for method INVITE - callid 5ee966a91d9c66793ab2b63b2a39c643@10.78.65.80:5060
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:28] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:28] VERBOSE[12211] app_dial.c:     -- Called SIP/20
[Feb 27 23:13:28] DEBUG[11359] app_queue.c: Extension '20@autohint' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 5ee966a91d9c66793ab2b63b2a39c643@10.78.65.80:5060 (Checking To) --From tag as59caace1 --To-tag   
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5ee966a91d9c66793ab2b63b2a39c643@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: SIP response 100 to standard invite
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 5ee966a91d9c66793ab2b63b2a39c643@10.78.65.80:5060 (Checking To) --From tag as59caace1 --To-tag 486696185  
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5ee966a91d9c66793ab2b63b2a39c643@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:28] DEBUG[11384] chan_sip.c: SIP response 180 to standard invite
[Feb 27 23:13:28] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:28] VERBOSE[12211] app_dial.c:     -- SIP/20-000000bf is ringing
[Feb 27 23:13:28] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:28] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 6 (Ringing)
[Feb 27 23:13:28] DEBUG[11358] devicestate.c: device 'SIP/20' state '6'
[Feb 27 23:13:28] DEBUG[12211] rtp_engine.c: Setting early bridge SDP of 'SIP/21-000000be' with that of 'SIP/20-000000bf'
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:28] DEBUG[12211] channel.c: Driver for channel 'SIP/21-000000be' does not support indication 3, emulating it
[Feb 27 23:13:28] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:28] DEBUG[12211] channel.c: Set channel SIP/21-000000be to write format slin
[Feb 27 23:13:28] DEBUG[12211] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Feb 27 23:13:28] DEBUG[12211] channel.c: Prodding channel 'SIP/21-000000be'
[Feb 27 23:13:28] DEBUG[12211] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Setting framing from config on incoming call
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: ** Our capability: 0x110c (ulaw|alaw|g729|g722) Video flag: True Text flag: True
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: ** Our prefcodec: 0x0 (nothing) 
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: -- Done with adding codecs to SDP
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Done building SDP. Settling with this capability: 0x110c (ulaw|alaw|g729|g722)
[Feb 27 23:13:28] DEBUG[12211] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:28] DEBUG[12211] res_rtp_asterisk.c: Received frame with no data for RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:28] DEBUG[12211] res_rtp_asterisk.c: Ooh, format changed from unknown to g722
[Feb 27 23:13:28] DEBUG[12211] res_rtp_asterisk.c: Created smoother: format: g722 ms: 20 len: 160
[Feb 27 23:13:28] DEBUG[12211] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xa3e3380'
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 1698705372@10.78.65.151 (Checking From) --From tag 1073600252 --To-tag   
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: **** Received CANCEL (14) - Command in SIP CANCEL
[Feb 27 23:13:29] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:29] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1698705372@10.78.65.151
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: Updating call counter for incoming call
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:29] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:29] DEBUG[11384] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa3e3380'
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 1 (Not in use)
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: device 'SIP/21' state '1'
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 487' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:29] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:29] DEBUG[12211] channel.c: Set channel SIP/21-000000be to write format g722
[Feb 27 23:13:29] DEBUG[12211] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Feb 27 23:13:29] DEBUG[12211] channel.c: Hanging up channel 'SIP/20-000000bf'
[Feb 27 23:13:29] DEBUG[12211] chan_sip.c: Hanging up zombie call. Be scared.
[Feb 27 23:13:29] DEBUG[12211] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:29] DEBUG[11359] app_queue.c: Extension '21@autohint' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:29] DEBUG[12211] chan_sip.c: Hanging up channel in state Ringing (not UP)
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:29] DEBUG[12211] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa460738'
[Feb 27 23:13:29] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:13:29] DEBUG[12211] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5ee966a91d9c66793ab2b63b2a39c643@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:13:29] DEBUG[12211] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:29] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:29] DEBUG[12211] app_dial.c: Exiting with DIALSTATUS=CANCEL.
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:29] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:29] DEBUG[12211] pbx.c: Spawn extension (internal_default,20,3) exited non-zero on 'SIP/21-000000be'
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:13:29] VERBOSE[12211] pbx.c:   == Spawn extension (internal_default, 20, 3) exited non-zero on 'SIP/21-000000be'
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:13:29] DEBUG[12211] channel.c: Soft-Hanging up channel 'SIP/21-000000be'
[Feb 27 23:13:29] DEBUG[12211] pbx.c: Launching 'Hangup'
[Feb 27 23:13:29] VERBOSE[12211] pbx.c:     -- Executing [h@internal_default:1] Hangup("SIP/21-000000be", "") in new stack
[Feb 27 23:13:29] DEBUG[11359] app_queue.c: Extension '20@autohint' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:29] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:29] DEBUG[12211] pbx.c: Spawn extension (internal_default,h,1) exited non-zero on 'SIP/21-000000be'
[Feb 27 23:13:29] VERBOSE[12211] pbx.c:   == Spawn extension (internal_default, h, 1) exited non-zero on 'SIP/21-000000be'
[Feb 27 23:13:29] DEBUG[12211] channel.c: Hanging up channel 'SIP/21-000000be'
[Feb 27 23:13:29] DEBUG[12211] chan_sip.c: Hanging up zombie call. Be scared.
[Feb 27 23:13:29] DEBUG[12211] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa3e3380'
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:29] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 1 (Not in use)
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: device 'SIP/21' state '1'
[Feb 27 23:13:29] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 1698705372@10.78.65.151 (Checking From) --From tag 1073600252 --To-tag as2a7ee005  
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: Stopping retransmission on '1698705372@10.78.65.151' of Response 2: Match Found
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: Destroying SIP dialog 1698705372@10.78.65.151
[Feb 27 23:13:29] DEBUG[11384] rtp_engine.c: Destroyed RTP instance '0xa3e3380'
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 5ee966a91d9c66793ab2b63b2a39c643@10.78.65.80:5060 (Checking To) --From tag as59caace1 --To-tag 486696185  
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: Acked pending invite 102
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: Stopping retransmission on '5ee966a91d9c66793ab2b63b2a39c643@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 5ee966a91d9c66793ab2b63b2a39c643@10.78.65.80:5060 (Checking To) --From tag as59caace1 --To-tag 486696185  
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: Stopping retransmission on '5ee966a91d9c66793ab2b63b2a39c643@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: SIP response 487 to standard invite
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: Trying to put 'ACK sip:20@' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5ee966a91d9c66793ab2b63b2a39c643@10.78.65.80:5060
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:29] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:13:29] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:13:29] DEBUG[11384] chan_sip.c: Destroying SIP dialog 5ee966a91d9c66793ab2b63b2a39c643@10.78.65.80:5060
[Feb 27 23:13:29] DEBUG[11384] rtp_engine.c: Destroyed RTP instance '0xa460738'
[Feb 27 23:13:29] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 2145734937@10.78.65.151 (Checking From) --From tag 53717407 --To-tag   
[Feb 27 23:13:36] DEBUG[11384] acl.c: For destination '10.78.65.151', our source address is '10.78.65.80'.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:36] VERBOSE[11384] netsock.c:   == Using UDPTL CoS mark 5
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Allocating new SIP dialog for 2145734937@10.78.65.151 - INVITE (No RTP)
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Feb 27 23:13:36] DEBUG[11384] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces"
[Feb 27 23:13:36] DEBUG[11384] sip/reqresp_parser.c: Found SIP option: -replaces-
[Feb 27 23:13:36] DEBUG[11384] sip/reqresp_parser.c: Matched SIP option: replaces
[Feb 27 23:13:36] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:36] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 2145734937@10.78.65.151 (Checking From) --From tag 53717407 --To-tag as511544f4  
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Stopping retransmission on '2145734937@10.78.65.151' of Response 1: Match Found
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 2145734937@10.78.65.151 (Checking From) --From tag 53717407 --To-tag   
[Feb 27 23:13:36] DEBUG[11384] netsock2.c: Splitting '10.78.65.80' into...
[Feb 27 23:13:36] DEBUG[11384] netsock2.c: ...host '10.78.65.80' and port ''.
[Feb 27 23:13:36] DEBUG[11384] netsock2.c: Splitting '10.78.65.80' into...
[Feb 27 23:13:36] DEBUG[11384] netsock2.c: ...host '10.78.65.80' and port ''.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Feb 27 23:13:36] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:36] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xa3e3380'
[Feb 27 23:13:36] DEBUG[11384] res_rtp_asterisk.c: Allocated port 10000 for RTP instance '0xa3e3380'
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: RTP instance '0xa3e3380' is setup and ready to go
[Feb 27 23:13:36] DEBUG[11384] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xa3e3380'
[Feb 27 23:13:36] VERBOSE[11384] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Setting NAT on RTP to Off
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing session-level SDP o=- 20083 20083 IN IP4 10.78.65.151... UNSUPPORTED.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing session-level SDP s=SDP data... UNSUPPORTED.
[Feb 27 23:13:36] DEBUG[11384] netsock2.c: Splitting '10.78.65.151' into...
[Feb 27 23:13:36] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port ''.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing session-level SDP c=IN IP4 10.78.65.151... OK.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Setting payload 8 based on m type on 0xaf668964
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Setting payload 0 based on m type on 0xaf668964
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Setting payload 18 based on m type on 0xaf668964
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Setting payload 9 based on m type on 0xaf668964
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Setting payload 102 based on m type on 0xaf668964
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Setting payload 101 based on m type on 0xaf668964
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:102 iLBC/8000... OK.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Incorporating payload 0 on 0xaf668964
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Incorporating payload 8 on 0xaf668964
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Incorporating payload 9 on 0xaf668964
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Incorporating payload 18 on 0xaf668964
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Incorporating payload 101 on 0xaf668964
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Incorporating payload 102 on 0xaf668964
[Feb 27 23:13:36] DEBUG[11384] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa3e3380'
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Copying payload 0 from 0xaf668964 to 0xa3e352c
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Copying payload 8 from 0xaf668964 to 0xa3e352c
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Copying payload 9 from 0xaf668964 to 0xa3e352c
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Copying payload 18 from 0xaf668964 to 0xa3e352c
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Copying payload 101 from 0xaf668964 to 0xa3e352c
[Feb 27 23:13:36] DEBUG[11384] rtp_engine.c: Copying payload 102 from 0xaf668964 to 0xa3e352c
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: We're settling with these formats: 0x110c (ulaw|alaw|g729|g722)
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Checking SIP call limits for device 21
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Updating call counter for incoming call
[Feb 27 23:13:36] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:36] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:36] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 2 (In use)
[Feb 27 23:13:36] DEBUG[11358] devicestate.c: device 'SIP/21' state '2'
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: *** Our native formats are 0x1000 (g722) 
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: *** Joint capabilities are 0x110c (ulaw|alaw|g729|g722) 
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: *** Our capabilities are 0x110e (gsm|ulaw|alaw|g729|g722) 
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) 
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: This channel will not be able to handle video.
[Feb 27 23:13:36] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: build_route: Contact hop: <sip:21@10.78.65.151:5062>
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: SIP/21-000000c0: New call is still down.... Trying... 
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:36] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:36] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:36] DEBUG[12212] pbx.c: Launching 'NoOp'
[Feb 27 23:13:36] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 2 (In use)
[Feb 27 23:13:36] VERBOSE[12212] pbx.c:     -- Executing [20@internal_default:1] NoOp("SIP/21-000000c0", "Internal SIP") in new stack
[Feb 27 23:13:36] DEBUG[11358] devicestate.c: device 'SIP/21' state '2'
[Feb 27 23:13:36] DEBUG[11359] app_queue.c: Extension '21@autohint' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:36] DEBUG[12212] pbx.c: Result of 'EXTEN' is '20'
[Feb 27 23:13:36] DEBUG[12212] pbx.c: Function result is '21'
[Feb 27 23:13:36] DEBUG[12212] pbx.c: Expression result is '0'
[Feb 27 23:13:36] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:36] DEBUG[12212] pbx.c: Launching 'ExecIf'
[Feb 27 23:13:36] VERBOSE[12212] pbx.c:     -- Executing [20@internal_default:2] ExecIf("SIP/21-000000c0", "0?Hangup(3)") in new stack
[Feb 27 23:13:36] DEBUG[12212] pbx.c: Result of 'EXTEN' is '20'
[Feb 27 23:13:36] DEBUG[12212] pbx.c: Launching 'Dial'
[Feb 27 23:13:36] VERBOSE[12212] pbx.c:     -- Executing [20@internal_default:3] Dial("SIP/21-000000c0", "SIP/20,60,j") in new stack
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Asked to create a SIP channel with formats: 0x1000 (g722)
[Feb 27 23:13:36] VERBOSE[12212] netsock.c:   == Using UDPTL CoS mark 5
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Allocating new SIP dialog for 25691cda3ef009912bfe1d12137eb238@10.78.65.80:5060 - INVITE (No RTP)
[Feb 27 23:13:36] DEBUG[12212] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xa460738'
[Feb 27 23:13:36] DEBUG[12212] res_rtp_asterisk.c: Allocated port 12922 for RTP instance '0xa460738'
[Feb 27 23:13:36] DEBUG[12212] rtp_engine.c: RTP instance '0xa460738' is setup and ready to go
[Feb 27 23:13:36] DEBUG[12212] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xa460738'
[Feb 27 23:13:36] VERBOSE[12212] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Setting NAT on RTP to Off
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:36] DEBUG[12212] acl.c: For destination '10.78.65.152', our source address is '10.78.65.80'.
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: *** Our native formats are 0x1000 (g722) 
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: *** Joint capabilities are 0x1000 (g722) 
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: *** Our capabilities are 0x110e (gsm|ulaw|alaw|g729|g722) 
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) 
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: *** Our preferred formats from the incoming channel are 0x1000 (g722) 
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: This channel will not be able to handle video.
[Feb 27 23:13:36] DEBUG[12212] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Feb 27 23:13:36] DEBUG[12212] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Feb 27 23:13:36] DEBUG[12212] rtp_engine.c: Seeded SDP of 'SIP/20-000000c1' with that of 'SIP/21-000000c0'
[Feb 27 23:13:36] DEBUG[12212] channel.c: Not copying variable DIALEDTIME.
[Feb 27 23:13:36] DEBUG[12212] channel.c: Not copying variable ANSWEREDTIME.
[Feb 27 23:13:36] DEBUG[12212] channel.c: Not copying variable DIALEDPEERNAME.
[Feb 27 23:13:36] DEBUG[12212] channel.c: Not copying variable DIALEDPEERNUMBER.
[Feb 27 23:13:36] DEBUG[12212] channel.c: Not copying variable DIALSTATUS.
[Feb 27 23:13:36] DEBUG[12212] channel.c: Not copying variable SIPCALLID.
[Feb 27 23:13:36] DEBUG[12212] channel.c: Not copying variable SIPDOMAIN.
[Feb 27 23:13:36] DEBUG[12212] channel.c: Not copying variable SIPURI.
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Outgoing Call for 20
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:36] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:36] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:36] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 6 (Ringing)
[Feb 27 23:13:36] DEBUG[11358] devicestate.c: device 'SIP/20' state '6'
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: ** Our capability: 0x100e (gsm|ulaw|alaw|g722) Video flag: False Text flag: False
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: ** Our prefcodec: 0x1000 (g722) 
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: -- Done with adding codecs to SDP
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Done building SDP. Settling with this capability: 0x100e (gsm|ulaw|alaw|g722)
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Initializing initreq for method INVITE - callid 0f2a680e1e3c146a7b1fce662523ddbf@10.78.65.80:5060
[Feb 27 23:13:36] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:36] VERBOSE[12212] app_dial.c:     -- Called SIP/20
[Feb 27 23:13:36] DEBUG[11359] app_queue.c: Extension '20@autohint' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 0f2a680e1e3c146a7b1fce662523ddbf@10.78.65.80:5060 (Checking To) --From tag as0d7dffa1 --To-tag   
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0f2a680e1e3c146a7b1fce662523ddbf@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: SIP response 100 to standard invite
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 0f2a680e1e3c146a7b1fce662523ddbf@10.78.65.80:5060 (Checking To) --From tag as0d7dffa1 --To-tag 1169685568  
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0f2a680e1e3c146a7b1fce662523ddbf@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:36] DEBUG[11384] chan_sip.c: SIP response 180 to standard invite
[Feb 27 23:13:36] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:36] VERBOSE[12212] app_dial.c:     -- SIP/20-000000c1 is ringing
[Feb 27 23:13:36] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:36] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 6 (Ringing)
[Feb 27 23:13:36] DEBUG[11358] devicestate.c: device 'SIP/20' state '6'
[Feb 27 23:13:36] DEBUG[12212] rtp_engine.c: Setting early bridge SDP of 'SIP/21-000000c0' with that of 'SIP/20-000000c1'
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:36] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:36] DEBUG[12212] channel.c: Driver for channel 'SIP/21-000000c0' does not support indication 3, emulating it
[Feb 27 23:13:36] DEBUG[12212] channel.c: Set channel SIP/21-000000c0 to write format slin
[Feb 27 23:13:36] DEBUG[12212] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Feb 27 23:13:36] DEBUG[12212] channel.c: Prodding channel 'SIP/21-000000c0'
[Feb 27 23:13:36] DEBUG[12212] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Setting framing from config on incoming call
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: ** Our capability: 0x110c (ulaw|alaw|g729|g722) Video flag: True Text flag: True
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: ** Our prefcodec: 0x0 (nothing) 
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: -- Done with adding codecs to SDP
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Done building SDP. Settling with this capability: 0x110c (ulaw|alaw|g729|g722)
[Feb 27 23:13:36] DEBUG[12212] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:36] DEBUG[12212] res_rtp_asterisk.c: Received frame with no data for RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:36] DEBUG[12212] res_rtp_asterisk.c: Ooh, format changed from unknown to g722
[Feb 27 23:13:36] DEBUG[12212] res_rtp_asterisk.c: Created smoother: format: g722 ms: 20 len: 160
[Feb 27 23:13:36] DEBUG[12212] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xa3e3380'
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:37] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: No remote address on RTP instance '0xa460738' so dropping frame
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 2145734937@10.78.65.151 (Checking From) --From tag 53717407 --To-tag   
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: **** Received CANCEL (14) - Command in SIP CANCEL
[Feb 27 23:13:38] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:38] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: Setting SIP_ALREADYGONE on dialog 2145734937@10.78.65.151
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: Updating call counter for incoming call
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:38] DEBUG[11384] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa3e3380'
[Feb 27 23:13:38] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 1 (Not in use)
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: device 'SIP/21' state '1'
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 487' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:38] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:38] DEBUG[12212] channel.c: Set channel SIP/21-000000c0 to write format g722
[Feb 27 23:13:38] DEBUG[12212] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Feb 27 23:13:38] DEBUG[12212] channel.c: Hanging up channel 'SIP/20-000000c1'
[Feb 27 23:13:38] DEBUG[12212] chan_sip.c: Hanging up zombie call. Be scared.
[Feb 27 23:13:38] DEBUG[12212] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:38] DEBUG[11359] app_queue.c: Extension '21@autohint' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:38] DEBUG[12212] chan_sip.c: Hanging up channel in state Ringing (not UP)
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa460738'
[Feb 27 23:13:38] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:38] DEBUG[12212] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0f2a680e1e3c146a7b1fce662523ddbf@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:13:38] DEBUG[12212] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:38] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:38] DEBUG[12212] app_dial.c: Exiting with DIALSTATUS=CANCEL.
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:38] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:13:38] DEBUG[12212] pbx.c: Spawn extension (internal_default,20,3) exited non-zero on 'SIP/21-000000c0'
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:13:38] VERBOSE[12212] pbx.c:   == Spawn extension (internal_default, 20, 3) exited non-zero on 'SIP/21-000000c0'
[Feb 27 23:13:38] DEBUG[12212] channel.c: Soft-Hanging up channel 'SIP/21-000000c0'
[Feb 27 23:13:38] DEBUG[12212] pbx.c: Launching 'Hangup'
[Feb 27 23:13:38] VERBOSE[12212] pbx.c:     -- Executing [h@internal_default:1] Hangup("SIP/21-000000c0", "") in new stack
[Feb 27 23:13:38] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:38] DEBUG[11359] app_queue.c: Extension '20@autohint' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:38] DEBUG[12212] pbx.c: Spawn extension (internal_default,h,1) exited non-zero on 'SIP/21-000000c0'
[Feb 27 23:13:38] VERBOSE[12212] pbx.c:   == Spawn extension (internal_default, h, 1) exited non-zero on 'SIP/21-000000c0'
[Feb 27 23:13:38] DEBUG[12212] channel.c: Hanging up channel 'SIP/21-000000c0'
[Feb 27 23:13:38] DEBUG[12212] chan_sip.c: Hanging up zombie call. Be scared.
[Feb 27 23:13:38] DEBUG[12212] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa3e3380'
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 0f2a680e1e3c146a7b1fce662523ddbf@10.78.65.80:5060 (Checking To) --From tag as0d7dffa1 --To-tag 1169685568  
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: Acked pending invite 102
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: Stopping retransmission on '0f2a680e1e3c146a7b1fce662523ddbf@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:38] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 1 (Not in use)
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: device 'SIP/21' state '1'
[Feb 27 23:13:38] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 0f2a680e1e3c146a7b1fce662523ddbf@10.78.65.80:5060 (Checking To) --From tag as0d7dffa1 --To-tag 1169685568  
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: Stopping retransmission on '0f2a680e1e3c146a7b1fce662523ddbf@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: SIP response 487 to standard invite
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: Trying to put 'ACK sip:20@' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: Setting SIP_ALREADYGONE on dialog 0f2a680e1e3c146a7b1fce662523ddbf@10.78.65.80:5060
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:38] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:13:38] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: Destroying SIP dialog 0f2a680e1e3c146a7b1fce662523ddbf@10.78.65.80:5060
[Feb 27 23:13:38] DEBUG[11384] rtp_engine.c: Destroyed RTP instance '0xa460738'
[Feb 27 23:13:38] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 2145734937@10.78.65.151 (Checking From) --From tag 53717407 --To-tag as6a01b0df  
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: Stopping retransmission on '2145734937@10.78.65.151' of Response 2: Match Found
[Feb 27 23:13:38] DEBUG[11384] chan_sip.c: Destroying SIP dialog 2145734937@10.78.65.151
[Feb 27 23:13:38] DEBUG[11384] rtp_engine.c: Destroyed RTP instance '0xa3e3380'
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Allocating new SIP dialog for 163e9c28398f081e1ddb6453053b5033@10.78.65.80:5060 - OPTIONS (No RTP)
[Feb 27 23:13:40] DEBUG[11384] acl.c: For destination '192.168.1.20', our source address is '10.78.65.80'.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Initializing initreq for method OPTIONS - callid 1435dca21980e02e031324f902cfafe4@10.78.65.80:5060
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.20:5060
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 1435dca21980e02e031324f902cfafe4@10.78.65.80:5060 (Checking To) --From tag as17d11e94 --To-tag t-7a0e0860695346053e3  
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Stopping retransmission on '1435dca21980e02e031324f902cfafe4@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Destroying SIP dialog 1435dca21980e02e031324f902cfafe4@10.78.65.80:5060
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 882129746@10.78.65.151 (Checking From) --From tag 1730297156 --To-tag   
[Feb 27 23:13:40] DEBUG[11384] acl.c: For destination '10.78.65.151', our source address is '10.78.65.80'.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:40] VERBOSE[11384] netsock.c:   == Using UDPTL CoS mark 5
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Allocating new SIP dialog for 882129746@10.78.65.151 - INVITE (No RTP)
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Feb 27 23:13:40] DEBUG[11384] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces"
[Feb 27 23:13:40] DEBUG[11384] sip/reqresp_parser.c: Found SIP option: -replaces-
[Feb 27 23:13:40] DEBUG[11384] sip/reqresp_parser.c: Matched SIP option: replaces
[Feb 27 23:13:40] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:40] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 882129746@10.78.65.151 (Checking From) --From tag 1730297156 --To-tag as2b35e265  
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Stopping retransmission on '882129746@10.78.65.151' of Response 1: Match Found
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 882129746@10.78.65.151 (Checking From) --From tag 1730297156 --To-tag   
[Feb 27 23:13:40] DEBUG[11384] netsock2.c: Splitting '10.78.65.80' into...
[Feb 27 23:13:40] DEBUG[11384] netsock2.c: ...host '10.78.65.80' and port ''.
[Feb 27 23:13:40] DEBUG[11384] netsock2.c: Splitting '10.78.65.80' into...
[Feb 27 23:13:40] DEBUG[11384] netsock2.c: ...host '10.78.65.80' and port ''.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Feb 27 23:13:40] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:40] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xaf162880'
[Feb 27 23:13:40] DEBUG[11384] res_rtp_asterisk.c: Allocated port 10130 for RTP instance '0xaf162880'
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: RTP instance '0xaf162880' is setup and ready to go
[Feb 27 23:13:40] DEBUG[11384] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xaf162880'
[Feb 27 23:13:40] VERBOSE[11384] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Setting NAT on RTP to Off
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing session-level SDP o=- 20084 20084 IN IP4 10.78.65.151... UNSUPPORTED.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing session-level SDP s=SDP data... UNSUPPORTED.
[Feb 27 23:13:40] DEBUG[11384] netsock2.c: Splitting '10.78.65.151' into...
[Feb 27 23:13:40] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port ''.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing session-level SDP c=IN IP4 10.78.65.151... OK.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Setting payload 8 based on m type on 0xaf668964
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Setting payload 0 based on m type on 0xaf668964
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Setting payload 18 based on m type on 0xaf668964
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Setting payload 9 based on m type on 0xaf668964
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Setting payload 102 based on m type on 0xaf668964
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Setting payload 101 based on m type on 0xaf668964
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:102 iLBC/8000... OK.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Incorporating payload 0 on 0xaf668964
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Incorporating payload 8 on 0xaf668964
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Incorporating payload 9 on 0xaf668964
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Incorporating payload 18 on 0xaf668964
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Incorporating payload 101 on 0xaf668964
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Incorporating payload 102 on 0xaf668964
[Feb 27 23:13:40] DEBUG[11384] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaf162880'
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Copying payload 0 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Copying payload 8 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Copying payload 9 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Copying payload 18 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Copying payload 101 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:40] DEBUG[11384] rtp_engine.c: Copying payload 102 from 0xaf668964 to 0xaf162a2c
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: We're settling with these formats: 0x110c (ulaw|alaw|g729|g722)
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Checking SIP call limits for device 21
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Updating call counter for incoming call
[Feb 27 23:13:40] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:40] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:40] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 2 (In use)
[Feb 27 23:13:40] DEBUG[11358] devicestate.c: device 'SIP/21' state '2'
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: *** Our native formats are 0x1000 (g722) 
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: *** Joint capabilities are 0x110c (ulaw|alaw|g729|g722) 
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: *** Our capabilities are 0x110e (gsm|ulaw|alaw|g729|g722) 
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) 
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: This channel will not be able to handle video.
[Feb 27 23:13:40] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: build_route: Contact hop: <sip:21@10.78.65.151:5062>
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: SIP/21-000000c2: New call is still down.... Trying... 
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:40] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:40] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:40] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 2 (In use)
[Feb 27 23:13:40] DEBUG[12213] pbx.c: Launching 'NoOp'
[Feb 27 23:13:40] DEBUG[11358] devicestate.c: device 'SIP/21' state '2'
[Feb 27 23:13:40] VERBOSE[12213] pbx.c:     -- Executing [20@internal_default:1] NoOp("SIP/21-000000c2", "Internal SIP") in new stack
[Feb 27 23:13:40] DEBUG[11359] app_queue.c: Extension '21@autohint' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:40] DEBUG[12213] pbx.c: Result of 'EXTEN' is '20'
[Feb 27 23:13:40] DEBUG[12213] pbx.c: Function result is '21'
[Feb 27 23:13:40] DEBUG[12213] pbx.c: Expression result is '0'
[Feb 27 23:13:40] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:40] DEBUG[12213] pbx.c: Launching 'ExecIf'
[Feb 27 23:13:40] VERBOSE[12213] pbx.c:     -- Executing [20@internal_default:2] ExecIf("SIP/21-000000c2", "0?Hangup(3)") in new stack
[Feb 27 23:13:40] DEBUG[12213] pbx.c: Result of 'EXTEN' is '20'
[Feb 27 23:13:40] DEBUG[12213] pbx.c: Launching 'Dial'
[Feb 27 23:13:40] VERBOSE[12213] pbx.c:     -- Executing [20@internal_default:3] Dial("SIP/21-000000c2", "SIP/20,60,j") in new stack
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Asked to create a SIP channel with formats: 0x1000 (g722)
[Feb 27 23:13:40] VERBOSE[12213] netsock.c:   == Using UDPTL CoS mark 5
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Allocating new SIP dialog for 18f00973645fc72b333bdce00dd99c02@10.78.65.80:5060 - INVITE (No RTP)
[Feb 27 23:13:40] DEBUG[12213] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xa3e3380'
[Feb 27 23:13:40] DEBUG[12213] res_rtp_asterisk.c: Allocated port 16618 for RTP instance '0xa3e3380'
[Feb 27 23:13:40] DEBUG[12213] rtp_engine.c: RTP instance '0xa3e3380' is setup and ready to go
[Feb 27 23:13:40] DEBUG[12213] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xa3e3380'
[Feb 27 23:13:40] VERBOSE[12213] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Setting NAT on RTP to Off
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:40] DEBUG[12213] acl.c: For destination '10.78.65.152', our source address is '10.78.65.80'.
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: *** Our native formats are 0x1000 (g722) 
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: *** Joint capabilities are 0x1000 (g722) 
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: *** Our capabilities are 0x110e (gsm|ulaw|alaw|g729|g722) 
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) 
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: *** Our preferred formats from the incoming channel are 0x1000 (g722) 
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: This channel will not be able to handle video.
[Feb 27 23:13:40] DEBUG[12213] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Feb 27 23:13:40] DEBUG[12213] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Feb 27 23:13:40] DEBUG[12213] rtp_engine.c: Seeded SDP of 'SIP/20-000000c3' with that of 'SIP/21-000000c2'
[Feb 27 23:13:40] DEBUG[12213] channel.c: Not copying variable DIALEDTIME.
[Feb 27 23:13:40] DEBUG[12213] channel.c: Not copying variable ANSWEREDTIME.
[Feb 27 23:13:40] DEBUG[12213] channel.c: Not copying variable DIALEDPEERNAME.
[Feb 27 23:13:40] DEBUG[12213] channel.c: Not copying variable DIALEDPEERNUMBER.
[Feb 27 23:13:40] DEBUG[12213] channel.c: Not copying variable DIALSTATUS.
[Feb 27 23:13:40] DEBUG[12213] channel.c: Not copying variable SIPCALLID.
[Feb 27 23:13:40] DEBUG[12213] channel.c: Not copying variable SIPDOMAIN.
[Feb 27 23:13:40] DEBUG[12213] channel.c: Not copying variable SIPURI.
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Outgoing Call for 20
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:40] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:40] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:40] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 6 (Ringing)
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: ** Our capability: 0x100e (gsm|ulaw|alaw|g722) Video flag: False Text flag: False
[Feb 27 23:13:40] DEBUG[11358] devicestate.c: device 'SIP/20' state '6'
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: ** Our prefcodec: 0x1000 (g722) 
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: -- Done with adding codecs to SDP
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Done building SDP. Settling with this capability: 0x100e (gsm|ulaw|alaw|g722)
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Initializing initreq for method INVITE - callid 12d016e30489075f747b2a7b0279bd49@10.78.65.80:5060
[Feb 27 23:13:40] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:40] VERBOSE[12213] app_dial.c:     -- Called SIP/20
[Feb 27 23:13:40] DEBUG[11359] app_queue.c: Extension '20@autohint' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 12d016e30489075f747b2a7b0279bd49@10.78.65.80:5060 (Checking To) --From tag as5b34985d --To-tag   
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '12d016e30489075f747b2a7b0279bd49@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: SIP response 100 to standard invite
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 12d016e30489075f747b2a7b0279bd49@10.78.65.80:5060 (Checking To) --From tag as5b34985d --To-tag 1382680134  
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '12d016e30489075f747b2a7b0279bd49@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:40] DEBUG[11384] chan_sip.c: SIP response 180 to standard invite
[Feb 27 23:13:40] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:40] VERBOSE[12213] app_dial.c:     -- SIP/20-000000c3 is ringing
[Feb 27 23:13:40] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:40] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 6 (Ringing)
[Feb 27 23:13:40] DEBUG[11358] devicestate.c: device 'SIP/20' state '6'
[Feb 27 23:13:40] DEBUG[12213] rtp_engine.c: Setting early bridge SDP of 'SIP/21-000000c2' with that of 'SIP/20-000000c3'
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:40] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:40] DEBUG[12213] channel.c: Driver for channel 'SIP/21-000000c2' does not support indication 3, emulating it
[Feb 27 23:13:40] DEBUG[12213] channel.c: Set channel SIP/21-000000c2 to write format slin
[Feb 27 23:13:40] DEBUG[12213] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Feb 27 23:13:40] DEBUG[12213] channel.c: Prodding channel 'SIP/21-000000c2'
[Feb 27 23:13:40] DEBUG[12213] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Setting framing from config on incoming call
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: ** Our capability: 0x110c (ulaw|alaw|g729|g722) Video flag: True Text flag: True
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: ** Our prefcodec: 0x0 (nothing) 
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: -- Done with adding codecs to SDP
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Done building SDP. Settling with this capability: 0x110c (ulaw|alaw|g729|g722)
[Feb 27 23:13:40] DEBUG[12213] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:40] DEBUG[12213] res_rtp_asterisk.c: Received frame with no data for RTP instance '0xaf162880' so dropping frame
[Feb 27 23:13:40] DEBUG[12213] res_rtp_asterisk.c: Ooh, format changed from unknown to g722
[Feb 27 23:13:40] DEBUG[12213] res_rtp_asterisk.c: Created smoother: format: g722 ms: 20 len: 160
[Feb 27 23:13:40] DEBUG[12213] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0xaf162880'
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:41] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[12213] res_rtp_asterisk.c: No remote address on RTP instance '0xa3e3380' so dropping frame
[Feb 27 23:13:42] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 882129746@10.78.65.151 (Checking From) --From tag 1730297156 --To-tag   
[Feb 27 23:13:42] DEBUG[11384] chan_sip.c: **** Received CANCEL (14) - Command in SIP CANCEL
[Feb 27 23:13:42] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:42] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:42] DEBUG[11384] chan_sip.c: Setting SIP_ALREADYGONE on dialog 882129746@10.78.65.151
[Feb 27 23:13:42] DEBUG[11384] chan_sip.c: Updating call counter for incoming call
[Feb 27 23:13:42] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:42] DEBUG[11384] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaf162880'
[Feb 27 23:13:42] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:42] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 1 (Not in use)
[Feb 27 23:13:42] DEBUG[11358] devicestate.c: device 'SIP/21' state '1'
[Feb 27 23:13:42] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 487' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:42] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:42] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:42] DEBUG[12213] channel.c: Set channel SIP/21-000000c2 to write format g722
[Feb 27 23:13:42] DEBUG[12213] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Feb 27 23:13:42] DEBUG[12213] res_timing_timerfd.c: Avoiding read on disarmed timerfd 31
[Feb 27 23:13:42] DEBUG[12213] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 0 instead
[Feb 27 23:13:42] DEBUG[11359] app_queue.c: Extension '21@autohint' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:42] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 882129746@10.78.65.151 (Checking From) --From tag 1730297156 --To-tag as4c10f291  
[Feb 27 23:13:42] DEBUG[11384] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Feb 27 23:13:42] DEBUG[11384] chan_sip.c: Stopping retransmission on '882129746@10.78.65.151' of Response 2: Match Found
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 1711996071@10.78.65.151 (Checking From) --From tag 1966316179 --To-tag   
[Feb 27 23:13:44] DEBUG[11384] acl.c: For destination '10.78.65.151', our source address is '10.78.65.80'.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:44] VERBOSE[11384] netsock.c:   == Using UDPTL CoS mark 5
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Allocating new SIP dialog for 1711996071@10.78.65.151 - INVITE (No RTP)
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Feb 27 23:13:44] DEBUG[11384] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces"
[Feb 27 23:13:44] DEBUG[11384] sip/reqresp_parser.c: Found SIP option: -replaces-
[Feb 27 23:13:44] DEBUG[11384] sip/reqresp_parser.c: Matched SIP option: replaces
[Feb 27 23:13:44] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:44] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 1711996071@10.78.65.151 (Checking From) --From tag 1966316179 --To-tag as6b1a5cc0  
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Stopping retransmission on '1711996071@10.78.65.151' of Response 1: Match Found
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 1711996071@10.78.65.151 (Checking From) --From tag 1966316179 --To-tag   
[Feb 27 23:13:44] DEBUG[11384] netsock2.c: Splitting '10.78.65.80' into...
[Feb 27 23:13:44] DEBUG[11384] netsock2.c: ...host '10.78.65.80' and port ''.
[Feb 27 23:13:44] DEBUG[11384] netsock2.c: Splitting '10.78.65.80' into...
[Feb 27 23:13:44] DEBUG[11384] netsock2.c: ...host '10.78.65.80' and port ''.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Feb 27 23:13:44] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:44] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xaee66520'
[Feb 27 23:13:44] DEBUG[11384] res_rtp_asterisk.c: Allocated port 13112 for RTP instance '0xaee66520'
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: RTP instance '0xaee66520' is setup and ready to go
[Feb 27 23:13:44] DEBUG[11384] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xaee66520'
[Feb 27 23:13:44] VERBOSE[11384] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Setting NAT on RTP to Off
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing session-level SDP o=- 20085 20085 IN IP4 10.78.65.151... UNSUPPORTED.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing session-level SDP s=SDP data... UNSUPPORTED.
[Feb 27 23:13:44] DEBUG[11384] netsock2.c: Splitting '10.78.65.151' into...
[Feb 27 23:13:44] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port ''.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing session-level SDP c=IN IP4 10.78.65.151... OK.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Setting payload 8 based on m type on 0xaf668964
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Setting payload 0 based on m type on 0xaf668964
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Setting payload 18 based on m type on 0xaf668964
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Setting payload 9 based on m type on 0xaf668964
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Setting payload 102 based on m type on 0xaf668964
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Setting payload 101 based on m type on 0xaf668964
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:102 iLBC/8000... OK.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Incorporating payload 0 on 0xaf668964
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Incorporating payload 8 on 0xaf668964
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Incorporating payload 9 on 0xaf668964
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Incorporating payload 18 on 0xaf668964
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Incorporating payload 101 on 0xaf668964
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Incorporating payload 102 on 0xaf668964
[Feb 27 23:13:44] DEBUG[11384] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaee66520'
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Copying payload 0 from 0xaf668964 to 0xaee666cc
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Copying payload 8 from 0xaf668964 to 0xaee666cc
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Copying payload 9 from 0xaf668964 to 0xaee666cc
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Copying payload 18 from 0xaf668964 to 0xaee666cc
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Copying payload 101 from 0xaf668964 to 0xaee666cc
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Copying payload 102 from 0xaf668964 to 0xaee666cc
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: We're settling with these formats: 0x110c (ulaw|alaw|g729|g722)
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Checking SIP call limits for device 21
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Updating call counter for incoming call
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:44] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 2 (In use)
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: device 'SIP/21' state '2'
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: *** Our native formats are 0x1000 (g722) 
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: *** Joint capabilities are 0x110c (ulaw|alaw|g729|g722) 
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: *** Our capabilities are 0x110e (gsm|ulaw|alaw|g729|g722) 
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) 
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: This channel will not be able to handle video.
[Feb 27 23:13:44] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: build_route: Contact hop: <sip:21@10.78.65.151:5062>
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: SIP/21-000000c4: New call is still down.... Trying... 
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:44] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 2 (In use)
[Feb 27 23:13:44] DEBUG[12214] pbx.c: Launching 'NoOp'
[Feb 27 23:13:44] DEBUG[11359] app_queue.c: Extension '21@autohint' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: device 'SIP/21' state '2'
[Feb 27 23:13:44] VERBOSE[12214] pbx.c:     -- Executing [20@internal_default:1] NoOp("SIP/21-000000c4", "Internal SIP") in new stack
[Feb 27 23:13:44] DEBUG[12214] pbx.c: Result of 'EXTEN' is '20'
[Feb 27 23:13:44] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:44] DEBUG[12214] pbx.c: Function result is '21'
[Feb 27 23:13:44] DEBUG[12214] pbx.c: Expression result is '0'
[Feb 27 23:13:44] DEBUG[12214] pbx.c: Launching 'ExecIf'
[Feb 27 23:13:44] VERBOSE[12214] pbx.c:     -- Executing [20@internal_default:2] ExecIf("SIP/21-000000c4", "0?Hangup(3)") in new stack
[Feb 27 23:13:44] DEBUG[12214] pbx.c: Result of 'EXTEN' is '20'
[Feb 27 23:13:44] DEBUG[12214] pbx.c: Launching 'Dial'
[Feb 27 23:13:44] VERBOSE[12214] pbx.c:     -- Executing [20@internal_default:3] Dial("SIP/21-000000c4", "SIP/20,60,j") in new stack
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: Asked to create a SIP channel with formats: 0x1000 (g722)
[Feb 27 23:13:44] VERBOSE[12214] netsock.c:   == Using UDPTL CoS mark 5
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: Allocating new SIP dialog for 20a6bbc924fc4be70759787923837905@10.78.65.80:5060 - INVITE (No RTP)
[Feb 27 23:13:44] DEBUG[12214] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xa460738'
[Feb 27 23:13:44] DEBUG[12214] res_rtp_asterisk.c: Allocated port 13736 for RTP instance '0xa460738'
[Feb 27 23:13:44] DEBUG[12214] rtp_engine.c: RTP instance '0xa460738' is setup and ready to go
[Feb 27 23:13:44] DEBUG[12214] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xa460738'
[Feb 27 23:13:44] VERBOSE[12214] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: Setting NAT on RTP to Off
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: Setting NAT on UDPTL to Off
[Feb 27 23:13:44] DEBUG[12214] acl.c: For destination '10.78.65.152', our source address is '10.78.65.80'.
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: *** Our native formats are 0x1000 (g722) 
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: *** Joint capabilities are 0x1000 (g722) 
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: *** Our capabilities are 0x110e (gsm|ulaw|alaw|g729|g722) 
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) 
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: *** Our preferred formats from the incoming channel are 0x1000 (g722) 
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: This channel will not be able to handle video.
[Feb 27 23:13:44] DEBUG[12214] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Feb 27 23:13:44] DEBUG[12214] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Feb 27 23:13:44] DEBUG[12214] rtp_engine.c: Seeded SDP of 'SIP/20-000000c5' with that of 'SIP/21-000000c4'
[Feb 27 23:13:44] DEBUG[12214] channel.c: Not copying variable DIALEDTIME.
[Feb 27 23:13:44] DEBUG[12214] channel.c: Not copying variable ANSWEREDTIME.
[Feb 27 23:13:44] DEBUG[12214] channel.c: Not copying variable DIALEDPEERNAME.
[Feb 27 23:13:44] DEBUG[12214] channel.c: Not copying variable DIALEDPEERNUMBER.
[Feb 27 23:13:44] DEBUG[12214] channel.c: Not copying variable DIALSTATUS.
[Feb 27 23:13:44] DEBUG[12214] channel.c: Not copying variable SIPCALLID.
[Feb 27 23:13:44] DEBUG[12214] channel.c: Not copying variable SIPDOMAIN.
[Feb 27 23:13:44] DEBUG[12214] channel.c: Not copying variable SIPURI.
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: Outgoing Call for 20
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:44] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 6 (Ringing)
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: ** Our capability: 0x100e (gsm|ulaw|alaw|g722) Video flag: False Text flag: False
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: device 'SIP/20' state '6'
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: ** Our prefcodec: 0x1000 (g722) 
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: -- Done with adding codecs to SDP
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: Done building SDP. Settling with this capability: 0x100e (gsm|ulaw|alaw|g722)
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: Initializing initreq for method INVITE - callid 50511a7045fb7601269de75148f6d8a2@10.78.65.80:5060
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:44] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:44] VERBOSE[12214] app_dial.c:     -- Called SIP/20
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 50511a7045fb7601269de75148f6d8a2@10.78.65.80:5060 (Checking To) --From tag as783eb243 --To-tag   
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '50511a7045fb7601269de75148f6d8a2@10.78.65.80:5060' Request 102: Found
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: SIP response 100 to standard invite
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 50511a7045fb7601269de75148f6d8a2@10.78.65.80:5060 (Checking To) --From tag as783eb243 --To-tag 160960065  
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Acked pending invite 102
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Stopping retransmission on '50511a7045fb7601269de75148f6d8a2@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:13:44] VERBOSE[11384] chan_sip.c:     -- Got SIP response 486 "Busy Here" back from 10.78.65.152:5062
[Feb 27 23:13:44] DEBUG[11384] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa460738'
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Trying to put 'ACK sip:20@' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Setting SIP_ALREADYGONE on dialog 50511a7045fb7601269de75148f6d8a2@10.78.65.80:5060
[Feb 27 23:13:44] VERBOSE[12214] app_dial.c:     -- SIP/20-000000c5 is busy
[Feb 27 23:13:44] DEBUG[12214] channel.c: Hanging up channel 'SIP/20-000000c5'
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: Hanging up zombie call. Be scared.
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:13:44] DEBUG[12214] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa460738'
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:44] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 6 (Ringing)
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: device 'SIP/20' state '6'
[Feb 27 23:13:44] VERBOSE[12214] app_dial.c:   == Everyone is busy/congested at this time (1:1/0/0)
[Feb 27 23:13:44] DEBUG[12214] app_dial.c: Exiting with DIALSTATUS=BUSY.
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:13:44] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 6 (Ringing)
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: device 'SIP/20' state '6'
[Feb 27 23:13:44] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:44] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 1711996071@10.78.65.151 (Checking From) --From tag 1966316179 --To-tag   
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: **** Received CANCEL (14) - Command in SIP CANCEL
[Feb 27 23:13:44] DEBUG[11384] netsock2.c: Splitting '10.78.65.151:5062' into...
[Feb 27 23:13:44] DEBUG[11384] netsock2.c: ...host '10.78.65.151' and port '5062'.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Setting SIP_ALREADYGONE on dialog 1711996071@10.78.65.151
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Updating call counter for incoming call
[Feb 27 23:13:44] DEBUG[11384] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaee66520'
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:44] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 1 (Not in use)
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: device 'SIP/21' state '1'
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 487' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:13:44] WARNING[12214] pbx.c: Don't know what to do with 'SIP/21-000000c4'
[Feb 27 23:13:44] DEBUG[12214] channel.c: Soft-Hanging up channel 'SIP/21-000000c4'
[Feb 27 23:13:44] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:44] DEBUG[12214] pbx.c: Launching 'Hangup'
[Feb 27 23:13:44] VERBOSE[12214] pbx.c:     -- Executing [h@internal_default:1] Hangup("SIP/21-000000c4", "") in new stack
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Destroying SIP dialog 50511a7045fb7601269de75148f6d8a2@10.78.65.80:5060
[Feb 27 23:13:44] DEBUG[12214] pbx.c: Spawn extension (internal_default,h,1) exited non-zero on 'SIP/21-000000c4'
[Feb 27 23:13:44] VERBOSE[12214] pbx.c:   == Spawn extension (internal_default, h, 1) exited non-zero on 'SIP/21-000000c4'
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Destroyed RTP instance '0xa460738'
[Feb 27 23:13:44] DEBUG[12214] channel.c: Hanging up channel 'SIP/21-000000c4'
[Feb 27 23:13:44] DEBUG[12214] chan_sip.c: Hanging up zombie call. Be scared.
[Feb 27 23:13:44] DEBUG[12214] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaee66520'
[Feb 27 23:13:44] DEBUG[11359] app_queue.c: Extension '21@autohint' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:13:44] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 1 (Not in use)
[Feb 27 23:13:44] DEBUG[11358] devicestate.c: device 'SIP/21' state '1'
[Feb 27 23:13:44] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 1711996071@10.78.65.151 (Checking From) --From tag 1966316179 --To-tag as0753e721  
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Stopping retransmission on '1711996071@10.78.65.151' of Response 2: Match Found
[Feb 27 23:13:44] DEBUG[11384] chan_sip.c: Destroying SIP dialog 1711996071@10.78.65.151
[Feb 27 23:13:44] DEBUG[11384] rtp_engine.c: Destroyed RTP instance '0xaee66520'
[Feb 27 23:13:45] DEBUG[11384] chan_sip.c: Auto destroying SIP dialog '4938efa65c1c5bf3753cc07c5103b9ea@netforce.ath.cx'
[Feb 27 23:13:45] DEBUG[11384] chan_sip.c: Destroying SIP dialog 4938efa65c1c5bf3753cc07c5103b9ea@netforce.ath.cx
[Feb 27 23:13:51] DEBUG[11384] chan_sip.c: Allocating new SIP dialog for 1106d71b3040aa00403b82252707c4aa@10.78.65.80:5060 - OPTIONS (No RTP)
[Feb 27 23:13:51] DEBUG[11384] acl.c: For destination '10.78.65.152', our source address is '10.78.65.80'.
[Feb 27 23:13:51] DEBUG[11384] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:13:51] DEBUG[11384] chan_sip.c: Initializing initreq for method OPTIONS - callid 658bc4bd4e1e42e125e3bdfc3d62f684@10.78.65.80:5060
[Feb 27 23:13:51] DEBUG[11384] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:13:51] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 658bc4bd4e1e42e125e3bdfc3d62f684@10.78.65.80:5060 (Checking To) --From tag as4f6b2b1e --To-tag 1271361473  
[Feb 27 23:13:51] DEBUG[11384] chan_sip.c: Stopping retransmission on '658bc4bd4e1e42e125e3bdfc3d62f684@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:13:51] DEBUG[11384] chan_sip.c: Destroying SIP dialog 658bc4bd4e1e42e125e3bdfc3d62f684@10.78.65.80:5060
[Feb 27 23:14:01] DEBUG[11384] chan_sip.c: Allocating new SIP dialog for 122c1304304ceb2c26e37e18124eef71@10.78.65.80:5060 - OPTIONS (No RTP)
[Feb 27 23:14:01] DEBUG[11384] acl.c: For destination '10.78.65.153', our source address is '10.78.65.80'.
[Feb 27 23:14:01] DEBUG[11384] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:14:01] DEBUG[11384] chan_sip.c: Initializing initreq for method OPTIONS - callid 500214bc33d20dd44d6b23c91d2ea081@10.78.65.80:5060
[Feb 27 23:14:01] DEBUG[11384] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.78.65.153:5060
[Feb 27 23:14:01] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 500214bc33d20dd44d6b23c91d2ea081@10.78.65.80:5060 (Checking To) --From tag as6644ab67 --To-tag 3989879984  
[Feb 27 23:14:01] DEBUG[11384] chan_sip.c: Stopping retransmission on '500214bc33d20dd44d6b23c91d2ea081@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:14:01] DEBUG[11384] chan_sip.c: Destroying SIP dialog 500214bc33d20dd44d6b23c91d2ea081@10.78.65.80:5060
[Feb 27 23:14:03] DEBUG[11384] chan_sip.c: Allocating new SIP dialog for 324794ba7608ae5205ea5645539770cd@10.78.65.80:5060 - OPTIONS (No RTP)
[Feb 27 23:14:03] DEBUG[11384] acl.c: For destination '10.78.65.151', our source address is '10.78.65.80'.
[Feb 27 23:14:03] DEBUG[11384] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:14:03] DEBUG[11384] chan_sip.c: Initializing initreq for method OPTIONS - callid 33cc14fa3ce2302159328d9f63e9dab9@10.78.65.80:5060
[Feb 27 23:14:03] DEBUG[11384] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.78.65.151:5062
[Feb 27 23:14:03] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 33cc14fa3ce2302159328d9f63e9dab9@10.78.65.80:5060 (Checking To) --From tag as6326839e --To-tag 1813951392  
[Feb 27 23:14:03] DEBUG[11384] chan_sip.c: Stopping retransmission on '33cc14fa3ce2302159328d9f63e9dab9@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:14:03] DEBUG[11384] chan_sip.c: Destroying SIP dialog 33cc14fa3ce2302159328d9f63e9dab9@10.78.65.80:5060
[Feb 27 23:14:11] DEBUG[11384] chan_sip.c: Allocating new SIP dialog for 6a8439f65eddab03138bb6f46aed2cec@10.78.65.80:5060 - OPTIONS (No RTP)
[Feb 27 23:14:11] DEBUG[11384] acl.c: For destination '192.168.1.80', our source address is '10.78.65.80'.
[Feb 27 23:14:11] DEBUG[11384] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:14:11] DEBUG[11384] chan_sip.c: Initializing initreq for method OPTIONS - callid 4dd6c5834fd7b3af65d325a008806bbd@10.78.65.80:5060
[Feb 27 23:14:11] DEBUG[11384] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.80:5060
[Feb 27 23:14:11] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 4dd6c5834fd7b3af65d325a008806bbd@10.78.65.80:5060 (Checking To) --From tag as0df01860 --To-tag as70e13295  
[Feb 27 23:14:11] DEBUG[11384] chan_sip.c: Stopping retransmission on '4dd6c5834fd7b3af65d325a008806bbd@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:14:11] DEBUG[11384] chan_sip.c: Destroying SIP dialog 4dd6c5834fd7b3af65d325a008806bbd@10.78.65.80:5060
[Feb 27 23:14:13] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 4035f4e316b589eb2330d70b70927fbb@netforce.ath.cx (Checking From) --From tag as3d800bf9 --To-tag   
[Feb 27 23:14:13] DEBUG[11384] acl.c: For destination '192.168.1.80', our source address is '10.78.65.80'.
[Feb 27 23:14:13] DEBUG[11384] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:14:13] DEBUG[11384] chan_sip.c: Allocating new SIP dialog for 4035f4e316b589eb2330d70b70927fbb@netforce.ath.cx - OPTIONS (No RTP)
[Feb 27 23:14:13] DEBUG[11384] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
[Feb 27 23:14:13] DEBUG[11384] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 192.168.1.80:5060
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Allocating new SIP dialog for 21ea91b71d3ae4c72490db4e124d3208@10.78.65.80:5060 - OPTIONS (No RTP)
[Feb 27 23:14:40] DEBUG[11384] acl.c: For destination '192.168.1.20', our source address is '10.78.65.80'.
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 10.78.65.80:5060
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Initializing initreq for method OPTIONS - callid 6e742bb0715ba5ca1718f51d2489e2dc@10.78.65.80:5060
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.1.20:5060
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 6e742bb0715ba5ca1718f51d2489e2dc@10.78.65.80:5060 (Checking To) --From tag as64822b69 --To-tag t-7a0e56807af9c12f3e4  
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Stopping retransmission on '6e742bb0715ba5ca1718f51d2489e2dc@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Destroying SIP dialog 6e742bb0715ba5ca1718f51d2489e2dc@10.78.65.80:5060
[Feb 27 23:14:40] VERBOSE[12213] app_dial.c:     -- Nobody picked up in 60000 ms
[Feb 27 23:14:40] DEBUG[12213] channel.c: Hanging up channel 'SIP/20-000000c3'
[Feb 27 23:14:40] DEBUG[12213] chan_sip.c: Hanging up zombie call. Be scared.
[Feb 27 23:14:40] DEBUG[12213] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:14:40] DEBUG[12213] chan_sip.c: Hanging up channel in state Ringing (not UP)
[Feb 27 23:14:40] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:14:40] DEBUG[12213] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xa3e3380'
[Feb 27 23:14:40] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:14:40] DEBUG[12213] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '12d016e30489075f747b2a7b0279bd49@10.78.65.80:5060' Request 102: Found
[Feb 27 23:14:40] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:14:40] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:14:40] DEBUG[12213] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:14:40] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:14:40] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:14:40] DEBUG[12213] app_dial.c: Exiting with DIALSTATUS=NOANSWER.
[Feb 27 23:14:40] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:14:40] DEBUG[12213] pbx.c: Extension 20, priority 3 returned normally even though call was hung up
[Feb 27 23:14:40] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:14:40] DEBUG[12213] channel.c: Soft-Hanging up channel 'SIP/21-000000c2'
[Feb 27 23:14:40] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:14:40] DEBUG[12213] pbx.c: Launching 'Hangup'
[Feb 27 23:14:40] VERBOSE[12213] pbx.c:     -- Executing [h@internal_default:1] Hangup("SIP/21-000000c2", "") in new stack
[Feb 27 23:14:40] DEBUG[12213] pbx.c: Spawn extension (internal_default,h,1) exited non-zero on 'SIP/21-000000c2'
[Feb 27 23:14:40] DEBUG[11359] app_queue.c: Extension '20@autohint' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:14:40] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:14:40] VERBOSE[12213] pbx.c:   == Spawn extension (internal_default, h, 1) exited non-zero on 'SIP/21-000000c2'
[Feb 27 23:14:40] DEBUG[12213] channel.c: Hanging up channel 'SIP/21-000000c2'
[Feb 27 23:14:40] DEBUG[12213] chan_sip.c: Hanging up zombie call. Be scared.
[Feb 27 23:14:40] DEBUG[12213] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xaf162880'
[Feb 27 23:14:40] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 21
[Feb 27 23:14:40] DEBUG[11358] chan_sip.c: Checking device state for peer 21
[Feb 27 23:14:40] DEBUG[11358] devicestate.c: Changing state for SIP/21 - state 1 (Not in use)
[Feb 27 23:14:40] DEBUG[11358] devicestate.c: device 'SIP/21' state '1'
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 12d016e30489075f747b2a7b0279bd49@10.78.65.80:5060 (Checking To) --From tag as5b34985d --To-tag 1382680134  
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Acked pending invite 102
[Feb 27 23:14:40] DEBUG[11394] app_queue.c: Device 'SIP/21' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Stopping retransmission on '12d016e30489075f747b2a7b0279bd49@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Destroying SIP dialog 882129746@10.78.65.151
[Feb 27 23:14:40] DEBUG[11384] rtp_engine.c: Destroyed RTP instance '0xaf162880'
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: = Looking for  Call ID: 12d016e30489075f747b2a7b0279bd49@10.78.65.80:5060 (Checking To) --From tag as5b34985d --To-tag 1382680134  
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Stopping retransmission on '12d016e30489075f747b2a7b0279bd49@10.78.65.80:5060' of Request 102: Match Found
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: SIP response 487 to standard invite
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Trying to put 'ACK sip:20@' onto UDP socket destined for 10.78.65.152:5062
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Updating call counter for outgoing call
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Setting SIP_ALREADYGONE on dialog 12d016e30489075f747b2a7b0279bd49@10.78.65.80:5060
[Feb 27 23:14:40] DEBUG[11358] devicestate.c: No provider found, checking channel drivers for SIP - 20
[Feb 27 23:14:40] DEBUG[11358] chan_sip.c: Checking device state for peer 20
[Feb 27 23:14:40] DEBUG[11358] devicestate.c: Changing state for SIP/20 - state 1 (Not in use)
[Feb 27 23:14:40] DEBUG[11358] devicestate.c: device 'SIP/20' state '1'
[Feb 27 23:14:40] DEBUG[11384] chan_sip.c: Destroying SIP dialog 12d016e30489075f747b2a7b0279bd49@10.78.65.80:5060
[Feb 27 23:14:40] DEBUG[11384] rtp_engine.c: Destroyed RTP instance '0xa3e3380'
[Feb 27 23:14:40] DEBUG[11394] app_queue.c: Device 'SIP/20' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Feb 27 23:14:44] VERBOSE[12208] asterisk.c:     -- Remote UNIX connection disconnected
[Feb 27 23:14:45] DEBUG[11384] chan_sip.c: Auto destroying SIP dialog '4035f4e316b589eb2330d70b70927fbb@netforce.ath.cx'
[Feb 27 23:14:45] DEBUG[11384] chan_sip.c: Destroying SIP dialog 4035f4e316b589eb2330d70b70927fbb@netforce.ath.cx
[Feb 27 23:14:47] VERBOSE[11354] asterisk.c:     -- Remote UNIX connection