[Oct  7 17:15:12] Asterisk SVN-branch-12-r400661 built by root @ <hostname> on a i686 running Linux on 2013-10-01 16:12:22 UTC
[Oct  7 17:15:12] VERBOSE[2566] config.c:   == Parsing '/etc/asterisk/logger.conf': Found
[Oct  7 17:15:12] VERBOSE[2566] logger.c:  Asterisk Queue Logger restarted
[Oct  7 17:15:19] VERBOSE[2433] res_pjsip_registrar.c:     -- Added contact 'sip:phone_A@10.24.19.97:5060;ob' to AOR 'phone_A' with expiration of 300 seconds
[Oct  7 17:15:19] DEBUG[2419] devicestate.c: Processing device state change for 'PJSIP/phone_A'
[Oct  7 17:15:19] DEBUG[2419] devicestate.c: Adding per-server state of 'Not in use' for 'PJSIP/phone_A'
[Oct  7 17:15:19] DEBUG[2419] devicestate.c: Aggregate devstate result is 'Not in use' for 'PJSIP/phone_A'
[Oct  7 17:15:19] DEBUG[2419] devicestate.c: Aggregate state for device 'PJSIP/phone_A' has changed to 'Not in use'
[Oct  7 17:15:19] WARNING[2433] res_pjsip_pubsub.c: Subscriptions not permitted for endpoint phone_A.
[Oct  7 17:15:19] WARNING[2433] res_pjsip_pubsub.c: Subscriptions not permitted for endpoint phone_A.
[Oct  7 17:15:21] VERBOSE[2433] res_pjsip_registrar.c:     -- Added contact 'sip:phone_B@10.24.18.165:5060;ob' to AOR 'phone_B' with expiration of 300 seconds
[Oct  7 17:15:21] DEBUG[2419] devicestate.c: Processing device state change for 'PJSIP/phone_B'
[Oct  7 17:15:21] DEBUG[2419] devicestate.c: Adding per-server state of 'Not in use' for 'PJSIP/phone_B'
[Oct  7 17:15:21] DEBUG[2419] devicestate.c: Aggregate devstate result is 'Not in use' for 'PJSIP/phone_B'
[Oct  7 17:15:21] DEBUG[2419] devicestate.c: Aggregate state for device 'PJSIP/phone_B' has changed to 'Not in use'
[Oct  7 17:15:21] WARNING[2433] res_pjsip_pubsub.c: Subscriptions not permitted for endpoint phone_B.
[Oct  7 17:15:21] WARNING[2433] res_pjsip_pubsub.c: Subscriptions not permitted for endpoint phone_B.
[Oct  7 17:15:26] DEBUG[2433] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb67685ec'
[Oct  7 17:15:26] DEBUG[2433] res_rtp_asterisk.c: Allocated port 13348 for RTP instance '0xb67685ec'
[Oct  7 17:15:26] DEBUG[2433] rtp_engine.c: RTP instance '0xb67685ec' is setup and ready to go
[Oct  7 17:15:26] DEBUG[2433] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb67685ec'
[Oct  7 17:15:26] DEBUG[2433] rtp_engine.c: Setting payload 0 based on m type on 0xb5866548
[Oct  7 17:15:26] DEBUG[2433] rtp_engine.c: Setting payload 8 based on m type on 0xb5866548
[Oct  7 17:15:26] DEBUG[2433] rtp_engine.c: Setting payload 9 based on m type on 0xb5866548
[Oct  7 17:15:26] DEBUG[2433] rtp_engine.c: Setting payload 111 based on m type on 0xb5866548
[Oct  7 17:15:26] DEBUG[2433] rtp_engine.c: Setting payload 18 based on m type on 0xb5866548
[Oct  7 17:15:26] DEBUG[2433] rtp_engine.c: Setting payload 58 based on m type on 0xb5866548
[Oct  7 17:15:26] DEBUG[2433] rtp_engine.c: Setting payload 118 based on m type on 0xb5866548
[Oct  7 17:15:26] DEBUG[2433] rtp_engine.c: Setting payload 58 based on m type on 0xb5866548
[Oct  7 17:15:26] DEBUG[2433] rtp_engine.c: Setting payload 96 based on m type on 0xb5866548
[Oct  7 17:15:26] DEBUG[2569][C-00000000] pbx.c: Launching 'NoOp'
[Oct  7 17:15:26] VERBOSE[2569][C-00000000] pbx.c:     -- Executing [1003@internal:1] NoOp("PJSIP/phone_A-00000000", "CALLERID NAME: Phone A") in new stack
[Oct  7 17:15:26] DEBUG[2569][C-00000000] pbx.c: Launching 'NoOp'
[Oct  7 17:15:26] VERBOSE[2569][C-00000000] pbx.c:     -- Executing [1003@internal:2] NoOp("PJSIP/phone_A-00000000", "CALLERID NUMBER: 1001") in new stack
[Oct  7 17:15:26] DEBUG[2569][C-00000000] pbx.c: Launching 'Dial'
[Oct  7 17:15:26] VERBOSE[2569][C-00000000] pbx.c:     -- Executing [1003@internal:3] Dial("PJSIP/phone_A-00000000", "SIP/phone_C") in new stack
[Oct  7 17:15:26] DEBUG[2569][C-00000000] chan_sip.c: Asked to create a SIP channel with formats: (ulaw)
[Oct  7 17:15:26] DEBUG[2569][C-00000000] chan_sip.c: Allocating new SIP dialog for 74fbde2856eeaddc5218d47e6ea8d333@127.0.0.1:5061 - INVITE (No RTP)
[Oct  7 17:15:26] DEBUG[2569][C-00000000] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb295ab44'
[Oct  7 17:15:26] DEBUG[2569][C-00000000] res_rtp_asterisk.c: Allocated port 17576 for RTP instance '0xb295ab44'
[Oct  7 17:15:26] DEBUG[2569][C-00000000] rtp_engine.c: RTP instance '0xb295ab44' is setup and ready to go
[Oct  7 17:15:26] DEBUG[2569][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb295ab44'
[Oct  7 17:15:26] VERBOSE[2569][C-00000000] netsock2.c:   == Using SIP RTP CoS mark 5
[Oct  7 17:15:26] DEBUG[2569][C-00000000] chan_sip.c: Setting NAT on RTP to Off
[Oct  7 17:15:26] DEBUG[2569][C-00000000] chan_sip.c: Setting NAT on RTP to Off
[Oct  7 17:15:26] DEBUG[2569][C-00000000] chan_sip.c: SIP call-id changed from '74fbde2856eeaddc5218d47e6ea8d333@127.0.0.1:5061' to '4dc4bd0d1659df7e0a83fb810820befe@10.24.18.161:5061'
[Oct  7 17:15:26] DEBUG[2569][C-00000000] rtp_engine.c: Seeded SDP of 'SIP/phone_C-00000000' with that of 'PJSIP/phone_A-00000000'
[Oct  7 17:15:26] DEBUG[2569][C-00000000] channel.c: Not copying variable DIALEDTIME.
[Oct  7 17:15:26] DEBUG[2569][C-00000000] channel.c: Not copying variable ANSWEREDTIME.
[Oct  7 17:15:26] DEBUG[2569][C-00000000] channel.c: Not copying variable DIALEDPEERNAME.
[Oct  7 17:15:26] DEBUG[2569][C-00000000] channel.c: Not copying variable DIALEDPEERNUMBER.
[Oct  7 17:15:26] DEBUG[2569][C-00000000] channel.c: Not copying variable DIALSTATUS.
[Oct  7 17:15:26] DEBUG[2569][C-00000000] chan_sip.c: Outgoing Call for phone_C
[Oct  7 17:15:26] DEBUG[2569][C-00000000] chan_sip.c: ** Our capability: (ulaw|g722) Video flag: False Text flag: False
[Oct  7 17:15:26] DEBUG[2569][C-00000000] chan_sip.c: ** Our prefcodec: (ulaw) 
[Oct  7 17:15:26] DEBUG[2569][C-00000000] chan_sip.c: Initializing initreq for method INVITE - callid 4dc4bd0d1659df7e0a83fb810820befe@10.24.18.161:5061
[Oct  7 17:15:26] VERBOSE[2569][C-00000000] app_dial.c:     -- Called SIP/phone_C
[Oct  7 17:15:26] DEBUG[2569][C-00000000] channel.c: PJSIP/phone_A-00000000: Dropping redundant connected line update "Phone C" <1003>.
[Oct  7 17:15:26] DEBUG[2472][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4dc4bd0d1659df7e0a83fb810820befe@10.24.18.161:5061' Request 102: Found
[Oct  7 17:15:26] DEBUG[2472][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4dc4bd0d1659df7e0a83fb810820befe@10.24.18.161:5061' Request 102: Found
[Oct  7 17:15:26] VERBOSE[2569][C-00000000] app_dial.c:     -- SIP/phone_C-00000000 is ringing
[Oct  7 17:15:26] DEBUG[2419] devicestate.c: Processing device state change for 'SIP/phone_C'
[Oct  7 17:15:26] DEBUG[2419] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_C'
[Oct  7 17:15:26] DEBUG[2419] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_C'
[Oct  7 17:15:26] DEBUG[2419] devicestate.c: Aggregate state for device 'SIP/phone_C' has changed to 'Not in use'
[Oct  7 17:15:26] DEBUG[2419] devicestate.c: Processing device state change for 'PJSIP/phone_A'
[Oct  7 17:15:26] DEBUG[2419] devicestate.c: Adding per-server state of 'In use' for 'PJSIP/phone_A'
[Oct  7 17:15:26] DEBUG[2419] devicestate.c: Aggregate devstate result is 'In use' for 'PJSIP/phone_A'
[Oct  7 17:15:26] DEBUG[2419] devicestate.c: Aggregate state for device 'PJSIP/phone_A' has changed to 'In use'
[Oct  7 17:15:26] DEBUG[2472][C-00000000] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4dc4bd0d1659df7e0a83fb810820befe@10.24.18.161:5061' Request 102: Found
[Oct  7 17:15:26] VERBOSE[2569][C-00000000] app_dial.c:     -- SIP/phone_C-00000000 is ringing
[Oct  7 17:15:27] DEBUG[2472][C-00000000] chan_sip.c: Acked pending invite 102
[Oct  7 17:15:27] DEBUG[2472][C-00000000] chan_sip.c: Stopping retransmission on '4dc4bd0d1659df7e0a83fb810820befe@10.24.18.161:5061' of Request 102: Match Found
[Oct  7 17:15:27] DEBUG[2472][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0xb2c96ca8
[Oct  7 17:15:27] DEBUG[2472][C-00000000] rtp_engine.c: Setting payload 9 based on m type on 0xb2c96ca8
[Oct  7 17:15:27] DEBUG[2472][C-00000000] rtp_engine.c: Setting payload 101 based on m type on 0xb2c96ca8
[Oct  7 17:15:27] DEBUG[2472][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb295ab44'
[Oct  7 17:15:27] DEBUG[2472][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb295ab44'
[Oct  7 17:15:27] DEBUG[2472][C-00000000] channel.c: Set channel SIP/phone_C-00000000 to read format ulaw
[Oct  7 17:15:27] DEBUG[2472][C-00000000] channel.c: Set channel SIP/phone_C-00000000 to write format ulaw
[Oct  7 17:15:27] DEBUG[2569][C-00000000] channel.c: PJSIP/phone_A-00000000: Dropping redundant connected line update "Phone C" <1003>.
[Oct  7 17:15:27] VERBOSE[2569][C-00000000] app_dial.c:     -- SIP/phone_C-00000000 answered PJSIP/phone_A-00000000
[Oct  7 17:15:27] DEBUG[2419] devicestate.c: Processing device state change for 'SIP/phone_C'
[Oct  7 17:15:27] DEBUG[2419] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_C'
[Oct  7 17:15:27] DEBUG[2419] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_C'
[Oct  7 17:15:27] DEBUG[2419] devicestate.c: Aggregate state for device 'SIP/phone_C' has not changed from 'Not in use'
[Oct  7 17:15:27] DEBUG[2569][C-00000000] features.c: Removing dialed interfaces datastore on SIP/phone_C-00000000 since we're bridging
[Oct  7 17:15:27] DEBUG[2419] devicestate.c: Processing device state change for 'PJSIP/phone_A'
[Oct  7 17:15:27] DEBUG[2419] devicestate.c: Adding per-server state of 'In use' for 'PJSIP/phone_A'
[Oct  7 17:15:27] DEBUG[2569][C-00000000] dahdi/bridge_native_dahdi.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: Cannot use native DAHDI.  Must have two channels.
[Oct  7 17:15:27] DEBUG[2419] devicestate.c: Aggregate devstate result is 'In use' for 'PJSIP/phone_A'
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge.
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping.
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge_native_rtp.c: Bridge '72aac1de-4e35-42f4-9615-4ceb22e0fa84' can not use native RTP bridge as two channels are required
[Oct  7 17:15:27] DEBUG[2419] devicestate.c: Aggregate state for device 'PJSIP/phone_A' has not changed from 'In use'
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge.
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want.
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Chose bridge technology simple_bridge
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: calling simple_bridge technology constructor
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: calling simple_bridge technology start
[Oct  7 17:15:27] DEBUG[2433] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb67685ec'
[Oct  7 17:15:27] DEBUG[2433] rtp_engine.c: Setting payload 0 based on m type on 0xb5865fd8
[Oct  7 17:15:27] DEBUG[2433] rtp_engine.c: Setting payload 8 based on m type on 0xb5865fd8
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge_channel.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: 0x86fb3a4(PJSIP/phone_A-00000000) is joining
[Oct  7 17:15:27] DEBUG[2433] rtp_engine.c: Setting payload 96 based on m type on 0xb5865fd8
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge_channel.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: 0x86b164c(SIP/phone_C-00000000) is joining
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge_channel.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: pushing 0x86fb3a4(PJSIP/phone_A-00000000)
[Oct  7 17:15:27] VERBOSE[2569][C-00000000] bridge_channel.c:     -- Channel PJSIP/phone_A-00000000 joined 'simple_bridge' basic-bridge <72aac1de-4e35-42f4-9615-4ceb22e0fa84>
[Oct  7 17:15:27] DEBUG[2569][C-00000000] dahdi/bridge_native_dahdi.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: Cannot use native DAHDI.  Must have two channels.
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge.
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want.
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge_native_rtp.c: Bridge '72aac1de-4e35-42f4-9615-4ceb22e0fa84' can not use native RTP bridge as two channels are required
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge.
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want.
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Chose bridge technology simple_bridge
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84 is already using the new technology.
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84 is happy that channel PJSIP/phone_A-00000000 already has read format ulaw
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84 is happy that channel PJSIP/phone_A-00000000 already has write format ulaw
[Oct  7 17:15:27] DEBUG[2569][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: 0x86fb3a4(PJSIP/phone_A-00000000) is joining simple_bridge technology
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge_channel.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: pushing 0x86b164c(SIP/phone_C-00000000)
[Oct  7 17:15:27] VERBOSE[2570][C-00000000] bridge_channel.c:     -- Channel SIP/phone_C-00000000 joined 'simple_bridge' basic-bridge <72aac1de-4e35-42f4-9615-4ceb22e0fa84>
[Oct  7 17:15:27] DEBUG[2423] cdr.c: Finalized CDR for SIP/phone_C-00000000 - start 1381184126.051174 answer 1381184127.637148 end 1381184127.639132 dispo ANSWERED
[Oct  7 17:15:27] DEBUG[2570][C-00000000] dahdi/bridge_native_dahdi.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: Cannot use native DAHDI.  Channel 'PJSIP/phone_A-00000000' not compatible.
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge.
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge technology softmix does not have any capabilities we want.
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge technology holding_bridge does not have any capabilities we want.
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Chose bridge technology native_rtp
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: calling native_rtp technology constructor
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: calling simple_bridge technology stop
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: 0x86fb3a4(PJSIP/phone_A-00000000) is leaving simple_bridge technology (dummy)
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84 is happy that channel PJSIP/phone_A-00000000 already has read format ulaw
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84 is happy that channel PJSIP/phone_A-00000000 already has write format ulaw
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: 0x86fb3a4(PJSIP/phone_A-00000000) is joining native_rtp technology
[Oct  7 17:15:27] DEBUG[2570][C-00000000] chan_sip.c: ** Our capability: (ulaw|g722) Video flag: True Text flag: True
[Oct  7 17:15:27] DEBUG[2570][C-00000000] chan_sip.c: ** Our prefcodec: (ulaw) 
[Oct  7 17:15:27] DEBUG[2570][C-00000000] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw)
[Oct  7 17:15:27] DEBUG[2570][C-00000000] chan_sip.c: Initializing already initialized SIP dialog 4dc4bd0d1659df7e0a83fb810820befe@10.24.18.161:5061 (presumably reinvite)
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84 is happy that channel SIP/phone_C-00000000 already has read format ulaw
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84 is happy that channel SIP/phone_C-00000000 already has write format ulaw
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: 0x86b164c(SIP/phone_C-00000000) is joining native_rtp technology
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: calling native_rtp technology start
[Oct  7 17:15:27] DEBUG[2570][C-00000000] bridge.c: Bridge 72aac1de-4e35-42f4-9615-4ceb22e0fa84: calling simple_bridge technology destructor
[Oct  7 17:15:27] DEBUG[2570][C-00000000] res_rtp_asterisk.c: 0xb53bd380 -- Probation learning mode pass with source address 10.24.17.17:63530
[Oct  7 17:15:27] DEBUG[2570][C-00000000] chan_sip.c: Oooh, format changed to ulaw
[Oct  7 17:15:27] DEBUG[2570][C-00000000] channel.c: Set channel SIP/phone_C-00000000 to read format ulaw
[Oct  7 17:15:27] DEBUG[2570][C-00000000] channel.c: Set channel SIP/phone_C-00000000 to write format ulaw
[Oct  7 17:15:27] DEBUG[2569][C-00000000] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw
[Oct  7 17:15:27] DEBUG[2569][C-00000000] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160
[Oct  7 17:15:27] DEBUG[2569][C-00000000] res_rtp_asterisk.c: 0xb5391600 -- Probation learning mode pass with source address 10.24.19.97:4046
[Oct  7 17:15:27] DEBUG[2570][C-00000000] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw
[Oct  7 17:15:27] DEBUG[2570][C-00000000] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160
[Oct  7 17:15:28] DEBUG[2472][C-00000000] chan_sip.c: Acked pending invite 103
[Oct  7 17:15:28] DEBUG[2472][C-00000000] chan_sip.c: Stopping retransmission on '4dc4bd0d1659df7e0a83fb810820befe@10.24.18.161:5061' of Request 103: Match Found
[Oct  7 17:15:28] DEBUG[2472][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0xb2c96ca8
[Oct  7 17:15:28] DEBUG[2472][C-00000000] rtp_engine.c: Setting payload 101 based on m type on 0xb2c96ca8
[Oct  7 17:15:28] DEBUG[2472][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb295ab44'
[Oct  7 17:15:28] DEBUG[2472][C-00000000] chan_sip.c: Stopping retransmission on '4dc4bd0d1659df7e0a83fb810820befe@10.24.18.161:5061' of Request 103: Match Not Found
[Oct  7 17:15:28] DEBUG[2433] rtp_engine.c: Setting payload 0 based on m type on 0xb5865cd8
[Oct  7 17:15:28] DEBUG[2433] rtp_engine.c: Setting payload 101 based on m type on 0xb5865cd8
[Oct  7 17:15:28] DEBUG[2433] rtp_engine.c: Setting payload 0 based on m type on 0xb5865cd8
[Oct  7 17:15:28] DEBUG[2433] rtp_engine.c: Setting payload 101 based on m type on 0xb5865cd8
[Oct  7 17:15:40] DEBUG[2472][C-00000000] rtp_engine.c: Setting payload 9 based on m type on 0xb2c96c78
[Oct  7 17:15:40] DEBUG[2472][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0xb2c96c78
[Oct  7 17:15:40] DEBUG[2472][C-00000000] rtp_engine.c: Setting payload 8 based on m type on 0xb2c96c78
[Oct  7 17:15:40] DEBUG[2472][C-00000000] rtp_engine.c: Setting payload 3 based on m type on 0xb2c96c78
[Oct  7 17:15:40] DEBUG[2472][C-00000000] rtp_engine.c: Setting payload 99 based on m type on 0xb2c96c78
[Oct  7 17:15:40] DEBUG[2472][C-00000000] rtp_engine.c: Setting payload 108 based on m type on 0xb2c96c78
[Oct  7 17:15:40] DEBUG[2472][C-00000000] rtp_engine.c: Setting payload 18 based on m type on 0xb2c96c78
[Oct  7 17:15:40] DEBUG[2472][C-00000000] rtp_engine.c: Setting payload 101 based on m type on 0xb2c96c78
[Oct  7 17:15:40] DEBUG[2472][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb295ab44'
[Oct  7 17:15:40] DEBUG[2472][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb295ab44'
[Oct  7 17:15:40] DEBUG[2472][C-00000000] channel.c: Set channel SIP/phone_C-00000000 to read format ulaw
[Oct  7 17:15:40] DEBUG[2472][C-00000000] channel.c: Set channel SIP/phone_C-00000000 to write format ulaw
[Oct  7 17:15:40] DEBUG[2472][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb295ab44'
[Oct  7 17:15:40] DEBUG[2472][C-00000000] chan_sip.c: Setting framing from config on incoming call
[Oct  7 17:15:40] DEBUG[2472][C-00000000] chan_sip.c: ** Our capability: (ulaw|g722) Video flag: True Text flag: True
[Oct  7 17:15:40] DEBUG[2472][C-00000000] chan_sip.c: ** Our prefcodec: (ulaw) 
[Oct  7 17:15:40] DEBUG[2569][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb67685ec'
[Oct  7 17:15:40] VERBOSE[2569][C-00000000] res_musiconhold.c:     -- Started music on hold, class 'default', on PJSIP/phone_A-00000000
[Oct  7 17:15:40] DEBUG[2569][C-00000000] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Oct  7 17:15:40] DEBUG[2569][C-00000000] channel.c: Set channel PJSIP/phone_A-00000000 to write format slin
[Oct  7 17:15:40] DEBUG[2569][C-00000000] res_musiconhold.c: PJSIP/phone_A-00000000 Opened file 0 '/var/lib/asterisk/moh/manolo_camp-morning_coffee'
[Oct  7 17:15:40] DEBUG[2569][C-00000000] res_rtp_asterisk.c: 0xb5391600 -- Probation learning mode pass with source address 10.24.19.97:4046