Hi. I got call canceling so many times.
I couldn’t find the reason.
please save me. 
Thanks.
I don’t know what information you need to debug.
So If you need more information please tell me, im waiting.
Thanks.
I’m using Asterisk 11.7.0
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘memberUid’ is ‘228128’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘countryCode’ is ‘1’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘userNumber’ is NULL
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘callerRetryCount’ is ‘1’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘retryTimeSchedule’ is ‘0’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘sendMessageTime’ is ‘0’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘uniqueId’ is ‘2-71795541’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘guideVoiceType’ is ‘0’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘callMode’ is ‘live’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘sendtaskUid’ is ‘3-3-71795541’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘tomcatServerIp’ is ‘174.xx.xxx.xxx’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘callerNumber’ is ‘212xxxxxxx’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Launching ‘AGI’
[Feb 23 13:13:36] VERBOSE[11856][C-000000cc] pbx.c: – Executing [s@outbound:1] AGI(“Local/s@outbound-000000cc;2”, “agi://localhost/outbounddialout.agi?memberUid=228128&introPath=N/NYCCarpenters/137037187901_Record_intro-06-04-2013[11_51]&messagePath=N/NYCCarpenters/139284506480_Record_message-02-19-2014[13_24]&countryCode=1&userNumber=&callerRetryCount=1&retryTimeSchedule=0&sendMessageTime=0&uniqueId=2-71795541&guideVoiceType=0&callMode=live&sendtaskUid=3-3-71795541&tomcatServerIp=174.xx.xxx.xxx&callerNumber=212xxxxxxx”) in new stack
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] netsock2.c: Splitting ‘localhost’ into…
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] netsock2.c: …host ‘localhost’ and port ‘’.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] res_agi.c: Wow, connected!
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘countryCode’ is ‘1’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘toolData’ is ‘631xxxxxxx’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘callerNumber’ is ‘212xxxxxxx’
[Feb 23 13:13:36] VERBOSE[11856][C-000000cc] res_agi.c: – AGI Script Executing Application: (SIPAddHeader) Options: (X-Asterisk-callerid:212xxxxxxx)
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: SIP Header added “X-Asterisk-callerid:212xxxxxxx” as __SIPADDHEADER01
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] pbx.c: Result of ‘sendtaskUid’ is ‘3-3-71795541’
[Feb 23 13:13:36] VERBOSE[11856][C-000000cc] res_agi.c: – AGI Script Executing Application: (SIPAddHeader) Options: (X-Asterisk-sendtaskuid:3-3-71795541)
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: SIP Header added “X-Asterisk-sendtaskuid:3-3-71795541” as __SIPADDHEADER02
[Feb 23 13:13:36] VERBOSE[11856][C-000000cc] res_agi.c: – AGI Script Executing Application: (Dial) Options: (SIP/1631xxxxxxx@innodial)
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Asked to create a SIP channel with formats: (slin)
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Allocating new SIP dialog for 5e661c62321766651ac4bba07c28cedb@sip.trumpia.com - INVITE (No RTP)
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] rtp_engine.c: Using engine ‘asterisk’ for RTP instance ‘0x2aaad0017578’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] res_rtp_asterisk.c: Allocated port 26424 for RTP instance ‘0x2aaad0017578’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] netsock2.c: Splitting ‘192.xxx.xx.xx’ into…
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] netsock2.c: …host ‘192.xxx.xx.xx’ and port ‘’.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] netsock2.c: Splitting ‘174.xx.xxx.xxx’ into…
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] netsock2.c: …host ‘174.xx.xxx.xxx’ and port ‘’.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] rtp_engine.c: RTP instance ‘0x2aaad0017578’ is setup and ready to go
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] res_rtp_asterisk.c: Setup RTCP on RTP instance ‘0x2aaad0017578’
[Feb 23 13:13:36] VERBOSE[11856][C-000000cc] netsock2.c: == Using SIP RTP CoS mark 5
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Setting NAT on RTP to Off
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: OBPROXY: Not applying OBproxy to this call
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: SIP call-id changed from '5e661c62321766651ac4bba07c28cedb@sip.innxxxxx.com’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] acl.c: For destination ‘72.xx.xx.xxx’, our source address is ‘174.xx.xxx.xxx’.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 174.xx.xxx.xxx:5060
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Setting NAT on RTP to Off
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: SIP call-id changed from '5e661c62321766651ac4bba07c28cedb@sip.innxxxxx.com’ to ’ `@sip.innxxxxx.com’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: *** Our native formats are (ulaw)
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: *** Joint capabilities are (nothing)
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: *** Our capabilities are (ulaw)
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: *** Our preferred formats from the incoming channel are (slin)
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: This channel will not be able to handle video.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel_internal_api.c: Channel Call ID changing from [C-000000cc] to [C-000000cc]
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] rtp_engine.c: Can’t find native functions for channel ‘Local/s@outbound-000000cc;2’
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable DIALEDTIME.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable ANSWEREDTIME.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable DIALEDPEERNAME.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable DIALEDPEERNUMBER.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable DIALSTATUS.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Copying hard-transferable variable SIPADDHEADER02.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Copying hard-transferable variable SIPADDHEADER01.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable callerNumber.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable countryCode.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable toolData.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable tomcatServerIp.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable introPath.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable messagePath.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable uniqueId.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable sendMessageTime.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Copying hard-transferable variable AJ_TRACE_ID.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable memberUid.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable guideVoiceType.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable callMode.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable retryTimeSchedule.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable sendtaskUid.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable callerRetryCount.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Not copying variable localIp.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Outgoing Call for 1631xxxxxxx
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Updating call counter for outgoing call
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Call to peer ‘innodial’ is 3 out of 200
[Feb 23 13:13:36] DEBUG[16696] devicestate.c: No provider found, checking channel drivers for SIP - innodial
[Feb 23 13:13:36] DEBUG[16696] chan_sip.c: Checking device state for peer innodial
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Adding SIP Header “X-Asterisk-sendtaskuid” with content :3-3-71795541:
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Adding SIP Header “X-Asterisk-callerid” with content :212xxxxxxx:
[Feb 23 13:13:36] DEBUG[16696] devicestate.c: Changing state for SIP/innodial - state 7 (Ring+Inuse)
[Feb 23 13:13:36] DEBUG[16696] devicestate.c: device ‘SIP/innodial’ state ‘7’
Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: ** Our capability: (ulaw) Video flag: False Text flag: False
[Feb 23 13:13:36] DEBUG[16744] app_queue.c: Device ‘SIP/innodial’ changed to state ‘7’ (Ring+Inuse) but we don’t care because they’re not a member of any queue.
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: ** Our prefcodec: (slin)
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: – Done with adding codecs to SDP
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Done building SDP. Settling with this capability: (ulaw)
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Initializing initreq for method INVITE - callid 56fb766422ee62471cf3e0ce15e592b8@sip.innxxxxx.com
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 0 [ 47]: INVITE sip:1631xxxxxxx@sip.innxxxxx.com SIP/2.0
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 174.xx.xxx.xxx:5060;branch=z9hG4bK6a5e160b
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 3 [ 52]: From: sip:1004zzzz@sip.innxxxxx.com;tag=as09f2f25f
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 4 [ 38]: To: sip:1631xxxxxxx@sip.innxxxxx.com
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 5 [ 43]: Contact: sip:1004zzzz@174.xx.xxx.xxx:5060
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 6 [ 58]: Call-ID: 56fb766422ee62471cf3e0ce15e592b8@sip.innxxxxx.com
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 8 [ 31]: User-Agent: trumpia call server
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 9 [ 35]: Date: Sun, 23 Feb 2014 21:13:36 GMT
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 12 [ 36]: X-Asterisk-sendtaskuid: 3-3-71795541
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 13 [ 31]: X-Asterisk-callerid: 212xxxxxxx
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15737
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] chan_sip.c: Trying to put ‘INVITE sip:’ onto UDP socket destined for 72.xx.xx.xxx:5060
[Feb 23 13:13:36] VERBOSE[11856][C-000000cc] app_dial.c: – Called SIP/1631xxxxxxx@innodial
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Set channel SIP/innodial-000000cd to read format slin
[Feb 23 13:13:36] DEBUG[11856][C-000000cc] channel.c: Set channel SIP/innodial-000000cd to write format slin
[Feb 23 13:13:36] DEBUG[16706] manager.c: Running action ‘Command’
[Feb 23 13:13:36] DEBUG[16718] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying
[Feb 23 13:13:36] DEBUG[16718] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 174.xx.xxx.xxx:5060;branch=z9hG4bK6a5e160b
[Feb 23 13:13:36] DEBUG[16718] chan_sip.c: Header 2 [ 52]: From: sip:1004zzzz@sip.innxxxxx.com;tag=as09f2f25f
[Feb 23 13:13:36] DEBUG[16718] chan_sip.c: Header 3 [ 38]: To: sip:1631xxxxxxx@sip.innxxxxx.com
[Feb 23 13:13:36] DEBUG[16718] chan_sip.c: Header 4 [ 58]: Call-ID: 56fb766422ee62471cf3e0ce15e592b8@sip.innxxxxx.com
[Feb 23 13:13:36] DEBUG[16718] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE
[Feb 23 13:13:36] DEBUG[16718] chan_sip.c: Header 6 [ 30]: Server: INNODIAL PLOXY 0.2.0.0
[Feb 23 13:13:36] DEBUG[16718] chan_sip.c: Header 7 [ 17]: Content-Length: 0
[Feb 23 13:13:36] DEBUG[16718] chan_sip.c: = Looking for Call ID: 56fb766422ee62471cf3e0ce15e592b8@sip.innxxxxx.com (Checking To) --From tag as09f2f25f --To-tag
[Feb 23 13:13:36] DEBUG[16718][C-000000cc] chan_sip.c: *** SIP TIMER: Cancelling retransmission #15737 - INVITE (got response)
[Feb 23 13:13:36] DEBUG[16718][C-000000cc] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '56fb766422ee62471cf3e0ce15e592b8@sip.innxxxxx.com’ Request 102: Found
[Feb 23 13:13:36] DEBUG[16718][C-000000cc] chan_sip.c: SIP response 100 to standard invite
[Feb 23 13:13:36] DEBUG[16709] manager.c: Running action ‘Getvar’
[Feb 23 13:13:36] DEBUG[16709] pbx.c: Result of ‘AJ_TRACE_ID’ is ‘AJ_ORIGINATE_3843’
[Feb 23 13:13:36] DEBUG[16710] manager.c: Running action ‘Getvar’
[Feb 23 13:13:36] DEBUG[16710] pbx.c: Result of ‘AJ_TRACE_ID’ is ‘AJ_ORIGINATE_3843’
[Feb 23 13:13:36] DEBUG[16710] manager.c: Running action ‘Getvar’
[Feb 23 13:13:36] DEBUG[16710] pbx.c: Result of ‘AJ_TRACE_ID’ is ‘AJ_ORIGINATE_3843’
[Feb 23 13:13:36] DEBUG[16709] manager.c: Running action ‘Getvar’
[Feb 23 13:13:36] DEBUG[16709] pbx.c: Result of ‘AJ_TRACE_ID’ is ‘AJ_ORIGINATE_3843’
[Feb 23 13:13:36] DEBUG[16710] manager.c: Running action ‘Getvar’
[Feb 23 13:13:36] DEBUG[16710] pbx.c: Result of ‘AJ_TRACE_ID’ is ‘AJ_ORIGINATE_3843’
[Feb 23 13:13:36] DEBUG[16709] manager.c: Running action ‘Getvar’
[Feb 23 13:13:36] DEBUG[16709] pbx.c: Result of ‘AJ_TRACE_ID’ is ‘AJ_ORIGINATE_3843’
[Feb 23 13:13:37] DEBUG[11799] res_rtp_asterisk.c: Got RTCP report of 44 bytes
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Allocating new SIP dialog for 154eda042d43adaf58f4857a08ee1d1d@sip.trumpia.com - OPTIONS (No RTP)
[Feb 23 13:13:40] DEBUG[16718] acl.c: For destination ‘72.xx.xx.xxx’, our source address is ‘174.xx.xxx.xxx’.
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 174.xx.xxx.xxx:5060
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: SIP call-id changed from '3c19d7af480fc6b12324c77c58b097f3@sip.trumpia.com’
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Initializing initreq for method OPTIONS - callid 3c19d7af480fc6b12324c77c58b097f3@sip.trumpia.com
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 0 [ 36]: OPTIONS sip:sip.innxxxxx.com SIP/2.0
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 174.xx.xxx.xxx:5060;branch=z9hG4bK26248646
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 3 [ 62]: From: “asterisk” sip:1004zzzz@sip.trumpia.com;tag=as79cc5ba0
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 4 [ 26]: To: sip:sip.innxxxxx.com
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 5 [ 43]: Contact: sip:1004zzzz@174.xx.xxx.xxx:5060
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 6 [ 57]: Call-ID: 3c19d7af480fc6b12324c77c58b097f3@sip.trumpia.com
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 8 [ 31]: User-Agent: trumpia call server
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 9 [ 35]: Date: Sun, 23 Feb 2014 21:13:40 GMT
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15739
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Trying to put ‘OPTIONS sip’ onto UDP socket destined for 72.xx.xx.xxx:5060
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 174.xx.xxx.xxx:5060;branch=z9hG4bK26248646
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 2 [ 62]: From: “asterisk” sip:1004zzzz@sip.trumpia.com;tag=as79cc5ba0
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 3 [ 68]: To: sip:sip.innxxxxx.com;tag=622a53cd0e8c7a2dcb1e3ddf583a076c.d74b
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 4 [ 57]: Call-ID: 3c19d7af480fc6b12324c77c58b097f3@sip.trumpia.com
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 6 [ 30]: Server: INNODIAL PLOXY 0.2.0.0
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Header 7 [ 17]: Content-Length: 0
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: = Looking for Call ID: 3c19d7af480fc6b12324c77c58b097f3@sip.trumpia.com (Checking To) --From tag as79cc5ba0 --To-tag 622a53cd0e8c7a2dcb1e3ddf583a076c.d74b
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15739
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Stopping retransmission on '3c19d7af480fc6b12324c77c58b097f3@sip.trumpia.com’ of Request 102: Match Found
[Feb 23 13:13:40] DEBUG[16718] chan_sip.c: Destroying SIP dialog 3c19d7af480fc6b12324c77c58b097f3@sip.trumpia.com
[Feb 23 13:13:40] DEBUG[11811][C-000000cb] channel.c: Hanging up channel ‘Local/s@outbound-000000cb;1’
[Feb 23 13:14:01] DEBUG[11856][C-000000cc] chan_sip.c: Hangup call SIP/innodial-000000cd, SIP callid 56fb766422ee62471cf3e0ce15e592b8@sip.innxxxxx.com
[Feb 23 13:14:01] DEBUG[11856][C-000000cc] chan_sip.c: update_call_counter(1631xxxxxxx) - decrement call limit counter on hangup
[Feb 23 13:14:01] DEBUG[11856][C-000000cc] chan_sip.c: Updating call counter for outgoing call
[Feb 23 13:14:01] DEBUG[11856][C-000000cc] chan_sip.c: Call to peer ‘innodial’ removed from call limit 200
[Feb 23 13:14:01] DEBUG[11856][C-000000cc] chan_sip.c: Hanging up channel in state Down (not UP)
[Feb 23 13:14:01] DEBUG[11856][C-000000cc] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0x2aaad0017578’
[Feb 23 13:14:01] DEBUG[11856][C-000000cc] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '56fb766422ee62471cf3e0ce15e592b8@sip.innxxxxx.com’ Request 102: Found
[Feb 23 13:14:01] DEBUG[11856][C-000000cc] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15750
[Feb 23 13:14:01] DEBUG[11856][C-000000cc] chan_sip.c: Trying to put ‘CANCEL sip:’ onto UDP socket destined for 72.xx.xx.xxx:5060