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  1. [Sep 30 13:31:24] DEBUG[26040]: chan_sip.c:4422 __sip_autodestruct: Auto destroying SIP dialog '00131971-bbc80003-07021434-1c143efc@10.1.10.70'
  2. [Sep 30 13:31:24] DEBUG[26040]: chan_sip.c:6827 sip_destroy: Destroying SIP dialog 00131971-bbc80003-07021434-1c143efc@10.1.10.70
  3. Really destroying SIP dialog '00131971-bbc80003-07021434-1c143efc@10.1.10.70' Method: REGISTER
  4. [Sep 30 13:31:27] DEBUG[26138]: manager.c:5208 process_message: Running action 'Originate'
  5. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:29624 sip_request_call: Asked to create a SIP channel with formats: (slin)
  6. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:8764 sip_alloc: Allocating new SIP dialog for 34d74350621b11b254d01ac137addb74@127.0.1.1:5060 - INVITE (No RTP)
  7. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7fb0b000fe08'
  8. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: res_rtp_asterisk.c:1749 ast_rtp_new: Allocated port 12230 for RTP instance '0x7fb0b000fe08'
  9. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '10.1.10.150' into...
  10. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '10.1.10.150' and port ''.
  11. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.1.34' into...
  12. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.1.34' and port ''.
  13. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.106.1' into...
  14. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.106.1' and port ''.
  15. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.105.1' into...
  16. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.105.1' and port ''.
  17. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x7fb0b000fe08' is setup and ready to go
  18. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: res_rtp_asterisk.c:3879 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fb0b000fe08'
  19. == Using SIP RTP CoS mark 5
  20. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:5735 do_setnat: Setting NAT on RTP to On
  21. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:3641 obproxy_get: OBPROXY: Not applying OBproxy to this call
  22. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: acl.c:979 ast_ouraddrfor: For destination '10.1.10.70', our source address is '10.1.10.150'.
  23. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:4031 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 10.1.10.150:5060
  24. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:5735 do_setnat: Setting NAT on RTP to On
  25. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:8559 change_callid_pvt: SIP call-id changed from '34d74350621b11b254d01ac137addb74@127.0.1.1:5060' to '46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060'
  26. [Sep 30 13:31:27] DEBUG[26038]: manager.c:4789 match_filter: Examining event:
  27. Event: Newchannel
  28. Privileg3: call,all
  29. Channel: SIP/501-00000000
  30. ChannelState: 0
  31. ChannelStateDesc: Down
  32. CallerIDNum:
  33. CallerIDName:
  34. AccountCode:
  35. Exten:
  36. Context: default
  37. Uniqueid: 1380562287.0
  38.  
  39.  
  40. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:7945 sip_new: *** Our native formats are (ulaw)
  41. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:7946 sip_new: *** Joint capabilities are (nothing)
  42. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:7947 sip_new: *** Our capabilities are (ulaw)
  43. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:7948 sip_new: *** AST_CODEC_CHOOSE formats are ulaw
  44. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:7950 sip_new: *** Our preferred formats from the incoming channel are (slin)
  45. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:7976 sip_new: This channel will not be able to handle video.
  46. [Sep 30 13:31:27] DEBUG[26038]: manager.c:4789 match_filter: Examining event:
  47. Event: NewAccountCode
  48. Privileg3: call,all
  49. Channel: SIP/501-00000000
  50. Uniqueid: 1380562287.0
  51. AccountCode: 1
  52. OldAccountCode:
  53.  
  54.  
  55. [Sep 30 13:31:27] DEBUG[26138]: manager.c:4789 match_filter: Examining event:
  56. Event: Newchannel
  57. Privileg3: call,all
  58. Channel: SIP/501-00000000
  59. ChannelState: 0
  60. ChannelStateDesc: Down
  61. CallerIDNum:
  62. CallerIDName:
  63. AccountCode:
  64. Exten:
  65. Context: default
  66. Uniqueid: 1380562287.0
  67.  
  68.  
  69. [Sep 30 13:31:27] DEBUG[26138]: manager.c:4789 match_filter: Examining event:
  70. Event: NewAccountCode
  71. Privileg3: call,all
  72. Channel: SIP/501-00000000
  73. Uniqueid: 1380562287.0
  74. AccountCode: 1
  75. OldAccountCode:
  76.  
  77.  
  78. [Sep 30 13:31:27] DEBUG[26038]: manager.c:4789 match_filter: Examining event:
  79. Event: NewCallerid
  80. Privileg3: call,all
  81. Channel: SIP/501-00000000
  82. CallerIDNum: 123456789
  83. CallerIDName:
  84. Uniqueid: 1380562287.0
  85. CID-CallingPres: 0 (Presentation Allowed, Not Screened)
  86.  
  87.  
  88. [Sep 30 13:31:27] DEBUG[26138]: manager.c:4789 match_filter: Examining event:
  89. Event: NewCallerid
  90. Privileg3: call,all
  91. Channel: SIP/501-00000000
  92. CallerIDNum: 123456789
  93. CallerIDName:
  94. Uniqueid: 1380562287.0
  95. CID-CallingPres: 0 (Presentation Allowed, Not Screened)
  96.  
  97.  
  98. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:6355 sip_call: Outgoing Call for 501
  99. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:6679 update_call_counter: Updating call counter for outgoing call
  100. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:13118 add_sdp: ** Our capability: (ulaw) Video flag: False Text flag: False
  101. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:13119 add_sdp: ** Our prefcodec: (slin)
  102. Audio is at 12230
  103. Adding codec 100003 (ulaw) to SDP
  104. Adding non-codec 0x1 (telephone-event) to SDP
  105. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:13256 add_sdp: -- Done with adding codecs to SDP
  106. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:13454 add_sdp: Done building SDP. Settling with this capability: (ulaw)
  107. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:3517 initialize_initreq: Initializing initreq for method INVITE - callid 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060
  108. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:9611 parse_request: Header 0 [ 52]: INVITE sip:501@10.1.10.70:5060;transport=udp SIP/2.0
  109. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:9611 parse_request: Header 1 [ 62]: Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK59d734ae;rport
  110. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:9611 parse_request: Header 2 [ 16]: Max-Forwards: 70
  111. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:9611 parse_request: Header 3 [ 48]: From: <sip:123456789@10.1.10.150>;tag=as187358e6
  112. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:9611 parse_request: Header 4 [ 43]: To: <sip:501@10.1.10.70:5060;transport=udp>
  113. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:9611 parse_request: Header 5 [ 41]: Contact: <sip:123456789@10.1.10.150:5060>
  114. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:9611 parse_request: Header 6 [ 58]: Call-ID: 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060
  115. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:9611 parse_request: Header 7 [ 16]: CSeq: 102 INVITE
  116. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:9611 parse_request: Header 8 [ 31]: User-Agent: Asterisk PBX 11.5.0
  117. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:9611 parse_request: Header 9 [ 35]: Date: Mon, 30 Sep 2013 17:31:27 GMT
  118. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:9611 parse_request: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  119. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:9611 parse_request: Header 11 [ 26]: Supported: replaces, timer
  120. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:9611 parse_request: Header 12 [ 29]: Content-Type: application/sdp
  121. Reliably Transmitting (NAT) to 10.1.10.70:5060:
  122. INVITE sip:501@10.1.10.70:5060;transport=udp SIP/2.0
  123. Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK59d734ae;rport
  124. Max-Forwards: 70
  125. From: <sip:123456789@10.1.10.150>;tag=as187358e6
  126. To: <sip:501@10.1.10.70:5060;transport=udp>
  127. Contact: <sip:123456789@10.1.10.150:5060>
  128. Call-ID: 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060
  129. CSeq: 102 INVITE
  130. User-Agent: Asterisk PBX 11.5.0
  131. Date: Mon, 30 Sep 2013 17:31:27 GMT
  132. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  133. Supported: replaces, timer
  134. Content-Type: application/sdp
  135. Content-Length: 258
  136.  
  137. v=0
  138. o=root 826112350 826112350 IN IP4 10.1.10.150
  139. s=Asterisk PBX 11.5.0
  140. c=IN IP4 10.1.10.150
  141. t=0 0
  142. m=audio 12230 RTP/AVP 0 101
  143. a=rtpmap:0 PCMU/8000
  144. a=rtpmap:101 telephone-event/8000
  145. a=fmtp:101 0-16
  146. a=silenceSupp:off - - - -
  147. a=ptime:20
  148. a=sendrecv
  149.  
  150. ---
  151. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:4334 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #71
  152. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:3874 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 10.1.10.70:5060
  153. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: channel.c:2840 ast_hangup: Hanging up channel 'SIP/501-00000000'
  154. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:7059 sip_hangup: Hangup call SIP/501-00000000, SIP callid 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060
  155. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: chan_sip.c:7078 sip_hangup: Hanging up channel in state Down (not UP)
  156. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: res_rtp_asterisk.c:3924 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fb0b000fe08'
  157. Scheduling destruction of SIP dialog '46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060' in 13184 ms (Method: INVITE)
  158. [Sep 30 13:31:27] DEBUG[26038]: manager.c:4789 match_filter: Examining event:
  159. Event: Hangup
  160. Privileg3: call,all
  161. Channel: SIP/501-00000000
  162. Uniqueid: 1380562287.0
  163. CallerIDNum: 123456789
  164. CallerIDName: <unknown>
  165. ConnectedLineNum: 123456789
  166. ConnectedLineName: <unknown>
  167. AccountCode: 1
  168. Cause: 0
  169. Cause-txt: Unknown
  170.  
  171.  
  172. [Sep 30 13:31:27] DEBUG[26138]: manager.c:4789 match_filter: Examining event:
  173. Event: Hangup
  174. Privileg3: call,all
  175. Channel: SIP/501-00000000
  176. Uniqueid: 1380562287.0
  177. CallerIDNum: 123456789
  178. CallerIDName: <unknown>
  179. ConnectedLineNum: 123456789
  180. ConnectedLineName: <unknown>
  181. AccountCode: 1
  182. Cause: 0
  183. Cause-txt: Unknown
  184.  
  185.  
  186. > [INSERT INTO cdr ("calldate","clid","src","dst","dcontext","channel","dstchannel","lastapp","lastdata","duration","billsec","disposition","amaflags","accountcode","uniqueid","userfield") VALUES ('2013-09-30 17:31:27','123456789','123456789','s','machine_check','SIP/501-00000000','','Dial','SIP/501',0,0,'FAILED',3,'1','1380562287.0','')]
  187. [Sep 30 13:31:27] DEBUG[26278][C-00000000]: cdr_pgsql.c:344 pgsql_log: inserting a CDR record.
  188. [Sep 30 13:31:27] DEBUG[26013]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 501
  189. [Sep 30 13:31:27] DEBUG[26013]: chan_sip.c:29519 sip_devicestate: Checking device state for peer 501
  190. [Sep 30 13:31:27] DEBUG[26013]: devicestate.c:467 do_state_change: Changing state for SIP/501 - state 1 (Not in use)
  191. [Sep 30 13:31:27] DEBUG[26013]: devicestate.c:442 devstate_event: device 'SIP/501' state '1'
  192. [Sep 30 13:31:27] DEBUG[26054]: app_queue.c:1805 handle_statechange: Device 'SIP/501' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
  193. [Sep 30 13:31:27] DEBUG[26038]: manager.c:4789 match_filter: Examining event:
  194. Event: OriginateResponse
  195. Privileg3: call,all
  196. ActionID: 555
  197. Response: Failure
  198. Channel: SIP/501
  199. Context: machine_check
  200. Exten: s
  201. Reason: 3
  202. Uniqueid: <null>
  203. CallerIDNum: 123456789
  204. CallerIDName: <unknown>
  205.  
  206.  
  207. [Sep 30 13:31:27] DEBUG[26138]: manager.c:4789 match_filter: Examining event:
  208. Event: OriginateResponse
  209. Privileg3: call,all
  210. ActionID: 555
  211. Response: Failure
  212. Channel: SIP/501
  213. Context: machine_check
  214. Exten: s
  215. Reason: 3
  216. Uniqueid: <null>
  217. CallerIDNum: 123456789
  218. CallerIDName: <unknown>
  219.  
  220.  
  221.  
  222. <--- SIP read from UDP:10.1.10.70:5060 --->
  223. SIP/2.0 100 Trying
  224. Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK59d734ae;rport
  225. From: <sip:123456789@10.1.10.150>;tag=as187358e6
  226. To: <sip:501@10.1.10.70:5060;transport=udp>
  227. Call-ID: 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060
  228. CSeq: 102 INVITE
  229. Server: Cisco-CP7940G/8.0
  230. Contact: <sip:204@10.1.10.70:5060;transport=udp>
  231. Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
  232. Content-Length: 0
  233.  
  234. <------------->
  235. [Sep 30 13:31:27] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying
  236. [Sep 30 13:31:27] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 1 [ 62]: Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK59d734ae;rport
  237. [Sep 30 13:31:27] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 2 [ 48]: From: <sip:123456789@10.1.10.150>;tag=as187358e6
  238. [Sep 30 13:31:27] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 3 [ 43]: To: <sip:501@10.1.10.70:5060;transport=udp>
  239. [Sep 30 13:31:27] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 4 [ 58]: Call-ID: 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060
  240. [Sep 30 13:31:27] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 5 [ 16]: CSeq: 102 INVITE
  241. [Sep 30 13:31:27] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 6 [ 25]: Server: Cisco-CP7940G/8.0
  242. [Sep 30 13:31:27] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 7 [ 48]: Contact: <sip:204@10.1.10.70:5060;transport=udp>
  243. [Sep 30 13:31:27] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 8 [ 65]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
  244. [Sep 30 13:31:27] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 9 [ 17]: Content-Length: 0
  245. --- (10 headers 0 lines) ---
  246. [Sep 30 13:31:27] DEBUG[26040]: chan_sip.c:9161 find_call: = Looking for Call ID: 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060 (Checking To) --From tag as187358e6 --To-tag
  247. [Sep 30 13:31:27] DEBUG[26040][C-00000000]: chan_sip.c:4600 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #71 - INVITE (got response)
  248. [Sep 30 13:31:27] DEBUG[26040][C-00000000]: chan_sip.c:4607 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060' Request 102: Found
  249. [Sep 30 13:31:27] DEBUG[26040][C-00000000]: chan_sip.c:22550 handle_response_invite: SIP response 100 to standard invite
  250. Reliably Transmitting (NAT) to 10.1.10.70:5060:
  251. CANCEL sip:501@10.1.10.70:5060;transport=udp SIP/2.0
  252. Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK59d734ae;rport
  253. Max-Forwards: 70
  254. From: <sip:123456789@10.1.10.150>;tag=as187358e6
  255. To: <sip:501@10.1.10.70:5060;transport=udp>
  256. Call-ID: 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060
  257. CSeq: 102 CANCEL
  258. User-Agent: Asterisk PBX 11.5.0
  259. Content-Length: 0
  260.  
  261.  
  262. ---
  263. [Sep 30 13:31:27] DEBUG[26040][C-00000000]: chan_sip.c:4334 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #74
  264. [Sep 30 13:31:27] DEBUG[26040][C-00000000]: chan_sip.c:3874 __sip_xmit: Trying to put 'CANCEL sip:' onto UDP socket destined for 10.1.10.70:5060
  265. Scheduling destruction of SIP dialog '46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060' in 13184 ms (Method: INVITE)
  266.  
  267. <--- SIP read from UDP:10.1.10.70:5060 --->
  268. SIP/2.0 200 OK
  269. Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK59d734ae;rport
  270. From: <sip:123456789@10.1.10.150>;tag=as187358e6
  271. To: <sip:501@10.1.10.70:5060;transport=udp>;tag=00131971bbc8011231565a96-165300aa
  272. Call-ID: 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060
  273. CSeq: 102 CANCEL
  274. Server: Cisco-CP7940G/8.0
  275. Content-Length: 0
  276.  
  277. <------------->
  278. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 0 [ 14]: SIP/2.0 200 OK
  279. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 1 [ 62]: Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK59d734ae;rport
  280. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 2 [ 48]: From: <sip:123456789@10.1.10.150>;tag=as187358e6
  281. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 3 [ 81]: To: <sip:501@10.1.10.70:5060;transport=udp>;tag=00131971bbc8011231565a96-165300aa
  282. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 4 [ 58]: Call-ID: 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060
  283. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 5 [ 16]: CSeq: 102 CANCEL
  284. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 6 [ 25]: Server: Cisco-CP7940G/8.0
  285. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 7 [ 17]: Content-Length: 0
  286. --- (8 headers 0 lines) ---
  287. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9161 find_call: = Looking for Call ID: 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060 (Checking To) --From tag as187358e6 --To-tag 00131971bbc8011231565a96-165300aa
  288. [Sep 30 13:31:28] DEBUG[26040][C-00000000]: chan_sip.c:4528 __sip_ack: Acked pending invite 102
  289. [Sep 30 13:31:28] DEBUG[26040][C-00000000]: chan_sip.c:4533 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #74
  290. [Sep 30 13:31:28] DEBUG[26040][C-00000000]: chan_sip.c:4566 __sip_ack: Stopping retransmission on '46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060' of Request 102: Match Found
  291.  
  292. <--- SIP read from UDP:10.1.10.70:5060 --->
  293. SIP/2.0 487 Request Cancelled
  294. Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK59d734ae;rport
  295. From: <sip:123456789@10.1.10.150>;tag=as187358e6
  296. To: <sip:501@10.1.10.70:5060;transport=udp>;tag=00131971bbc8011231565a96-165300aa
  297. Call-ID: 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060
  298. CSeq: 102 INVITE
  299. Server: Cisco-CP7940G/8.0
  300. Contact: <sip:204@10.1.10.70:5060;transport=udp>
  301. Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
  302. Remote-Party-ID: "204" <sip:204@pbx.disklessworkstations.com>;party=called;id-type=subscriber;privacy=off;screen=yes
  303. Content-Length: 0
  304.  
  305. <------------->
  306. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 0 [ 29]: SIP/2.0 487 Request Cancelled
  307. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 1 [ 62]: Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK59d734ae;rport
  308. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 2 [ 48]: From: <sip:123456789@10.1.10.150>;tag=as187358e6
  309. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 3 [ 81]: To: <sip:501@10.1.10.70:5060;transport=udp>;tag=00131971bbc8011231565a96-165300aa
  310. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 4 [ 58]: Call-ID: 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060
  311. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 5 [ 16]: CSeq: 102 INVITE
  312. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 6 [ 25]: Server: Cisco-CP7940G/8.0
  313. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 7 [ 48]: Contact: <sip:204@10.1.10.70:5060;transport=udp>
  314. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 8 [ 65]: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
  315. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 9 [116]: Remote-Party-ID: "204" <sip:204@pbx.disklessworkstations.com>;party=called;id-type=subscriber;privacy=off;screen=yes
  316. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9611 parse_request: Header 10 [ 17]: Content-Length: 0
  317. --- (11 headers 0 lines) ---
  318. [Sep 30 13:31:28] DEBUG[26040]: chan_sip.c:9161 find_call: = Looking for Call ID: 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060 (Checking To) --From tag as187358e6 --To-tag 00131971bbc8011231565a96-165300aa
  319. [Sep 30 13:31:28] DEBUG[26040][C-00000000]: chan_sip.c:4566 __sip_ack: Stopping retransmission on '46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060' of Request 102: Match Found
  320. [Sep 30 13:31:28] DEBUG[26040][C-00000000]: chan_sip.c:22550 handle_response_invite: SIP response 487 to standard invite
  321. Transmitting (NAT) to 10.1.10.70:5060:
  322. ACK sip:501@10.1.10.70:5060;transport=udp SIP/2.0
  323. Via: SIP/2.0/UDP 10.1.10.150:5060;branch=z9hG4bK59d734ae;rport
  324. Max-Forwards: 70
  325. From: <sip:123456789@10.1.10.150>;tag=as187358e6
  326. To: <sip:501@10.1.10.70:5060;transport=udp>;tag=00131971bbc8011231565a96-165300aa
  327. Contact: <sip:123456789@10.1.10.150:5060>
  328. Call-ID: 46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060
  329. CSeq: 102 ACK
  330. User-Agent: Asterisk PBX 11.5.0
  331. Content-Length: 0
  332.  
  333.  
  334. ---
  335. [Sep 30 13:31:28] DEBUG[26040][C-00000000]: chan_sip.c:3874 __sip_xmit: Trying to put 'ACK sip:501' onto UDP socket destined for 10.1.10.70:5060
  336. [Sep 30 13:31:28] DEBUG[26040][C-00000000]: chan_sip.c:6679 update_call_counter: Updating call counter for outgoing call
  337. Scheduling destruction of SIP dialog '46368fa21dd95f38315fa42e658d66b7@10.1.10.150:5060' in 13184 ms (Method: INVITE)
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