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  1. [Oct 24 00:02:35] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
  2. [Oct 24 00:02:37] == Manager 'sendcron' logged off from 127.0.0.1
  3. [Oct 24 00:02:37] == Manager 'sendcron' logged off from 127.0.0.1
  4. [Oct 24 00:02:49] == Parsing '/etc/asterisk/manager.conf': [Oct 24 00:02:49] Found
  5. [Oct 24 00:02:49] == Manager 'sendcron' logged on from 127.0.0.1
  6. [Oct 24 00:02:55] > Channel SIP/gs102-000000a2 was answered.
  7. [Oct 24 00:02:55] -- Executing [8600051@default:1] MeetMe("SIP/gs102-000000a2", "8600051|F") in new stack
  8. [Oct 24 00:02:55] == Parsing '/etc/asterisk/meetme.conf': [Oct 24 00:02:55] Found
  9. [Oct 24 00:02:55] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Oct 24 00:02:55] Found
  10. [Oct 24 00:02:55] -- Created MeetMe conference 1023 for conference '8600051'
  11. [Oct 24 00:02:55] -- <SIP/gs102-000000a2> Playing 'conf-onlyperson' (language 'en')
  12. [Oct 24 00:02:57] == Manager 'sendcron' logged off from 127.0.0.1
  13. [Oct 24 00:03:01] == Parsing '/etc/asterisk/manager.conf': [Oct 24 00:03:01] Found
  14. [Oct 24 00:03:01] == Manager 'sendcron' logged on from 127.0.0.1
  15. [Oct 24 00:03:01] == Parsing '/etc/asterisk/manager.conf': [Oct 24 00:03:01] Found
  16. [Oct 24 00:03:01] == Manager 'sendcron' logged on from 127.0.0.1
  17. [Oct 24 00:03:01] == Manager 'sendcron' logged off from 127.0.0.1
  18. [Oct 24 00:03:01] == Manager 'sendcron' logged off from 127.0.0.1
  19. [Oct 24 00:03:06] == Parsing '/etc/asterisk/manager.conf': [Oct 24 00:03:06] Found
  20. [Oct 24 00:03:06] == Manager 'sendcron' logged on from 127.0.0.1
  21. [Oct 24 00:03:06] == Manager 'sendcron' logged off from 127.0.0.1
  22. [Oct 24 00:03:08] == Parsing '/etc/asterisk/manager.conf': [Oct 24 00:03:08] Found
  23. [Oct 24 00:03:08] == Manager 'sendcron' logged on from 127.0.0.1
  24. [Oct 24 00:03:08] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-70d2,2", "8600051|F") in new stack
  25. [Oct 24 00:03:08] > Channel Local/8600051@default-70d2,1 was answered.
  26. [Oct 24 00:03:08] -- Executing [12503802844@default:1] AGI("Local/8600051@default-70d2,1", "agi://127.0.0.1:4577/call_log") in new stack
  27. [Oct 24 00:03:08] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
  28. [Oct 24 00:03:08] -- Executing [12503802844@default:2] Dial("Local/8600051@default-70d2,1", "SIP/switch2voip/12503802844|10000|To") in new stack
  29. [Oct 24 00:03:08] Audio is at 192.168.1.135 port 16164
  30. [Oct 24 00:03:08] Adding codec 0x4 (ulaw) to SDP
  31. [Oct 24 00:03:08] Adding non-codec 0x1 (telephone-event) to SDP
  32. [Oct 24 00:03:08] Reliably Transmitting (NAT) to 66.33.147.150:5060:
  33. INVITE sip:[email protected];cpd=on SIP/2.0
  34. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK323e60da;rport
  35. From: "M0240003080000278525" <sip:[email protected]>;tag=as5d156ce4
  36. To: <sip:[email protected];cpd=on>
  37. Contact: <sip:[email protected]>
  38. Call-ID: [email protected]
  39. CSeq: 102 INVITE
  40. User-Agent: Asterisk PBX
  41. Max-Forwards: 70
  42. Remote-Party-ID: "M0240003080000278525" <sip:[email protected]>;privacy=off;screen=no
  43. Date: Mon, 24 Oct 2011 07:03:08 GMT
  44. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  45. Supported: replaces
  46. Content-Type: application/sdp
  47. Content-Length: 213
  48.  
  49. v=0
  50. o=root 3322 3322 IN IP4 192.168.1.135
  51. s=session
  52. c=IN IP4 192.168.1.135
  53. t=0 0
  54. m=audio 16164 RTP/AVP 0 101
  55. a=rtpmap:0 PCMU/8000
  56. a=rtpmap:101 telephone-event/8000
  57. a=fmtp:101 0-16
  58. a=ptime:20
  59. a=sendrecv
  60.  
  61. ---
  62. [Oct 24 00:03:08] -- Called switch2voip/12503802844
  63. [Oct 24 00:03:08]
  64. <--- SIP read from 66.33.147.150:5060 --->
  65. SIP/2.0 407 Unauthorized
  66. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK323e60da;received=24.69.86.249;rport=5060
  67. From: "M0240003080000278525" <sip:[email protected]>;tag=as5d156ce4
  68. To: <sip:[email protected];cpd=on>
  69. Call-ID: [email protected]
  70. CSeq: 102 INVITE
  71. Contact: <sip:66.33.147.150:5060>
  72. Server: Net2phone Carrier
  73. Proxy-Authenticate: Digest realm="net2phone",nonce="EA399F0C94498826D23E43A5C9AD2ADB"
  74. Content-Length: 0
  75.  
  76.  
  77. <------------->
  78. [Oct 24 00:03:08] --- (10 headers 0 lines) ---
  79. [Oct 24 00:03:08] Transmitting (NAT) to 66.33.147.150:5060:
  80. ACK sip:[email protected];cpd=on SIP/2.0
  81. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK323e60da;rport
  82. From: "M0240003080000278525" <sip:[email protected]>;tag=as5d156ce4
  83. To: <sip:[email protected];cpd=on>
  84. Contact: <sip:[email protected]>
  85. Call-ID: [email protected]
  86. CSeq: 102 ACK
  87. User-Agent: Asterisk PBX
  88. Max-Forwards: 70
  89. Remote-Party-ID: "M0240003080000278525" <sip:[email protected]>;privacy=off;screen=no
  90. Content-Length: 0
  91.  
  92.  
  93. ---
  94. [Oct 24 00:03:08] Audio is at 192.168.1.135 port 16164
  95. [Oct 24 00:03:08] Adding codec 0x4 (ulaw) to SDP
  96. [Oct 24 00:03:08] Adding non-codec 0x1 (telephone-event) to SDP
  97. [Oct 24 00:03:08] Reliably Transmitting (NAT) to 66.33.147.150:5060:
  98. INVITE sip:[email protected];cpd=on SIP/2.0
  99. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK3cd96ea7;rport
  100. From: "M0240003080000278525" <sip:[email protected]>;tag=as5d156ce4
  101. To: <sip:[email protected];cpd=on>
  102. Contact: <sip:[email protected]>
  103. Call-ID: [email protected]
  104. CSeq: 103 INVITE
  105. User-Agent: Asterisk PBX
  106. Max-Forwards: 70
  107. Remote-Party-ID: "M0240003080000278525" <sip:[email protected]>;privacy=off;screen=no
  108. Proxy-Authorization: Digest username="5208913412", realm="net2phone", algorithm=MD5, uri="sip:[email protected];cpd=on", nonce="EA399F0C94498826D23E43A5C9AD2ADB", response="2a030b288dff62dba555db470aee91b3"
  109. Date: Mon, 24 Oct 2011 07:03:08 GMT
  110. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  111. Supported: replaces
  112. Content-Type: application/sdp
  113. Content-Length: 213
  114.  
  115. v=0
  116. o=root 3322 3323 IN IP4 192.168.1.135
  117. s=session
  118. c=IN IP4 192.168.1.135
  119. t=0 0
  120. m=audio 16164 RTP/AVP 0 101
  121. a=rtpmap:0 PCMU/8000
  122. a=rtpmap:101 telephone-event/8000
  123. a=fmtp:101 0-16
  124. a=ptime:20
  125. a=sendrecv
  126.  
  127. ---
  128. [Oct 24 00:03:08]
  129. <--- SIP read from 66.33.147.150:5060 --->
  130. SIP/2.0 100 Trying
  131. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK3cd96ea7;received=24.69.86.249;rport=5060
  132. From: "M0240003080000278525" <sip:[email protected]>;tag=as5d156ce4
  133. To: <sip:[email protected];cpd=on>;tag=ccid-933870736-1-1877
  134. Call-ID: [email protected]
  135. CSeq: 103 INVITE
  136. Contact: <sip:66.33.147.150:5060>
  137. Server: Net2phone Carrier
  138. Content-Length: 0
  139.  
  140.  
  141. <------------->
  142. [Oct 24 00:03:08] --- (9 headers 0 lines) ---
  143. [Oct 24 00:03:09] == Refreshing DNS lookups.
  144. [Oct 24 00:03:10] == Manager 'sendcron' logged off from 127.0.0.1
  145. [Oct 24 00:03:10]
  146. <--- SIP read from 66.33.147.150:5060 --->
  147. SIP/2.0 183 Session Progress
  148. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK3cd96ea7;received=24.69.86.249;rport=5060
  149. From: "M0240003080000278525" <sip:[email protected]>;tag=as5d156ce4
  150. To: <sip:[email protected];cpd=on>;tag=ccid-933870736-1-1877
  151. Call-ID: [email protected]
  152. CSeq: 103 INVITE
  153. Contact: <sip:66.33.147.150:5060>
  154. Server: Net2phone Carrier
  155. Content-Length: 218
  156. Content-Type: application/sdp
  157.  
  158. v=0
  159. o=5208913412 933870736 933870736 IN IP4 66.33.166.133
  160. s=SIP Call
  161. c=IN IP4 66.33.166.133
  162. t=0 0
  163. m=audio 20368 RTP/AVP 0 101
  164. a=ptime:20
  165. a=rtpmap:0 PCMU/8000
  166. a=rtpmap:101 telephone-event/8000
  167. a=fmtp:101 0-11
  168.  
  169. <------------->
  170. [Oct 24 00:03:10] --- (10 headers 10 lines) ---
  171. [Oct 24 00:03:10] Found RTP audio format 0
  172. [Oct 24 00:03:10] Found RTP audio format 101
  173. [Oct 24 00:03:10] Found audio description format PCMU for ID 0
  174. [Oct 24 00:03:10] Found audio description format telephone-event for ID 101
  175. [Oct 24 00:03:10] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
  176. [Oct 24 00:03:10] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  177. [Oct 24 00:03:10] Peer audio RTP is at port 66.33.166.133:20368
  178. [Oct 24 00:03:10] -- SIP/switch2voip-000000a3 is making progress passing it to Local/8600051@default-70d2,1
  179. [Oct 24 00:03:11] Reliably Transmitting (NAT) to 66.33.147.150:5060:
  180. OPTIONS sip:66.33.147.150;cpd=on SIP/2.0
  181. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK6a1d6546;rport
  182. From: "asterisk" <sip:[email protected]>;tag=as6f7fc30f
  183. To: <sip:66.33.147.150;cpd=on>
  184. Contact: <sip:[email protected]>
  185. Call-ID: [email protected]
  186. CSeq: 102 OPTIONS
  187. User-Agent: Asterisk PBX
  188. Max-Forwards: 70
  189. Date: Mon, 24 Oct 2011 07:03:11 GMT
  190. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  191. Supported: replaces
  192. Content-Length: 0
  193.  
  194.  
  195. ---
  196. [Oct 24 00:03:11]
  197. <--- SIP read from 66.33.147.150:5060 --->
  198. SIP/2.0 200 OK
  199. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK6a1d6546;received=24.69.86.249;rport=5060
  200. From: "asterisk" <sip:[email protected]>;tag=as6f7fc30f
  201. To: <sip:66.33.147.150;cpd=on>
  202. Call-ID: [email protected]
  203. CSeq: 102 OPTIONS
  204. Contact: <sip:66.33.147.150:5060>
  205. Server: Net2Phone Carrier
  206. Content-Length: 0
  207.  
  208.  
  209. <------------->
  210. [Oct 24 00:03:11] --- (9 headers 0 lines) ---
  211. [Oct 24 00:03:11] Really destroying SIP dialog '[email protected]' Method: OPTIONS
  212. [Oct 24 00:03:11] NOTICE[3339]: chan_sip.c:8178 sip_reregister: -- Re-registration for [email protected]
  213. [Oct 24 00:03:11] REGISTER 12 headers, 0 lines
  214. [Oct 24 00:03:11] Reliably Transmitting (NAT) to 66.33.147.150:5060:
  215. REGISTER sip:66.33.147.150 SIP/2.0
  216. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK629b06fa;rport
  217. From: <sip:[email protected]>;tag=as1b35d174
  218. To: <sip:[email protected]>
  219. Call-ID: [email protected]
  220. CSeq: 126 REGISTER
  221. User-Agent: Asterisk PBX
  222. Max-Forwards: 70
  223. Authorization: Digest username="5208913412", realm="net2phone", algorithm=MD5, uri="sip:66.33.147.150", nonce="D85AE13B15FF9ED1B687B6669AAEEA0C", response="3c48bcf14de76aea714239e002b51608"
  224. Expires: 120
  225. Contact: <sip:[email protected]>
  226. Event: registration
  227. Content-Length: 0
  228.  
  229.  
  230. ---
  231. [Oct 24 00:03:12]
  232. <--- SIP read from 66.33.147.150:5060 --->
  233. SIP/2.0 407 Unauthorized
  234. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK629b06fa;received=24.69.86.249;rport=5060
  235. From: <sip:[email protected]>;tag=as1b35d174
  236. To: <sip:[email protected]>
  237. Call-ID: [email protected]
  238. CSeq: 126 REGISTER
  239. Contact: <sip:66.33.147.150:5060>
  240. Server: Net2Phone Carrier
  241. Proxy-Authenticate: Digest realm="net2phone",nonce="D5D174315E659793209D1463D4FC3058"
  242. Content-Length: 0
  243.  
  244.  
  245. <------------->
  246. [Oct 24 00:03:12] --- (10 headers 0 lines) ---
  247. [Oct 24 00:03:12] Responding to challenge, registration to domain/host name 66.33.147.150
  248. [Oct 24 00:03:12] REGISTER 12 headers, 0 lines
  249. [Oct 24 00:03:12] Reliably Transmitting (NAT) to 66.33.147.150:5060:
  250. REGISTER sip:66.33.147.150 SIP/2.0
  251. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK57742523;rport
  252. From: <sip:[email protected]>;tag=as763f7467
  253. To: <sip:[email protected]>
  254. Call-ID: [email protected]
  255. CSeq: 127 REGISTER
  256. User-Agent: Asterisk PBX
  257. Max-Forwards: 70
  258. Proxy-Authorization: Digest username="5208913412", realm="net2phone", algorithm=MD5, uri="sip:66.33.147.150", nonce="D5D174315E659793209D1463D4FC3058", response="b7415c90a739722830ca35334297448f"
  259. Expires: 120
  260. Contact: <sip:[email protected]>
  261. Event: registration
  262. Content-Length: 0
  263.  
  264.  
  265. ---
  266. [Oct 24 00:03:12]
  267. <--- SIP read from 66.33.147.150:5060 --->
  268. SIP/2.0 200 OK
  269. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK3cd96ea7;received=24.69.86.249;rport=5060
  270. From: "M0240003080000278525" <sip:[email protected]>;tag=as5d156ce4
  271. To: <sip:[email protected];cpd=on>;tag=ccid-933870736-1-1877
  272. Allow: ACK,BYE,CANCEL,INVITE,OPTIONS
  273. Call-ID: [email protected]
  274. CSeq: 103 INVITE
  275. Contact: <sip:66.33.147.150:5060>
  276. Server: Net2phone Carrier
  277. Content-Length: 218
  278. Content-Type: application/sdp
  279.  
  280. v=0
  281. o=5208913412 933870736 933870736 IN IP4 66.33.166.133
  282. s=SIP Call
  283. c=IN IP4 66.33.166.133
  284. t=0 0
  285. m=audio 20368 RTP/AVP 0 101
  286. a=ptime:20
  287. a=rtpmap:0 PCMU/8000
  288. a=rtpmap:101 telephone-event/8000
  289. a=fmtp:101 0-11
  290.  
  291. <------------->
  292. [Oct 24 00:03:12] --- (11 headers 10 lines) ---
  293. [Oct 24 00:03:12] Found RTP audio format 0
  294. [Oct 24 00:03:12] Found RTP audio format 101
  295. [Oct 24 00:03:12] Found audio description format PCMU for ID 0
  296. [Oct 24 00:03:12] Found audio description format telephone-event for ID 101
  297. [Oct 24 00:03:12] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
  298. [Oct 24 00:03:12] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  299. [Oct 24 00:03:12] Peer audio RTP is at port 66.33.166.133:20368
  300. [Oct 24 00:03:12] list_route: hop: <sip:66.33.147.150:5060>
  301. [Oct 24 00:03:12] set_destination: Parsing <sip:66.33.147.150:5060> for address/port to send to
  302. [Oct 24 00:03:12] set_destination: set destination to 66.33.147.150, port 5060
  303. [Oct 24 00:03:12] Transmitting (NAT) to 66.33.147.150:5060:
  304. ACK sip:66.33.147.150:5060 SIP/2.0
  305. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK289cb3ef;rport
  306. From: "M0240003080000278525" <sip:[email protected]>;tag=as5d156ce4
  307. To: <sip:[email protected];cpd=on>;tag=ccid-933870736-1-1877
  308. Contact: <sip:[email protected]>
  309. Call-ID: [email protected]
  310. CSeq: 103 ACK
  311. User-Agent: Asterisk PBX
  312. Max-Forwards: 70
  313. Remote-Party-ID: "M0240003080000278525" <sip:[email protected]>;privacy=off;screen=no
  314. Content-Length: 0
  315.  
  316.  
  317. ---
  318. [Oct 24 00:03:12] -- SIP/switch2voip-000000a3 answered Local/8600051@default-70d2,1
  319. [Oct 24 00:03:12]
  320. <--- SIP read from 66.33.147.150:5060 --->
  321. SIP/2.0 200 OK
  322. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK57742523;received=24.69.86.249;rport=5060
  323. From: <sip:[email protected]>;tag=as763f7467
  324. To: <sip:[email protected]>
  325. Call-ID: [email protected]
  326. Expires: 90
  327. CSeq: 127 REGISTER
  328. Contact: <sip:[email protected]>
  329. Server: Net2Phone Carrier
  330. Content-Length: 0
  331.  
  332.  
  333. <------------->
  334. [Oct 24 00:03:12] --- (10 headers 0 lines) ---
  335. [Oct 24 00:03:12] Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
  336. [Oct 24 00:03:12] NOTICE[3339]: chan_sip.c:13779 handle_response_register: Outbound Registration: Expiry for 66.33.147.150 is 90 sec (Scheduling reregistration in 75 s)
  337. [Oct 24 00:03:21] == Parsing '/etc/asterisk/manager.conf': [Oct 24 00:03:21] Found
  338. [Oct 24 00:03:21] == Manager 'sendcron' logged on from 127.0.0.1
  339. [Oct 24 00:03:21] -- Executing [h@default:1] DeadAGI("Local/8600051@default-70d2,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----13-----9") in new stack
  340. [Oct 24 00:03:21] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----13-----9 completed, returning 0
  341. [Oct 24 00:03:21] Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
  342. [Oct 24 00:03:21] set_destination: Parsing <sip:66.33.147.150:5060> for address/port to send to
  343. [Oct 24 00:03:21] set_destination: set destination to 66.33.147.150, port 5060
  344. [Oct 24 00:03:21] Reliably Transmitting (NAT) to 66.33.147.150:5060:
  345. BYE sip:66.33.147.150:5060 SIP/2.0
  346. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK4cd2c970;rport
  347. From: "M0240003080000278525" <sip:[email protected]>;tag=as5d156ce4
  348. To: <sip:[email protected];cpd=on>;tag=ccid-933870736-1-1877
  349. Call-ID: [email protected]
  350. CSeq: 104 BYE
  351. User-Agent: Asterisk PBX
  352. Max-Forwards: 70
  353. Remote-Party-ID: "M0240003080000278525" <sip:[email protected]>;privacy=off;screen=no
  354. Proxy-Authorization: Digest username="5208913412", realm="net2phone", algorithm=MD5, uri="sip:66.33.147.150:5060", nonce="EA399F0C94498826D23E43A5C9AD2ADB", response="633f963ca96385af356b002df9e3a570"
  355. X-Asterisk-HangupCause: Normal Clearing
  356. X-Asterisk-HangupCauseCode: 16
  357. Content-Length: 0
  358.  
  359.  
  360. ---
  361. [Oct 24 00:03:21] == Spawn extension (default, 12503802844, 2) exited non-zero on 'Local/8600051@default-70d2,1'
  362. [Oct 24 00:03:21] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-70d2,2'
  363. [Oct 24 00:03:21] -- Executing [h@default:1] DeadAGI("Local/8600051@default-70d2,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
  364. [Oct 24 00:03:21] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
  365. [Oct 24 00:03:21] == Parsing '/etc/asterisk/manager.conf': [Oct 24 00:03:21] Found
  366. [Oct 24 00:03:21] == Manager 'sendcron' logged on from 127.0.0.1
  367. [Oct 24 00:03:21]
  368. <--- SIP read from 66.33.147.150:5060 --->
  369. SIP/2.0 200 OK
  370. Via: SIP/2.0/UDP 192.168.1.135:5060;branch=z9hG4bK4cd2c970;received=24.69.86.249;rport=5060
  371. From: "M0240003080000278525" <sip:[email protected]>;tag=as5d156ce4
  372. To: <sip:[email protected];cpd=on>;tag=ccid-933870736-1-1877
  373. Call-ID: [email protected]
  374. CSeq: 104 BYE
  375. Content-Length: 0
  376.  
  377.  
  378. <------------->
  379. [Oct 24 00:03:21] --- (7 headers 0 lines) ---
  380. [Oct 24 00:03:21] Really destroying SIP dialog '[email protected]' Method: INVITE
  381. [Oct 24 00:03:23] == Manager 'sendcron' logged off from 127.0.0.1
  382. [Oct 24 00:03:23] == Manager 'sendcron' logged off from 127.0.0.1
  383. [Oct 24 00:03:34] == Parsing '/etc/asterisk/manager.conf': [Oct 24 00:03:34] Found
  384. [Oct 24 00:03:34] == Manager 'sendcron' logged on from 127.0.0.1
  385. [Oct 24 00:03:34] -- Hungup 'DAHDI/pseudo-2120088752'
  386. [Oct 24 00:03:34] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/gs102-000000a2'
  387. [Oct 24 00:03:34] -- Executing [h@default:1] DeadAGI("SIP/gs102-000000a2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
  388. [Oct 24 00:03:34] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
  389. [Oct 24 00:03:34] == Parsing '/etc/asterisk/manager.conf': [Oct 24 00:03:34] Found
  390. [Oct 24 00:03:34] == Manager 'sendcron' logged on from 127.0.0.1
  391. [Oct 24 00:03:34] -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-74a1,2", "8600051|K") in new stack
  392. [Oct 24 00:03:34] WARNING[7969]: app_meetme.c:3134 admin_exec: Conference number '8600051' not found!
  393. [Oct 24 00:03:34] -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-74a1,2", "") in new stack
  394. [Oct 24 00:03:34] == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-74a1,2'
  395. [Oct 24 00:03:34] -- Executing [h@default:1] DeadAGI("Local/55558600051@default-74a1,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
  396. [Oct 24 00:03:34] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
  397. [Oct 24 00:03:36] == Manager 'sendcron' logged off from 127.0.0.1
  398. [Oct 24 00:03:36] == Manager 'sendcron' logged off from 127.0.0.1
  399. [Oct 24 00:03:44] Really destroying SIP dialog '[email protected]' Method: REGISTER
  400. vicibox*CLI>
  401.  
  402.  
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