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  1. Scheduling destruction of SIP dialog '3bfe27191964963c057526493854649b@10.71.173.187:5060' in 32000 ms (Method: INVITE)
  2. -- Executing [failed@obd_promo:1] Set("OutgoingSpoolFailed", "RES=OK") in new stack
  3. -- Auto fallthrough, channel 'OutgoingSpoolFailed' status is 'UNKNOWN'
  4. [2015-01-22 12:14:13.830] NOTICE[41678]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (3) Remote end Ringing
  5. [2015-01-22 12:14:13.830] NOTICE[41678]: pbx_spool.c:392 attempt_thread: Queued call to SIP/+2349096852432@outgoing expired without completion after 0 attempts
  6. Retransmitting #6 (no NAT) to 10.67.64.4:5060:
  7. INVITE sip:+2349096852432@10.67.64.4 SIP/2.0
  8. Via: SIP/2.0/UDP 10.71.173.187:5060;branch=z9hG4bK0c96057f
  9. Max-Forwards: 70
  10. From: <sip:+23454883@10.71.173.187>;tag=as1ede5a0c
  11. To: <sip:+2349096852432@10.67.64.4>
  12. Contact: <sip:+23454883@10.71.173.187:5060>
  13. Call-ID: 3bfe27191964963c057526493854649b@10.71.173.187:5060
  14. CSeq: 102 INVITE
  15. User-Agent: Asterisk PBX 11.12.0
  16. Date: Thu, 22 Jan 2015 12:13:43 GMT
  17. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  18. Supported: replaces, timer
  19. Content-Type: application/sdp
  20. Content-Length: 285
  21.  
  22. v=0
  23. o=root 1424000242 1424000242 IN IP4 10.71.173.187
  24. s=Asterisk PBX 11.12.0
  25. c=IN IP4 10.71.173.187
  26. t=0 0
  27. m=audio 12536 RTP/AVP 8 3 0 101
  28. a=rtpmap:8 PCMA/8000
  29. a=rtpmap:3 GSM/8000
  30. a=rtpmap:0 PCMU/8000
  31. a=rtpmap:101 telephone-event/8000
  32. a=fmtp:101 0-16
  33. a=ptime:20
  34. a=sendrecv
  35.  
  36. ---
  37. [2015-01-22 12:14:15.794] WARNING[28085]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission 3bfe27191964963c057526493854649b@10.71.173.187:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  38. Packet timed out after 32000ms with no response
  39. Really destroying SIP dialog '3bfe27191964963c057526493854649b@10.71.173.187:5060' Method: INVITE
  40.  
  41. <--- SIP read from UDP:10.67.64.4:5064 --->
  42. OPTIONS sip:10.71.173.187:5060 SIP/2.0
  43. Via: SIP/2.0/UDP 10.67.64.4:5064;branch=z9hG4bKk0hvlvqjhjfq3wjiwi01ijq00;X-DptMsg=143
  44. Call-ID: 4ff2w3lhiqvjbw33fffw2fikblii34wh@10.18.5.64
  45. From: <sip:10.67.64.4:5060>;tag=bk12ibfm
  46. To: <sip:10.71.173.187:5060>
  47. CSeq: 1 OPTIONS
  48. Max-Forwards: 70
  49. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
  50. Content-Length: 0
  51.  
  52. <------------->
  53. --- (9 headers 0 lines) ---
  54. Sending to 10.67.64.4:5064 (no NAT)
  55. Looking for s in public (domain 10.71.173.187)
  56.  
  57. <--- Transmitting (no NAT) to 10.67.64.4:5064 --->
  58. SIP/2.0 404 Not Found
  59. Via: SIP/2.0/UDP 10.67.64.4:5064;branch=z9hG4bKk0hvlvqjhjfq3wjiwi01ijq00;X-DptMsg=143;received=10.67.64.4
  60. From: <sip:10.67.64.4:5060>;tag=bk12ibfm
  61. To: <sip:10.71.173.187:5060>;tag=as26a2d6a2
  62. Call-ID: 4ff2w3lhiqvjbw33fffw2fikblii34wh@10.18.5.64
  63. CSeq: 1 OPTIONS
  64. Server: Asterisk PBX 11.12.0
  65. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  66. Supported: replaces, timer
  67. Accept: application/sdp
  68. Content-Length: 0
  69.  
  70.  
  71. <------------>
  72. Scheduling destruction of SIP dialog '4ff2w3lhiqvjbw33fffw2fikblii34wh@10.18.5.64' in 32000 ms (Method: OPTIONS)
  73. Really destroying SIP dialog 'bqbmijh4wwl4qb3kimijhlm3jm04m2wj@10.18.5.64' Method: OPTIONS
  74.  
  75. <--- SIP read from UDP:10.67.64.4:5066 --->
  76. OPTIONS sip:10.71.173.187:5060 SIP/2.0
  77. Via: SIP/2.0/UDP 10.67.64.4:5066;branch=z9hG4bKw4qw0j2fq0m14kj1mk12fmmbk;X-DptMsg=145
  78. Call-ID: jl1mq3mj402k0vvim00lkfqqkv0kb4ii@10.18.5.64
  79. From: <sip:10.67.64.4:5060>;tag=0m04lvk3
  80. To: <sip:10.71.173.187:5060>
  81. CSeq: 1 OPTIONS
  82. Max-Forwards: 70
  83. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
  84. Content-Length: 0
  85.  
  86. <------------->
  87. --- (9 headers 0 lines) ---
  88. Sending to 10.67.64.4:5066 (no NAT)
  89. Looking for s in public (domain 10.71.173.187)
  90.  
  91. <--- Transmitting (no NAT) to 10.67.64.4:5066 --->
  92. SIP/2.0 404 Not Found
  93. Via: SIP/2.0/UDP 10.67.64.4:5066;branch=z9hG4bKw4qw0j2fq0m14kj1mk12fmmbk;X-DptMsg=145;received=10.67.64.4
  94. From: <sip:10.67.64.4:5060>;tag=0m04lvk3
  95. To: <sip:10.71.173.187:5060>;tag=as0f0d5fb0
  96. Call-ID: jl1mq3mj402k0vvim00lkfqqkv0kb4ii@10.18.5.64
  97. CSeq: 1 OPTIONS
  98. Server: Asterisk PBX 11.12.0
  99. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  100. Supported: replaces, timer
  101. Accept: application/sdp
  102. Content-Length: 0
  103.  
  104.  
  105. <------------>
  106. Scheduling destruction of SIP dialog 'jl1mq3mj402k0vvim00lkfqqkv0kb4ii@10.18.5.64' in 32000 ms (Method: OPTIONS)
  107. Really destroying SIP dialog '4ff2w3lhiqvjbw33fffw2fikblii34wh@10.18.5.64' Method: OPTIONS
  108.  
  109. <--- SIP read from UDP:10.67.64.4:5067 --->
  110. OPTIONS sip:10.71.173.187:5060 SIP/2.0
  111. Via: SIP/2.0/UDP 10.67.64.4:5067;branch=z9hG4bKflqf2ikkh0ki04hvqm22jw1wv;X-DptMsg=146
  112. Call-ID: 2hh31w2klbjj2qhvkhbw33wbhjmbjkv0@10.18.5.64
  113. From: <sip:10.67.64.4:5060>;tag=4hji0mii
  114. To: <sip:10.71.173.187:5060>
  115. CSeq: 1 OPTIONS
  116. Max-Forwards: 70
  117. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
  118. Content-Length: 0
  119.  
  120. <------------->
  121. --- (9 headers 0 lines) ---
  122. Sending to 10.67.64.4:5067 (no NAT)
  123. Looking for s in public (domain 10.71.173.187)
  124.  
  125. <--- Transmitting (no NAT) to 10.67.64.4:5067 --->
  126. SIP/2.0 404 Not Found
  127. Via: SIP/2.0/UDP 10.67.64.4:5067;branch=z9hG4bKflqf2ikkh0ki04hvqm22jw1wv;X-DptMsg=146;received=10.67.64.4
  128. From: <sip:10.67.64.4:5060>;tag=4hji0mii
  129. To: <sip:10.71.173.187:5060>;tag=as777b755d
  130. Call-ID: 2hh31w2klbjj2qhvkhbw33wbhjmbjkv0@10.18.5.64
  131. CSeq: 1 OPTIONS
  132. Server: Asterisk PBX 11.12.0
  133. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  134. Supported: replaces, timer
  135. Accept: application/sdp
  136. Content-Length: 0
  137.  
  138.  
  139. <------------>
  140. Scheduling destruction of SIP dialog '2hh31w2klbjj2qhvkhbw33wbhjmbjkv0@10.18.5.64' in 32000 ms (Method: OPTIONS)
  141. Really destroying SIP dialog 'jl1mq3mj402k0vvim00lkfqqkv0kb4ii@10.18.5.64' Method: OPTIONS
  142.  
  143. <--- SIP read from UDP:10.67.64.4:5069 --->
  144. OPTIONS sip:10.71.173.187:5060 SIP/2.0
  145. Via: SIP/2.0/UDP 10.67.64.4:5069;branch=z9hG4bK2vf42mm2q1wl0bhw1fmwq4mbi;X-DptMsg=148
  146. Call-ID: v23i2wihqih34fvffm0fv32vh3milfvm@10.18.5.64
  147. From: <sip:10.67.64.4:5060>;tag=kkjl4fvb
  148. To: <sip:10.71.173.187:5060>
  149. CSeq: 1 OPTIONS
  150. Max-Forwards: 70
  151. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
  152. Content-Length: 0
  153.  
  154. <------------->
  155. --- (9 headers 0 lines) ---
  156. Sending to 10.67.64.4:5069 (no NAT)
  157. Looking for s in public (domain 10.71.173.187)
  158.  
  159. <--- Transmitting (no NAT) to 10.67.64.4:5069 --->
  160. SIP/2.0 404 Not Found
  161. Via: SIP/2.0/UDP 10.67.64.4:5069;branch=z9hG4bK2vf42mm2q1wl0bhw1fmwq4mbi;X-DptMsg=148;received=10.67.64.4
  162. From: <sip:10.67.64.4:5060>;tag=kkjl4fvb
  163. To: <sip:10.71.173.187:5060>;tag=as5a82d63d
  164. Call-ID: v23i2wihqih34fvffm0fv32vh3milfvm@10.18.5.64
  165. CSeq: 1 OPTIONS
  166. Server: Asterisk PBX 11.12.0
  167. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  168. Supported: replaces, timer
  169. Accept: application/sdp
  170. Content-Length: 0
  171.  
  172.  
  173. <------------>
  174. Scheduling destruction of SIP dialog 'v23i2wihqih34fvffm0fv32vh3milfvm@10.18.5.64' in 32000 ms (Method: OPTIONS)
  175. Really destroying SIP dialog '2hh31w2klbjj2qhvkhbw33wbhjmbjkv0@10.18.5.64' Method: OPTIONS
  176.  
  177.  
  178.  
  179. rancardivr-etisalatng-app2*CLI>
  180. -- Attempting call on SIP/+2349096852432@outgoing for start@obd_promo:1 (Retry 1)
  181. == Using SIP RTP CoS mark 5
  182. Audio is at 28264
  183. Adding codec 100004 (alaw) to SDP
  184. Adding codec 100002 (gsm) to SDP
  185. Adding codec 100003 (ulaw) to SDP
  186. Adding non-codec 0x1 (telephone-event) to SDP
  187. Reliably Transmitting (no NAT) to 10.67.64.4:5060:
  188. INVITE sip:+2349096852432@10.67.64.4 SIP/2.0
  189. Via: SIP/2.0/UDP 10.71.173.187:5060;branch=z9hG4bK323a9aaa
  190. Max-Forwards: 70
  191. From: <sip:+23454883@10.71.173.187>;tag=as28b91fbe
  192. To: <sip:+2349096852432@10.67.64.4>
  193. Contact: <sip:+23454883@10.71.173.187:5060>
  194. Call-ID: 37e3e9b51e4f26f038956be5533ed56d@10.71.173.187:5060
  195. CSeq: 102 INVITE
  196. User-Agent: Asterisk PBX 11.12.0
  197. Date: Thu, 22 Jan 2015 12:16:23 GMT
  198. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  199. Supported: replaces, timer
  200. Content-Type: application/sdp
  201. Content-Length: 285
  202.  
  203. v=0
  204. o=root 1260239039 1260239039 IN IP4 10.71.173.187
  205. s=Asterisk PBX 11.12.0
  206. c=IN IP4 10.71.173.187
  207. t=0 0
  208. m=audio 28264 RTP/AVP 8 3 0 101
  209. a=rtpmap:8 PCMA/8000
  210. a=rtpmap:3 GSM/8000
  211. a=rtpmap:0 PCMU/8000
  212. a=rtpmap:101 telephone-event/8000
  213. a=fmtp:101 0-16
  214. a=ptime:20
  215. a=sendrecv
  216.  
  217. ---
  218. Retransmitting #1 (no NAT) to 10.67.64.4:5060:
  219. INVITE sip:+2349096852432@10.67.64.4 SIP/2.0
  220. Via: SIP/2.0/UDP 10.71.173.187:5060;branch=z9hG4bK323a9aaa
  221. Max-Forwards: 70
  222. From: <sip:+23454883@10.71.173.187>;tag=as28b91fbe
  223. To: <sip:+2349096852432@10.67.64.4>
  224. Contact: <sip:+23454883@10.71.173.187:5060>
  225. Call-ID: 37e3e9b51e4f26f038956be5533ed56d@10.71.173.187:5060
  226. CSeq: 102 INVITE
  227. User-Agent: Asterisk PBX 11.12.0
  228. Date: Thu, 22 Jan 2015 12:16:23 GMT
  229. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  230. Supported: replaces, timer
  231. Content-Type: application/sdp
  232. Content-Length: 285
  233.  
  234. v=0
  235. o=root 1260239039 1260239039 IN IP4 10.71.173.187
  236. s=Asterisk PBX 11.12.0
  237. c=IN IP4 10.71.173.187
  238. t=0 0
  239. m=audio 28264 RTP/AVP 8 3 0 101
  240. a=rtpmap:8 PCMA/8000
  241. a=rtpmap:3 GSM/8000
  242. a=rtpmap:0 PCMU/8000
  243. a=rtpmap:101 telephone-event/8000
  244. a=fmtp:101 0-16
  245. a=ptime:20
  246. a=sendrecv
  247.  
  248. ---
  249. Retransmitting #2 (no NAT) to 10.67.64.4:5060:
  250. INVITE sip:+2349096852432@10.67.64.4 SIP/2.0
  251. Via: SIP/2.0/UDP 10.71.173.187:5060;branch=z9hG4bK323a9aaa
  252. Max-Forwards: 70
  253. From: <sip:+23454883@10.71.173.187>;tag=as28b91fbe
  254. To: <sip:+2349096852432@10.67.64.4>
  255. Contact: <sip:+23454883@10.71.173.187:5060>
  256. Call-ID: 37e3e9b51e4f26f038956be5533ed56d@10.71.173.187:5060
  257. CSeq: 102 INVITE
  258. User-Agent: Asterisk PBX 11.12.0
  259. Date: Thu, 22 Jan 2015 12:16:23 GMT
  260. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  261. Supported: replaces, timer
  262. Content-Type: application/sdp
  263. Content-Length: 285
  264.  
  265. v=0
  266. o=root 1260239039 1260239039 IN IP4 10.71.173.187
  267. s=Asterisk PBX 11.12.0
  268. c=IN IP4 10.71.173.187
  269. t=0 0
  270. m=audio 28264 RTP/AVP 8 3 0 101
  271. a=rtpmap:8 PCMA/8000
  272. a=rtpmap:3 GSM/8000
  273. a=rtpmap:0 PCMU/8000
  274. a=rtpmap:101 telephone-event/8000
  275. a=fmtp:101 0-16
  276. a=ptime:20
  277. a=sendrecv
  278.  
  279. ---
  280. Retransmitting #3 (no NAT) to 10.67.64.4:5060:
  281. INVITE sip:+2349096852432@10.67.64.4 SIP/2.0
  282. Via: SIP/2.0/UDP 10.71.173.187:5060;branch=z9hG4bK323a9aaa
  283. Max-Forwards: 70
  284. From: <sip:+23454883@10.71.173.187>;tag=as28b91fbe
  285. To: <sip:+2349096852432@10.67.64.4>
  286. Contact: <sip:+23454883@10.71.173.187:5060>
  287. Call-ID: 37e3e9b51e4f26f038956be5533ed56d@10.71.173.187:5060
  288. CSeq: 102 INVITE
  289. User-Agent: Asterisk PBX 11.12.0
  290. Date: Thu, 22 Jan 2015 12:16:23 GMT
  291. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  292. Supported: replaces, timer
  293. Content-Type: application/sdp
  294. Content-Length: 285
  295.  
  296. v=0
  297. o=root 1260239039 1260239039 IN IP4 10.71.173.187
  298. s=Asterisk PBX 11.12.0
  299. c=IN IP4 10.71.173.187
  300. t=0 0
  301. m=audio 28264 RTP/AVP 8 3 0 101
  302. a=rtpmap:8 PCMA/8000
  303. a=rtpmap:3 GSM/8000
  304. a=rtpmap:0 PCMU/8000
  305. a=rtpmap:101 telephone-event/8000
  306. a=fmtp:101 0-16
  307. a=ptime:20
  308. a=sendrecv
  309.  
  310. ---
  311. Retransmitting #4 (no NAT) to 10.67.64.4:5060:
  312. INVITE sip:+2349096852432@10.67.64.4 SIP/2.0
  313. Via: SIP/2.0/UDP 10.71.173.187:5060;branch=z9hG4bK323a9aaa
  314. Max-Forwards: 70
  315. From: <sip:+23454883@10.71.173.187>;tag=as28b91fbe
  316. To: <sip:+2349096852432@10.67.64.4>
  317. Contact: <sip:+23454883@10.71.173.187:5060>
  318. Call-ID: 37e3e9b51e4f26f038956be5533ed56d@10.71.173.187:5060
  319. CSeq: 102 INVITE
  320. User-Agent: Asterisk PBX 11.12.0
  321. Date: Thu, 22 Jan 2015 12:16:23 GMT
  322. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  323. Supported: replaces, timer
  324. Content-Type: application/sdp
  325. Content-Length: 285
  326.  
  327. v=0
  328. o=root 1260239039 1260239039 IN IP4 10.71.173.187
  329. s=Asterisk PBX 11.12.0
  330. c=IN IP4 10.71.173.187
  331. t=0 0
  332. m=audio 28264 RTP/AVP 8 3 0 101
  333. a=rtpmap:8 PCMA/8000
  334. a=rtpmap:3 GSM/8000
  335. a=rtpmap:0 PCMU/8000
  336. a=rtpmap:101 telephone-event/8000
  337. a=fmtp:101 0-16
  338. a=ptime:20
  339. a=sendrecv
  340.  
  341. ---
  342.  
  343. <--- SIP read from UDP:10.67.64.4:5061 --->
  344. OPTIONS sip:10.71.173.187:5060 SIP/2.0
  345. Via: SIP/2.0/UDP 10.67.64.4:5061;branch=z9hG4bKkk0qm4k4vkii24fji3bjf0i2h;X-DptMsg=140
  346. Call-ID: m2vlf2j3qwv3wbkiv3b0w3q2fjwf24q1@10.18.5.64
  347. From: <sip:10.67.64.4:5060>;tag=hw12i123
  348. To: <sip:10.71.173.187:5060>
  349. CSeq: 1 OPTIONS
  350. Max-Forwards: 70
  351. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
  352. Content-Length: 0
  353.  
  354. <------------->
  355. --- (9 headers 0 lines) ---
  356. Sending to 10.67.64.4:5061 (no NAT)
  357. Looking for s in public (domain 10.71.173.187)
  358.  
  359. <--- Transmitting (no NAT) to 10.67.64.4:5061 --->
  360. SIP/2.0 404 Not Found
  361. Via: SIP/2.0/UDP 10.67.64.4:5061;branch=z9hG4bKkk0qm4k4vkii24fji3bjf0i2h;X-DptMsg=140;received=10.67.64.4
  362. From: <sip:10.67.64.4:5060>;tag=hw12i123
  363. To: <sip:10.71.173.187:5060>;tag=as21db8366
  364. Call-ID: m2vlf2j3qwv3wbkiv3b0w3q2fjwf24q1@10.18.5.64
  365. CSeq: 1 OPTIONS
  366. Server: Asterisk PBX 11.12.0
  367. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  368. Supported: replaces, timer
  369. Accept: application/sdp
  370. Content-Length: 0
  371.  
  372.  
  373. <------------>
  374. Scheduling destruction of SIP dialog 'm2vlf2j3qwv3wbkiv3b0w3q2fjwf24q1@10.18.5.64' in 32000 ms (Method: OPTIONS)
  375. Really destroying SIP dialog 'v23i2wihqih34fvffm0fv32vh3milfvm@10.18.5.64' Method: OPTIONS
  376. Retransmitting #5 (no NAT) to 10.67.64.4:5060:
  377. INVITE sip:+2349096852432@10.67.64.4 SIP/2.0
  378. Via: SIP/2.0/UDP 10.71.173.187:5060;branch=z9hG4bK323a9aaa
  379. Max-Forwards: 70
  380. From: <sip:+23454883@10.71.173.187>;tag=as28b91fbe
  381. To: <sip:+2349096852432@10.67.64.4>
  382. Contact: <sip:+23454883@10.71.173.187:5060>
  383. Call-ID: 37e3e9b51e4f26f038956be5533ed56d@10.71.173.187:5060
  384. CSeq: 102 INVITE
  385. User-Agent: Asterisk PBX 11.12.0
  386. Date: Thu, 22 Jan 2015 12:16:23 GMT
  387. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  388. Supported: replaces, timer
  389. Content-Type: application/sdp
  390. Content-Length: 285
  391.  
  392. v=0
  393. o=root 1260239039 1260239039 IN IP4 10.71.173.187
  394. s=Asterisk PBX 11.12.0
  395. c=IN IP4 10.71.173.187
  396. t=0 0
  397. m=audio 28264 RTP/AVP 8 3 0 101
  398. a=rtpmap:8 PCMA/8000
  399. a=rtpmap:3 GSM/8000
  400. a=rtpmap:0 PCMU/8000
  401. a=rtpmap:101 telephone-event/8000
  402. a=fmtp:101 0-16
  403. a=ptime:20
  404. a=sendrecv
  405.  
  406. ---
  407. Scheduling destruction of SIP dialog '37e3e9b51e4f26f038956be5533ed56d@10.71.173.187:5060' in 32000 ms (Method: INVITE)
  408. -- Executing [failed@obd_promo:1] Set("OutgoingSpoolFailed", "RES=OK") in new stack
  409. -- Auto fallthrough, channel 'OutgoingSpoolFailed' status is 'UNKNOWN'
  410. [2015-01-22 12:16:53.882] NOTICE[45226]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (3) Remote end Ringing
  411. [2015-01-22 12:16:53.882] NOTICE[45226]: pbx_spool.c:392 attempt_thread: Queued call to SIP/+2349096852432@outgoing expired without completion after 0 attempts
  412. Retransmitting #6 (no NAT) to 10.67.64.4:5060:
  413. INVITE sip:+2349096852432@10.67.64.4 SIP/2.0
  414. Via: SIP/2.0/UDP 10.71.173.187:5060;branch=z9hG4bK323a9aaa
  415. Max-Forwards: 70
  416. From: <sip:+23454883@10.71.173.187>;tag=as28b91fbe
  417. To: <sip:+2349096852432@10.67.64.4>
  418. Contact: <sip:+23454883@10.71.173.187:5060>
  419. Call-ID: 37e3e9b51e4f26f038956be5533ed56d@10.71.173.187:5060
  420. CSeq: 102 INVITE
  421. User-Agent: Asterisk PBX 11.12.0
  422. Date: Thu, 22 Jan 2015 12:16:23 GMT
  423. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  424. Supported: replaces, timer
  425. Content-Type: application/sdp
  426. Content-Length: 285
  427.  
  428. v=0
  429. o=root 1260239039 1260239039 IN IP4 10.71.173.187
  430. s=Asterisk PBX 11.12.0
  431. c=IN IP4 10.71.173.187
  432. t=0 0
  433. m=audio 28264 RTP/AVP 8 3 0 101
  434. a=rtpmap:8 PCMA/8000
  435. a=rtpmap:3 GSM/8000
  436. a=rtpmap:0 PCMU/8000
  437. a=rtpmap:101 telephone-event/8000
  438. a=fmtp:101 0-16
  439. a=ptime:20
  440. a=sendrecv
  441.  
  442. ---
  443. [2015-01-22 12:16:55.847] WARNING[28085]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission 37e3e9b51e4f26f038956be5533ed56d@10.71.173.187:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  444. Packet timed out after 32000ms with no response
  445. Really destroying SIP dialog '37e3e9b51e4f26f038956be5533ed56d@10.71.173.187:5060' Method: INVITE
  446.  
  447. <--- SIP read from UDP:10.67.64.4:5062 --->
  448. OPTIONS sip:10.71.173.187:5060 SIP/2.0
  449. Via: SIP/2.0/UDP 10.67.64.4:5062;branch=z9hG4bK3qvkjbqvk0kjjbhhb3vvm4qfk;X-DptMsg=141
  450. Call-ID: fwqbwijjkkjmbhvjhf2m140jwbqfiv2i@10.18.5.64
  451. From: <sip:10.67.64.4:5060>;tag=mkfw2iw3
  452. To: <sip:10.71.173.187:5060>
  453. CSeq: 1 OPTIONS
  454. Max-Forwards: 70
  455. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
  456. Content-Length: 0
  457.  
  458. <------------->
  459. --- (9 headers 0 lines) ---
  460. Sending to 10.67.64.4:5062 (no NAT)
  461. Looking for s in public (domain 10.71.173.187)
  462.  
  463. <--- Transmitting (no NAT) to 10.67.64.4:5062 --->
  464. SIP/2.0 404 Not Found
  465. Via: SIP/2.0/UDP 10.67.64.4:5062;branch=z9hG4bK3qvkjbqvk0kjjbhhb3vvm4qfk;X-DptMsg=141;received=10.67.64.4
  466. From: <sip:10.67.64.4:5060>;tag=mkfw2iw3
  467. To: <sip:10.71.173.187:5060>;tag=as3dc53525
  468. Call-ID: fwqbwijjkkjmbhvjhf2m140jwbqfiv2i@10.18.5.64
  469. CSeq: 1 OPTIONS
  470. Server: Asterisk PBX 11.12.0
  471. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  472. Supported: replaces, timer
  473. Accept: application/sdp
  474. Content-Length: 0
  475.  
  476.  
  477. <------------>
  478. Scheduling destruction of SIP dialog 'fwqbwijjkkjmbhvjhf2m140jwbqfiv2i@10.18.5.64' in 32000 ms (Method: OPTIONS)
  479. rancardivr-etisalatng-app2*CLI>
  480. Disconnected from Asterisk server
  481. Asterisk cleanly ending (0).
  482. Executing last minute cleanups
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