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  1. <--- SIP read from UDP:95.26.67.247:5062 --->
  2. INVITE sip:79250287897@sip2.bad-times.wtf SIP/2.0
  3. Via: SIP/2.0/UDP 95.26.67.247:5062;rport;branch=z9hG4bK498352626
  4. From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
  5. To: <sip:79250287897@sip2.bad-times.wtf>
  6. Call-ID: 1621070911@192.168.0.25
  7. CSeq: 20 INVITE
  8. Contact: <sip:root@95.26.67.247:5062>
  9. Max-Forwards: 70
  10. User-Agent: qutecom/rev-g-trunk
  11. Expires: 120
  12. Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
  13. Content-Type: application/sdp
  14. Content-Length: 370
  15.  
  16. v=0
  17. o=userX 20000001 20000001 IN IP4 95.26.67.247
  18. s=A call
  19. c=IN IP4 95.26.67.247
  20. t=1335435751 1335439351
  21. m=audio 10600 RTP/AVP 0 8 3 9 101
  22. a=rtpmap:0 PCMU/8000/1
  23. a=rtpmap:8 PCMA/8000/1
  24. a=rtpmap:3 GSM/8000/1
  25. a=rtpmap:9 G722/8000/1
  26. a=rtpmap:101 telephone-event/8000/1
  27. a=ptime:20
  28. m=video 10702 RTP/AVP 34 31
  29. a=rtpmap:34 H263/90000/1
  30. a=rtpmap:31 H261/90000/1
  31. <------------->
  32. --- (13 headers 15 lines) ---
  33. Sending to 95.26.67.247:5062 (NAT)
  34. Using INVITE request as basis request - 1621070911@192.168.0.25
  35. Found peer 'root' for 'root' from 95.26.67.247:5062
  36. == Using SIP RTP CoS mark 5
  37. Found RTP audio format 0
  38. Found RTP audio format 8
  39. Found RTP audio format 3
  40. Found RTP audio format 9
  41. Found RTP audio format 101
  42. Found audio description format PCMU for ID 0
  43. Found audio description format PCMA for ID 8
  44. Found audio description format GSM for ID 3
  45. Found audio description format G722 for ID 9
  46. Found audio description format telephone-event for ID 101
  47. Found RTP video format 34
  48. Found RTP video format 31
  49. Found video description format H263 for ID 34
  50. Found video description format H261 for ID 31
  51. Capabilities: us - (ulaw|alaw|g729), peer - audio=(gsm|ulaw|alaw|g722)/video=(h261|h263)/text=(nothing), combined - (ulaw|alaw)
  52. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  53. Peer audio RTP is at port 95.26.67.247:10600
  54. Looking for 79250287897 in sip (domain sip2.bad-times.wtf)
  55. list_route: hop: <sip:root@95.26.67.247:5062>
  56.  
  57. <--- Transmitting (NAT) to 95.26.67.247:5062 --->
  58. SIP/2.0 100 Trying
  59. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
  60. From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
  61. To: <sip:79250287897@sip2.bad-times.wtf>
  62. Call-ID: 1621070911@192.168.0.25
  63. CSeq: 20 INVITE
  64. Server: Asterisk PBX 10.3.1
  65. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  66. Supported: replaces, timer
  67. Contact: <sip:79250287897@22.22.22.22:5060>
  68. Content-Length: 0
  69.  
  70.  
  71. <------------>
  72. -- Executing [79250287897@sip:1] Dial("SIP/root-00000004", "SIP/skypeost/79250287897") in new stack
  73. == Using SIP RTP CoS mark 5
  74. -- Called SIP/skypeost/79250287897
  75. -- SIP/skypeost-00000005 is ringing
  76.  
  77. <--- Transmitting (NAT) to 95.26.67.247:5062 --->
  78. SIP/2.0 180 Ringing
  79. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
  80. From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
  81. To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
  82. Call-ID: 1621070911@192.168.0.25
  83. CSeq: 20 INVITE
  84. Server: Asterisk PBX 10.3.1
  85. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  86. Supported: replaces, timer
  87. Contact: <sip:79250287897@22.22.22.22:5060>
  88. Content-Length: 0
  89.  
  90.  
  91. <------------>
  92. -- SIP/skypeost-00000005 answered SIP/root-00000004
  93. Audio is at 18214
  94. Adding codec 100003 (ulaw) to SDP
  95. Adding codec 100004 (alaw) to SDP
  96. Adding non-codec 0x1 (telephone-event) to SDP
  97.  
  98. <--- Reliably Transmitting (NAT) to 95.26.67.247:5062 --->
  99. SIP/2.0 200 OK
  100. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
  101. From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
  102. To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
  103. Call-ID: 1621070911@192.168.0.25
  104. CSeq: 20 INVITE
  105. Server: Asterisk PBX 10.3.1
  106. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  107. Supported: replaces, timer
  108. Contact: <sip:79250287897@22.22.22.22:5060>
  109. Content-Type: application/sdp
  110. Content-Length: 284
  111.  
  112. v=0
  113. o=root 226525189 226525189 IN IP4 22.22.22.22
  114. s=Asterisk PBX 10.3.1
  115. c=IN IP4 22.22.22.22
  116. t=0 0
  117. m=audio 18214 RTP/AVP 0 8 101
  118. a=rtpmap:0 PCMU/8000
  119. a=rtpmap:8 PCMA/8000
  120. a=rtpmap:101 telephone-event/8000
  121. a=fmtp:101 0-16
  122. a=ptime:20
  123. a=sendrecv
  124. m=video 0 RTP/AVP 34 31
  125.  
  126. <------------>
  127. -- Locally bridging SIP/root-00000004 and SIP/skypeost-00000005
  128. Retransmitting #1 (NAT) to 95.26.67.247:5062:
  129. SIP/2.0 200 OK
  130. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
  131. From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
  132. To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
  133. Call-ID: 1621070911@192.168.0.25
  134. CSeq: 20 INVITE
  135. Server: Asterisk PBX 10.3.1
  136. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  137. Supported: replaces, timer
  138. Contact: <sip:79250287897@22.22.22.22:5060>
  139. Content-Type: application/sdp
  140. Content-Length: 284
  141.  
  142. v=0
  143. o=root 226525189 226525189 IN IP4 22.22.22.22
  144. s=Asterisk PBX 10.3.1
  145. c=IN IP4 22.22.22.22
  146. t=0 0
  147. m=audio 18214 RTP/AVP 0 8 101
  148. a=rtpmap:0 PCMU/8000
  149. a=rtpmap:8 PCMA/8000
  150. a=rtpmap:101 telephone-event/8000
  151. a=fmtp:101 0-16
  152. a=ptime:20
  153. a=sendrecv
  154. m=video 0 RTP/AVP 34 31
  155.  
  156. ---
  157. Retransmitting #2 (NAT) to 95.26.67.247:5062:
  158. SIP/2.0 200 OK
  159. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
  160. From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
  161. To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
  162. Call-ID: 1621070911@192.168.0.25
  163. CSeq: 20 INVITE
  164. Server: Asterisk PBX 10.3.1
  165. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  166. Supported: replaces, timer
  167. Contact: <sip:79250287897@22.22.22.22:5060>
  168. Content-Type: application/sdp
  169. Content-Length: 284
  170.  
  171. v=0
  172. o=root 226525189 226525189 IN IP4 22.22.22.22
  173. s=Asterisk PBX 10.3.1
  174. c=IN IP4 22.22.22.22
  175. t=0 0
  176. m=audio 18214 RTP/AVP 0 8 101
  177. a=rtpmap:0 PCMU/8000
  178. a=rtpmap:8 PCMA/8000
  179. a=rtpmap:101 telephone-event/8000
  180. a=fmtp:101 0-16
  181. a=ptime:20
  182. a=sendrecv
  183. m=video 0 RTP/AVP 34 31
  184.  
  185. ---
  186.  
  187. <--- SIP read from UDP:95.26.67.247:5062 --->
  188. OPTIONS sip:root@sip2.bad-times.wtf SIP/2.0
  189. Via: SIP/2.0/UDP 95.26.67.247:5062;rport;branch=z9hG4bK1534818002
  190. From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=1936133642
  191. To: <sip:root@sip2.bad-times.wtf>
  192. Call-ID: 393535928@192.168.0.25
  193. CSeq: 20 OPTIONS
  194. Max-Forwards: 70
  195. User-Agent: qutecom/rev-g-trunk
  196. Expires: 120
  197. Accept: application/sdp
  198. Content-Length: 0
  199.  
  200. <------------->
  201. --- (11 headers 0 lines) ---
  202. Looking for root in default (domain sip2.bad-times.wtf)
  203.  
  204. <--- Transmitting (NAT) to 95.26.67.247:5062 --->
  205. SIP/2.0 404 Not Found
  206. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK1534818002;received=95.26.67.247;rport=5062
  207. From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=1936133642
  208. To: <sip:root@sip2.bad-times.wtf>;tag=as4bbf5af1
  209. Call-ID: 393535928@192.168.0.25
  210. CSeq: 20 OPTIONS
  211. Server: Asterisk PBX 10.3.1
  212. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  213. Supported: replaces, timer
  214. Accept: application/sdp
  215. Content-Length: 0
  216.  
  217.  
  218. <------------>
  219. Scheduling destruction of SIP dialog '393535928@192.168.0.25' in 32000 ms (Method: OPTIONS)
  220. Retransmitting #3 (NAT) to 95.26.67.247:5062:
  221. SIP/2.0 200 OK
  222. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
  223. From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
  224. To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
  225. Call-ID: 1621070911@192.168.0.25
  226. CSeq: 20 INVITE
  227. Server: Asterisk PBX 10.3.1
  228. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  229. Supported: replaces, timer
  230. Contact: <sip:79250287897@22.22.22.22:5060>
  231. Content-Type: application/sdp
  232. Content-Length: 284
  233.  
  234. v=0
  235. o=root 226525189 226525189 IN IP4 22.22.22.22
  236. s=Asterisk PBX 10.3.1
  237. c=IN IP4 22.22.22.22
  238. t=0 0
  239. m=audio 18214 RTP/AVP 0 8 101
  240. a=rtpmap:0 PCMU/8000
  241. a=rtpmap:8 PCMA/8000
  242. a=rtpmap:101 telephone-event/8000
  243. a=fmtp:101 0-16
  244. a=ptime:20
  245. a=sendrecv
  246. m=video 0 RTP/AVP 34 31
  247.  
  248. ---
  249. Really destroying SIP dialog '1287725240@192.168.0.25' Method: OPTIONS
  250. Retransmitting #4 (NAT) to 95.26.67.247:5062:
  251. SIP/2.0 200 OK
  252. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
  253. From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
  254. To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
  255. Call-ID: 1621070911@192.168.0.25
  256. CSeq: 20 INVITE
  257. Server: Asterisk PBX 10.3.1
  258. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  259. Supported: replaces, timer
  260. Contact: <sip:79250287897@22.22.22.22:5060>
  261. Content-Type: application/sdp
  262. Content-Length: 284
  263.  
  264. v=0
  265. o=root 226525189 226525189 IN IP4 22.22.22.22
  266. s=Asterisk PBX 10.3.1
  267. c=IN IP4 22.22.22.22
  268. t=0 0
  269. m=audio 18214 RTP/AVP 0 8 101
  270. a=rtpmap:0 PCMU/8000
  271. a=rtpmap:8 PCMA/8000
  272. a=rtpmap:101 telephone-event/8000
  273. a=fmtp:101 0-16
  274. a=ptime:20
  275. a=sendrecv
  276. m=video 0 RTP/AVP 34 31
  277.  
  278. ---
  279. Retransmitting #5 (NAT) to 95.26.67.247:5062:
  280. SIP/2.0 200 OK
  281. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
  282. From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
  283. To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
  284. Call-ID: 1621070911@192.168.0.25
  285. CSeq: 20 INVITE
  286. Server: Asterisk PBX 10.3.1
  287. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  288. Supported: replaces, timer
  289. Contact: <sip:79250287897@22.22.22.22:5060>
  290. Content-Type: application/sdp
  291. Content-Length: 284
  292.  
  293. v=0
  294. o=root 226525189 226525189 IN IP4 22.22.22.22
  295. s=Asterisk PBX 10.3.1
  296. c=IN IP4 22.22.22.22
  297. t=0 0
  298. m=audio 18214 RTP/AVP 0 8 101
  299. a=rtpmap:0 PCMU/8000
  300. a=rtpmap:8 PCMA/8000
  301. a=rtpmap:101 telephone-event/8000
  302. a=fmtp:101 0-16
  303. a=ptime:20
  304. a=sendrecv
  305. m=video 0 RTP/AVP 34 31
  306.  
  307. ---
  308. Retransmitting #6 (NAT) to 95.26.67.247:5062:
  309. SIP/2.0 200 OK
  310. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK498352626;received=95.26.67.247;rport=5062
  311. From: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
  312. To: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
  313. Call-ID: 1621070911@192.168.0.25
  314. CSeq: 20 INVITE
  315. Server: Asterisk PBX 10.3.1
  316. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  317. Supported: replaces, timer
  318. Contact: <sip:79250287897@22.22.22.22:5060>
  319. Content-Type: application/sdp
  320. Content-Length: 284
  321.  
  322. v=0
  323. o=root 226525189 226525189 IN IP4 22.22.22.22
  324. s=Asterisk PBX 10.3.1
  325. c=IN IP4 22.22.22.22
  326. t=0 0
  327. m=audio 18214 RTP/AVP 0 8 101
  328. a=rtpmap:0 PCMU/8000
  329. a=rtpmap:8 PCMA/8000
  330. a=rtpmap:101 telephone-event/8000
  331. a=fmtp:101 0-16
  332. a=ptime:20
  333. a=sendrecv
  334. m=video 0 RTP/AVP 34 31
  335.  
  336. ---
  337. [Apr 26 14:22:33] WARNING[3194]: chan_sip.c:3663 retrans_pkt: Retransmission timeout reached on transmission 1621070911@192.168.0.25 for seqno 20 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  338. Packet timed out after 6400ms with no response
  339. [Apr 26 14:22:33] WARNING[3194]: chan_sip.c:3692 retrans_pkt: Hanging up call 1621070911@192.168.0.25 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  340. == Spawn extension (sip, 79250287897, 1) exited non-zero on 'SIP/root-00000004'
  341. Scheduling destruction of SIP dialog '1621070911@192.168.0.25' in 6400 ms (Method: INVITE)
  342. set_destination: Parsing <sip:root@95.26.67.247:5062> for address/port to send to
  343. set_destination: set destination to 95.26.67.247:5062
  344. Reliably Transmitting (NAT) to 95.26.67.247:5062:
  345. BYE sip:root@95.26.67.247:5062 SIP/2.0
  346. Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK71b6681e;rport
  347. Max-Forwards: 70
  348. From: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
  349. To: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
  350. Call-ID: 1621070911@192.168.0.25
  351. CSeq: 102 BYE
  352. User-Agent: Asterisk PBX 10.3.1
  353. X-Asterisk-HangupCause: Protocol error, unspecified
  354. X-Asterisk-HangupCauseCode: 111
  355. Content-Length: 0
  356.  
  357.  
  358. ---
  359.  
  360. <--- SIP read from UDP:95.26.67.247:5062 --->
  361. SIP/2.0 100 Trying
  362. Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK71b6681e;rport=5060
  363. From: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
  364. To: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
  365. Call-ID: 1621070911@192.168.0.25
  366. CSeq: 102 BYE
  367. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
  368. Content-Length: 0
  369.  
  370. <------------->
  371. --- (8 headers 0 lines) ---
  372.  
  373. <--- SIP read from UDP:95.26.67.247:5062 --->
  374. SIP/2.0 200 OK
  375. Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK71b6681e;rport=5060
  376. From: <sip:79250287897@sip2.bad-times.wtf>;tag=as7ad143ca
  377. To: root_sip2.bad-times.wtf <sip:root@sip2.bad-times.wtf>;tag=389538903
  378. Call-ID: 1621070911@192.168.0.25
  379. CSeq: 102 BYE
  380. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
  381. Content-Length: 0
  382.  
  383. <------------->
  384. --- (8 headers 0 lines) ---
  385. SIP Response message for INCOMING dialog BYE arrived
  386. Really destroying SIP dialog '1621070911@192.168.0.25' Method: INVITE
  387. ster*CLI>
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