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  1. <--- SIP read from UDP:93.80.9.48:5062 --->
  2. INVITE sip:79250287897@sip.mydomain.lol SIP/2.0
  3. Via: SIP/2.0/UDP 93.80.9.48:5062;rport;branch=z9hG4bK2069899694
  4. From: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=1282093941
  5. To: <sip:79250287897@sip.mydomain.lol>
  6. Call-ID: 1236664015@192.168.0.25
  7. CSeq: 20 INVITE
  8. Contact: <sip:root@93.80.9.48:5062>
  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: 366
  15.  
  16. v=0
  17. o=userX 20000001 20000001 IN IP4 93.80.9.48
  18. s=A call
  19. c=IN IP4 93.80.9.48
  20. t=1335397677 1335401277
  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 93.80.9.48:5062 (NAT)
  34. Using INVITE request as basis request - 1236664015@192.168.0.25
  35. Found peer 'root' for 'root' from 93.80.9.48:5062
  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 93.80.9.48:10600
  54. Looking for 79250287897 in sip (domain sip.mydomain.lol)
  55. list_route: hop: <sip:root@93.80.9.48:5062>
  56.  
  57. <--- Transmitting (NAT) to 93.80.9.48:5062 --->
  58. SIP/2.0 100 Trying
  59. Via: SIP/2.0/UDP 93.80.9.48:5062;branch=z9hG4bK2069899694;received=93.80.9.48;rport=5062
  60. From: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=1282093941
  61. To: <sip:79250287897@sip.mydomain.lol>
  62. Call-ID: 1236664015@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@11.11.11.11:5060>
  68. Content-Length: 0
  69.  
  70.  
  71. <------------>
  72. -- Executing [79250287897@sip:1] Dial("SIP/root-0000000a", "SIP/skypeost/79250287897") in new stack
  73. == Using SIP RTP CoS mark 5
  74. -- Called SIP/skypeost/79250287897
  75.  
  76. <--- SIP read from UDP:93.80.9.48:5062 --->
  77. OPTIONS sip:root@sip.mydomain.lol SIP/2.0
  78. Via: SIP/2.0/UDP 93.80.9.48:5062;rport;branch=z9hG4bK406771120
  79. From: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=234295660
  80. To: <sip:root@sip.mydomain.lol>
  81. Call-ID: 402667159@192.168.0.25
  82. CSeq: 20 OPTIONS
  83. Max-Forwards: 70
  84. User-Agent: qutecom/rev-g-trunk
  85. Expires: 120
  86. Accept: application/sdp
  87. Content-Length: 0
  88.  
  89. <------------->
  90. --- (11 headers 0 lines) ---
  91. Looking for root in default (domain sip.mydomain.lol)
  92.  
  93. <--- Transmitting (NAT) to 93.80.9.48:5062 --->
  94. SIP/2.0 404 Not Found
  95. Via: SIP/2.0/UDP 93.80.9.48:5062;branch=z9hG4bK406771120;received=93.80.9.48;rport=5062
  96. From: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=234295660
  97. To: <sip:root@sip.mydomain.lol>;tag=as2205d542
  98. Call-ID: 402667159@192.168.0.25
  99. CSeq: 20 OPTIONS
  100. Server: Asterisk PBX 10.3.1
  101. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  102. Supported: replaces, timer
  103. Accept: application/sdp
  104. Content-Length: 0
  105.  
  106.  
  107. <------------>
  108. Scheduling destruction of SIP dialog '402667159@192.168.0.25' in 32000 ms (Method: OPTIONS)
  109. Really destroying SIP dialog '1653682233@192.168.0.25' Method: OPTIONS
  110. -- SIP/skypeost-0000000b is ringing
  111.  
  112. <--- Transmitting (NAT) to 93.80.9.48:5062 --->
  113. SIP/2.0 180 Ringing
  114. Via: SIP/2.0/UDP 93.80.9.48:5062;branch=z9hG4bK2069899694;received=93.80.9.48;rport=5062
  115. From: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=1282093941
  116. To: <sip:79250287897@sip.mydomain.lol>;tag=as11e31197
  117. Call-ID: 1236664015@192.168.0.25
  118. CSeq: 20 INVITE
  119. Server: Asterisk PBX 10.3.1
  120. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  121. Supported: replaces, timer
  122. Contact: <sip:79250287897@11.11.11.11:5060>
  123. Content-Length: 0
  124.  
  125.  
  126. <------------>
  127. -- SIP/skypeost-0000000b answered SIP/root-0000000a
  128. Audio is at 18254
  129. Adding codec 100003 (ulaw) to SDP
  130. Adding codec 100004 (alaw) to SDP
  131. Adding non-codec 0x1 (telephone-event) to SDP
  132.  
  133. <--- Reliably Transmitting (NAT) to 93.80.9.48:5062 --->
  134. SIP/2.0 200 OK
  135. Via: SIP/2.0/UDP 93.80.9.48:5062;branch=z9hG4bK2069899694;received=93.80.9.48;rport=5062
  136. From: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=1282093941
  137. To: <sip:79250287897@sip.mydomain.lol>;tag=as11e31197
  138. Call-ID: 1236664015@192.168.0.25
  139. CSeq: 20 INVITE
  140. Server: Asterisk PBX 10.3.1
  141. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  142. Supported: replaces, timer
  143. Contact: <sip:79250287897@11.11.11.11:5060>
  144. Content-Type: application/sdp
  145. Content-Length: 286
  146.  
  147. v=0
  148. o=root 1871591395 1871591395 IN IP4 11.11.11.11
  149. s=Asterisk PBX 10.3.1
  150. c=IN IP4 11.11.11.11
  151. t=0 0
  152. m=audio 18254 RTP/AVP 0 8 101
  153. a=rtpmap:0 PCMU/8000
  154. a=rtpmap:8 PCMA/8000
  155. a=rtpmap:101 telephone-event/8000
  156. a=fmtp:101 0-16
  157. a=ptime:20
  158. a=sendrecv
  159. m=video 0 RTP/AVP 34 31
  160.  
  161. <------------>
  162. -- Locally bridging SIP/root-0000000a and SIP/skypeost-0000000b
  163. Retransmitting #1 (NAT) to 93.80.9.48:5062:
  164. SIP/2.0 200 OK
  165. Via: SIP/2.0/UDP 93.80.9.48:5062;branch=z9hG4bK2069899694;received=93.80.9.48;rport=5062
  166. From: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=1282093941
  167. To: <sip:79250287897@sip.mydomain.lol>;tag=as11e31197
  168. Call-ID: 1236664015@192.168.0.25
  169. CSeq: 20 INVITE
  170. Server: Asterisk PBX 10.3.1
  171. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  172. Supported: replaces, timer
  173. Contact: <sip:79250287897@11.11.11.11:5060>
  174. ontent-Type: application/sdp
  175. Content-Length: 286
  176.  
  177. v=0
  178. o=root 1871591395 1871591395 IN IP4 11.11.11.11
  179. s=Asterisk PBX 10.3.1
  180. c=IN IP4 11.11.11.11
  181. t=0 0
  182. m=audio 18254 RTP/AVP 0 8 101
  183. a=rtpmap:0 PCMU/8000
  184. a=rtpmap:8 PCMA/8000
  185. a=rtpmap:101 telephone-event/8000
  186. a=fmtp:101 0-16
  187. a=ptime:20
  188. a=sendrecv
  189. m=video 0 RTP/AVP 34 31
  190.  
  191. ---
  192. Retransmitting #2 (NAT) to 93.80.9.48:5062:
  193. SIP/2.0 200 OK
  194. Via: SIP/2.0/UDP 93.80.9.48:5062;branch=z9hG4bK2069899694;received=93.80.9.48;rport=5062
  195. From: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=1282093941
  196. To: <sip:79250287897@sip.mydomain.lol>;tag=as11e31197
  197. Call-ID: 1236664015@192.168.0.25
  198. CSeq: 20 INVITE
  199. Server: Asterisk PBX 10.3.1
  200. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  201. Supported: replaces, timer
  202. Contact: <sip:79250287897@11.11.11.11:5060>
  203. ontent-Type: application/sdp
  204. Content-Length: 286
  205.  
  206. v=0
  207. o=root 1871591395 1871591395 IN IP4 11.11.11.11
  208. s=Asterisk PBX 10.3.1
  209. c=IN IP4 11.11.11.11
  210. t=0 0
  211. m=audio 18254 RTP/AVP 0 8 101
  212. a=rtpmap:0 PCMU/8000
  213. a=rtpmap:8 PCMA/8000
  214. a=rtpmap:101 telephone-event/8000
  215. a=fmtp:101 0-16
  216. a=ptime:20
  217. a=sendrecv
  218. m=video 0 RTP/AVP 34 31
  219.  
  220. ---
  221. Retransmitting #3 (NAT) to 93.80.9.48:5062:
  222. SIP/2.0 200 OK
  223. Via: SIP/2.0/UDP 93.80.9.48:5062;branch=z9hG4bK2069899694;received=93.80.9.48;rport=5062
  224. From: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=1282093941
  225. To: <sip:79250287897@sip.mydomain.lol>;tag=as11e31197
  226. Call-ID: 1236664015@192.168.0.25
  227. CSeq: 20 INVITE
  228. Server: Asterisk PBX 10.3.1
  229. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  230. Supported: replaces, timer
  231. Contact: <sip:79250287897@11.11.11.11:5060>
  232. ontent-Type: application/sdp
  233. Content-Length: 286
  234.  
  235. v=0
  236. o=root 1871591395 1871591395 IN IP4 11.11.11.11
  237. s=Asterisk PBX 10.3.1
  238. c=IN IP4 11.11.11.11
  239. t=0 0
  240. m=audio 18254 RTP/AVP 0 8 101
  241. a=rtpmap:0 PCMU/8000
  242. a=rtpmap:8 PCMA/8000
  243. a=rtpmap:101 telephone-event/8000
  244. a=fmtp:101 0-16
  245. a=ptime:20
  246. a=sendrecv
  247. m=video 0 RTP/AVP 34 31
  248.  
  249. ---
  250. Retransmitting #4 (NAT) to 93.80.9.48:5062:
  251. SIP/2.0 200 OK
  252. Via: SIP/2.0/UDP 93.80.9.48:5062;branch=z9hG4bK2069899694;received=93.80.9.48;rport=5062
  253. From: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=1282093941
  254. To: <sip:79250287897@sip.mydomain.lol>;tag=as11e31197
  255. Call-ID: 1236664015@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@11.11.11.11:5060>
  261. ontent-Type: application/sdp
  262. Content-Length: 286
  263.  
  264. v=0
  265. o=root 1871591395 1871591395 IN IP4 11.11.11.11
  266. s=Asterisk PBX 10.3.1
  267. c=IN IP4 11.11.11.11
  268. t=0 0
  269. m=audio 18254 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 93.80.9.48:5062:
  280. SIP/2.0 200 OK
  281. Via: SIP/2.0/UDP 93.80.9.48:5062;branch=z9hG4bK2069899694;received=93.80.9.48;rport=5062
  282. From: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=1282093941
  283. To: <sip:79250287897@sip.mydomain.lol>;tag=as11e31197
  284. Call-ID: 1236664015@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@11.11.11.11:5060>
  290. ontent-Type: application/sdp
  291. Content-Length: 286
  292.  
  293. v=0
  294. o=root 1871591395 1871591395 IN IP4 11.11.11.11
  295. s=Asterisk PBX 10.3.1
  296. c=IN IP4 11.11.11.11
  297. t=0 0
  298. m=audio 18254 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 93.80.9.48:5062:
  309. SIP/2.0 200 OK
  310. Via: SIP/2.0/UDP 93.80.9.48:5062;branch=z9hG4bK2069899694;received=93.80.9.48;rport=5062
  311. From: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=1282093941
  312. To: <sip:79250287897@sip.mydomain.lol>;tag=as11e31197
  313. Call-ID: 1236664015@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@11.11.11.11:5060>
  319. ontent-Type: application/sdp
  320. Content-Length: 286
  321.  
  322. v=0
  323. o=root 1871591395 1871591395 IN IP4 11.11.11.11
  324. s=Asterisk PBX 10.3.1
  325. c=IN IP4 11.11.11.11
  326. t=0 0
  327. m=audio 18254 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 03:47:58] WARNING[2242]: chan_sip.c:3663 retrans_pkt: Retransmission timeout reached on transmission 1236664015@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 03:47:58] WARNING[2242]: chan_sip.c:3692 retrans_pkt: Hanging up call 1236664015@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-0000000a'
  341. Scheduling destruction of SIP dialog '1236664015@192.168.0.25' in 6400 ms (Method: INVITE)
  342. set_destination: Parsing <sip:root@93.80.9.48:5062> for address/port to send to
  343. set_destination: set destination to 93.80.9.48:5062
  344. Reliably Transmitting (NAT) to 93.80.9.48:5062:
  345. BYE sip:root@93.80.9.48:5062 SIP/2.0
  346. Via: SIP/2.0/UDP 11.11.11.11:5060;branch=z9hG4bK1729357b;rport
  347. Max-Forwards: 70
  348. From: <sip:79250287897@sip.mydomain.lol>;tag=as11e31197
  349. To: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=1282093941
  350. Call-ID: 1236664015@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:93.80.9.48:5062 --->
  361. SIP/2.0 100 Trying
  362. Via: SIP/2.0/UDP 11.11.11.11:5060;branch=z9hG4bK1729357b;rport=5060
  363. From: <sip:79250287897@sip.mydomain.lol>;tag=as11e31197
  364. To: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=1282093941
  365. Call-ID: 1236664015@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:93.80.9.48:5062 --->
  374. SIP/2.0 200 OK
  375. Via: SIP/2.0/UDP 11.11.11.11:5060;branch=z9hG4bK1729357b;rport=5060
  376. From: <sip:79250287897@sip.mydomain.lol>;tag=as11e31197
  377. To: root_sip.mydomain.lol <sip:root@sip.mydomain.lol>;tag=1282093941
  378. Call-ID: 1236664015@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 '1236664015@192.168.0.25' Method: INVITE
  387. ster*CLI>
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