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  1. <--- Transmitting (NAT) to 192.168.75.10:5060 --->
  2. SIP/2.0 100 Trying
  3. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-4aaa91cd;received=192.168.75.10;rport=5060
  4. From: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  5. To: <sip:16136008510@152.48.1.105>
  6. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  7. CSeq: 102 INVITE
  8. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  9. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  10. Supported: replaces, timer
  11. Contact: <sip:16136008510@152.48.1.105:5060>
  12. Content-Length: 0
  13.  
  14.  
  15. <------------>
  16. -- Executing [16136008510@father-out:1] Dial("SIP/100-0000000e", "SIP/16136008510@19199129730") in new stack
  17. == Using SIP RTP CoS mark 5
  18. -- Called SIP/16136008510@19199129730
  19. -- SIP/19199129730-0000000f is ringing
  20.  
  21. <--- Transmitting (NAT) to 192.168.75.10:5060 --->
  22. SIP/2.0 180 Ringing
  23. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-4aaa91cd;received=192.168.75.10;rport=5060
  24. From: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  25. To: <sip:16136008510@152.48.1.105>;tag=as40544425
  26. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  27. CSeq: 102 INVITE
  28. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  29. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  30. Supported: replaces, timer
  31. Contact: <sip:16136008510@152.48.1.105:5060>
  32. Content-Length: 0
  33.  
  34.  
  35. <------------>
  36. -- SIP/19199129730-0000000f is ringing
  37.  
  38. <--- SIP read from UDP:192.168.75.10:5060 --->
  39. NOTIFY sip:152.48.1.105 SIP/2.0
  40. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-da5e8099
  41. From: 100 <sip:100@152.48.1.105>;tag=e2d4e37a564416c8o0
  42. To: <sip:152.48.1.105>
  43. Call-ID: 41d06fc2-6ae0b720@192.168.75.10
  44. CSeq: 84 NOTIFY
  45. Max-Forwards: 70
  46. Event: keep-alive
  47. User-Agent: Linksys/PAP2T-5.1.6(LS)
  48. Content-Length: 0
  49.  
  50. <------------->
  51. --- (10 headers 0 lines) ---
  52.  
  53. <--- Transmitting (no NAT) to 192.168.75.10:5060 --->
  54. SIP/2.0 200 OK
  55. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-da5e8099;received=192.168.75.10
  56. From: 100 <sip:100@152.48.1.105>;tag=e2d4e37a564416c8o0
  57. To: <sip:152.48.1.105>;tag=as66262241
  58. Call-ID: 41d06fc2-6ae0b720@192.168.75.10
  59. CSeq: 84 NOTIFY
  60. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  61. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  62. Supported: replaces, timer
  63. Content-Length: 0
  64.  
  65.  
  66. <------------>
  67. Scheduling destruction of SIP dialog '41d06fc2-6ae0b720@192.168.75.10' in 32000 ms (Method: NOTIFY)
  68. -- SIP/19199129730-0000000f answered SIP/100-0000000e
  69. Audio is at 16770
  70. Adding codec 0x4 (ulaw) to SDP
  71. Adding codec 0x100 (g729) to SDP
  72. Adding non-codec 0x1 (telephone-event) to SDP
  73.  
  74. <--- Reliably Transmitting (NAT) to 192.168.75.10:5060 --->
  75. SIP/2.0 200 OK
  76. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-4aaa91cd;received=192.168.75.10;rport=5060
  77. From: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  78. To: <sip:16136008510@152.48.1.105>;tag=as40544425
  79. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  80. CSeq: 102 INVITE
  81. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  82. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  83. Supported: replaces, timer
  84. Contact: <sip:16136008510@152.48.1.105:5060>
  85. Content-Type: application/sdp
  86. Content-Length: 296
  87.  
  88. v=0
  89. o=root 756031082 756031082 IN IP4 152.48.1.105
  90. s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  91. c=IN IP4 152.48.1.105
  92. t=0 0
  93. m=audio 16770 RTP/AVP 0 18 101
  94. a=rtpmap:0 PCMU/8000
  95. a=rtpmap:18 G729/8000
  96. a=fmtp:18 annexb=no
  97. a=rtpmap:101 telephone-event/8000
  98. a=fmtp:101 0-16
  99. a=ptime:20
  100. a=sendrecv
  101.  
  102. <------------>
  103. -- Locally bridging SIP/100-0000000e and SIP/19199129730-0000000f
  104. Retransmitting #1 (NAT) to 192.168.75.10:5060:
  105. SIP/2.0 200 OK
  106. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-4aaa91cd;received=192.168.75.10;rport=5060
  107. From: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  108. To: <sip:16136008510@152.48.1.105>;tag=as40544425
  109. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  110. CSeq: 102 INVITE
  111. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  112. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  113. Supported: replaces, timer
  114. Contact: <sip:16136008510@152.48.1.105:5060>
  115. Content-Type: application/sdp
  116. Content-Length: 296
  117.  
  118. v=0
  119. o=root 756031082 756031082 IN IP4 152.48.1.105
  120. s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  121. c=IN IP4 152.48.1.105
  122. t=0 0
  123. m=audio 16770 RTP/AVP 0 18 101
  124. a=rtpmap:0 PCMU/8000
  125. a=rtpmap:18 G729/8000
  126. a=fmtp:18 annexb=no
  127. a=rtpmap:101 telephone-event/8000
  128. a=fmtp:101 0-16
  129. a=ptime:20
  130. a=sendrecv
  131.  
  132. ---
  133. Retransmitting #2 (NAT) to 192.168.75.10:5060:
  134. SIP/2.0 200 OK
  135. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-4aaa91cd;received=192.168.75.10;rport=5060
  136. From: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  137. To: <sip:16136008510@152.48.1.105>;tag=as40544425
  138. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  139. CSeq: 102 INVITE
  140. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  141. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  142. Supported: replaces, timer
  143. Contact: <sip:16136008510@152.48.1.105:5060>
  144. Content-Type: application/sdp
  145. Content-Length: 296
  146.  
  147. v=0
  148. o=root 756031082 756031082 IN IP4 152.48.1.105
  149. s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  150. c=IN IP4 152.48.1.105
  151. t=0 0
  152. m=audio 16770 RTP/AVP 0 18 101
  153. a=rtpmap:0 PCMU/8000
  154. a=rtpmap:18 G729/8000
  155. a=fmtp:18 annexb=no
  156. a=rtpmap:101 telephone-event/8000
  157. a=fmtp:101 0-16
  158. a=ptime:20
  159. a=sendrecv
  160.  
  161. ---
  162. [Oct 18 10:31:54] NOTICE[29728]: chan_sip.c:14440 check_auth: Correct auth, but based on stale nonce received from '<sip:200@sakhter.net>;tag=SP1394be37c3c7376ab'
  163. Retransmitting #3 (NAT) to 192.168.75.10:5060:
  164. SIP/2.0 200 OK
  165. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-4aaa91cd;received=192.168.75.10;rport=5060
  166. From: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  167. To: <sip:16136008510@152.48.1.105>;tag=as40544425
  168. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  169. CSeq: 102 INVITE
  170. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  171. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  172. Supported: replaces, timer
  173. Contact: <sip:16136008510@152.48.1.105:5060>
  174. Content-Type: application/sdp
  175. Content-Length: 296
  176.  
  177. v=0
  178. o=root 756031082 756031082 IN IP4 152.48.1.105
  179. s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  180. c=IN IP4 152.48.1.105
  181. t=0 0
  182. m=audio 16770 RTP/AVP 0 18 101
  183. a=rtpmap:0 PCMU/8000
  184. a=rtpmap:18 G729/8000
  185. a=fmtp:18 annexb=no
  186. a=rtpmap:101 telephone-event/8000
  187. a=fmtp:101 0-16
  188. a=ptime:20
  189. a=sendrecv
  190.  
  191. ---
  192. Retransmitting #4 (NAT) to 192.168.75.10:5060:
  193. SIP/2.0 200 OK
  194. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-4aaa91cd;received=192.168.75.10;rport=5060
  195. From: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  196. To: <sip:16136008510@152.48.1.105>;tag=as40544425
  197. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  198. CSeq: 102 INVITE
  199. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  200. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  201. Supported: replaces, timer
  202. Contact: <sip:16136008510@152.48.1.105:5060>
  203. Content-Type: application/sdp
  204. Content-Length: 296
  205.  
  206. v=0
  207. o=root 756031082 756031082 IN IP4 152.48.1.105
  208. s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  209. c=IN IP4 152.48.1.105
  210. t=0 0
  211. m=audio 16770 RTP/AVP 0 18 101
  212. a=rtpmap:0 PCMU/8000
  213. a=rtpmap:18 G729/8000
  214. a=fmtp:18 annexb=no
  215. a=rtpmap:101 telephone-event/8000
  216. a=fmtp:101 0-16
  217. a=ptime:20
  218. a=sendrecv
  219.  
  220. ---
  221. [Oct 18 10:31:59] NOTICE[29728]: chan_sip.c:14440 check_auth: Correct auth, but based on stale nonce received from '<sip:400@sakhter.net>;tag=SP24592031616cfdc6'
  222. [Oct 18 10:32:01] NOTICE[29728]: chan_sip.c:13058 sip_reregister: -- Re-registration for 17772708539@callcentric.com
  223. > doing dnsmgr_lookup for 'callcentric.com'
  224. > ast_get_srv: SRV lookup for '_sip._udp.callcentric.com' mapped to host alpha6.callcentric.com, port 5080
  225. [Oct 18 10:32:01] NOTICE[29728]: chan_sip.c:13058 sip_reregister: -- Re-registration for GV19199129730@gvgw2.simonics.com
  226. > doing dnsmgr_lookup for 'gvgw2.simonics.com'
  227. > doing dnsmgr_lookup for 'callcentric.com'
  228. > ast_get_srv: SRV lookup for '_sip._udp.callcentric.com' mapped to host alpha11.callcentric.com, port 5080
  229. > doing dnsmgr_lookup for 'gvgw2.simonics.com'
  230. [Oct 18 10:32:01] NOTICE[29728]: chan_sip.c:20714 handle_response_register: Outbound Registration: Expiry for callcentric.com is 120 sec (Scheduling reregistration in 105 s)
  231. [Oct 18 10:32:01] NOTICE[29728]: chan_sip.c:20714 handle_response_register: Outbound Registration: Expiry for gvgw2.simonics.com is 120 sec (Scheduling reregistration in 105 s)
  232. Retransmitting #5 (NAT) to 192.168.75.10:5060:
  233. SIP/2.0 200 OK
  234. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-4aaa91cd;received=192.168.75.10;rport=5060
  235. From: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  236. To: <sip:16136008510@152.48.1.105>;tag=as40544425
  237. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  238. CSeq: 102 INVITE
  239. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  240. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  241. Supported: replaces, timer
  242. Contact: <sip:16136008510@152.48.1.105:5060>
  243. Content-Type: application/sdp
  244. Content-Length: 296
  245.  
  246. v=0
  247. o=root 756031082 756031082 IN IP4 152.48.1.105
  248. s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  249. c=IN IP4 152.48.1.105
  250. t=0 0
  251. m=audio 16770 RTP/AVP 0 18 101
  252. a=rtpmap:0 PCMU/8000
  253. a=rtpmap:18 G729/8000
  254. a=fmtp:18 annexb=no
  255. a=rtpmap:101 telephone-event/8000
  256. a=fmtp:101 0-16
  257. a=ptime:20
  258. a=sendrecv
  259.  
  260. ---
  261. Reliably Transmitting (NAT) to 192.168.75.10:5060:
  262. OPTIONS sip:100@192.168.75.10:5060 SIP/2.0
  263. Via: SIP/2.0/UDP 152.48.1.105:5060;branch=z9hG4bK0c1b7c46;rport
  264. Max-Forwards: 70
  265. From: "asterisk" <sip:asterisk@152.48.1.105>;tag=as021229c5
  266. To: <sip:100@192.168.75.10:5060>
  267. Contact: <sip:asterisk@152.48.1.105:5060>
  268. Call-ID: 55cf4b1b6f0e2030315aca6047009128@152.48.1.105:5060
  269. CSeq: 102 OPTIONS
  270. User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  271. Date: Thu, 18 Oct 2012 14:32:01 GMT
  272. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  273. Supported: replaces, timer
  274. Content-Length: 0
  275.  
  276.  
  277. ---
  278.  
  279. <--- SIP read from UDP:192.168.75.10:5060 --->
  280. SIP/2.0 200 OK
  281. To: <sip:100@192.168.75.10:5060>;tag=e1527ee24a0f280i0
  282. From: "asterisk" <sip:asterisk@152.48.1.105>;tag=as021229c5
  283. Call-ID: 55cf4b1b6f0e2030315aca6047009128@152.48.1.105:5060
  284. CSeq: 102 OPTIONS
  285. Via: SIP/2.0/UDP 152.48.1.105:5060;branch=z9hG4bK0c1b7c46
  286. Server: Linksys/PAP2T-5.1.6(LS)
  287. Content-Length: 0
  288. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  289. Supported: x-sipura, replaces
  290.  
  291. <------------->
  292. --- (10 headers 0 lines) ---
  293. Really destroying SIP dialog '55cf4b1b6f0e2030315aca6047009128@152.48.1.105:5060' Method: OPTIONS
  294.  
  295. <--- SIP read from UDP:192.168.75.10:5060 --->
  296. NOTIFY sip:152.48.1.105 SIP/2.0
  297. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-add2ca42
  298. From: 100 <sip:100@152.48.1.105>;tag=e2d4e37a564416c8o0
  299. To: <sip:152.48.1.105>
  300. Call-ID: 41d06fc2-6ae0b720@192.168.75.10
  301. CSeq: 85 NOTIFY
  302. Max-Forwards: 70
  303. Event: keep-alive
  304. User-Agent: Linksys/PAP2T-5.1.6(LS)
  305. Content-Length: 0
  306.  
  307. <------------->
  308. --- (10 headers 0 lines) ---
  309.  
  310. <--- Transmitting (no NAT) to 192.168.75.10:5060 --->
  311. SIP/2.0 200 OK
  312. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-add2ca42;received=192.168.75.10
  313. From: 100 <sip:100@152.48.1.105>;tag=e2d4e37a564416c8o0
  314. To: <sip:152.48.1.105>;tag=as66262241
  315. Call-ID: 41d06fc2-6ae0b720@192.168.75.10
  316. CSeq: 85 NOTIFY
  317. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  318. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  319. Supported: replaces, timer
  320. Content-Length: 0
  321.  
  322.  
  323. <------------>
  324. Scheduling destruction of SIP dialog '41d06fc2-6ae0b720@192.168.75.10' in 32000 ms (Method: NOTIFY)
  325. Retransmitting #6 (NAT) to 192.168.75.10:5060:
  326. SIP/2.0 200 OK
  327. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-4aaa91cd;received=192.168.75.10;rport=5060
  328. From: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  329. To: <sip:16136008510@152.48.1.105>;tag=as40544425
  330. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  331. CSeq: 102 INVITE
  332. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  333. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  334. Supported: replaces, timer
  335. Contact: <sip:16136008510@152.48.1.105:5060>
  336. Content-Type: application/sdp
  337. Content-Length: 296
  338.  
  339. v=0
  340. o=root 756031082 756031082 IN IP4 152.48.1.105
  341. s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  342. c=IN IP4 152.48.1.105
  343. t=0 0
  344. m=audio 16770 RTP/AVP 0 18 101
  345. a=rtpmap:0 PCMU/8000
  346. a=rtpmap:18 G729/8000
  347. a=fmtp:18 annexb=no
  348. a=rtpmap:101 telephone-event/8000
  349. a=fmtp:101 0-16
  350. a=ptime:20
  351. a=sendrecv
  352.  
  353. ---
  354. Retransmitting #7 (NAT) to 192.168.75.10:5060:
  355. SIP/2.0 200 OK
  356. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-4aaa91cd;received=192.168.75.10;rport=5060
  357. From: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  358. To: <sip:16136008510@152.48.1.105>;tag=as40544425
  359. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  360. CSeq: 102 INVITE
  361. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  362. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  363. Supported: replaces, timer
  364. Contact: <sip:16136008510@152.48.1.105:5060>
  365. Content-Type: application/sdp
  366. Content-Length: 296
  367.  
  368. v=0
  369. o=root 756031082 756031082 IN IP4 152.48.1.105
  370. s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  371. c=IN IP4 152.48.1.105
  372. t=0 0
  373. m=audio 16770 RTP/AVP 0 18 101
  374. a=rtpmap:0 PCMU/8000
  375. a=rtpmap:18 G729/8000
  376. a=fmtp:18 annexb=no
  377. a=rtpmap:101 telephone-event/8000
  378. a=fmtp:101 0-16
  379. a=ptime:20
  380. a=sendrecv
  381.  
  382. ---
  383. sakhter-nlr*CLI>
  384. sakhter-nlr*CLI>
  385. sakhter-nlr*CLI>
  386. sakhter-nlr*CLI>
  387. sakhter-nlr*CLI>
  388. sakhter-nlr*CLI>
  389. sakhter-nlr*CLI>
  390. sakhter-nlr*CLI>
  391. sakhter-nlr*CLI>
  392. sakhter-nlr*CLI>
  393. sakhter-nlr*CLI>
  394. sakhter-nlr*CLI>
  395. sakhter-nlr*CLI>
  396. sakhter-nlr*CLI>
  397. sakhter-nlr*CLI>
  398. Retransmitting #8 (NAT) to 192.168.75.10:5060:
  399. SIP/2.0 200 OK
  400. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-4aaa91cd;received=192.168.75.10;rport=5060
  401. From: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  402. To: <sip:16136008510@152.48.1.105>;tag=as40544425
  403. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  404. CSeq: 102 INVITE
  405. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  406. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  407. Supported: replaces, timer
  408. Contact: <sip:16136008510@152.48.1.105:5060>
  409. Content-Type: application/sdp
  410. Content-Length: 296
  411.  
  412. v=0
  413. o=root 756031082 756031082 IN IP4 152.48.1.105
  414. s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  415. c=IN IP4 152.48.1.105
  416. t=0 0
  417. m=audio 16770 RTP/AVP 0 18 101
  418. a=rtpmap:0 PCMU/8000
  419. a=rtpmap:18 G729/8000
  420. a=fmtp:18 annexb=no
  421. a=rtpmap:101 telephone-event/8000
  422. a=fmtp:101 0-16
  423. a=ptime:20
  424. a=sendrecv
  425.  
  426. ---
  427. [Oct 18 10:32:13] WARNING[29728]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission 5f9a738b-3c22d7ed@192.168.75.10 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  428. Packet timed out after 21248ms with no response
  429. [Oct 18 10:32:13] WARNING[29728]: chan_sip.c:3670 retrans_pkt: Hanging up call 5f9a738b-3c22d7ed@192.168.75.10 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  430. -- Executing [h@father-out:1] Hangup("SIP/100-0000000e", "") in new stack
  431. == Spawn extension (father-out, h, 1) exited non-zero on 'SIP/100-0000000e'
  432. == Spawn extension (father-out, 16136008510, 1) exited non-zero on 'SIP/100-0000000e'
  433. Scheduling destruction of SIP dialog '5f9a738b-3c22d7ed@192.168.75.10' in 21248 ms (Method: INVITE)
  434. set_destination: Parsing <sip:100@192.168.75.10:5060> for address/port to send to
  435. set_destination: set destination to 192.168.75.10:5060
  436. Reliably Transmitting (NAT) to 192.168.75.10:5060:
  437. BYE sip:100@192.168.75.10:5060 SIP/2.0
  438. Via: SIP/2.0/UDP 152.48.1.105:5060;branch=z9hG4bK6ea0e58d;rport
  439. Max-Forwards: 70
  440. From: <sip:16136008510@152.48.1.105>;tag=as40544425
  441. To: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  442. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  443. CSeq: 102 BYE
  444. User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  445. Proxy-Authorization: Digest username="100", realm="asterisk", algorithm=MD5, uri="sip:152.48.1.105", nonce="", response="686f77e412a7b2e96785f900d9f34a14"
  446. X-Asterisk-HangupCause: Protocol error, unspecified
  447. X-Asterisk-HangupCauseCode: 111
  448. Content-Length: 0
  449.  
  450.  
  451. ---
  452. Retransmitting #1 (NAT) to 192.168.75.10:5060:
  453. BYE sip:100@192.168.75.10:5060 SIP/2.0
  454. Via: SIP/2.0/UDP 152.48.1.105:5060;branch=z9hG4bK6ea0e58d;rport
  455. Max-Forwards: 70
  456. From: <sip:16136008510@152.48.1.105>;tag=as40544425
  457. To: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  458. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  459. CSeq: 102 BYE
  460. User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  461. Proxy-Authorization: Digest username="100", realm="asterisk", algorithm=MD5, uri="sip:152.48.1.105", nonce="", response="686f77e412a7b2e96785f900d9f34a14"
  462. X-Asterisk-HangupCause: Protocol error, unspecified
  463. X-Asterisk-HangupCauseCode: 111
  464. Content-Length: 0
  465.  
  466.  
  467. ---
  468.  
  469. <--- SIP read from UDP:192.168.75.10:5060 --->
  470. SIP/2.0 200 OK
  471. To: 100 <sip:100@152.48.1.105>;tag=3845eef3b32b53a5o0
  472. From: <sip:16136008510@152.48.1.105>;tag=as40544425
  473. Call-ID: 5f9a738b-3c22d7ed@192.168.75.10
  474. CSeq: 102 BYE
  475. Via: SIP/2.0/UDP 152.48.1.105:5060;branch=z9hG4bK6ea0e58d
  476. Server: Linksys/PAP2T-5.1.6(LS)
  477. Content-Length: 0
  478.  
  479. <------------->
  480. --- (8 headers 0 lines) ---
  481. SIP Response message for INCOMING dialog BYE arrived
  482. Really destroying SIP dialog '5f9a738b-3c22d7ed@192.168.75.10' Method: INVITE
  483.  
  484. <--- SIP read from UDP:192.168.75.10:5060 --->
  485. NOTIFY sip:152.48.1.105 SIP/2.0
  486. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-4800269e
  487. From: 100 <sip:100@152.48.1.105>;tag=e2d4e37a564416c8o0
  488. To: <sip:152.48.1.105>
  489. Call-ID: 41d06fc2-6ae0b720@192.168.75.10
  490. CSeq: 86 NOTIFY
  491. Max-Forwards: 70
  492. Event: keep-alive
  493. User-Agent: Linksys/PAP2T-5.1.6(LS)
  494. Content-Length: 0
  495.  
  496. <------------->
  497. --- (10 headers 0 lines) ---
  498.  
  499. <--- Transmitting (no NAT) to 192.168.75.10:5060 --->
  500. SIP/2.0 200 OK
  501. Via: SIP/2.0/UDP 192.168.75.10:5060;branch=z9hG4bK-4800269e;received=192.168.75.10
  502. From: 100 <sip:100@152.48.1.105>;tag=e2d4e37a564416c8o0
  503. To: <sip:152.48.1.105>;tag=as66262241
  504. Call-ID: 41d06fc2-6ae0b720@192.168.75.10
  505. CSeq: 86 NOTIFY
  506. Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  507. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  508. Supported: replaces, timer
  509. Content-Length: 0
  510.  
  511.  
  512. <------------>
  513. Scheduling destruction of SIP dialog '41d06fc2-6ae0b720@192.168.75.10' in 32000 ms (Method: NOTIFY)
  514. [Oct 18 10:32:24] NOTICE[29728]: chan_sip.c:14440 check_auth: Correct auth, but based on stale nonce received from '<sip:200@sakhter.net>;tag=SP1394be37c3c7376ab'
  515. sakhter-nlr*CLI> exit
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