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  1. <-- SIP read from 208.1.87.235:5060:
  2. INVITE sip:s@208.100.1.33 SIP/2.0
  3. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK466c650c;rport
  4. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as2372c817
  5. To: <sip:s@208.100.1.33>
  6. Contact: <sip:8159093011@208.1.87.235>
  7. Call-ID: 73de34ce2c2778863080eb474c7efe86@208.1.87.235
  8. CSeq: 102 INVITE
  9. User-Agent: Asterisk PBX
  10. Max-Forwards: 70
  11. Date: Thu, 15 Jan 2009 18:32:00 GMT
  12. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  13. Supported: replaces
  14. Content-Type: application/sdp
  15. Content-Length: 262
  16.  
  17. v=0
  18. o=root 5712 5712 IN IP4 208.1.87.235
  19. s=session
  20. c=IN IP4 208.1.87.235
  21. t=0 0
  22. m=audio 18082 RTP/AVP 0 8 101
  23. a=rtpmap:0 PCMU/8000
  24. a=rtpmap:8 PCMA/8000
  25. a=rtpmap:101 telephone-event/8000
  26. a=fmtp:101 0-16
  27. a=silenceSupp:off - - - -
  28. a=ptime:20
  29. a=sendrecv
  30.  
  31. --- (14 headers 13 lines)---
  32. Using INVITE request as basis request - 73de34ce2c2778863080eb474c7efe86@208.1.87.235
  33. Sending to 208.1.87.235 : 5060 (NAT)
  34. Found peer '8159911010'
  35. Found RTP audio format 0
  36. Found RTP audio format 8
  37. Found RTP audio format 101
  38. Peer audio RTP is at port 208.1.87.235:18082
  39. Found description format PCMU
  40. Found description format PCMA
  41. Found description format telephone-event
  42. Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
  43. Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  44. Looking for s in DID-incoming (domain 208.100.1.33)
  45. Reliably Transmitting (no NAT) to 208.1.87.235:5060:
  46. SIP/2.0 404 Not Found
  47. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK466c650c;rport;received=208.1.87.235
  48. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as2372c817
  49. To: <sip:s@208.100.1.33>;tag=as0a443b7e
  50. Call-ID: 73de34ce2c2778863080eb474c7efe86@208.1.87.235
  51. CSeq: 102 INVITE
  52. User-Agent: Asterisk PBX
  53. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  54. Contact: <sip:s@208.100.1.33>
  55. Content-Length: 0
  56.  
  57.  
  58. ---
  59. ds00209*CLI>
  60. <-- SIP read from 208.1.87.235:5060:
  61. ACK sip:s@208.100.1.33 SIP/2.0
  62. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK466c650c;rport
  63. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as2372c817
  64. To: <sip:s@208.100.1.33>;tag=as0a443b7e
  65. Contact: <sip:8159093011@208.1.87.235>
  66. Call-ID: 73de34ce2c2778863080eb474c7efe86@208.1.87.235
  67. CSeq: 102 ACK
  68. User-Agent: Asterisk PBX
  69. Max-Forwards: 70
  70. Content-Length: 0
  71.  
  72.  
  73. --- (10 headers 0 lines)---
  74. Destroying call '73de34ce2c2778863080eb474c7efe86@208.1.87.235'
  75. ds00209*CLI>
  76. <-- SIP read from 208.1.87.235:5060:
  77. OPTIONS sip:s@208.100.1.33 SIP/2.0
  78. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK50bc24dd;rport
  79. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as614d943d
  80. To: <sip:s@208.100.1.33>
  81. Contact: <sip:Unknown@208.1.87.235>
  82. Call-ID: 015d11846944110558ccdb9d1ca2d3c7@208.1.87.235
  83. CSeq: 102 OPTIONS
  84. User-Agent: Asterisk PBX
  85. Max-Forwards: 70
  86. Date: Thu, 15 Jan 2009 18:32:02 GMT
  87. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  88. Supported: replaces
  89. Content-Length: 0
  90.  
  91.  
  92. --- (13 headers 0 lines)---
  93. Looking for s in DID-incoming (domain 208.100.1.33)
  94. Transmitting (no NAT) to 208.1.87.235:5060:
  95. SIP/2.0 404 Not Found
  96. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK50bc24dd;rport;received=208.1.87.235
  97. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as614d943d
  98. To: <sip:s@208.100.1.33>;tag=as13ca41b4
  99. Call-ID: 015d11846944110558ccdb9d1ca2d3c7@208.1.87.235
  100. CSeq: 102 OPTIONS
  101. User-Agent: Asterisk PBX
  102. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  103. Contact: <sip:208.100.1.33>
  104. Accept: application/sdp
  105. Content-Length: 0
  106.  
  107.  
  108. ---
  109. Destroying call '015d11846944110558ccdb9d1ca2d3c7@208.1.87.235'
  110. ds00209*CLI>
  111. <-- SIP read from 208.1.87.235:5060:
  112. OPTIONS sip:s@208.100.1.33 SIP/2.0
  113. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK0dbcc6e7;rport
  114. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as413ccc69
  115. To: <sip:s@208.100.1.33>
  116. Contact: <sip:Unknown@208.1.87.235>
  117. Call-ID: 4c441dfe3ab9a36607c3d983725fd045@208.1.87.235
  118. CSeq: 102 OPTIONS
  119. User-Agent: Asterisk PBX
  120. Max-Forwards: 70
  121. Date: Thu, 15 Jan 2009 18:32:02 GMT
  122. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  123. Supported: replaces
  124. Content-Length: 0
  125.  
  126.  
  127. --- (13 headers 0 lines)---
  128. Looking for s in DID-incoming (domain 208.100.1.33)
  129. Transmitting (no NAT) to 208.1.87.235:5060:
  130. SIP/2.0 404 Not Found
  131. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK0dbcc6e7;rport;received=208.1.87.235
  132. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as413ccc69
  133. To: <sip:s@208.100.1.33>;tag=as7dbcd0a4
  134. Call-ID: 4c441dfe3ab9a36607c3d983725fd045@208.1.87.235
  135. CSeq: 102 OPTIONS
  136. User-Agent: Asterisk PBX
  137. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  138. Contact: <sip:208.100.1.33>
  139. Accept: application/sdp
  140. Content-Length: 0
  141.  
  142.  
  143. ---
  144. Destroying call '4c441dfe3ab9a36607c3d983725fd045@208.1.87.235'
  145. ds00209*CLI>
  146. <-- SIP read from 208.1.87.235:5060:
  147. INVITE sip:s@208.100.1.33 SIP/2.0
  148. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK3f9f519a;rport
  149. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as78f48c39
  150. To: <sip:s@208.100.1.33>
  151. Contact: <sip:8159093011@208.1.87.235>
  152. Call-ID: 2989c00c19e1d8cb28cce1b070278e70@208.1.87.235
  153. CSeq: 102 INVITE
  154. User-Agent: Asterisk PBX
  155. Max-Forwards: 70
  156. Date: Thu, 15 Jan 2009 18:32:13 GMT
  157. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  158. Supported: replaces
  159. Content-Type: application/sdp
  160. Content-Length: 262
  161.  
  162. v=0
  163. o=root 5712 5712 IN IP4 208.1.87.235
  164. s=session
  165. c=IN IP4 208.1.87.235
  166. t=0 0
  167. m=audio 11530 RTP/AVP 0 8 101
  168. a=rtpmap:0 PCMU/8000
  169. a=rtpmap:8 PCMA/8000
  170. a=rtpmap:101 telephone-event/8000
  171. a=fmtp:101 0-16
  172. a=silenceSupp:off - - - -
  173. a=ptime:20
  174. a=sendrecv
  175.  
  176. --- (14 headers 13 lines)---
  177. Using INVITE request as basis request - 2989c00c19e1d8cb28cce1b070278e70@208.1.87.235
  178. Sending to 208.1.87.235 : 5060 (NAT)
  179. Found peer '8159911010'
  180. Found RTP audio format 0
  181. Found RTP audio format 8
  182. Found RTP audio format 101
  183. Peer audio RTP is at port 208.1.87.235:11530
  184. Found description format PCMU
  185. Found description format PCMA
  186. Found description format telephone-event
  187. Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
  188. Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  189. Looking for s in DID-incoming (domain 208.100.1.33)
  190. Reliably Transmitting (no NAT) to 208.1.87.235:5060:
  191. SIP/2.0 404 Not Found
  192. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK3f9f519a;rport;received=208.1.87.235
  193. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as78f48c39
  194. To: <sip:s@208.100.1.33>;tag=as3f10be63
  195. Call-ID: 2989c00c19e1d8cb28cce1b070278e70@208.1.87.235
  196. CSeq: 102 INVITE
  197. User-Agent: Asterisk PBX
  198. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  199. Contact: <sip:s@208.100.1.33>
  200. Content-Length: 0
  201.  
  202.  
  203. ---
  204. ds00209*CLI>
  205. <-- SIP read from 208.1.87.235:5060:
  206. ACK sip:s@208.100.1.33 SIP/2.0
  207. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK3f9f519a;rport
  208. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as78f48c39
  209. To: <sip:s@208.100.1.33>;tag=as3f10be63
  210. Contact: <sip:8159093011@208.1.87.235>
  211. Call-ID: 2989c00c19e1d8cb28cce1b070278e70@208.1.87.235
  212. CSeq: 102 ACK
  213. User-Agent: Asterisk PBX
  214. Max-Forwards: 70
  215. Content-Length: 0
  216.  
  217.  
  218. --- (10 headers 0 lines)---
  219. Destroying call '2989c00c19e1d8cb28cce1b070278e70@208.1.87.235'
  220. Jan 15 12:33:01 NOTICE[28744]: chan_sip.c:5357 sip_reregister: -- Re-registration for 8152641125@www2.t6voice.com
  221. REGISTER 13 headers, 0 lines
  222. Reliably Transmitting (no NAT) to 208.1.87.235:5060:
  223. REGISTER sip:www2.t6voice.com SIP/2.0
  224. Via: SIP/2.0/UDP 208.100.1.33:5060;branch=z9hG4bK16106109;rport
  225. From: <sip:8152641125@www2.t6voice.com>;tag=as6af7b5a0
  226. To: <sip:8152641125@www2.t6voice.com>
  227. Call-ID: 208349ca6390adb36f24cb415bc5449e@216.86.146.10
  228. CSeq: 106 REGISTER
  229. User-Agent: Asterisk PBX
  230. Max-Forwards: 70
  231. Authorization: Digest username="8152641125", realm="asterisk", algorithm=MD5, uri="sip:www2.t6voice.com", nonce="3ede7caf", response="d797a25fc378bcbd3e4da3875ecdd03a", opaque=""
  232. Expires: 1200
  233. Contact: <sip:s@208.100.1.33>
  234. Event: registration
  235. Content-Length: 0
  236.  
  237.  
  238. ---
  239. Jan 15 12:33:01 NOTICE[28744]: chan_sip.c:5357 sip_reregister: -- Re-registration for 8159911010@www2.t6voice.com
  240. REGISTER 13 headers, 0 lines
  241. Reliably Transmitting (no NAT) to 208.1.87.235:5060:
  242. REGISTER sip:www2.t6voice.com SIP/2.0
  243. Via: SIP/2.0/UDP 208.100.1.33:5060;branch=z9hG4bK6c020021;rport
  244. From: <sip:8159911010@www2.t6voice.com>;tag=as38c14b33
  245. To: <sip:8159911010@www2.t6voice.com>
  246. Call-ID: 153fc34e083252316dd635721e1794ef@216.86.146.10
  247. CSeq: 106 REGISTER
  248. User-Agent: Asterisk PBX
  249. Max-Forwards: 70
  250. Authorization: Digest username="8159911010", realm="asterisk", algorithm=MD5, uri="sip:www2.t6voice.com", nonce="369f67c6", response="8609d0b0cffddaa20821956bd78c9ed9", opaque=""
  251. Expires: 1200
  252. Contact: <sip:s@208.100.1.33>
  253. Event: registration
  254. Content-Length: 0
  255.  
  256.  
  257. ---
  258. ds00209*CLI>
  259. <-- SIP read from 208.1.87.235:5060:
  260. SIP/2.0 100 Trying
  261. Via: SIP/2.0/UDP 208.100.1.33:5060;branch=z9hG4bK16106109;received=208.100.1.33;rport=5060
  262. From: <sip:8152641125@www2.t6voice.com>;tag=as6af7b5a0
  263. To: <sip:8152641125@www2.t6voice.com>
  264. Call-ID: 208349ca6390adb36f24cb415bc5449e@216.86.146.10
  265. CSeq: 106 REGISTER
  266. User-Agent: Asterisk PBX
  267. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  268. Supported: replaces
  269. Contact: <sip:8152641125@208.1.87.235>
  270. Content-Length: 0
  271.  
  272.  
  273. --- (11 headers 0 lines)---
  274. ds00209*CLI>
  275. <-- SIP read from 208.1.87.235:5060:
  276. SIP/2.0 401 Unauthorized
  277. Via: SIP/2.0/UDP 208.100.1.33:5060;branch=z9hG4bK16106109;received=208.100.1.33;rport=5060
  278. From: <sip:8152641125@www2.t6voice.com>;tag=as6af7b5a0
  279. To: <sip:8152641125@www2.t6voice.com>;tag=as71d64d2f
  280. Call-ID: 208349ca6390adb36f24cb415bc5449e@216.86.146.10
  281. CSeq: 106 REGISTER
  282. User-Agent: Asterisk PBX
  283. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  284. Supported: replaces
  285. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4cf8393d"
  286. Content-Length: 0
  287.  
  288.  
  289. --- (11 headers 0 lines)---
  290. Responding to challenge, registration to domain/host name www2.t6voice.com
  291. REGISTER 13 headers, 0 lines
  292. Reliably Transmitting (no NAT) to 208.1.87.235:5060:
  293. REGISTER sip:www2.t6voice.com SIP/2.0
  294. Via: SIP/2.0/UDP 208.100.1.33:5060;branch=z9hG4bK0634941b;rport
  295. From: <sip:8152641125@www2.t6voice.com>;tag=as2357265f
  296. To: <sip:8152641125@www2.t6voice.com>
  297. Call-ID: 208349ca6390adb36f24cb415bc5449e@216.86.146.10
  298. CSeq: 107 REGISTER
  299. User-Agent: Asterisk PBX
  300. Max-Forwards: 70
  301. Authorization: Digest username="8152641125", realm="asterisk", algorithm=MD5, uri="sip:www2.t6voice.com", nonce="4cf8393d", response="0928851add7357eec26c76397bce3a19", opaque=""
  302. Expires: 1200
  303. Contact: <sip:s@208.100.1.33>
  304. Event: registration
  305. Content-Length: 0
  306.  
  307.  
  308. ---
  309. ds00209*CLI>
  310. <-- SIP read from 208.1.87.235:5060:
  311. SIP/2.0 100 Trying
  312. Via: SIP/2.0/UDP 208.100.1.33:5060;branch=z9hG4bK6c020021;received=208.100.1.33;rport=5060
  313. From: <sip:8159911010@www2.t6voice.com>;tag=as38c14b33
  314. To: <sip:8159911010@www2.t6voice.com>
  315. Call-ID: 153fc34e083252316dd635721e1794ef@216.86.146.10
  316. CSeq: 106 REGISTER
  317. User-Agent: Asterisk PBX
  318. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  319. Supported: replaces
  320. Contact: <sip:8159911010@208.1.87.235>
  321. Content-Length: 0
  322.  
  323.  
  324. --- (11 headers 0 lines)---
  325.  
  326. <-- SIP read from 208.1.87.235:5060:
  327. SIP/2.0 401 Unauthorized
  328. Via: SIP/2.0/UDP 208.100.1.33:5060;branch=z9hG4bK6c020021;received=208.100.1.33;rport=5060
  329. From: <sip:8159911010@www2.t6voice.com>;tag=as38c14b33
  330. To: <sip:8159911010@www2.t6voice.com>;tag=as16dec810
  331. Call-ID: 153fc34e083252316dd635721e1794ef@216.86.146.10
  332. CSeq: 106 REGISTER
  333. User-Agent: Asterisk PBX
  334. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  335. Supported: replaces
  336. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f9a25fa"
  337. Content-Length: 0
  338.  
  339.  
  340. --- (11 headers 0 lines)---
  341. Responding to challenge, registration to domain/host name www2.t6voice.com
  342. REGISTER 13 headers, 0 lines
  343. Reliably Transmitting (no NAT) to 208.1.87.235:5060:
  344. REGISTER sip:www2.t6voice.com SIP/2.0
  345. Via: SIP/2.0/UDP 208.100.1.33:5060;branch=z9hG4bK51b81a7c;rport
  346. From: <sip:8159911010@www2.t6voice.com>;tag=as337fcfcf
  347. To: <sip:8159911010@www2.t6voice.com>
  348. Call-ID: 153fc34e083252316dd635721e1794ef@216.86.146.10
  349. CSeq: 107 REGISTER
  350. User-Agent: Asterisk PBX
  351. Max-Forwards: 70
  352. Authorization: Digest username="8159911010", realm="asterisk", algorithm=MD5, uri="sip:www2.t6voice.com", nonce="6f9a25fa", response="969f1a9df3d6261db05e76911e32e1c9", opaque=""
  353. Expires: 1200
  354. Contact: <sip:s@208.100.1.33>
  355. Event: registration
  356. Content-Length: 0
  357.  
  358.  
  359. ---
  360.  
  361. <-- SIP read from 208.1.87.235:5060:
  362. SIP/2.0 100 Trying
  363. Via: SIP/2.0/UDP 208.100.1.33:5060;branch=z9hG4bK0634941b;received=208.100.1.33;rport=5060
  364. From: <sip:8152641125@www2.t6voice.com>;tag=as2357265f
  365. To: <sip:8152641125@www2.t6voice.com>
  366. Call-ID: 208349ca6390adb36f24cb415bc5449e@216.86.146.10
  367. CSeq: 107 REGISTER
  368. User-Agent: Asterisk PBX
  369. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  370. Supported: replaces
  371. Contact: <sip:8152641125@208.1.87.235>
  372. Content-Length: 0
  373.  
  374.  
  375. --- (11 headers 0 lines)---
  376.  
  377. <-- SIP read from 208.1.87.235:5060:
  378. SIP/2.0 200 OK
  379. Via: SIP/2.0/UDP 208.100.1.33:5060;branch=z9hG4bK0634941b;received=208.100.1.33;rport=5060
  380. From: <sip:8152641125@www2.t6voice.com>;tag=as2357265f
  381. To: <sip:8152641125@www2.t6voice.com>;tag=as71d64d2f
  382. Call-ID: 208349ca6390adb36f24cb415bc5449e@216.86.146.10
  383. CSeq: 107 REGISTER
  384. User-Agent: Asterisk PBX
  385. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  386. Supported: replaces
  387. Expires: 1200
  388. Contact: <sip:s@208.100.1.33>;expires=1200
  389. Date: Thu, 15 Jan 2009 18:32:14 GMT
  390. Content-Length: 0
  391.  
  392.  
  393. --- (13 headers 0 lines)---
  394. Scheduling destruction of call '208349ca6390adb36f24cb415bc5449e@216.86.146.10' in 32000 ms
  395. Jan 15 12:33:01 NOTICE[28744]: chan_sip.c:9854 handle_response_register: Outbound Registration: Expiry for www2.t6voice.com is 1200 sec (Scheduling reregistration in 1185 s)
  396. ds00209*CLI>
  397. <-- SIP read from 208.1.87.235:5060:
  398. SIP/2.0 100 Trying
  399. Via: SIP/2.0/UDP 208.100.1.33:5060;branch=z9hG4bK51b81a7c;received=208.100.1.33;rport=5060
  400. From: <sip:8159911010@www2.t6voice.com>;tag=as337fcfcf
  401. To: <sip:8159911010@www2.t6voice.com>
  402. Call-ID: 153fc34e083252316dd635721e1794ef@216.86.146.10
  403. CSeq: 107 REGISTER
  404. User-Agent: Asterisk PBX
  405. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  406. Supported: replaces
  407. Contact: <sip:8159911010@208.1.87.235>
  408. Content-Length: 0
  409.  
  410.  
  411. --- (11 headers 0 lines)---
  412. ds00209*CLI>
  413. <-- SIP read from 208.1.87.235:5060:
  414. SIP/2.0 200 OK
  415. Via: SIP/2.0/UDP 208.100.1.33:5060;branch=z9hG4bK51b81a7c;received=208.100.1.33;rport=5060
  416. From: <sip:8159911010@www2.t6voice.com>;tag=as337fcfcf
  417. To: <sip:8159911010@www2.t6voice.com>;tag=as16dec810
  418. Call-ID: 153fc34e083252316dd635721e1794ef@216.86.146.10
  419. CSeq: 107 REGISTER
  420. User-Agent: Asterisk PBX
  421. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  422. Supported: replaces
  423. Expires: 1200
  424. Contact: <sip:s@208.100.1.33>;expires=1200
  425. Date: Thu, 15 Jan 2009 18:32:14 GMT
  426. Content-Length: 0
  427.  
  428.  
  429. --- (13 headers 0 lines)---
  430. Scheduling destruction of call '153fc34e083252316dd635721e1794ef@216.86.146.10' in 32000 ms
  431. Jan 15 12:33:01 NOTICE[28744]: chan_sip.c:9854 handle_response_register: Outbound Registration: Expiry for www2.t6voice.com is 1200 sec (Scheduling reregistration in 1185 s)
  432. ds00209*CLI>
  433. <-- SIP read from 208.1.87.235:5060:
  434. NOTIFY sip:s@208.100.1.33 SIP/2.0
  435. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK5e4b457d;rport
  436. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as4c4bb935
  437. To: <sip:s@208.100.1.33>
  438. Contact: <sip:Unknown@208.1.87.235>
  439. Call-ID: 77dbccaa01093a42293a38924c3fdbc8@208.1.87.235
  440. CSeq: 102 NOTIFY
  441. User-Agent: Asterisk PBX
  442. Max-Forwards: 70
  443. Event: message-summary
  444. Content-Type: application/simple-message-summary
  445. Content-Length: 87
  446.  
  447. Messages-Waiting: no
  448. Message-Account: sip:*97@208.1.87.235
  449. Voice-Message: 0/0 (0/0)
  450.  
  451. --- (12 headers 3 lines)---
  452. Transmitting (no NAT) to 208.1.87.235:5060:
  453. SIP/2.0 200 OK
  454. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK5e4b457d;rport;received=208.1.87.235
  455. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as4c4bb935
  456. To: <sip:s@208.100.1.33>;tag=as23bad2b2
  457. Call-ID: 77dbccaa01093a42293a38924c3fdbc8@208.1.87.235
  458. CSeq: 102 NOTIFY
  459. User-Agent: Asterisk PBX
  460. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  461. Content-Length: 0
  462.  
  463.  
  464. ---
  465. Destroying call '77dbccaa01093a42293a38924c3fdbc8@208.1.87.235'
  466. ds00209*CLI>
  467. <-- SIP read from 208.1.87.235:5060:
  468. NOTIFY sip:s@208.100.1.33 SIP/2.0
  469. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK64835035;rport
  470. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as2cb25d8b
  471. To: <sip:s@208.100.1.33>
  472. Contact: <sip:Unknown@208.1.87.235>
  473. Call-ID: 5d0403d65182e4f1248437aa5e0676c8@208.1.87.235
  474. CSeq: 102 NOTIFY
  475. User-Agent: Asterisk PBX
  476. Max-Forwards: 70
  477. Event: message-summary
  478. Content-Type: application/simple-message-summary
  479. Content-Length: 87
  480.  
  481. Messages-Waiting: no
  482. Message-Account: sip:*97@208.1.87.235
  483. Voice-Message: 0/0 (0/0)
  484.  
  485. --- (12 headers 3 lines)---
  486. Transmitting (no NAT) to 208.1.87.235:5060:
  487. SIP/2.0 200 OK
  488. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK64835035;rport;received=208.1.87.235
  489. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as2cb25d8b
  490. To: <sip:s@208.100.1.33>;tag=as6c539522
  491. Call-ID: 5d0403d65182e4f1248437aa5e0676c8@208.1.87.235
  492. CSeq: 102 NOTIFY
  493. User-Agent: Asterisk PBX
  494. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  495. Content-Length: 0
  496.  
  497.  
  498. ---
  499. Destroying call '5d0403d65182e4f1248437aa5e0676c8@208.1.87.235'
  500. ds00209*CLI>
  501. <-- SIP read from 208.1.87.235:5060:
  502. INVITE sip:s@208.100.1.33 SIP/2.0
  503. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK08ab4349;rport
  504. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as7207850d
  505. To: <sip:s@208.100.1.33>
  506. Contact: <sip:8159093011@208.1.87.235>
  507. Call-ID: 3c7c5b08106b55877c9d20fd3bce4ac8@208.1.87.235
  508. CSeq: 102 INVITE
  509. User-Agent: Asterisk PBX
  510. Max-Forwards: 70
  511. Date: Thu, 15 Jan 2009 18:32:26 GMT
  512. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  513. Supported: replaces
  514. Content-Type: application/sdp
  515. Content-Length: 262
  516.  
  517. v=0
  518. o=root 5712 5712 IN IP4 208.1.87.235
  519. s=session
  520. c=IN IP4 208.1.87.235
  521. t=0 0
  522. m=audio 13090 RTP/AVP 0 8 101
  523. a=rtpmap:0 PCMU/8000
  524. a=rtpmap:8 PCMA/8000
  525. a=rtpmap:101 telephone-event/8000
  526. a=fmtp:101 0-16
  527. a=silenceSupp:off - - - -
  528. a=ptime:20
  529. a=sendrecv
  530.  
  531. --- (14 headers 13 lines)---
  532. Using INVITE request as basis request - 3c7c5b08106b55877c9d20fd3bce4ac8@208.1.87.235
  533. Sending to 208.1.87.235 : 5060 (NAT)
  534. Found peer '8159911010'
  535. Found RTP audio format 0
  536. Found RTP audio format 8
  537. Found RTP audio format 101
  538. Peer audio RTP is at port 208.1.87.235:13090
  539. Found description format PCMU
  540. Found description format PCMA
  541. Found description format telephone-event
  542. Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
  543. Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  544. Looking for s in DID-incoming (domain 208.100.1.33)
  545. Reliably Transmitting (no NAT) to 208.1.87.235:5060:
  546. SIP/2.0 404 Not Found
  547. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK08ab4349;rport;received=208.1.87.235
  548. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as7207850d
  549. To: <sip:s@208.100.1.33>;tag=as43b487f3
  550. Call-ID: 3c7c5b08106b55877c9d20fd3bce4ac8@208.1.87.235
  551. CSeq: 102 INVITE
  552. User-Agent: Asterisk PBX
  553. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  554. Contact: <sip:s@208.100.1.33>
  555. Content-Length: 0
  556.  
  557.  
  558. ---
  559. ds00209*CLI>
  560. <-- SIP read from 208.1.87.235:5060:
  561. ACK sip:s@208.100.1.33 SIP/2.0
  562. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK08ab4349;rport
  563. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as7207850d
  564. To: <sip:s@208.100.1.33>;tag=as43b487f3
  565. Contact: <sip:8159093011@208.1.87.235>
  566. Call-ID: 3c7c5b08106b55877c9d20fd3bce4ac8@208.1.87.235
  567. CSeq: 102 ACK
  568. User-Agent: Asterisk PBX
  569. Max-Forwards: 70
  570. Content-Length: 0
  571.  
  572.  
  573. --- (10 headers 0 lines)---
  574. Destroying call '3c7c5b08106b55877c9d20fd3bce4ac8@208.1.87.235'
  575. ds00209*CLI>
  576. <-- SIP read from 208.1.87.235:5060:
  577. INVITE sip:s@208.100.1.33 SIP/2.0
  578. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK38b06127;rport
  579. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as69c61c34
  580. To: <sip:s@208.100.1.33>
  581. Contact: <sip:8159093011@208.1.87.235>
  582. Call-ID: 64d48729186fc8240fef2ba276b61559@208.1.87.235
  583. CSeq: 102 INVITE
  584. User-Agent: Asterisk PBX
  585. Max-Forwards: 70
  586. Date: Thu, 15 Jan 2009 18:32:38 GMT
  587. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  588. Supported: replaces
  589. Content-Type: application/sdp
  590. Content-Length: 262
  591.  
  592. v=0
  593. o=root 5712 5712 IN IP4 208.1.87.235
  594. s=session
  595. c=IN IP4 208.1.87.235
  596. t=0 0
  597. m=audio 19544 RTP/AVP 0 8 101
  598. a=rtpmap:0 PCMU/8000
  599. a=rtpmap:8 PCMA/8000
  600. a=rtpmap:101 telephone-event/8000
  601. a=fmtp:101 0-16
  602. a=silenceSupp:off - - - -
  603. a=ptime:20
  604. a=sendrecv
  605.  
  606. --- (14 headers 13 lines)---
  607. Using INVITE request as basis request - 64d48729186fc8240fef2ba276b61559@208.1.87.235
  608. Sending to 208.1.87.235 : 5060 (NAT)
  609. Found peer '8159911010'
  610. Found RTP audio format 0
  611. Found RTP audio format 8
  612. Found RTP audio format 101
  613. Peer audio RTP is at port 208.1.87.235:19544
  614. Found description format PCMU
  615. Found description format PCMA
  616. Found description format telephone-event
  617. Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
  618. Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  619. Looking for s in DID-incoming (domain 208.100.1.33)
  620. Reliably Transmitting (no NAT) to 208.1.87.235:5060:
  621. SIP/2.0 404 Not Found
  622. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK38b06127;rport;received=208.1.87.235
  623. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as69c61c34
  624. To: <sip:s@208.100.1.33>;tag=as5552c5e5
  625. Call-ID: 64d48729186fc8240fef2ba276b61559@208.1.87.235
  626. CSeq: 102 INVITE
  627. User-Agent: Asterisk PBX
  628. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  629. Contact: <sip:s@208.100.1.33>
  630. Content-Length: 0
  631.  
  632.  
  633. ---
  634. ds00209*CLI>
  635. <-- SIP read from 208.1.87.235:5060:
  636. ACK sip:s@208.100.1.33 SIP/2.0
  637. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK38b06127;rport
  638. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as69c61c34
  639. To: <sip:s@208.100.1.33>;tag=as5552c5e5
  640. Contact: <sip:8159093011@208.1.87.235>
  641. Call-ID: 64d48729186fc8240fef2ba276b61559@208.1.87.235
  642. CSeq: 102 ACK
  643. User-Agent: Asterisk PBX
  644. Max-Forwards: 70
  645. Content-Length: 0
  646.  
  647.  
  648. --- (10 headers 0 lines)---
  649. Destroying call '64d48729186fc8240fef2ba276b61559@208.1.87.235'
  650. Destroying call '208349ca6390adb36f24cb415bc5449e@216.86.146.10'
  651. Destroying call '153fc34e083252316dd635721e1794ef@216.86.146.10'
  652. ds00209*CLI>
  653. <-- SIP read from 208.1.87.235:5060:
  654. INVITE sip:s@208.100.1.33 SIP/2.0
  655. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK0638b4d0;rport
  656. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as0d0528da
  657. To: <sip:s@208.100.1.33>
  658. Contact: <sip:8159093011@208.1.87.235>
  659. Call-ID: 3b37c57c752cfc6f152b29273b08ef5e@208.1.87.235
  660. CSeq: 102 INVITE
  661. User-Agent: Asterisk PBX
  662. Max-Forwards: 70
  663. Date: Thu, 15 Jan 2009 18:32:52 GMT
  664. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  665. Supported: replaces
  666. Content-Type: application/sdp
  667. Content-Length: 262
  668.  
  669. v=0
  670. o=root 5712 5712 IN IP4 208.1.87.235
  671. s=session
  672. c=IN IP4 208.1.87.235
  673. t=0 0
  674. m=audio 15288 RTP/AVP 0 8 101
  675. a=rtpmap:0 PCMU/8000
  676. a=rtpmap:8 PCMA/8000
  677. a=rtpmap:101 telephone-event/8000
  678. a=fmtp:101 0-16
  679. a=silenceSupp:off - - - -
  680. a=ptime:20
  681. a=sendrecv
  682.  
  683. --- (14 headers 13 lines)---
  684. Using INVITE request as basis request - 3b37c57c752cfc6f152b29273b08ef5e@208.1.87.235
  685. Sending to 208.1.87.235 : 5060 (NAT)
  686. Found peer '8159911010'
  687. Found RTP audio format 0
  688. Found RTP audio format 8
  689. Found RTP audio format 101
  690. Peer audio RTP is at port 208.1.87.235:15288
  691. Found description format PCMU
  692. Found description format PCMA
  693. Found description format telephone-event
  694. Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
  695. Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  696. Looking for s in DID-incoming (domain 208.100.1.33)
  697. Reliably Transmitting (no NAT) to 208.1.87.235:5060:
  698. SIP/2.0 404 Not Found
  699. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK0638b4d0;rport;received=208.1.87.235
  700. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as0d0528da
  701. To: <sip:s@208.100.1.33>;tag=as145b011b
  702. Call-ID: 3b37c57c752cfc6f152b29273b08ef5e@208.1.87.235
  703. CSeq: 102 INVITE
  704. User-Agent: Asterisk PBX
  705. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  706. Contact: <sip:s@208.100.1.33>
  707. Content-Length: 0
  708.  
  709.  
  710. ---
  711. ds00209*CLI>
  712. <-- SIP read from 208.1.87.235:5060:
  713. ACK sip:s@208.100.1.33 SIP/2.0
  714. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK0638b4d0;rport
  715. From: "Mike Hammett" <sip:8159093011@208.1.87.235>;tag=as0d0528da
  716. To: <sip:s@208.100.1.33>;tag=as145b011b
  717. Contact: <sip:8159093011@208.1.87.235>
  718. Call-ID: 3b37c57c752cfc6f152b29273b08ef5e@208.1.87.235
  719. CSeq: 102 ACK
  720. User-Agent: Asterisk PBX
  721. Max-Forwards: 70
  722. Content-Length: 0
  723.  
  724.  
  725. --- (10 headers 0 lines)---
  726. Destroying call '3b37c57c752cfc6f152b29273b08ef5e@208.1.87.235'
  727. ds00209*CLI>
  728. <-- SIP read from 208.1.87.235:5060:
  729. OPTIONS sip:s@208.100.1.33 SIP/2.0
  730. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK4d802dfb;rport
  731. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as17cb37d9
  732. To: <sip:s@208.100.1.33>
  733. Contact: <sip:Unknown@208.1.87.235>
  734. Call-ID: 030535ad50a46dc8138695521f27542c@208.1.87.235
  735. CSeq: 102 OPTIONS
  736. User-Agent: Asterisk PBX
  737. Max-Forwards: 70
  738. Date: Thu, 15 Jan 2009 18:33:02 GMT
  739. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  740. Supported: replaces
  741. Content-Length: 0
  742.  
  743.  
  744. --- (13 headers 0 lines)---
  745. Looking for s in DID-incoming (domain 208.100.1.33)
  746. Transmitting (no NAT) to 208.1.87.235:5060:
  747. SIP/2.0 404 Not Found
  748. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK4d802dfb;rport;received=208.1.87.235
  749. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as17cb37d9
  750. To: <sip:s@208.100.1.33>;tag=as66f53d5b
  751. Call-ID: 030535ad50a46dc8138695521f27542c@208.1.87.235
  752. CSeq: 102 OPTIONS
  753. User-Agent: Asterisk PBX
  754. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  755. Contact: <sip:208.100.1.33>
  756. Accept: application/sdp
  757. Content-Length: 0
  758.  
  759.  
  760. ---
  761. Destroying call '030535ad50a46dc8138695521f27542c@208.1.87.235'
  762. ds00209*CLI>
  763. <-- SIP read from 208.1.87.235:5060:
  764. OPTIONS sip:s@208.100.1.33 SIP/2.0
  765. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK6c609e4c;rport
  766. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as0315ec4d
  767. To: <sip:s@208.100.1.33>
  768. Contact: <sip:Unknown@208.1.87.235>
  769. Call-ID: 15842d8c4319d2b90d7318c473b8aba5@208.1.87.235
  770. CSeq: 102 OPTIONS
  771. User-Agent: Asterisk PBX
  772. Max-Forwards: 70
  773. Date: Thu, 15 Jan 2009 18:33:02 GMT
  774. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  775. Supported: replaces
  776. Content-Length: 0
  777.  
  778.  
  779. --- (13 headers 0 lines)---
  780. Looking for s in DID-incoming (domain 208.100.1.33)
  781. Transmitting (no NAT) to 208.1.87.235:5060:
  782. SIP/2.0 404 Not Found
  783. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK6c609e4c;rport;received=208.1.87.235
  784. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as0315ec4d
  785. To: <sip:s@208.100.1.33>;tag=as02a4421d
  786. Call-ID: 15842d8c4319d2b90d7318c473b8aba5@208.1.87.235
  787. CSeq: 102 OPTIONS
  788. User-Agent: Asterisk PBX
  789. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  790. Contact: <sip:208.100.1.33>
  791. Accept: application/sdp
  792. Content-Length: 0
  793.  
  794.  
  795. ---
  796. Destroying call '15842d8c4319d2b90d7318c473b8aba5@208.1.87.235'
  797. ds00209*CLI>
  798. <-- SIP read from 208.1.87.235:5060:
  799. OPTIONS sip:s@208.100.1.33 SIP/2.0
  800. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK1a97333a;rport
  801. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as4149685e
  802. To: <sip:s@208.100.1.33>
  803. Contact: <sip:Unknown@208.1.87.235>
  804. Call-ID: 5ae9eefe09f308db1b9bfaed71745def@208.1.87.235
  805. CSeq: 102 OPTIONS
  806. User-Agent: Asterisk PBX
  807. Max-Forwards: 70
  808. Date: Thu, 15 Jan 2009 18:34:02 GMT
  809. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  810. Supported: replaces
  811. Content-Length: 0
  812.  
  813.  
  814. --- (13 headers 0 lines)---
  815. Looking for s in DID-incoming (domain 208.100.1.33)
  816. Transmitting (no NAT) to 208.1.87.235:5060:
  817. SIP/2.0 404 Not Found
  818. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK1a97333a;rport;received=208.1.87.235
  819. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as4149685e
  820. To: <sip:s@208.100.1.33>;tag=as59013176
  821. Call-ID: 5ae9eefe09f308db1b9bfaed71745def@208.1.87.235
  822. CSeq: 102 OPTIONS
  823. User-Agent: Asterisk PBX
  824. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  825. Contact: <sip:208.100.1.33>
  826. Accept: application/sdp
  827. Content-Length: 0
  828.  
  829.  
  830. ---
  831. Destroying call '5ae9eefe09f308db1b9bfaed71745def@208.1.87.235'
  832. ds00209*CLI>
  833. <-- SIP read from 208.1.87.235:5060:
  834. OPTIONS sip:s@208.100.1.33 SIP/2.0
  835. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK2f6b1330;rport
  836. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as19863cc4
  837. To: <sip:s@208.100.1.33>
  838. Contact: <sip:Unknown@208.1.87.235>
  839. Call-ID: 07dee20e6d5a4f3f5bc3c24e7c914dde@208.1.87.235
  840. CSeq: 102 OPTIONS
  841. User-Agent: Asterisk PBX
  842. Max-Forwards: 70
  843. Date: Thu, 15 Jan 2009 18:34:02 GMT
  844. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  845. Supported: replaces
  846. Content-Length: 0
  847.  
  848.  
  849. --- (13 headers 0 lines)---
  850. Looking for s in DID-incoming (domain 208.100.1.33)
  851. Transmitting (no NAT) to 208.1.87.235:5060:
  852. SIP/2.0 404 Not Found
  853. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK2f6b1330;rport;received=208.1.87.235
  854. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as19863cc4
  855. To: <sip:s@208.100.1.33>;tag=as7fbe51ff
  856. Call-ID: 07dee20e6d5a4f3f5bc3c24e7c914dde@208.1.87.235
  857. CSeq: 102 OPTIONS
  858. User-Agent: Asterisk PBX
  859. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  860. Contact: <sip:208.100.1.33>
  861. Accept: application/sdp
  862. Content-Length: 0
  863.  
  864.  
  865. ---
  866. Destroying call '07dee20e6d5a4f3f5bc3c24e7c914dde@208.1.87.235'
  867. ds00209*CLI>
  868. <-- SIP read from 208.1.87.235:5060:
  869. OPTIONS sip:s@208.100.1.33 SIP/2.0
  870. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK4c49b0bc;rport
  871. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as22c2d98a
  872. To: <sip:s@208.100.1.33>
  873. Contact: <sip:Unknown@208.1.87.235>
  874. Call-ID: 05d6914f3b490aff61bedae865585852@208.1.87.235
  875. CSeq: 102 OPTIONS
  876. User-Agent: Asterisk PBX
  877. Max-Forwards: 70
  878. Date: Thu, 15 Jan 2009 18:35:02 GMT
  879. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  880. Supported: replaces
  881. Content-Length: 0
  882.  
  883.  
  884. --- (13 headers 0 lines)---
  885. Looking for s in DID-incoming (domain 208.100.1.33)
  886. Transmitting (no NAT) to 208.1.87.235:5060:
  887. SIP/2.0 404 Not Found
  888. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK4c49b0bc;rport;received=208.1.87.235
  889. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as22c2d98a
  890. To: <sip:s@208.100.1.33>;tag=as3e64ae51
  891. Call-ID: 05d6914f3b490aff61bedae865585852@208.1.87.235
  892. CSeq: 102 OPTIONS
  893. User-Agent: Asterisk PBX
  894. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  895. Contact: <sip:208.100.1.33>
  896. Accept: application/sdp
  897. Content-Length: 0
  898.  
  899.  
  900. ---
  901. Destroying call '05d6914f3b490aff61bedae865585852@208.1.87.235'
  902. ds00209*CLI>
  903. <-- SIP read from 208.1.87.235:5060:
  904. OPTIONS sip:s@208.100.1.33 SIP/2.0
  905. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK7287eb48;rport
  906. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as73ff88a1
  907. To: <sip:s@208.100.1.33>
  908. Contact: <sip:Unknown@208.1.87.235>
  909. Call-ID: 0a6606051f2f588c635c247c09d8ab8f@208.1.87.235
  910. CSeq: 102 OPTIONS
  911. User-Agent: Asterisk PBX
  912. Max-Forwards: 70
  913. Date: Thu, 15 Jan 2009 18:35:02 GMT
  914. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  915. Supported: replaces
  916. Content-Length: 0
  917.  
  918.  
  919. --- (13 headers 0 lines)---
  920. Looking for s in DID-incoming (domain 208.100.1.33)
  921. Transmitting (no NAT) to 208.1.87.235:5060:
  922. SIP/2.0 404 Not Found
  923. Via: SIP/2.0/UDP 208.1.87.235:5060;branch=z9hG4bK7287eb48;rport;received=208.1.87.235
  924. From: "Unknown" <sip:Unknown@208.1.87.235>;tag=as73ff88a1
  925. To: <sip:s@208.100.1.33>;tag=as2aa2f888
  926. Call-ID: 0a6606051f2f588c635c247c09d8ab8f@208.1.87.235
  927. CSeq: 102 OPTIONS
  928. User-Agent: Asterisk PBX
  929. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  930. Contact: <sip:208.100.1.33>
  931. Accept: application/sdp
  932. Content-Length: 0
  933.  
  934.  
  935. ---
  936. Destroying call '0a6606051f2f588c635c247c09d8ab8f@208.1.87.235'
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