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  1.  
  2. REGISTER 12 headers, 0 lines
  3. Reliably Transmitting (NAT) to 204.11.192.163:5060:
  4. REGISTER sip:callcentric.com SIP/2.0
  5. Via: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK62e2c472;rport
  6. Max-Forwards: 70
  7. From: <sip:17772409788@callcentric.com>;tag=as7ac5e878
  8. To: <sip:17772409788@callcentric.com>
  9. Call-ID: 46e4335774b9b7735a8923f846485aee@192.168.1.207
  10. CSeq: 110 REGISTER
  11. Supported: replaces, timer
  12. User-Agent: Asterisk PBX 13.6.0
  13. Authorization: Digest username="17772409788", realm="callcentric.com", algorithm=MD5, uri="sip:sip:callcentric.com", nonce="9632af4f209bb5dd833cd95f13f9fa88", response="6580a677be786b8da38ed6c8ff7edff1"
  14. Expires: 120
  15. Contact: <sip:s@142.134.91.178:5060>
  16. Content-Length: 0
  17.  
  18.  
  19. ---
  20.  
  21. <--- SIP read from UDP:204.11.192.163:5060 --->
  22. SIP/2.0 200 Ok
  23. v: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK62e2c472;rport=62265
  24. f: <sip:17772409788@callcentric.com>;tag=as7ac5e878
  25. t: <sip:17772409788@callcentric.com>
  26. i: 46e4335774b9b7735a8923f846485aee@192.168.1.207
  27. CSeq: 110 REGISTER
  28. m: <sip:s@142.134.91.178:5060>;expires=62
  29. l: 0
  30.  
  31. <------------->
  32. --- (8 headers 0 lines) ---
  33. Really destroying SIP dialog '46e4335774b9b7735a8923f846485aee@192.168.1.207' Method: REGISTER
  34.  
  35. <--- SIP read from UDP:192.168.1.102:5070 --->
  36. INVITE sip:14183172685@192.168.1.207 SIP/2.0
  37. Via: SIP/2.0/UDP 192.168.1.102:5070;rport;branch=z9hG4bK1956108
  38. To: <sip:14183172685@192.168.1.207>
  39. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8339
  40. Call-ID: 1447514428-6108-GAMING-PC@192.168.1.102
  41. CSeq: 766 INVITE
  42. Max-Forwards: 70
  43. User-Agent: NCH Software Express Talk 4.35
  44. Contact: <sip:0000FFFF0001@192.168.1.102:5070>
  45. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
  46. Supported: replaces
  47. Content-Type: application/sdp
  48. Content-Length: 354
  49.  
  50. v=0
  51. o=NCHSoftware-Talk 1447514436 1447514463 IN IP4 192.168.1.102
  52. s=Express Talk Call
  53. c=IN IP4 192.168.1.102
  54. t=0 0
  55. m=audio 8000 RTP/AVP 0 8 96 3 13 101
  56. a=rtpmap:0 PCMU/8000
  57. a=rtpmap:8 PCMA/8000
  58. a=rtpmap:96 G726-32/8000
  59. a=rtpmap:3 GSM/8000
  60. a=rtpmap:13 CN/8000
  61. a=rtpmap:101 telephone-event/8000
  62. a=fmtp:101 0-16
  63. a=sendrecv
  64. a=direction:active
  65. <------------->
  66. --- (13 headers 15 lines) ---
  67. Sending to 192.168.1.102:5070 (NAT)
  68. Sending to 192.168.1.102:5070 (NAT)
  69. Using INVITE request as basis request - 1447514428-6108-GAMING-PC@192.168.1.102
  70. Found peer '0000FFFF0001' for '0000FFFF0001' from 192.168.1.102:5070
  71.  
  72. <--- Reliably Transmitting (NAT) to 192.168.1.102:5070 --->
  73. SIP/2.0 401 Unauthorized
  74. Via: SIP/2.0/UDP 192.168.1.102:5070;branch=z9hG4bK1956108;received=192.168.1.102;rport=5070
  75. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8339
  76. To: <sip:14183172685@192.168.1.207>;tag=as1db677bf
  77. Call-ID: 1447514428-6108-GAMING-PC@192.168.1.102
  78. CSeq: 766 INVITE
  79. Server: Asterisk PBX 13.6.0
  80. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  81. Supported: replaces, timer
  82. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="24a24850"
  83. Content-Length: 0
  84.  
  85.  
  86. <------------>
  87. Scheduling destruction of SIP dialog '1447514428-6108-GAMING-PC@192.168.1.102' in 32000 ms (Method: INVITE)
  88.  
  89. <--- SIP read from UDP:192.168.1.102:5070 --->
  90. ACK sip:14183172685@192.168.1.207 SIP/2.0
  91. Via: SIP/2.0/UDP 192.168.1.102:5070;rport;branch=z9hG4bK1956108
  92. To: <sip:14183172685@192.168.1.207>;tag=as1db677bf
  93. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8339
  94. Call-ID: 1447514428-6108-GAMING-PC@192.168.1.102
  95. CSeq: 766 ACK
  96. Max-Forwards: 20
  97. User-Agent: NCH Software Express Talk 4.35
  98. Content-Length: 0
  99.  
  100. <------------->
  101. --- (9 headers 0 lines) ---
  102.  
  103. <--- SIP read from UDP:192.168.1.102:5070 --->
  104. INVITE sip:14183172685@192.168.1.207 SIP/2.0
  105. Via: SIP/2.0/UDP 192.168.1.102:5070;rport;branch=z9hG4bK1966108
  106. To: <sip:14183172685@192.168.1.207>
  107. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8339
  108. Call-ID: 1447514428-6108-GAMING-PC@192.168.1.102
  109. CSeq: 767 INVITE
  110. Max-Forwards: 20
  111. User-Agent: NCH Software Express Talk 4.35
  112. Contact: <sip:0000FFFF0001@192.168.1.102:5070>
  113. Authorization: Digest username="0000FFFF0001",realm="asterisk",nonce="24a24850",uri="sip:14183172685@192.168.1.207",response="f0d156ff9a6e04f51fd99846f18e2b8c",opaque="",algorithm=MD5
  114. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
  115. Supported: replaces
  116. Content-Type: application/sdp
  117. Content-Length: 354
  118.  
  119. v=0
  120. o=NCHSoftware-Talk 1447514436 1447514463 IN IP4 192.168.1.102
  121. s=Express Talk Call
  122. c=IN IP4 192.168.1.102
  123. t=0 0
  124. m=audio 8000 RTP/AVP 0 8 96 3 13 101
  125. a=rtpmap:0 PCMU/8000
  126. a=rtpmap:8 PCMA/8000
  127. a=rtpmap:96 G726-32/8000
  128. a=rtpmap:3 GSM/8000
  129. a=rtpmap:13 CN/8000
  130. a=rtpmap:101 telephone-event/8000
  131. a=fmtp:101 0-16
  132. a=sendrecv
  133. a=direction:active
  134. <------------->
  135. --- (14 headers 15 lines) ---
  136. Sending to 192.168.1.102:5070 (NAT)
  137. Using INVITE request as basis request - 1447514428-6108-GAMING-PC@192.168.1.102
  138. Found peer '0000FFFF0001' for '0000FFFF0001' from 192.168.1.102:5070
  139. == Using SIP RTP CoS mark 5
  140. Found RTP audio format 0
  141. Found RTP audio format 8
  142. Found RTP audio format 96
  143. Found RTP audio format 3
  144. Found RTP audio format 13
  145. Found RTP audio format 101
  146. Found audio description format PCMU for ID 0
  147. Found audio description format PCMA for ID 8
  148. Found audio description format G726-32 for ID 96
  149. Found audio description format GSM for ID 3
  150. Found audio description format CN for ID 13
  151. Found audio description format telephone-event for ID 101
  152. Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|gsm|alaw|g726)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  153. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
  154. Peer audio RTP is at port 192.168.1.102:8000
  155. Looking for 14183172685 in LocalSets (domain 192.168.1.207)
  156. sip_route_dump: route/path hop: <sip:0000FFFF0001@192.168.1.102:5070>
  157.  
  158. <--- Transmitting (NAT) to 192.168.1.102:5070 --->
  159. SIP/2.0 100 Trying
  160. Via: SIP/2.0/UDP 192.168.1.102:5070;branch=z9hG4bK1966108;received=192.168.1.102;rport=5070
  161. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8339
  162. To: <sip:14183172685@192.168.1.207>
  163. Call-ID: 1447514428-6108-GAMING-PC@192.168.1.102
  164. CSeq: 767 INVITE
  165. Server: Asterisk PBX 13.6.0
  166. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  167. Supported: replaces, timer
  168. Contact: <sip:14183172685@192.168.1.207:5060>
  169. Content-Length: 0
  170.  
  171.  
  172. <------------>
  173. -- Executing [14183172685@LocalSets:1] Log("SIP/0000FFFF0001-00000006", "NOTICE, Dialing out from "Stephen" <0000FFFF0001> to 4183172685 through Foo Provider") in new stack
  174. -- Executing [14183172685@LocalSets:2] Dial("SIP/0000FFFF0001-00000006", "SIP/14183172685@callcentric") in new stack
  175. == Using SIP RTP CoS mark 5
  176. Audio is at 25812
  177. Adding codec ulaw to SDP
  178. Adding non-codec 0x1 (telephone-event) to SDP
  179. Reliably Transmitting (no NAT) to 204.11.192.39:5080:
  180. INVITE sip:14183172685@callcentric.com SIP/2.0
  181. Via: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK40a26109
  182. Max-Forwards: 70
  183. From: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  184. To: <sip:14183172685@callcentric.com>
  185. Contact: <sip:17772409788@142.134.91.178:5060>
  186. Call-ID: 4d2699751664361c1a2c819e06af92df@callcentric.com
  187. CSeq: 102 INVITE
  188. User-Agent: Asterisk PBX 13.6.0
  189. Date: Sat, 14 Nov 2015 21:03:05 GMT
  190. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  191. Supported: replaces, timer
  192. Content-Type: application/sdp
  193. Content-Length: 243
  194.  
  195. v=0
  196. o=root 1942138452 1942138452 IN IP4 142.134.91.178
  197. s=Asterisk PBX 13.6.0
  198. c=IN IP4 142.134.91.178
  199. t=0 0
  200. m=audio 25812 RTP/AVP 0 101
  201. a=rtpmap:0 PCMU/8000
  202. a=rtpmap:101 telephone-event/8000
  203. a=fmtp:101 0-16
  204. a=maxptime:150
  205. a=sendrecv
  206.  
  207. ---
  208. -- Called SIP/14183172685@callcentric
  209.  
  210. <--- SIP read from UDP:204.11.192.39:5080 --->
  211. SIP/2.0 407 Proxy Authentication Required
  212. v: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK40a26109
  213. f: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  214. t: <sip:14183172685@callcentric.com>
  215. i: 4d2699751664361c1a2c819e06af92df@callcentric.com
  216. CSeq: 102 INVITE
  217. Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="7d8c988c6fe4a01478392c6595b6334a", opaque="", stale=TRUE, algorithm=MD5
  218. l: 0
  219.  
  220. <------------->
  221. --- (8 headers 0 lines) ---
  222. Transmitting (no NAT) to 204.11.192.39:5080:
  223. ACK sip:14183172685@callcentric.com SIP/2.0
  224. Via: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK40a26109
  225. Max-Forwards: 70
  226. From: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  227. To: <sip:14183172685@callcentric.com>
  228. Contact: <sip:17772409788@142.134.91.178:5060>
  229. Call-ID: 4d2699751664361c1a2c819e06af92df@callcentric.com
  230. CSeq: 102 ACK
  231. User-Agent: Asterisk PBX 13.6.0
  232. Content-Length: 0
  233.  
  234.  
  235. ---
  236. Audio is at 25812
  237. Adding codec ulaw to SDP
  238. Adding non-codec 0x1 (telephone-event) to SDP
  239. Reliably Transmitting (no NAT) to 204.11.192.39:5080:
  240. INVITE sip:14183172685@callcentric.com SIP/2.0
  241. Via: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK51660b21
  242. Max-Forwards: 70
  243. From: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  244. To: <sip:14183172685@callcentric.com>
  245. Contact: <sip:17772409788@142.134.91.178:5060>
  246. Call-ID: 4d2699751664361c1a2c819e06af92df@callcentric.com
  247. CSeq: 103 INVITE
  248. User-Agent: Asterisk PBX 13.6.0
  249. Proxy-Authorization: Digest username="17772409788", realm="callcentric.com", algorithm=MD5, uri="sip:sip:callcentric.com", nonce="7d8c988c6fe4a01478392c6595b6334a", response="f94801db90b2fdff0e3c98066633e1f0"
  250. Date: Sat, 14 Nov 2015 21:03:05 GMT
  251. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  252. Supported: replaces, timer
  253. Content-Type: application/sdp
  254. Content-Length: 243
  255.  
  256. v=0
  257. o=root 1942138452 1942138453 IN IP4 142.134.91.178
  258. s=Asterisk PBX 13.6.0
  259. c=IN IP4 142.134.91.178
  260. t=0 0
  261. m=audio 25812 RTP/AVP 0 101
  262. a=rtpmap:0 PCMU/8000
  263. a=rtpmap:101 telephone-event/8000
  264. a=fmtp:101 0-16
  265. a=maxptime:150
  266. a=sendrecv
  267.  
  268. ---
  269.  
  270. <--- SIP read from UDP:204.11.192.39:5080 --->
  271. SIP/2.0 100 Trying
  272. v: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK51660b21
  273. f: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  274. t: <sip:14183172685@callcentric.com>
  275. i: 4d2699751664361c1a2c819e06af92df@callcentric.com
  276. CSeq: 103 INVITE
  277. l: 0
  278.  
  279. <------------->
  280. --- (7 headers 0 lines) ---
  281.  
  282. <--- SIP read from UDP:204.11.192.39:5080 --->
  283. SIP/2.0 183 Session Progress
  284. v: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK51660b21
  285. f: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  286. t: <sip:14183172685@callcentric.com>;tag=l8cldz860o
  287. i: 4d2699751664361c1a2c819e06af92df@callcentric.com
  288. CSeq: 103 INVITE
  289. m: <sip:d3d2f28969567d89093eddc3315ceb1d@204.11.192.39:5080;transport=udp>
  290. Supported: timer, 100rel, replaces
  291. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
  292. c: application/sdp
  293. l: 252
  294.  
  295. v=0
  296. o=root 359878195 359878195 IN IP4 204.11.192.39
  297. s=call
  298. c=IN IP4 204.11.192.39
  299. t=0 0
  300. m=audio 57200 RTP/AVP 0 101
  301. a=rtpmap:0 pcmu/8000
  302. a=rtpmap:101 telephone-event/8000
  303. a=fmtp:101 0-15
  304. a=sendrecv
  305. a=silenceSupp:off - - - -
  306. a=setup:actpass
  307. <------------->
  308. --- (11 headers 12 lines) ---
  309. sip_route_dump: route/path hop: <sip:d3d2f28969567d89093eddc3315ceb1d@204.11.192.39:5080;transport=udp>
  310. Found RTP audio format 0
  311. Found RTP audio format 101
  312. Found audio description format pcmu for ID 0
  313. Found audio description format telephone-event for ID 101
  314. Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
  315. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  316. Peer audio RTP is at port 204.11.192.39:57200
  317. -- SIP/callcentric-00000007 is making progress passing it to SIP/0000FFFF0001-00000006
  318. Audio is at 18538
  319. Adding codec ulaw to SDP
  320. Adding codec alaw to SDP
  321. Adding non-codec 0x1 (telephone-event) to SDP
  322.  
  323. <--- Transmitting (NAT) to 192.168.1.102:5070 --->
  324. SIP/2.0 183 Session Progress
  325. Via: SIP/2.0/UDP 192.168.1.102:5070;branch=z9hG4bK1966108;received=192.168.1.102;rport=5070
  326. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8339
  327. To: <sip:14183172685@192.168.1.207>;tag=as3b133f47
  328. Call-ID: 1447514428-6108-GAMING-PC@192.168.1.102
  329. CSeq: 767 INVITE
  330. Server: Asterisk PBX 13.6.0
  331. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  332. Supported: replaces, timer
  333. Contact: <sip:14183172685@192.168.1.207:5060>
  334. Content-Type: application/sdp
  335. Content-Length: 263
  336.  
  337. v=0
  338. o=root 556121366 556121366 IN IP4 192.168.1.207
  339. s=Asterisk PBX 13.6.0
  340. c=IN IP4 192.168.1.207
  341. t=0 0
  342. m=audio 18538 RTP/AVP 0 8 101
  343. a=rtpmap:0 PCMU/8000
  344. a=rtpmap:8 PCMA/8000
  345. a=rtpmap:101 telephone-event/8000
  346. a=fmtp:101 0-16
  347. a=maxptime:150
  348. a=sendrecv
  349.  
  350. <------------>
  351.  
  352. <--- SIP read from UDP:204.11.192.39:5080 --->
  353. SIP/2.0 183 Session Progress
  354. v: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK51660b21
  355. f: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  356. t: <sip:14183172685@callcentric.com>;tag=l8cldz860o
  357. i: 4d2699751664361c1a2c819e06af92df@callcentric.com
  358. CSeq: 103 INVITE
  359. m: <sip:d3d2f28969567d89093eddc3315ceb1d@204.11.192.39:5080;transport=udp>
  360. Supported: timer, 100rel, replaces
  361. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
  362. c: application/sdp
  363. l: 252
  364.  
  365. v=0
  366. o=root 359878195 359878195 IN IP4 204.11.192.39
  367. s=call
  368. c=IN IP4 204.11.192.39
  369. t=0 0
  370. m=audio 57200 RTP/AVP 0 101
  371. a=rtpmap:0 pcmu/8000
  372. a=rtpmap:101 telephone-event/8000
  373. a=fmtp:101 0-15
  374. a=sendrecv
  375. a=silenceSupp:off - - - -
  376. a=setup:actpass
  377. <------------->
  378. --- (11 headers 12 lines) ---
  379. sip_route_dump: route/path hop: <sip:d3d2f28969567d89093eddc3315ceb1d@204.11.192.39:5080;transport=udp>
  380. -- SIP/callcentric-00000007 is making progress passing it to SIP/0000FFFF0001-00000006
  381.  
  382. <--- SIP read from UDP:204.11.192.39:5080 --->
  383. SIP/2.0 183 Session Progress
  384. v: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK51660b21
  385. f: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  386. t: <sip:14183172685@callcentric.com>;tag=l8cldz860o
  387. i: 4d2699751664361c1a2c819e06af92df@callcentric.com
  388. CSeq: 103 INVITE
  389. m: <sip:d3d2f28969567d89093eddc3315ceb1d@204.11.192.39:5080;transport=udp>
  390. Supported: timer, 100rel, replaces
  391. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
  392. c: application/sdp
  393. l: 252
  394.  
  395. v=0
  396. o=root 359878195 359878195 IN IP4 204.11.192.39
  397. s=call
  398. c=IN IP4 204.11.192.39
  399. t=0 0
  400. m=audio 57200 RTP/AVP 0 101
  401. a=rtpmap:0 pcmu/8000
  402. a=rtpmap:101 telephone-event/8000
  403. a=fmtp:101 0-15
  404. a=sendrecv
  405. a=silenceSupp:off - - - -
  406. a=setup:actpass
  407. <------------->
  408. --- (11 headers 12 lines) ---
  409. sip_route_dump: route/path hop: <sip:d3d2f28969567d89093eddc3315ceb1d@204.11.192.39:5080;transport=udp>
  410. -- SIP/callcentric-00000007 is making progress passing it to SIP/0000FFFF0001-00000006
  411.  
  412. <--- SIP read from UDP:204.11.192.39:5080 --->
  413. SIP/2.0 183 Session Progress
  414. v: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK51660b21
  415. f: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  416. t: <sip:14183172685@callcentric.com>;tag=l8cldz860o
  417. i: 4d2699751664361c1a2c819e06af92df@callcentric.com
  418. CSeq: 103 INVITE
  419. m: <sip:d3d2f28969567d89093eddc3315ceb1d@204.11.192.39:5080;transport=udp>
  420. Supported: timer, 100rel, replaces
  421. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
  422. c: application/sdp
  423. l: 252
  424.  
  425. v=0
  426. o=root 359878195 359878195 IN IP4 204.11.192.39
  427. s=call
  428. c=IN IP4 204.11.192.39
  429. t=0 0
  430. m=audio 57200 RTP/AVP 0 101
  431. a=rtpmap:0 pcmu/8000
  432. a=rtpmap:101 telephone-event/8000
  433. a=fmtp:101 0-15
  434. a=sendrecv
  435. a=silenceSupp:off - - - -
  436. a=setup:actpass
  437. <------------->
  438. --- (11 headers 12 lines) ---
  439. sip_route_dump: route/path hop: <sip:d3d2f28969567d89093eddc3315ceb1d@204.11.192.39:5080;transport=udp>
  440. -- SIP/callcentric-00000007 is making progress passing it to SIP/0000FFFF0001-00000006
  441.  
  442. <--- SIP read from UDP:204.11.192.39:5080 --->
  443. SIP/2.0 183 Session Progress
  444. v: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK51660b21
  445. f: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  446. t: <sip:14183172685@callcentric.com>;tag=l8cldz860o
  447. i: 4d2699751664361c1a2c819e06af92df@callcentric.com
  448. CSeq: 103 INVITE
  449. m: <sip:d3d2f28969567d89093eddc3315ceb1d@204.11.192.39:5080;transport=udp>
  450. Supported: timer, 100rel, replaces
  451. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
  452. c: application/sdp
  453. l: 252
  454.  
  455. v=0
  456. o=root 359878195 359878195 IN IP4 204.11.192.39
  457. s=call
  458. c=IN IP4 204.11.192.39
  459. t=0 0
  460. m=audio 57200 RTP/AVP 0 101
  461. a=rtpmap:0 pcmu/8000
  462. a=rtpmap:101 telephone-event/8000
  463. a=fmtp:101 0-15
  464. a=sendrecv
  465. a=silenceSupp:off - - - -
  466. a=setup:actpass
  467. <------------->
  468. --- (11 headers 12 lines) ---
  469. sip_route_dump: route/path hop: <sip:d3d2f28969567d89093eddc3315ceb1d@204.11.192.39:5080;transport=udp>
  470. -- SIP/callcentric-00000007 is making progress passing it to SIP/0000FFFF0001-00000006
  471.  
  472. <--- SIP read from UDP:192.168.1.102:5070 --->
  473. CANCEL sip:14183172685@192.168.1.207 SIP/2.0
  474. Via: SIP/2.0/UDP 192.168.1.102:5070;rport;branch=z9hG4bK1966108
  475. To: <sip:14183172685@192.168.1.207>
  476. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8339
  477. Call-ID: 1447514428-6108-GAMING-PC@192.168.1.102
  478. CSeq: 767 CANCEL
  479. Max-Forwards: 20
  480. User-Agent: NCH Software Express Talk 4.35
  481. Authorization: Digest username="0000FFFF0001",realm="asterisk",nonce="24a24850",uri="sip:14183172685@192.168.1.207",response="f0d156ff9a6e04f51fd99846f18e2b8c",opaque="",algorithm=MD5
  482. Content-Length: 0
  483.  
  484. <------------->
  485. --- (10 headers 0 lines) ---
  486. Sending to 192.168.1.102:5070 (NAT)
  487.  
  488. <--- Reliably Transmitting (NAT) to 192.168.1.102:5070 --->
  489. SIP/2.0 487 Request Terminated
  490. Via: SIP/2.0/UDP 192.168.1.102:5070;branch=z9hG4bK1966108;received=192.168.1.102;rport=5070
  491. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8339
  492. To: <sip:14183172685@192.168.1.207>;tag=as3b133f47
  493. Call-ID: 1447514428-6108-GAMING-PC@192.168.1.102
  494. CSeq: 767 INVITE
  495. Server: Asterisk PBX 13.6.0
  496. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  497. Supported: replaces, timer
  498. Content-Length: 0
  499.  
  500.  
  501. <------------>
  502.  
  503. <--- Transmitting (NAT) to 192.168.1.102:5070 --->
  504. SIP/2.0 200 OK
  505. Via: SIP/2.0/UDP 192.168.1.102:5070;branch=z9hG4bK1966108;received=192.168.1.102;rport=5070
  506. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8339
  507. To: <sip:14183172685@192.168.1.207>;tag=as3b133f47
  508. Call-ID: 1447514428-6108-GAMING-PC@192.168.1.102
  509. CSeq: 767 CANCEL
  510. Server: Asterisk PBX 13.6.0
  511. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  512. Supported: replaces, timer
  513. Content-Length: 0
  514.  
  515.  
  516. <------------>
  517.  
  518. <--- SIP read from UDP:192.168.1.102:5070 --->
  519. ACK sip:14183172685@192.168.1.207 SIP/2.0
  520. Via: SIP/2.0/UDP 192.168.1.102:5070;rport;branch=z9hG4bK1966108
  521. To: <sip:14183172685@192.168.1.207>;tag=as3b133f47
  522. From: "Stephen" <sip:0000FFFF0001@192.168.1.207>;tag=8339
  523. Call-ID: 1447514428-6108-GAMING-PC@192.168.1.102
  524. CSeq: 767 ACK
  525. Max-Forwards: 20
  526. User-Agent: NCH Software Express Talk 4.35
  527. Authorization: Digest username="0000FFFF0001",realm="asterisk",nonce="24a24850",uri="sip:14183172685@192.168.1.207",response="f0d156ff9a6e04f51fd99846f18e2b8c",opaque="",algorithm=MD5
  528. Content-Length: 0
  529.  
  530. <------------->
  531. --- (10 headers 0 lines) ---
  532. Scheduling destruction of SIP dialog '4d2699751664361c1a2c819e06af92df@callcentric.com' in 32000 ms (Method: INVITE)
  533. Reliably Transmitting (no NAT) to 204.11.192.39:5080:
  534. CANCEL sip:14183172685@callcentric.com SIP/2.0
  535. Via: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK51660b21
  536. Max-Forwards: 70
  537. From: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  538. To: <sip:14183172685@callcentric.com>
  539. Call-ID: 4d2699751664361c1a2c819e06af92df@callcentric.com
  540. CSeq: 103 CANCEL
  541. User-Agent: Asterisk PBX 13.6.0
  542. Content-Length: 0
  543.  
  544.  
  545. ---
  546. Scheduling destruction of SIP dialog '4d2699751664361c1a2c819e06af92df@callcentric.com' in 32000 ms (Method: INVITE)
  547. == Spawn extension (LocalSets, 14183172685, 2) exited non-zero on 'SIP/0000FFFF0001-00000006'
  548.  
  549. <--- SIP read from UDP:204.11.192.39:5080 --->
  550. SIP/2.0 200 OK
  551. v: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK51660b21
  552. f: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  553. t: <sip:14183172685@callcentric.com>;tag=l8cldz860o
  554. i: 4d2699751664361c1a2c819e06af92df@callcentric.com
  555. CSeq: 103 CANCEL
  556. l: 0
  557.  
  558. <------------->
  559. --- (7 headers 0 lines) ---
  560. Really destroying SIP dialog '1447514428-6108-GAMING-PC@192.168.1.102' Method: ACK
  561.  
  562. <--- SIP read from UDP:204.11.192.39:5080 --->
  563. SIP/2.0 487 Request Terminated
  564. v: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK51660b21
  565. f: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  566. t: <sip:14183172685@callcentric.com>;tag=l8cldz860o
  567. i: 4d2699751664361c1a2c819e06af92df@callcentric.com
  568. CSeq: 103 INVITE
  569. m: <sip:f808c804f0edc4d34356b23bfb947120@204.11.192.39:5080;transport=udp>
  570. l: 0
  571.  
  572. <------------->
  573. --- (8 headers 0 lines) ---
  574. Transmitting (no NAT) to 204.11.192.39:5080:
  575. ACK sip:d3d2f28969567d89093eddc3315ceb1d@204.11.192.39:5080;transport=udp SIP/2.0
  576. Via: SIP/2.0/UDP 142.134.91.178:5060;branch=z9hG4bK51660b21
  577. Max-Forwards: 70
  578. From: "Stephen" <sip:17772409788@callcentric.com>;tag=as77f921fe
  579. To: <sip:14183172685@callcentric.com>;tag=l8cldz860o
  580. Contact: <sip:17772409788@142.134.91.178:5060>
  581. Call-ID: 4d2699751664361c1a2c819e06af92df@callcentric.com
  582. CSeq: 103 ACK
  583. User-Agent: Asterisk PBX 13.6.0
  584. Content-Length: 0
  585.  
  586.  
  587. ---
  588. Scheduling destruction of SIP dialog '4d2699751664361c1a2c819e06af92df@callcentric.com' in 32000 ms (Method: INVITE)
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