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  1. <--- SIP read from UDP:172.56.16.80:60073 --->
  2. INVITE sip:609846@sip.ghostcall.in SIP/2.0
  3. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  4. CSeq: 8280 INVITE
  5. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  6. To: <sip:609846@sip.ghostcall.in>
  7. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK60919b3205094e26ed2380e3b0c48892313236;rport
  8. Max-Forwards: 70
  9. Contact: <sip:85492-85489@25.98.154.0:58663;transport=udp>
  10. Content-Type: application/sdp
  11. Content-Length: 295
  12.  
  13. v=0
  14. o=- 1442958744953 1442958744973 IN IP4 25.98.154.0
  15. s=-
  16. c=IN IP4 25.98.154.0
  17. t=0 0
  18. m=audio 36158 RTP/AVP 96 97 3 0 8 127
  19. a=rtpmap:96 GSM-EFR/8000
  20. a=rtpmap:97 AMR/8000
  21. a=rtpmap:3 GSM/8000
  22. a=rtpmap:0 PCMU/8000
  23. a=rtpmap:8 PCMA/8000
  24. a=rtpmap:127 telephone-event/8000
  25. a=fmtp:127 0-15
  26. <------------->
  27. --- (10 headers 13 lines) ---
  28. Sending to 172.56.16.80:60073 (NAT)
  29. Sending to 172.56.16.80:60073 (NAT)
  30. Using INVITE request as basis request - be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  31. Found peer '85492-85489' for '85492-85489' from 172.56.16.80:60073
  32.  
  33. <--- Reliably Transmitting (NAT) to 172.56.16.80:60073 --->
  34. SIP/2.0 401 Unauthorized
  35. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK60919b3205094e26ed2380e3b0c48892313236;received=172.56.16.80;rport=60073
  36. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  37. To: <sip:609846@sip.ghostcall.in>;tag=as2d0f2766
  38. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  39. CSeq: 8280 INVITE
  40. Server: Asterisk PBX 11.17.1
  41. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  42. Supported: replaces
  43. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="20008198"
  44. Content-Length: 0
  45.  
  46.  
  47. <------------>
  48. Scheduling destruction of SIP dialog 'be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0' in 6400 ms (Method: INVITE)
  49. Reliably Transmitting (NAT) to 172.56.16.80:60073:
  50. OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  51. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK6b07d449;rport
  52. Max-Forwards: 70
  53. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as73113299
  54. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  55. Contact: <sip:asterisk@54.201.41.128:5060>
  56. Call-ID: 745734b85b1c701a32daf6ea1fd466c9@54.201.41.128:5060
  57. CSeq: 102 OPTIONS
  58. User-Agent: Asterisk PBX 11.17.1
  59. Date: Tue, 22 Sep 2015 21:54:16 GMT
  60. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  61. Supported: replaces
  62. Content-Length: 0
  63.  
  64.  
  65. ---
  66.  
  67. <--- SIP read from UDP:172.56.16.80:60073 --->
  68. ACK sip:609846@sip.ghostcall.in SIP/2.0
  69. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  70. Max-Forwards: 70
  71. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  72. To: <sip:609846@sip.ghostcall.in>;tag=as2d0f2766
  73. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK60919b3205094e26ed2380e3b0c48892313236;rport
  74. CSeq: 8280 ACK
  75. Content-Length: 0
  76.  
  77. <------------->
  78. --- (8 headers 0 lines) ---
  79.  
  80. <--- SIP read from UDP:172.56.16.80:60073 --->
  81. SIP/2.0 400 Bad Request (OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  82.  
  83. Unexpected Token : (at offset 41))
  84. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK6b07d449;rport
  85. CSeq: 102 OPTIONS
  86. Call-ID: 745734b85b1c701a32daf6ea1fd466c9@54.201.41.128:5060
  87. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as73113299
  88. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  89. Content-Type: message/sipfrag
  90.  
  91. Content-Length: 626
  92.  
  93.  
  94. OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  95. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK6b07d449;rport
  96. Max-Forwards: 70
  97. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as73113299
  98. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  99. Contact: <sip:asterisk@54.201.41.128:5060>
  100. Call-ID: 745734b85b1c701a32daf6ea1fd466c9@54.201.41.128:5060
  101. CSeq: 102 OPTIONS
  102. User-Agent: Asterisk PBX 11.17.1
  103. Date: Tue, 22 Sep 2015 21:54:16 GMT
  104. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  105. Supported: replaces
  106. Content-Length: 0
  107.  
  108. <------------->
  109. --- (1 headers 24 lines) ---
  110.  
  111. <--- SIP read from UDP:172.56.16.80:60073 --->
  112. INVITE sip:609846@sip.ghostcall.in:5060 SIP/2.0
  113. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  114. CSeq: 8281 INVITE
  115. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  116. To: <sip:609846@sip.ghostcall.in>
  117. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK5e4ec4eeecf6f9e38719994e36d8eb64313236;rport
  118. Max-Forwards: 70
  119. Contact: <sip:85492-85489@25.98.154.0:58663;transport=udp>
  120. Content-Type: application/sdp
  121. Authorization: Digest username="85492-85489",realm="asterisk",nonce="20008198",uri="sip:609846@sip.ghostcall.in:5060",response="7cf3531b953be071ff57ebc6aca08641",algorithm=MD5
  122. Content-Length: 295
  123.  
  124. v=0
  125. o=- 1442958744953 1442958744973 IN IP4 25.98.154.0
  126. s=-
  127. c=IN IP4 25.98.154.0
  128. t=0 0
  129. m=audio 36158 RTP/AVP 96 97 3 0 8 127
  130. a=rtpmap:96 GSM-EFR/8000
  131. a=rtpmap:97 AMR/8000
  132. a=rtpmap:3 GSM/8000
  133. a=rtpmap:0 PCMU/8000
  134. a=rtpmap:8 PCMA/8000
  135. a=rtpmap:127 telephone-event/8000
  136. a=fmtp:127 0-15
  137. <------------->
  138. --- (11 headers 13 lines) ---
  139. Sending to 172.56.16.80:60073 (NAT)
  140. Sending to 172.56.16.80:60073 (NAT)
  141. Using INVITE request as basis request - be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  142. Found peer '85492-85489' for '85492-85489' from 172.56.16.80:60073
  143.  
  144. <--- Reliably Transmitting (NAT) to 172.56.16.80:60073 --->
  145. SIP/2.0 401 Unauthorized
  146. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK5e4ec4eeecf6f9e38719994e36d8eb64313236;received=172.56.16.80;rport=60073
  147. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  148. To: <sip:609846@sip.ghostcall.in>;tag=as10398a34
  149. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  150. CSeq: 8281 INVITE
  151. Server: Asterisk PBX 11.17.1
  152. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  153. Supported: replaces
  154. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="52def069"
  155. Content-Length: 0
  156.  
  157.  
  158. <------------>
  159. Scheduling destruction of SIP dialog 'be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0' in 6400 ms (Method: INVITE)
  160. [2015-09-22 21:54:16] NOTICE[9413]: chan_sip.c:29710 sip_poke_peer: Still have a QUALIFY dialog active, deleting
  161. Really destroying SIP dialog '745734b85b1c701a32daf6ea1fd466c9@54.201.41.128:5060' Method: OPTIONS
  162. Reliably Transmitting (NAT) to 172.56.16.80:60073:
  163. OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  164. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK72f406d9;rport
  165. Max-Forwards: 70
  166. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as07f3c691
  167. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  168. Contact: <sip:asterisk@54.201.41.128:5060>
  169. Call-ID: 019b040d2773469e63a950ad4a5674ac@54.201.41.128:5060
  170. CSeq: 102 OPTIONS
  171. User-Agent: Asterisk PBX 11.17.1
  172. Date: Tue, 22 Sep 2015 21:54:16 GMT
  173. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  174. Supported: replaces
  175. Content-Length: 0
  176.  
  177.  
  178. ---
  179.  
  180. <--- SIP read from UDP:172.56.16.80:60073 --->
  181. ACK sip:609846@sip.ghostcall.in:5060 SIP/2.0
  182. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  183. Max-Forwards: 70
  184. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  185. To: <sip:609846@sip.ghostcall.in>;tag=as10398a34
  186. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK5e4ec4eeecf6f9e38719994e36d8eb64313236;rport
  187. CSeq: 8281 ACK
  188. Content-Length: 0
  189.  
  190. <------------->
  191. --- (8 headers 0 lines) ---
  192.  
  193. <--- SIP read from UDP:172.56.16.80:60073 --->
  194. SIP/2.0 400 Bad Request (OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  195.  
  196. Unexpected Token : (at offset 41))
  197. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK72f406d9;rport
  198. CSeq: 102 OPTIONS
  199. Call-ID: 019b040d2773469e63a950ad4a5674ac@54.201.41.128:5060
  200. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as07f3c691
  201. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  202. Content-Type: message/sipfrag
  203.  
  204. Content-Length: 626
  205.  
  206.  
  207. OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  208. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK72f406d9;rport
  209. Max-Forwards: 70
  210. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as07f3c691
  211. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  212. Contact: <sip:asterisk@54.201.41.128:5060>
  213. Call-ID: 019b040d2773469e63a950ad4a5674ac@54.201.41.128:5060
  214. CSeq: 102 OPTIONS
  215. User-Agent: Asterisk PBX 11.17.1
  216. Date: Tue, 22 Sep 2015 21:54:16 GMT
  217. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  218. Supported: replaces
  219. Content-Length: 0
  220.  
  221. <------------->
  222. --- (1 headers 24 lines) ---
  223.  
  224. <--- SIP read from UDP:172.56.16.80:60073 --->
  225. INVITE sip:609846@sip.ghostcall.in:5060 SIP/2.0
  226. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  227. CSeq: 8282 INVITE
  228. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  229. To: <sip:609846@sip.ghostcall.in>
  230. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK445043af377279a75178ccfdac8265b8313236;rport
  231. Max-Forwards: 70
  232. Contact: <sip:85492-85489@25.98.154.0:58663;transport=udp>
  233. Content-Type: application/sdp
  234. Authorization: Digest username="85492-85489",realm="asterisk",nonce="52def069",uri="sip:609846@sip.ghostcall.in:5060",response="372e15e6c97686d94635f642315e6579",algorithm=MD5
  235. Content-Length: 295
  236.  
  237. v=0
  238. o=- 1442958744953 1442958744973 IN IP4 25.98.154.0
  239. s=-
  240. c=IN IP4 25.98.154.0
  241. t=0 0
  242. m=audio 36158 RTP/AVP 96 97 3 0 8 127
  243. a=rtpmap:96 GSM-EFR/8000
  244. a=rtpmap:97 AMR/8000
  245. a=rtpmap:3 GSM/8000
  246. a=rtpmap:0 PCMU/8000
  247. a=rtpmap:8 PCMA/8000
  248. a=rtpmap:127 telephone-event/8000
  249. a=fmtp:127 0-15
  250. <------------->
  251. --- (11 headers 13 lines) ---
  252. Sending to 172.56.16.80:60073 (NAT)
  253. Using INVITE request as basis request - be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  254. Found peer '85492-85489' for '85492-85489' from 172.56.16.80:60073
  255. == Using SIP RTP CoS mark 5
  256. Found RTP audio format 96
  257. Found RTP audio format 97
  258. Found RTP audio format 3
  259. Found RTP audio format 0
  260. Found RTP audio format 8
  261. Found RTP audio format 127
  262. Found unknown media description format GSM-EFR for ID 96
  263. Found unknown media description format AMR for ID 97
  264. Found audio description format GSM for ID 3
  265. Found audio description format PCMU for ID 0
  266. Found audio description format PCMA for ID 8
  267. Found audio description format telephone-event for ID 127
  268. Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  269. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  270. Peer audio RTP is at port 25.98.154.0:36158
  271. Looking for 609846 in sipcall (domain sip.ghostcall.in)
  272. list_route: hop: <sip:85492-85489@25.98.154.0:58663;transport=udp>
  273.  
  274. <--- Transmitting (NAT) to 172.56.16.80:60073 --->
  275. SIP/2.0 100 Trying
  276. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK445043af377279a75178ccfdac8265b8313236;received=172.56.16.80;rport=60073
  277. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  278. To: <sip:609846@sip.ghostcall.in>
  279. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  280. CSeq: 8282 INVITE
  281. Server: Asterisk PBX 11.17.1
  282. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  283. Supported: replaces
  284. Contact: <sip:609846@54.201.41.128:5060>
  285. Content-Length: 0
  286.  
  287.  
  288. <------------>
  289. [2015-09-22 21:54:16] NOTICE[9413]: chan_sip.c:29710 sip_poke_peer: Still have a QUALIFY dialog active, deleting
  290. Really destroying SIP dialog '019b040d2773469e63a950ad4a5674ac@54.201.41.128:5060' Method: OPTIONS
  291. Reliably Transmitting (NAT) to 172.56.16.80:60073:
  292. OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  293. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK433410d8;rport
  294. Max-Forwards: 70
  295. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as4ca4d8cd
  296. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  297. Contact: <sip:asterisk@54.201.41.128:5060>
  298. Call-ID: 3023ec0d5a540b5d622245ec3cc9de84@54.201.41.128:5060
  299. CSeq: 102 OPTIONS
  300. User-Agent: Asterisk PBX 11.17.1
  301. Date: Tue, 22 Sep 2015 21:54:16 GMT
  302. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  303. Supported: replaces
  304. Content-Length: 0
  305.  
  306.  
  307. ---
  308. -- Executing [609846@sipcall:1] AGI("SIP/85492-85489-00000004", "agi.php,sipcall") in new stack
  309. -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php
  310. -- AGI Script Executing Application: (Dial) Options: (local/18585003770@lcr/,,m(ringing)M(callConnected,609846))
  311. -- Called local/18585003770@lcr/
  312. -- Started music on hold, class 'ringing', on SIP/85492-85489-00000004
  313. Audio is at 16460
  314. Adding codec 100003 (ulaw) to SDP
  315. Adding codec 100004 (alaw) to SDP
  316. Adding non-codec 0x1 (telephone-event) to SDP
  317.  
  318. <--- Transmitting (NAT) to 172.56.16.80:60073 --->
  319. SIP/2.0 183 Session Progress
  320. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK445043af377279a75178ccfdac8265b8313236;received=172.56.16.80;rport=60073
  321. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  322. To: <sip:609846@sip.ghostcall.in>;tag=as17faa035
  323. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  324. CSeq: 8282 INVITE
  325. Server: Asterisk PBX 11.17.1
  326. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  327. Supported: replaces
  328. Contact: <sip:609846@54.201.41.128:5060>
  329. Content-Type: application/sdp
  330. Content-Length: 260
  331.  
  332. v=0
  333. o=root 351845085 351845085 IN IP4 54.201.41.128
  334. s=Asterisk PBX 11.17.1
  335. c=IN IP4 54.201.41.128
  336. t=0 0
  337. m=audio 16460 RTP/AVP 0 8 127
  338. a=rtpmap:0 PCMU/8000
  339. a=rtpmap:8 PCMA/8000
  340. a=rtpmap:127 telephone-event/8000
  341. a=fmtp:127 0-16
  342. a=ptime:20
  343. a=sendrecv
  344.  
  345. <------------>
  346. == Using SIP RTP CoS mark 5
  347. Audio is at 13938
  348. Adding codec 100003 (ulaw) to SDP
  349. Adding codec 100004 (alaw) to SDP
  350. Adding non-codec 0x1 (telephone-event) to SDP
  351. Reliably Transmitting (no NAT) to 67.231.13.113:5060:
  352. INVITE sip:18585003770@67.231.13.113 SIP/2.0
  353. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK460c44d3
  354. Max-Forwards: 70
  355. From: <sip:19512493802@54.201.41.128>;tag=as29d77572
  356. To: <sip:18585003770@67.231.13.113>
  357. Contact: <sip:19512493802@54.201.41.128:5060>
  358. Call-ID: 4838e5a66ae6e33c5b8efd2c51e6b5cb@54.201.41.128:5060
  359. CSeq: 102 INVITE
  360. User-Agent: Asterisk PBX 11.17.1
  361. Date: Tue, 22 Sep 2015 21:54:16 GMT
  362. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  363. Supported: replaces
  364. Remote-Party-ID: "19512493802" <sip:19512493802@54.201.41.128>;party=calling;privacy=off;screen=no
  365. Content-Type: application/sdp
  366. Content-Length: 262
  367.  
  368. v=0
  369. o=root 1561207795 1561207795 IN IP4 54.201.41.128
  370. s=Asterisk PBX 11.17.1
  371. c=IN IP4 54.201.41.128
  372. t=0 0
  373. m=audio 13938 RTP/AVP 0 8 101
  374. a=rtpmap:0 PCMU/8000
  375. a=rtpmap:8 PCMA/8000
  376. a=rtpmap:101 telephone-event/8000
  377. a=fmtp:101 0-16
  378. a=ptime:20
  379. a=sendrecv
  380.  
  381. ---
  382.  
  383. <--- SIP read from UDP:172.56.16.80:60073 --->
  384. SIP/2.0 400 Bad Request (OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  385.  
  386. Unexpected Token : (at offset 41))
  387. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK433410d8;rport
  388. CSeq: 102 OPTIONS
  389. Call-ID: 3023ec0d5a540b5d622245ec3cc9de84@54.201.41.128:5060
  390. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as4ca4d8cd
  391. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  392. Content-Type: message/sipfrag
  393.  
  394. Content-Length: 626
  395.  
  396.  
  397. OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  398. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK433410d8;rport
  399. Max-Forwards: 70
  400. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as4ca4d8cd
  401. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  402. Contact: <sip:asterisk@54.201.41.128:5060>
  403. Call-ID: 3023ec0d5a540b5d622245ec3cc9de84@54.201.41.128:5060
  404. CSeq: 102 OPTIONS
  405. User-Agent: Asterisk PBX 11.17.1
  406. Date: Tue, 22 Sep 2015 21:54:16 GMT
  407. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  408. Supported: replaces
  409. Content-Length: 0
  410.  
  411. <------------->
  412. --- (1 headers 24 lines) ---
  413.  
  414. <--- SIP read from UDP:67.231.13.113:5060 --->
  415. SIP/2.0 100 Trying
  416. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK460c44d3
  417. From: <sip:19512493802@54.201.41.128>;tag=as29d77572
  418. To: <sip:18585003770@67.231.13.113>;tag=gK00f3f4c9
  419. Call-ID: 4838e5a66ae6e33c5b8efd2c51e6b5cb@54.201.41.128:5060
  420. CSeq: 102 INVITE
  421. Content-Length: 0
  422.  
  423. <------------->
  424. --- (7 headers 0 lines) ---
  425. Retransmitting #1 (NAT) to 172.56.16.80:60073:
  426. OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  427. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK433410d8;rport
  428. Max-Forwards: 70
  429. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as4ca4d8cd
  430. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  431. Contact: <sip:asterisk@54.201.41.128:5060>
  432. Call-ID: 3023ec0d5a540b5d622245ec3cc9de84@54.201.41.128:5060
  433. CSeq: 102 OPTIONS
  434. User-Agent: Asterisk PBX 11.17.1
  435. Date: Tue, 22 Sep 2015 21:54:16 GMT
  436. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  437. Supported: replaces
  438. Content-Length: 0
  439.  
  440.  
  441. ---
  442.  
  443. <--- SIP read from UDP:67.231.13.113:5060 --->
  444. SIP/2.0 183 Session Progress
  445. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK460c44d3
  446. From: <sip:19512493802@54.201.41.128>;tag=as29d77572
  447. To: <sip:18585003770@67.231.13.113>;tag=gK00f3f4c9
  448. Call-ID: 4838e5a66ae6e33c5b8efd2c51e6b5cb@54.201.41.128:5060
  449. CSeq: 102 INVITE
  450. Contact: <sip:18585003770@67.231.13.113:5060>
  451. Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS
  452. Content-Length: 244
  453. Content-Disposition: session; handling=required
  454. Content-Type: application/sdp
  455.  
  456. v=0
  457. o=Sonus_UAC 1472675637 1221840643 IN IP4 67.231.13.113
  458. s=SIP Media Capabilities
  459. c=IN IP4 67.231.13.79
  460. t=0 0
  461. m=audio 34328 RTP/AVP 0 101
  462. a=rtpmap:0 PCMU/8000
  463. a=rtpmap:101 telephone-event/8000
  464. a=fmtp:101 0-15
  465. a=sendrecv
  466. a=ptime:20
  467. <------------->
  468. --- (11 headers 11 lines) ---
  469. list_route: hop: <sip:18585003770@67.231.13.113:5060>
  470. Found RTP audio format 0
  471. Found RTP audio format 101
  472. Found audio description format PCMU for ID 0
  473. Found audio description format telephone-event for ID 101
  474. Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
  475. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  476. Peer audio RTP is at port 67.231.13.79:34328
  477. -- SIP/bandwidth-00000005 is making progress passing it to Local/18585003770@lcr-00000002;2
  478. -- Local/18585003770@lcr-00000002;1 is making progress passing it to SIP/85492-85489-00000004
  479. > 0x7f8e9c0062e0 -- Probation passed - setting RTP source address to 67.231.13.79:34328
  480.  
  481. <--- SIP read from UDP:67.231.13.113:5060 --->
  482. SIP/2.0 183 Session Progress
  483. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK460c44d3
  484. From: <sip:19512493802@54.201.41.128>;tag=as29d77572
  485. To: <sip:18585003770@67.231.13.113>;tag=gK00f3f4c9
  486. Call-ID: 4838e5a66ae6e33c5b8efd2c51e6b5cb@54.201.41.128:5060
  487. CSeq: 102 INVITE
  488. Contact: <sip:18585003770@67.231.13.113:5060>
  489. Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS
  490. Content-Length: 244
  491. Content-Disposition: session; handling=required
  492. Content-Type: application/sdp
  493.  
  494. v=0
  495. o=Sonus_UAC 1472675637 1221840643 IN IP4 67.231.13.113
  496. s=SIP Media Capabilities
  497. c=IN IP4 67.231.13.79
  498. t=0 0
  499. m=audio 34328 RTP/AVP 0 101
  500. a=rtpmap:0 PCMU/8000
  501. a=rtpmap:101 telephone-event/8000
  502. a=fmtp:101 0-15
  503. a=sendrecv
  504. a=ptime:20
  505. <------------->
  506. --- (11 headers 11 lines) ---
  507. list_route: hop: <sip:18585003770@67.231.13.113:5060>
  508. -- SIP/bandwidth-00000005 is making progress passing it to Local/18585003770@lcr-00000002;2
  509. -- Local/18585003770@lcr-00000002;1 is making progress passing it to SIP/85492-85489-00000004
  510.  
  511. <--- SIP read from UDP:172.56.16.80:60073 --->
  512. SIP/2.0 400 Bad Request (OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  513.  
  514. Unexpected Token : (at offset 41))
  515. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK433410d8;rport
  516. CSeq: 102 OPTIONS
  517. Call-ID: 3023ec0d5a540b5d622245ec3cc9de84@54.201.41.128:5060
  518. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as4ca4d8cd
  519. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  520. Content-Type: message/sipfrag
  521.  
  522. Content-Length: 626
  523.  
  524.  
  525. OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  526. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK433410d8;rport
  527. Max-Forwards: 70
  528. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as4ca4d8cd
  529. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  530. Contact: <sip:asterisk@54.201.41.128:5060>
  531. Call-ID: 3023ec0d5a540b5d622245ec3cc9de84@54.201.41.128:5060
  532. CSeq: 102 OPTIONS
  533. User-Agent: Asterisk PBX 11.17.1
  534. Date: Tue, 22 Sep 2015 21:54:16 GMT
  535. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  536. Supported: replaces
  537. Content-Length: 0
  538.  
  539. <------------->
  540. --- (1 headers 24 lines) ---
  541. Retransmitting #2 (NAT) to 172.56.16.80:60073:
  542. OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  543. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK433410d8;rport
  544. Max-Forwards: 70
  545. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as4ca4d8cd
  546. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  547. Contact: <sip:asterisk@54.201.41.128:5060>
  548. Call-ID: 3023ec0d5a540b5d622245ec3cc9de84@54.201.41.128:5060
  549. CSeq: 102 OPTIONS
  550. User-Agent: Asterisk PBX 11.17.1
  551. Date: Tue, 22 Sep 2015 21:54:16 GMT
  552. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  553. Supported: replaces
  554. Content-Length: 0
  555.  
  556.  
  557. ---
  558.  
  559. <--- SIP read from UDP:172.56.16.80:60073 --->
  560. SIP/2.0 400 Bad Request (OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  561.  
  562. Unexpected Token : (at offset 41))
  563. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK433410d8;rport
  564. CSeq: 102 OPTIONS
  565. Call-ID: 3023ec0d5a540b5d622245ec3cc9de84@54.201.41.128:5060
  566. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as4ca4d8cd
  567. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  568. Content-Type: message/sipfrag
  569.  
  570. Content-Length: 626
  571.  
  572.  
  573. OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  574. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK433410d8;rport
  575. Max-Forwards: 70
  576. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as4ca4d8cd
  577. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  578. Contact: <sip:asterisk@54.201.41.128:5060>
  579. Call-ID: 3023ec0d5a540b5d622245ec3cc9de84@54.201.41.128:5060
  580. CSeq: 102 OPTIONS
  581. User-Agent: Asterisk PBX 11.17.1
  582. Date: Tue, 22 Sep 2015 21:54:16 GMT
  583. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  584. Supported: replaces
  585. Content-Length: 0
  586.  
  587. <------------->
  588. --- (1 headers 24 lines) ---
  589.  
  590. <--- SIP read from UDP:67.231.13.113:5060 --->
  591. SIP/2.0 200 OK
  592. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK460c44d3
  593. From: <sip:19512493802@54.201.41.128>;tag=as29d77572
  594. To: <sip:18585003770@67.231.13.113>;tag=gK00f3f4c9
  595. Call-ID: 4838e5a66ae6e33c5b8efd2c51e6b5cb@54.201.41.128:5060
  596. CSeq: 102 INVITE
  597. Accept: application/sdp
  598. Contact: <sip:18585003770@67.231.13.113:5060>
  599. Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS
  600. Supported: replaces
  601. Content-Length: 244
  602. Content-Disposition: session; handling=required
  603. Content-Type: application/sdp
  604.  
  605. v=0
  606. o=Sonus_UAC 1472675637 1221840643 IN IP4 67.231.13.113
  607. s=SIP Media Capabilities
  608. c=IN IP4 67.231.13.79
  609. t=0 0
  610. m=audio 34328 RTP/AVP 0 101
  611. a=rtpmap:0 PCMU/8000
  612. a=rtpmap:101 telephone-event/8000
  613. a=fmtp:101 0-15
  614. a=sendrecv
  615. a=ptime:20
  616. <------------->
  617. --- (13 headers 11 lines) ---
  618. list_route: hop: <sip:18585003770@67.231.13.113:5060>
  619. set_destination: Parsing <sip:18585003770@67.231.13.113:5060> for address/port to send to
  620. set_destination: set destination to 67.231.13.113:5060
  621. Transmitting (no NAT) to 67.231.13.113:5060:
  622. ACK sip:18585003770@67.231.13.113:5060 SIP/2.0
  623. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK7f7c6fa1
  624. Max-Forwards: 70
  625. From: <sip:19512493802@54.201.41.128>;tag=as29d77572
  626. To: <sip:18585003770@67.231.13.113>;tag=gK00f3f4c9
  627. Contact: <sip:19512493802@54.201.41.128:5060>
  628. Call-ID: 4838e5a66ae6e33c5b8efd2c51e6b5cb@54.201.41.128:5060
  629. CSeq: 102 ACK
  630. User-Agent: Asterisk PBX 11.17.1
  631. Content-Length: 0
  632.  
  633.  
  634. ---
  635. -- SIP/bandwidth-00000005 answered Local/18585003770@lcr-00000002;2
  636. -- Local/18585003770@lcr-00000002;1 answered SIP/85492-85489-00000004
  637. -- Executing [s@macro-callConnected:1] AGI("Local/18585003770@lcr-00000002;1", "agi.php,call_status,609846") in new stack
  638. -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php
  639. -- Executing [h@lcr:1] AGI("Local/18585003770@lcr-00000002;2", "agi.php,hangup") in new stack
  640. -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php
  641. agi.php,call_status,609846: Update call status to connected for 609846
  642. -- <SIP/bandwidth-00000005>AGI Script agi.php completed, returning 0
  643. -- Executing [s@macro-callConnected:2] AGI("SIP/bandwidth-00000005", "googletts.agi,,en") in new stack
  644. -- Launched AGI Script /var/lib/asterisk/agi-bin/googletts.agi
  645. agi.php,hangup: HANGUP - HANGING UP THE CALL
  646. agi.php,hangup: SKIPPING HANGUP
  647. -- <Local/18585003770@lcr-00000002;2>AGI Script agi.php completed, returning 0
  648. == Spawn extension (lcr, 18585003770, 1) exited non-zero on 'Local/18585003770@lcr-00000002;2'
  649. -- <SIP/bandwidth-00000005>AGI Script googletts.agi completed, returning 0
  650. -- Stopped music on hold on SIP/85492-85489-00000004
  651. Audio is at 16460
  652. Adding codec 100003 (ulaw) to SDP
  653. Adding codec 100004 (alaw) to SDP
  654. Adding non-codec 0x1 (telephone-event) to SDP
  655.  
  656. <--- Reliably Transmitting (NAT) to 172.56.16.80:60073 --->
  657. SIP/2.0 200 OK
  658. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK445043af377279a75178ccfdac8265b8313236;received=172.56.16.80;rport=60073
  659. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  660. To: <sip:609846@sip.ghostcall.in>;tag=as17faa035
  661. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  662. CSeq: 8282 INVITE
  663. Server: Asterisk PBX 11.17.1
  664. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  665. Supported: replaces
  666. Contact: <sip:609846@54.201.41.128:5060>
  667. Content-Type: application/sdp
  668. Content-Length: 260
  669.  
  670. v=0
  671. o=root 351845085 351845085 IN IP4 54.201.41.128
  672. s=Asterisk PBX 11.17.1
  673. c=IN IP4 54.201.41.128
  674. t=0 0
  675. m=audio 16460 RTP/AVP 0 8 127
  676. a=rtpmap:0 PCMU/8000
  677. a=rtpmap:8 PCMA/8000
  678. a=rtpmap:127 telephone-event/8000
  679. a=fmtp:127 0-16
  680. a=ptime:20
  681. a=sendrecv
  682.  
  683. <------------>
  684. -- Locally bridging SIP/85492-85489-00000004 and SIP/bandwidth-00000005
  685. Retransmitting #1 (NAT) to 172.56.16.80:60073:
  686. SIP/2.0 200 OK
  687. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK445043af377279a75178ccfdac8265b8313236;received=172.56.16.80;rport=60073
  688. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  689. To: <sip:609846@sip.ghostcall.in>;tag=as17faa035
  690. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  691. CSeq: 8282 INVITE
  692. Server: Asterisk PBX 11.17.1
  693. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  694. Supported: replaces
  695. Contact: <sip:609846@54.201.41.128:5060>
  696. Content-Type: application/sdp
  697. Content-Length: 260
  698.  
  699. v=0
  700. o=root 351845085 351845085 IN IP4 54.201.41.128
  701. s=Asterisk PBX 11.17.1
  702. c=IN IP4 54.201.41.128
  703. t=0 0
  704. m=audio 16460 RTP/AVP 0 8 127
  705. a=rtpmap:0 PCMU/8000
  706. a=rtpmap:8 PCMA/8000
  707. a=rtpmap:127 telephone-event/8000
  708. a=fmtp:127 0-16
  709. a=ptime:20
  710. a=sendrecv
  711.  
  712. ---
  713. Retransmitting #2 (NAT) to 172.56.16.80:60073:
  714. SIP/2.0 200 OK
  715. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK445043af377279a75178ccfdac8265b8313236;received=172.56.16.80;rport=60073
  716. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  717. To: <sip:609846@sip.ghostcall.in>;tag=as17faa035
  718. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  719. CSeq: 8282 INVITE
  720. Server: Asterisk PBX 11.17.1
  721. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  722. Supported: replaces
  723. Contact: <sip:609846@54.201.41.128:5060>
  724. Content-Type: application/sdp
  725. Content-Length: 260
  726.  
  727. v=0
  728. o=root 351845085 351845085 IN IP4 54.201.41.128
  729. s=Asterisk PBX 11.17.1
  730. c=IN IP4 54.201.41.128
  731. t=0 0
  732. m=audio 16460 RTP/AVP 0 8 127
  733. a=rtpmap:0 PCMU/8000
  734. a=rtpmap:8 PCMA/8000
  735. a=rtpmap:127 telephone-event/8000
  736. a=fmtp:127 0-16
  737. a=ptime:20
  738. a=sendrecv
  739.  
  740. ---
  741.  
  742. <--- SIP read from UDP:172.56.16.80:60073 --->
  743. ACK sip:609846@54.201.41.128:5060 SIP/2.0
  744. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  745. CSeq: 8282 ACK
  746. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK3e3c3461ba32c10bbe9d965d74508c61313236
  747. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  748. To: <sip:609846@sip.ghostcall.in>;tag=as17faa035
  749. Max-Forwards: 70
  750. Authorization: Digest username="85492-85489",realm="asterisk",nonce="52def069",uri="sip:609846@sip.ghostcall.in:5060",response="372e15e6c97686d94635f642315e6579",algorithm=MD5
  751. Content-Length: 0
  752.  
  753. <------------->
  754. --- (9 headers 0 lines) ---
  755.  
  756. <--- SIP read from UDP:172.56.16.80:60073 --->
  757. OPTIONS sip:609846@sip.ghostcall.in SIP/2.0
  758. Call-ID: 4e718bb30b8a49724f0011d355e06d34@25.98.154.0
  759. CSeq: 966 OPTIONS
  760. From: <sip:85492-85489@sip.ghostcall.in>;tag=182449340
  761. To: <sip:609846@sip.ghostcall.in>
  762. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK2459ed2e1c6e9d33e18fdd7fd54ae366313236;rport
  763. Max-Forwards: 70
  764. Contact: <sip:85492-85489@25.98.154.0:58663;transport=udp>
  765. Content-Length: 0
  766.  
  767. <------------->
  768. --- (9 headers 0 lines) ---
  769. Sending to 172.56.16.80:60073 (NAT)
  770. Looking for 609846 in default (domain sip.ghostcall.in)
  771.  
  772. <--- Transmitting (NAT) to 172.56.16.80:60073 --->
  773. SIP/2.0 404 Not Found
  774. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK2459ed2e1c6e9d33e18fdd7fd54ae366313236;received=172.56.16.80;rport=60073
  775. From: <sip:85492-85489@sip.ghostcall.in>;tag=182449340
  776. To: <sip:609846@sip.ghostcall.in>;tag=as516972d0
  777. Call-ID: 4e718bb30b8a49724f0011d355e06d34@25.98.154.0
  778. CSeq: 966 OPTIONS
  779. Server: Asterisk PBX 11.17.1
  780. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  781. Supported: replaces
  782. Accept: application/sdp
  783. Content-Length: 0
  784.  
  785.  
  786. <------------>
  787. Scheduling destruction of SIP dialog '4e718bb30b8a49724f0011d355e06d34@25.98.154.0' in 32000 ms (Method: OPTIONS)
  788.  
  789. <--- SIP read from UDP:172.56.16.80:60073 --->
  790. ACK sip:609846@54.201.41.128:5060 SIP/2.0
  791. Call-ID: be1f67e1800c373fb7ddf255b7d134e8@25.98.154.0
  792. CSeq: 8282 ACK
  793. Via: SIP/2.0/UDP 25.98.154.0:58663;branch=z9hG4bK3e3c3461ba32c10bbe9d965d74508c61313236
  794. From: <sip:85492-85489@sip.ghostcall.in>;tag=605493810
  795. To: <sip:609846@sip.ghostcall.in>;tag=as17faa035
  796. Max-Forwards: 70
  797. Authorization: Digest username="85492-85489",realm="asterisk",nonce="52def069",uri="sip:609846@sip.ghostcall.in:5060",response="372e15e6c97686d94635f642315e6579",algorithm=MD5
  798. Content-Length: 0
  799.  
  800. <------------->
  801. --- (9 headers 0 lines) ---
  802. > 0x7f8ea4132e50 -- Probation passed - setting RTP source address to 172.56.16.80:64170
  803. Retransmitting #3 (NAT) to 172.56.16.80:60073:
  804. OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  805. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK433410d8;rport
  806. Max-Forwards: 70
  807. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as4ca4d8cd
  808. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  809. Contact: <sip:asterisk@54.201.41.128:5060>
  810. Call-ID: 3023ec0d5a540b5d622245ec3cc9de84@54.201.41.128:5060
  811. CSeq: 102 OPTIONS
  812. User-Agent: Asterisk PBX 11.17.1
  813. Date: Tue, 22 Sep 2015 21:54:16 GMT
  814. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  815. Supported: replaces
  816. Content-Length: 0
  817.  
  818.  
  819. ---
  820.  
  821. <--- SIP read from UDP:172.56.16.80:60073 --->
  822. SIP/2.0 400 Bad Request (OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  823.  
  824. Unexpected Token : (at offset 41))
  825. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK433410d8;rport
  826. CSeq: 102 OPTIONS
  827. Call-ID: 3023ec0d5a540b5d622245ec3cc9de84@54.201.41.128:5060
  828. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as4ca4d8cd
  829. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  830. Content-Type: message/sipfrag
  831.  
  832. Content-Length: 626
  833.  
  834.  
  835. OPTIONS sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52 SIP/2.0
  836. Via: SIP/2.0/UDP 54.201.41.128:5060;branch=z9hG4bK433410d8;rport
  837. Max-Forwards: 70
  838. From: "asterisk" <sip:asterisk@54.201.41.128>;tag=as4ca4d8cd
  839. To: <sip:85492-85489@25.98.154.0:586634f941be444bdf2fa0158f06c0d52>
  840. Contact: <sip:asterisk@54.201.41.128:5060>
  841. Call-ID: 3023ec0d5a540b5d622245ec3cc9de84@54.201.41.128:5060
  842. CSeq: 102 OPTIONS
  843. User-Agent: Asterisk PBX 11.17.1
  844. Date: Tue, 22 Sep 2015 21:54:16 GMT
  845. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  846. Supported: replaces
  847. Content-Length: 0
  848.  
  849. <------------->
  850. --- (1 headers 24 lines) ---
  851.  
  852. <--- SIP read from UDP:24.90.118.72:46123 --->
  853. OPTIONS sip:sip.ghostcall.in SIP/2.0
  854. Call-ID: bf14e7cb73f7e64b235f4d378e4d58da@10.0.1.165
  855. CSeq: 7240 OPTIONS
  856. From: <sip:211703-211700@sip.ghostcall.in>;tag=3949595897
  857. To: <sip:211703-211700@sip.ghostcall.in>
  858. Via: SIP/2.0/UDP 10.0.1.165:39328;branch=z9hG4bK2cf840c7cde0da362e83309efeea6ee9333330;rport
  859. Max-Forwards: 70
  860. User-Agent: SIPAUA/0.1.001
  861. Content-Length: 0
  862.  
  863. <------------->
  864. --- (9 headers 0 lines) ---
  865. Sending to 24.90.118.72:46123 (NAT)
  866. Looking for s in default (domain sip.ghostcall.in)
  867.  
  868. <--- Transmitting (NAT) to 24.90.118.72:46123 --->
  869. SIP/2.0 404 Not Found
  870. Via: SIP/2.0/UDP 10.0.1.165:39328;branch=z9hG4bK2cf840c7cde0da362e83309efeea6ee9333330;received=24.90.118.72;rport=46123
  871. From: <sip:211703-211700@sip.ghostcall.in>;tag=3949595897
  872. To: <sip:211703-211700@sip.ghostcall.in>;tag=as04570f5f
  873. Call-ID: bf14e7cb73f7e64b235f4d378e4d58da@10.0.1.165
  874. CSeq: 7240 OPTIONS
  875. Server: Asterisk PBX 11.17.1
  876. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  877. Supported: replaces
  878. Accept: application/sdp
  879. Content-Length: 0
  880.  
  881.  
  882. <------------>
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