Advertisement
Guest User

Untitled

a guest
Mar 10th, 2014
124
Never
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
text 11.43 KB | None
  1. Audio is at 20148
  2. Adding codec 0x100 (g729) to SDP
  3. Adding non-codec 0x1 (telephone-event) to SDP
  4.  
  5. <--- Transmitting (NAT) to 190.113.109.189:5060 --->
  6. SIP/2.0 183 Session Progress
  7. Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
  8. From: <sip:500@201.237.180.158>;tag=1826651759
  9. To: <sip:83226044@201.237.180.158>;tag=as1776630f
  10. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  11. CSeq: 21 INVITE
  12. Server: FPBX-2.8.1(1.8.20.0)
  13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  14. Supported: replaces, timer
  15. Contact: <sip:83226044@201.237.180.158:5060>
  16. Content-Type: application/sdp
  17. Content-Length: 264
  18.  
  19. v=0
  20. o=root 422912795 422912795 IN IP4 201.237.180.158
  21. s=Asterisk PBX 1.8.20.0
  22. c=IN IP4 201.237.180.158
  23. t=0 0
  24. m=audio 20148 RTP/AVP 18 101
  25. a=rtpmap:18 G729/8000
  26. a=fmtp:18 annexb=no
  27. a=rtpmap:101 telephone-event/8000
  28. a=fmtp:101 0-16
  29. a=ptime:20
  30. a=sendrecv
  31.  
  32. <------------>
  33. -- SIP/gateway-00000071 answered SIP/500-00000070
  34. Audio is at 20148
  35. Adding codec 0x100 (g729) to SDP
  36. Adding non-codec 0x1 (telephone-event) to SDP
  37.  
  38. <--- Reliably Transmitting (NAT) to 190.113.109.189:5060 --->
  39. SIP/2.0 200 OK
  40. Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
  41. From: <sip:500@201.237.180.158>;tag=1826651759
  42. To: <sip:83226044@201.237.180.158>;tag=as1776630f
  43. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  44. CSeq: 21 INVITE
  45. Server: FPBX-2.8.1(1.8.20.0)
  46. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  47. Supported: replaces, timer
  48. Contact: <sip:83226044@201.237.180.158:5060>
  49. Content-Type: application/sdp
  50. Content-Length: 264
  51.  
  52. v=0
  53. o=root 422912795 422912796 IN IP4 201.237.180.158
  54. s=Asterisk PBX 1.8.20.0
  55. c=IN IP4 201.237.180.158
  56. t=0 0
  57. m=audio 20148 RTP/AVP 18 101
  58. a=rtpmap:18 G729/8000
  59. a=fmtp:18 annexb=no
  60. a=rtpmap:101 telephone-event/8000
  61. a=fmtp:101 0-16
  62. a=ptime:20
  63. a=sendrecv
  64.  
  65. <------------>
  66. -- Locally bridging SIP/500-00000070 and SIP/gateway-00000071
  67. Retransmitting #1 (NAT) to 190.113.109.189:5060:
  68. SIP/2.0 200 OK
  69. Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
  70. From: <sip:500@201.237.180.158>;tag=1826651759
  71. To: <sip:83226044@201.237.180.158>;tag=as1776630f
  72. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  73. CSeq: 21 INVITE
  74. Server: FPBX-2.8.1(1.8.20.0)
  75. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  76. Supported: replaces, timer
  77. Contact: <sip:83226044@201.237.180.158:5060>
  78. ontent-Type: application/sdp
  79. Content-Length: 264
  80.  
  81. v=0
  82. o=root 422912795 422912796 IN IP4 201.237.180.158
  83. s=Asterisk PBX 1.8.20.0
  84. c=IN IP4 201.237.180.158
  85. t=0 0
  86. m=audio 20148 RTP/AVP 18 101
  87. a=rtpmap:18 G729/8000
  88. a=fmtp:18 annexb=no
  89. a=rtpmap:101 telephone-event/8000
  90. a=fmtp:101 0-16
  91. a=ptime:20
  92. a=sendrecv
  93.  
  94.  
  95. ---
  96. Retransmitting #2 (NAT) to 190.113.109.189:5060:
  97. SIP/2.0 200 OK
  98. Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
  99. From: <sip:500@201.237.180.158>;tag=1826651759
  100. To: <sip:83226044@201.237.180.158>;tag=as1776630f
  101. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  102. CSeq: 21 INVITE
  103. Server: FPBX-2.8.1(1.8.20.0)
  104. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  105. Supported: replaces, timer
  106. Contact: <sip:83226044@201.237.180.158:5060>
  107. ontent-Type: application/sdp
  108. Content-Length: 264
  109.  
  110. v=0
  111. o=root 422912795 422912796 IN IP4 201.237.180.158
  112. s=Asterisk PBX 1.8.20.0
  113. c=IN IP4 201.237.180.158
  114. t=0 0
  115. m=audio 20148 RTP/AVP 18 101
  116. a=rtpmap:18 G729/8000
  117. a=fmtp:18 annexb=no
  118. a=rtpmap:101 telephone-event/8000
  119. a=fmtp:101 0-16
  120. a=ptime:20
  121. a=sendrecv
  122.  
  123.  
  124. ---
  125. Retransmitting #3 (NAT) to 190.113.109.189:5060:
  126. SIP/2.0 200 OK
  127. Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
  128. From: <sip:500@201.237.180.158>;tag=1826651759
  129. To: <sip:83226044@201.237.180.158>;tag=as1776630f
  130. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  131. CSeq: 21 INVITE
  132. Server: FPBX-2.8.1(1.8.20.0)
  133. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  134. Supported: replaces, timer
  135. Contact: <sip:83226044@201.237.180.158:5060>
  136. ontent-Type: application/sdp
  137. Content-Length: 264
  138.  
  139. v=0
  140. o=root 422912795 422912796 IN IP4 201.237.180.158
  141. s=Asterisk PBX 1.8.20.0
  142. c=IN IP4 201.237.180.158
  143. t=0 0
  144. m=audio 20148 RTP/AVP 18 101
  145. a=rtpmap:18 G729/8000
  146. a=fmtp:18 annexb=no
  147. a=rtpmap:101 telephone-event/8000
  148. a=fmtp:101 0-16
  149. a=ptime:20
  150. a=sendrecv
  151.  
  152.  
  153. ---
  154. -- Registered SIP '500' at 190.113.109.189:35225
  155. > Saved useragent "Zoiper r21367" for peer 500
  156. Retransmitting #4 (NAT) to 190.113.109.189:5060:
  157. SIP/2.0 200 OK
  158. Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
  159. From: <sip:500@201.237.180.158>;tag=1826651759
  160. To: <sip:83226044@201.237.180.158>;tag=as1776630f
  161. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  162. CSeq: 21 INVITE
  163. Server: FPBX-2.8.1(1.8.20.0)
  164. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  165. Supported: replaces, timer
  166. Contact: <sip:83226044@201.237.180.158:5060>
  167. ontent-Type: application/sdp
  168. Content-Length: 264
  169.  
  170. v=0
  171. o=root 422912795 422912796 IN IP4 201.237.180.158
  172. s=Asterisk PBX 1.8.20.0
  173. c=IN IP4 201.237.180.158
  174. t=0 0
  175. m=audio 20148 RTP/AVP 18 101
  176. a=rtpmap:18 G729/8000
  177. a=fmtp:18 annexb=no
  178. a=rtpmap:101 telephone-event/8000
  179. a=fmtp:101 0-16
  180. a=ptime:20
  181. a=sendrecv
  182.  
  183.  
  184. ---
  185. Retransmitting #5 (NAT) to 190.113.109.189:5060:
  186. SIP/2.0 200 OK
  187. Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
  188. From: <sip:500@201.237.180.158>;tag=1826651759
  189. To: <sip:83226044@201.237.180.158>;tag=as1776630f
  190. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  191. CSeq: 21 INVITE
  192. Server: FPBX-2.8.1(1.8.20.0)
  193. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  194. Supported: replaces, timer
  195. Contact: <sip:83226044@201.237.180.158:5060>
  196. ontent-Type: application/sdp
  197. Content-Length: 264
  198.  
  199. v=0
  200. o=root 422912795 422912796 IN IP4 201.237.180.158
  201. s=Asterisk PBX 1.8.20.0
  202. c=IN IP4 201.237.180.158
  203. t=0 0
  204. m=audio 20148 RTP/AVP 18 101
  205. a=rtpmap:18 G729/8000
  206. a=fmtp:18 annexb=no
  207. a=rtpmap:101 telephone-event/8000
  208. a=fmtp:101 0-16
  209. a=ptime:20
  210. a=sendrecv
  211.  
  212.  
  213. ---
  214. Retransmitting #6 (NAT) to 190.113.109.189:5060:
  215. SIP/2.0 200 OK
  216. Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
  217. From: <sip:500@201.237.180.158>;tag=1826651759
  218. To: <sip:83226044@201.237.180.158>;tag=as1776630f
  219. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  220. CSeq: 21 INVITE
  221. Server: FPBX-2.8.1(1.8.20.0)
  222. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  223. Supported: replaces, timer
  224. Contact: <sip:83226044@201.237.180.158:5060>
  225. ontent-Type: application/sdp
  226. Content-Length: 264
  227.  
  228. v=0
  229. o=root 422912795 422912796 IN IP4 201.237.180.158
  230. s=Asterisk PBX 1.8.20.0
  231. c=IN IP4 201.237.180.158
  232. t=0 0
  233. m=audio 20148 RTP/AVP 18 101
  234. a=rtpmap:18 G729/8000
  235. a=fmtp:18 annexb=no
  236. a=rtpmap:101 telephone-event/8000
  237. a=fmtp:101 0-16
  238. a=ptime:20
  239. a=sendrecv
  240.  
  241. ---
  242. Retransmitting #6 (NAT) to 190.113.109.189:5060:
  243. SIP/2.0 200 OK
  244. Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
  245. From: <sip:500@201.237.180.158>;tag=1826651759
  246. To: <sip:83226044@201.237.180.158>;tag=as1776630f
  247. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  248. CSeq: 21 INVITE
  249. Server: FPBX-2.8.1(1.8.20.0)
  250. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  251. Supported: replaces, timer
  252. Contact: <sip:83226044@201.237.180.158:5060>
  253. ontent-Type: application/sdp
  254. Content-Length: 264
  255.  
  256. v=0
  257. o=root 422912795 422912796 IN IP4 201.237.180.158
  258. s=Asterisk PBX 1.8.20.0
  259. c=IN IP4 201.237.180.158
  260. t=0 0
  261. m=audio 20148 RTP/AVP 18 101
  262. a=rtpmap:18 G729/8000
  263. a=fmtp:18 annexb=no
  264. a=rtpmap:101 telephone-event/8000
  265. a=fmtp:101 0-16
  266. a=ptime:20
  267. a=sendrecv
  268.  
  269.  
  270. ---
  271. Retransmitting #8 (NAT) to 190.113.109.189:5060:
  272. SIP/2.0 200 OK
  273. Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
  274. From: <sip:500@201.237.180.158>;tag=1826651759
  275. To: <sip:83226044@201.237.180.158>;tag=as1776630f
  276. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  277. CSeq: 21 INVITE
  278. Server: FPBX-2.8.1(1.8.20.0)
  279. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  280. Supported: replaces, timer
  281. Contact: <sip:83226044@201.237.180.158:5060>
  282. ontent-Type: application/sdp
  283. Content-Length: 264
  284.  
  285. v=0
  286. o=root 422912795 422912796 IN IP4 201.237.180.158
  287. s=Asterisk PBX 1.8.20.0
  288. c=IN IP4 201.237.180.158
  289. t=0 0
  290. m=audio 20148 RTP/AVP 18 101
  291. a=rtpmap:18 G729/8000
  292. a=fmtp:18 annexb=no
  293. a=rtpmap:101 telephone-event/8000
  294. a=fmtp:101 0-16
  295. a=ptime:20
  296. a=sendrecv
  297.  
  298.  
  299. ---
  300. Retransmitting #9 (NAT) to 190.113.109.189:5060:
  301. SIP/2.0 200 OK
  302. Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1374081301;received=190.113.109.189;rport=5060
  303. From: <sip:500@201.237.180.158>;tag=1826651759
  304. To: <sip:83226044@201.237.180.158>;tag=as1776630f
  305. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  306. CSeq: 21 INVITE
  307. Server: FPBX-2.8.1(1.8.20.0)
  308. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  309. Supported: replaces, timer
  310. Contact: <sip:83226044@201.237.180.158:5060>
  311. ontent-Type: application/sdp
  312. Content-Length: 264
  313.  
  314. v=0
  315. o=root 422912795 422912796 IN IP4 201.237.180.158
  316. s=Asterisk PBX 1.8.20.0
  317. c=IN IP4 201.237.180.158
  318. t=0 0
  319. m=audio 20148 RTP/AVP 18 101
  320. a=rtpmap:18 G729/8000
  321. a=fmtp:18 annexb=no
  322. a=rtpmap:101 telephone-event/8000
  323. a=fmtp:101 0-16
  324. a=ptime:20
  325. a=sendrecv
  326.  
  327. ---
  328.  
  329. Scheduling destruction of SIP dialog '705414843-5060-3@BJC.BGI.A.BAB' in 27968 ms (Method: INVITE)
  330. set_destination: Parsing <sip:500@190.113.109.189:5060> for address/port to send to
  331. set_destination: set destination to 190.113.109.189:5060
  332. Reliably Transmitting (NAT) to 190.113.109.189:5060:
  333. BYE sip:500@190.113.109.189:5060 SIP/2.0
  334. Via: SIP/2.0/UDP 201.237.180.158:5060;branch=z9hG4bK1e937f44;rport
  335. Max-Forwards: 70
  336. From: <sip:83226044@201.237.180.158>;tag=as1776630f
  337. To: <sip:500@201.237.180.158>;tag=1826651759
  338. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  339. CSeq: 102 BYE
  340. User-Agent: FPBX-2.8.1(1.8.20.0)
  341. Proxy-Authorization: Digest username="500", realm="asterisk", algorithm=MD5, uri="sip:201.237.180.158", nonce="", response="62d27937fdf6d838f8331e598518d361"
  342. X-Asterisk-HangupCause: No user responding
  343. X-Asterisk-HangupCauseCode: 18
  344. Content-Length: 0
  345.  
  346.  
  347. ---
  348. Retransmitting #1 (NAT) to 190.113.109.189:5060:
  349. BYE sip:500@190.113.109.189:5060 SIP/2.0
  350. Via: SIP/2.0/UDP 201.237.180.158:5060;branch=z9hG4bK1e937f44;rport
  351. Max-Forwards: 70
  352. From: <sip:83226044@201.237.180.158>;tag=as1776630f
  353. To: <sip:500@201.237.180.158>;tag=1826651759
  354. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  355. CSeq: 102 BYE
  356. User-Agent: FPBX-2.8.1(1.8.20.0)
  357. Proxy-Authorization: Digest username="500", realm="asterisk", algorithm=MD5, uri="sip:201.237.180.158", nonce="", response="62d27937fdf6d838f8331e598518d361"
  358. X-Asterisk-HangupCause: No user responding
  359. X-Asterisk-HangupCauseCode: 18
  360. Content-Length: 0
  361.  
  362.  
  363. ---
  364.  
  365. <--- SIP read from UDP:190.113.109.189:5060 --->
  366. SIP/2.0 200 OK
  367. Via: SIP/2.0/UDP 190.113.109.189:5060;branch=z9hG4bK1e937f44;rport=5060
  368. From: <sip:83226044@201.237.180.158>;tag=as1776630f
  369. To: <sip:500@201.237.180.158>;tag=1826651759
  370. Call-ID: 705414843-5060-3@BJC.BGI.A.BAB
  371. CSeq: 102 BYE
  372. Contact: <sip:500@190.113.109.189:5060>
  373. Supported: replaces, path, timer
  374. User-Agent: Grandstream GXP1160 1.0.5.15
  375. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  376. Content-Length: 0
Advertisement
RAW Paste Data Copied
Advertisement