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  1. <--- SIP read from UDP:10.10.10.189:5060 --->
  2. INVITE sip:104@10.10.10.6 SIP/2.0
  3. Via: SIP/2.0/UDP 10.10.10.189:5060;branch=z9hG4bK364935080;rport
  4. From: <sip:103@10.10.10.6>;tag=1134179135
  5. To: <sip:104@10.10.10.6>
  6. Call-ID: 1844477604-5060-24@BA.BA.BA.BIJ
  7. CSeq: 230 INVITE
  8. Contact: <sip:103@10.10.10.189:5060>
  9. Max-Forwards: 70
  10. User-Agent: Grandstream GXP1625 1.0.2.4
  11. Privacy: none
  12. P-Preferred-Identity: <sip:103@10.10.10.6>
  13. Supported: replaces, path, timer
  14. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  15. Content-Type: application/sdp
  16. Accept: application/sdp, application/dtmf-relay
  17. Content-Length: 332
  18.  
  19. v=0
  20. o=103 8000 8000 IN IP4 10.10.10.189
  21. s=SIP Call
  22. c=IN IP4 10.10.10.189
  23. t=0 0
  24. m=audio 5004 RTP/AVP 0 8 18 9 2 101
  25. a=sendrecv
  26. a=rtpmap:0 PCMU/8000
  27. a=ptime:20
  28. a=rtpmap:8 PCMA/8000
  29. a=rtpmap:18 G729/8000
  30. a=fmtp:18 annexb=no
  31. a=rtpmap:9 G722/8000
  32. a=rtpmap:2 G726-32/8000
  33. a=rtpmap:101 telephone-event/8000
  34. a=fmtp:101 0-15
  35. <------------->
  36. --- (16 headers 16 lines) ---
  37. Sending to 10.10.10.189:5060 (no NAT)
  38. Sending to 10.10.10.189:5060 (no NAT)
  39. Using INVITE request as basis request - 1844477604-5060-24@BA.BA.BA.BIJ
  40. Found peer '103' for '103' from 10.10.10.189:5060
  41.  
  42. <--- Reliably Transmitting (no NAT) to 10.10.10.189:5060 --->
  43. SIP/2.0 401 Unauthorized
  44. Via: SIP/2.0/UDP 10.10.10.189:5060;branch=z9hG4bK364935080;received=10.10.10.189;rport=5060
  45. From: <sip:103@10.10.10.6>;tag=1134179135
  46. To: <sip:104@10.10.10.6>;tag=as3ee5888a
  47. Call-ID: 1844477604-5060-24@BA.BA.BA.BIJ
  48. CSeq: 230 INVITE
  49. Server: Asterisk PBX 13.6.0
  50. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  51. Supported: replaces, timer
  52. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37eb957f"
  53. Content-Length: 0
  54.  
  55.  
  56. <------------>
  57. Scheduling destruction of SIP dialog '1844477604-5060-24@BA.BA.BA.BIJ' in 6400 ms (Method: INVITE)
  58.  
  59. <--- SIP read from UDP:10.10.10.189:5060 --->
  60. ACK sip:104@10.10.10.6 SIP/2.0
  61. Via: SIP/2.0/UDP 10.10.10.189:5060;branch=z9hG4bK364935080;rport
  62. From: <sip:103@10.10.10.6>;tag=1134179135
  63. To: <sip:104@10.10.10.6>;tag=as3ee5888a
  64. Call-ID: 1844477604-5060-24@BA.BA.BA.BIJ
  65. CSeq: 230 ACK
  66. Content-Length: 0
  67.  
  68. <------------->
  69. --- (7 headers 0 lines) ---
  70.  
  71. <--- SIP read from UDP:10.10.10.189:5060 --->
  72. INVITE sip:104@10.10.10.6 SIP/2.0
  73. Via: SIP/2.0/UDP 10.10.10.189:5060;branch=z9hG4bK165226484;rport
  74. From: <sip:103@10.10.10.6>;tag=1134179135
  75. To: <sip:104@10.10.10.6>
  76. Call-ID: 1844477604-5060-24@BA.BA.BA.BIJ
  77. CSeq: 231 INVITE
  78. Contact: <sip:103@10.10.10.189:5060>
  79. Authorization: Digest username="103", realm="asterisk", nonce="37eb957f", uri="sip:104@10.10.10.6", response="096ec291ab312dbf70d0e4ee47645f8e", algorithm=MD5
  80. Max-Forwards: 70
  81. User-Agent: Grandstream GXP1625 1.0.2.4
  82. Privacy: none
  83. P-Preferred-Identity: <sip:103@10.10.10.6>
  84. Supported: replaces, path, timer
  85. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  86. Content-Type: application/sdp
  87. Accept: application/sdp, application/dtmf-relay
  88. Content-Length: 332
  89.  
  90. v=0
  91. o=103 8000 8000 IN IP4 10.10.10.189
  92. s=SIP Call
  93. c=IN IP4 10.10.10.189
  94. t=0 0
  95. m=audio 5004 RTP/AVP 0 8 18 9 2 101
  96. a=sendrecv
  97. a=rtpmap:0 PCMU/8000
  98. a=ptime:20
  99. a=rtpmap:8 PCMA/8000
  100. a=rtpmap:18 G729/8000
  101. a=fmtp:18 annexb=no
  102. a=rtpmap:9 G722/8000
  103. a=rtpmap:2 G726-32/8000
  104. a=rtpmap:101 telephone-event/8000
  105. a=fmtp:101 0-15
  106. <------------->
  107. --- (17 headers 16 lines) ---
  108. Sending to 10.10.10.189:5060 (no NAT)
  109. Using INVITE request as basis request - 1844477604-5060-24@BA.BA.BA.BIJ
  110. Found peer '103' for '103' from 10.10.10.189:5060
  111. == Using SIP RTP CoS mark 5
  112. Found RTP audio format 0
  113. Found RTP audio format 8
  114. Found RTP audio format 18
  115. Found RTP audio format 9
  116. Found RTP audio format 2
  117. Found RTP audio format 101
  118. Found audio description format PCMU for ID 0
  119. Found audio description format PCMA for ID 8
  120. Found audio description format G729 for ID 18
  121. Found audio description format G722 for ID 9
  122. Found audio description format G726-32 for ID 2
  123. Found audio description format telephone-event for ID 101
  124. Capabilities: us - (g729|alaw|gsm), peer - audio=(ulaw|g726|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (g729|alaw)
  125. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  126. Peer audio RTP is at port 10.10.10.189:5004
  127. Looking for 104 in tecnico (domain 10.10.10.6)
  128. sip_route_dump: route/path hop: <sip:103@10.10.10.189:5060>
  129.  
  130. <--- Transmitting (no NAT) to 10.10.10.189:5060 --->
  131. SIP/2.0 100 Trying
  132. Via: SIP/2.0/UDP 10.10.10.189:5060;branch=z9hG4bK165226484;received=10.10.10.189;rport=5060
  133. From: <sip:103@10.10.10.6>;tag=1134179135
  134. To: <sip:104@10.10.10.6>
  135. Call-ID: 1844477604-5060-24@BA.BA.BA.BIJ
  136. CSeq: 231 INVITE
  137. Server: Asterisk PBX 13.6.0
  138. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  139. Supported: replaces, timer
  140. Session-Expires: 1800;refresher=uas
  141. Contact: <sip:104@10.10.10.6:5060>
  142. Content-Length: 0
  143.  
  144.  
  145. <------------>
  146. -- Executing [104@tecnico:1] Answer("SIP/103-00000014", "") in new stack
  147. Audio is at 13426
  148. Adding codec g729 to SDP
  149. Adding codec alaw to SDP
  150. Adding codec gsm to SDP
  151. Adding non-codec 0x1 (telephone-event) to SDP
  152.  
  153. <--- Reliably Transmitting (no NAT) to 10.10.10.189:5060 --->
  154. SIP/2.0 200 OK
  155. Via: SIP/2.0/UDP 10.10.10.189:5060;branch=z9hG4bK165226484;received=10.10.10.189;rport=5060
  156. From: <sip:103@10.10.10.6>;tag=1134179135
  157. To: <sip:104@10.10.10.6>;tag=as530b3741
  158. Call-ID: 1844477604-5060-24@BA.BA.BA.BIJ
  159. CSeq: 231 INVITE
  160. Server: Asterisk PBX 13.6.0
  161. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  162. Supported: replaces, timer
  163. Session-Expires: 1800;refresher=uas
  164. Contact: <sip:104@10.10.10.6:5060>
  165. Content-Type: application/sdp
  166. Require: timer
  167. Content-Length: 315
  168.  
  169. v=0
  170. o=root 186851750 186851750 IN IP4 10.10.10.6
  171. s=Asterisk PBX 13.6.0
  172. c=IN IP4 10.10.10.6
  173. t=0 0
  174. m=audio 13426 RTP/AVP 18 8 3 101
  175. a=rtpmap:18 G729/8000
  176. a=fmtp:18 annexb=no
  177. a=rtpmap:8 PCMA/8000
  178. a=rtpmap:3 GSM/8000
  179. a=rtpmap:101 telephone-event/8000
  180. a=fmtp:101 0-16
  181. a=ptime:20
  182. a=maxptime:150
  183. a=sendrecv
  184.  
  185. <------------>
  186.  
  187. <--- SIP read from UDP:10.10.10.189:5060 --->
  188. ACK sip:104@10.10.10.6:5060 SIP/2.0
  189. Via: SIP/2.0/UDP 10.10.10.189:5060;branch=z9hG4bK559882522;rport
  190. From: <sip:103@10.10.10.6>;tag=1134179135
  191. To: <sip:104@10.10.10.6>;tag=as530b3741
  192. Call-ID: 1844477604-5060-24@BA.BA.BA.BIJ
  193. CSeq: 231 ACK
  194. Contact: <sip:103@10.10.10.189:5060>
  195. Max-Forwards: 70
  196. Supported: replaces, path, timer
  197. User-Agent: Grandstream GXP1625 1.0.2.4
  198. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  199. Content-Length: 0
  200.  
  201. <------------->
  202. --- (12 headers 0 lines) ---
  203. > 0x7f20780142f0 -- Probation passed - setting RTP source address to 10.10.10.189:5004
  204. -- Executing [104@tecnico:2] Wait("SIP/103-00000014", "2") in new stack
  205. -- Executing [104@tecnico:3] Playback("SIP/103-00000014", "conf-now-recording") in new stack
  206. -- <SIP/103-00000014> Playing 'conf-now-recording.g729' (language 'es')
  207. Really destroying SIP dialog '2W1waAaRbXA8kO6pDTHymnB.yKMD25tF' Method: REGISTER
  208. -- Executing [104@tecnico:4] Monitor("SIP/103-00000014", "wav,/recordings/2015/11/25/,m") in new stack
  209. -- Executing [104@tecnico:5] Dial("SIP/103-00000014", "SIP/104,10,mxXwW") in new stack
  210. == Using SIP RTP CoS mark 5
  211. Audio is at 17482
  212. Adding codec g729 to SDP
  213. Adding codec alaw to SDP
  214. Adding codec gsm to SDP
  215. Adding non-codec 0x1 (telephone-event) to SDP
  216. Reliably Transmitting (no NAT) to 10.10.10.190:5060:
  217. INVITE sip:104@10.10.10.190:5060 SIP/2.0
  218. Via: SIP/2.0/UDP 10.10.10.6:5060;branch=z9hG4bK62bf9025
  219. Max-Forwards: 70
  220. From: "103" <sip:103@10.10.10.6>;tag=as5ffded2f
  221. To: <sip:104@10.10.10.190:5060>
  222. Contact: <sip:103@10.10.10.6:5060>
  223. Call-ID: 4fafeafd24fbb164250db2456d005d0e@10.10.10.6:5060
  224. CSeq: 102 INVITE
  225. User-Agent: Asterisk PBX 13.6.0
  226. Date: Wed, 25 Nov 2015 20:05:43 GMT
  227. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  228. Supported: replaces, timer
  229. Content-Type: application/sdp
  230. Content-Length: 315
  231.  
  232. v=0
  233. o=root 718887876 718887876 IN IP4 10.10.10.6
  234. s=Asterisk PBX 13.6.0
  235. c=IN IP4 10.10.10.6
  236. t=0 0
  237. m=audio 17482 RTP/AVP 18 8 3 101
  238. a=rtpmap:18 G729/8000
  239. a=fmtp:18 annexb=no
  240. a=rtpmap:8 PCMA/8000
  241. a=rtpmap:3 GSM/8000
  242. a=rtpmap:101 telephone-event/8000
  243. a=fmtp:101 0-16
  244. a=ptime:20
  245. a=maxptime:150
  246. a=sendrecv
  247.  
  248. ---
  249. -- Called SIP/104
  250. -- Started music on hold, class 'default', on channel 'SIP/103-00000014'
  251.  
  252. <--- SIP read from UDP:10.10.10.190:5060 --->
  253. SIP/2.0 100 Trying
  254. Via: SIP/2.0/UDP 10.10.10.6:5060;branch=z9hG4bK62bf9025
  255. From: "103" <sip:103@10.10.10.6>;tag=as5ffded2f
  256. To: <sip:104@10.10.10.190:5060>
  257. Call-ID: 4fafeafd24fbb164250db2456d005d0e@10.10.10.6:5060
  258. CSeq: 102 INVITE
  259. Supported: replaces, path, timer
  260. User-Agent: Grandstream GXP1625 1.0.2.4
  261. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  262. Content-Length: 0
  263.  
  264. <------------->
  265. --- (10 headers 0 lines) ---
  266.  
  267. <--- SIP read from UDP:10.10.10.190:5060 --->
  268. SIP/2.0 180 Ringing
  269. Via: SIP/2.0/UDP 10.10.10.6:5060;branch=z9hG4bK62bf9025
  270. From: "103" <sip:103@10.10.10.6>;tag=as5ffded2f
  271. To: <sip:104@10.10.10.190:5060>;tag=1355083839
  272. Call-ID: 4fafeafd24fbb164250db2456d005d0e@10.10.10.6:5060
  273. CSeq: 102 INVITE
  274. Contact: <sip:104@10.10.10.190:5060>
  275. Supported: replaces, path, timer
  276. User-Agent: Grandstream GXP1625 1.0.2.4
  277. Allow-Events: talk, hold
  278. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  279. Content-Length: 0
  280.  
  281. <------------->
  282. --- (12 headers 0 lines) ---
  283. sip_route_dump: route/path hop: <sip:104@10.10.10.190:5060>
  284. -- SIP/104-00000015 is ringing
  285.  
  286. <--- SIP read from UDP:10.10.10.190:5060 --->
  287. SIP/2.0 200 OK
  288. Via: SIP/2.0/UDP 10.10.10.6:5060;branch=z9hG4bK62bf9025
  289. From: "103" <sip:103@10.10.10.6>;tag=as5ffded2f
  290. To: <sip:104@10.10.10.190:5060>;tag=1355083839
  291. Call-ID: 4fafeafd24fbb164250db2456d005d0e@10.10.10.6:5060
  292. CSeq: 102 INVITE
  293. Contact: <sip:104@10.10.10.190:5060>
  294. Supported: replaces, path, timer
  295. User-Agent: Grandstream GXP1625 1.0.2.4
  296. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  297. Content-Type: application/sdp
  298. Content-Length: 257
  299.  
  300. v=0
  301. o=104 8000 8000 IN IP4 10.10.10.190
  302. s=SIP Call
  303. c=IN IP4 10.10.10.190
  304. t=0 0
  305. m=audio 5004 RTP/AVP 18 8 101
  306. a=sendrecv
  307. a=rtpmap:18 G729/8000
  308. a=fmtp:18 annexb=no
  309. a=ptime:20
  310. a=rtpmap:8 PCMA/8000
  311. a=rtpmap:101 telephone-event/8000
  312. a=fmtp:101 0-15
  313. <------------->
  314. --- (12 headers 13 lines) ---
  315. Found RTP audio format 18
  316. Found RTP audio format 8
  317. Found RTP audio format 101
  318. Found audio description format G729 for ID 18
  319. Found audio description format PCMA for ID 8
  320. Found audio description format telephone-event for ID 101
  321. Capabilities: us - (g729|alaw|gsm), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (g729|alaw)
  322. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  323. Peer audio RTP is at port 10.10.10.190:5004
  324. sip_route_dump: route/path hop: <sip:104@10.10.10.190:5060>
  325. set_destination: Parsing <sip:104@10.10.10.190:5060> for address/port to send to
  326. set_destination: set destination to 10.10.10.190:5060
  327. Transmitting (no NAT) to 10.10.10.190:5060:
  328. ACK sip:104@10.10.10.190:5060 SIP/2.0
  329. Via: SIP/2.0/UDP 10.10.10.6:5060;branch=z9hG4bK79884ec9
  330. Max-Forwards: 70
  331. From: "103" <sip:103@10.10.10.6>;tag=as5ffded2f
  332. To: <sip:104@10.10.10.190:5060>;tag=1355083839
  333. Contact: <sip:103@10.10.10.6:5060>
  334. Call-ID: 4fafeafd24fbb164250db2456d005d0e@10.10.10.6:5060
  335. CSeq: 102 ACK
  336. User-Agent: Asterisk PBX 13.6.0
  337. Content-Length: 0
  338.  
  339.  
  340. ---
  341. -- SIP/104-00000015 answered SIP/103-00000014
  342. -- Stopped music on hold on SIP/103-00000014
  343. -- Channel SIP/104-00000015 joined 'simple_bridge' basic-bridge <1b41ec96-19cc-4963-90d6-e18c38425965>
  344. -- Channel SIP/103-00000014 joined 'simple_bridge' basic-bridge <1b41ec96-19cc-4963-90d6-e18c38425965>
  345. > 0x7f206c2cec70 -- Probation passed - setting RTP source address to 10.10.10.190:5004
  346. -- Channel SIP/104-00000015 left 'simple_bridge' basic-bridge <1b41ec96-19cc-4963-90d6-e18c38425965>
  347. Scheduling destruction of SIP dialog '4fafeafd24fbb164250db2456d005d0e@10.10.10.6:5060' in 6400 ms (Method: INVITE)
  348. set_destination: Parsing <sip:104@10.10.10.190:5060> for address/port to send to
  349. set_destination: set destination to 10.10.10.190:5060
  350. Reliably Transmitting (no NAT) to 10.10.10.190:5060:
  351. BYE sip:104@10.10.10.190:5060 SIP/2.0
  352. Via: SIP/2.0/UDP 10.10.10.6:5060;branch=z9hG4bK1aa8620c
  353. Max-Forwards: 70
  354. From: "103" <sip:103@10.10.10.6>;tag=as5ffded2f
  355. To: <sip:104@10.10.10.190:5060>;tag=1355083839
  356. Call-ID: 4fafeafd24fbb164250db2456d005d0e@10.10.10.6:5060
  357. CSeq: 103 BYE
  358. User-Agent: Asterisk PBX 13.6.0
  359. X-Asterisk-HangupCause: Normal Clearing
  360. X-Asterisk-HangupCauseCode: 16
  361. Content-Length: 0
  362.  
  363.  
  364. ---
  365. -- Channel SIP/103-00000014 left 'simple_bridge' basic-bridge <1b41ec96-19cc-4963-90d6-e18c38425965>
  366. == Spawn extension (tecnico, 104, 5) exited non-zero on 'SIP/103-00000014'
  367. Scheduling destruction of SIP dialog '1844477604-5060-24@BA.BA.BA.BIJ' in 6400 ms (Method: ACK)
  368. set_destination: Parsing <sip:103@10.10.10.189:5060> for address/port to send to
  369. set_destination: set destination to 10.10.10.189:5060
  370. Reliably Transmitting (no NAT) to 10.10.10.189:5060:
  371. BYE sip:103@10.10.10.189:5060 SIP/2.0
  372. Via: SIP/2.0/UDP 10.10.10.6:5060;branch=z9hG4bK56fe7293;rport
  373. Max-Forwards: 70
  374. From: <sip:104@10.10.10.6>;tag=as530b3741
  375. To: <sip:103@10.10.10.6>;tag=1134179135
  376. Call-ID: 1844477604-5060-24@BA.BA.BA.BIJ
  377. CSeq: 102 BYE
  378. User-Agent: Asterisk PBX 13.6.0
  379. Proxy-Authorization: Digest username="103", realm="asterisk", algorithm=MD5, uri="sip:10.10.10.6", nonce="37eb957f", response="ae42fde7aeaf63d523611ca46194b061"
  380. X-Asterisk-HangupCause: Normal Clearing
  381. X-Asterisk-HangupCauseCode: 16
  382. Content-Length: 0
  383.  
  384.  
  385. ---
  386.  
  387. <--- SIP read from UDP:10.10.10.190:5060 --->
  388. SIP/2.0 200 OK
  389. Via: SIP/2.0/UDP 10.10.10.6:5060;branch=z9hG4bK1aa8620c
  390. From: "103" <sip:103@10.10.10.6>;tag=as5ffded2f
  391. To: <sip:104@10.10.10.190:5060>;tag=1355083839
  392. Call-ID: 4fafeafd24fbb164250db2456d005d0e@10.10.10.6:5060
  393. CSeq: 103 BYE
  394. Contact: <sip:104@10.10.10.190:5060>
  395. Supported: replaces, path, timer
  396. User-Agent: Grandstream GXP1625 1.0.2.4
  397. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  398. Content-Length: 0
  399.  
  400. <------------->
  401. --- (11 headers 0 lines) ---
  402. Really destroying SIP dialog '4fafeafd24fbb164250db2456d005d0e@10.10.10.6:5060' Method: INVITE
  403.  
  404. <--- SIP read from UDP:10.10.10.189:5060 --->
  405. SIP/2.0 200 OK
  406. Via: SIP/2.0/UDP 10.10.10.6:5060;branch=z9hG4bK56fe7293;rport=5060
  407. From: <sip:104@10.10.10.6>;tag=as530b3741
  408. To: <sip:103@10.10.10.6>;tag=1134179135
  409. Call-ID: 1844477604-5060-24@BA.BA.BA.BIJ
  410. CSeq: 102 BYE
  411. Contact: <sip:103@10.10.10.189:5060>
  412. Supported: replaces, path, timer
  413. User-Agent: Grandstream GXP1625 1.0.2.4
  414. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
  415. Content-Length: 0
  416.  
  417. <------------->
  418. --- (11 headers 0 lines) ---
  419. SIP Response message for INCOMING dialog BYE arrived
  420. Really destroying SIP dialog '1844477604-5060-24@BA.BA.BA.BIJ' Method: ACK
  421. Really destroying SIP dialog 'KldCIW3z.xT7QmieTGhhpDlHVHOHw6gY' Method: REGISTER
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