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  1. <--- SIP read from UDP:85.114.2.202:5060 --->
  2. INVITE sip:6221762@91.142.85.146:5060 SIP/2.0
  3. Via: SIP/2.0/UDP 85.114.2.202;rport;branch=z9hG4bK1XKem6a18aN8H
  4. Max-Forwards: 68
  5. From: "8129488743" <sip:8129488743@85.114.2.202>;tag=XcvQ519Q6aB5N
  6. To: <sip:6221762@91.142.85.146:5060>
  7. Call-ID: 2a9c7627-8731-1235-cb94-005056b55727
  8. CSeq: 104623489 INVITE
  9. Contact: <sip:gateway@85.114.2.202:5060>
  10. User-Agent: Obit-GW-L
  11. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
  12. Supported: path, replaces
  13. Allow-Events: talk, hold, conference, refer
  14. Content-Type: application/sdp
  15. Content-Disposition: session
  16. Content-Length: 289
  17. X-Origin-Server: eltex-02-spb
  18. X-FS-Support: update_display,send_info
  19. Remote-Party-ID: "8129488743" <sip:8129488743@85.114.2.202>;party=calling;screen=yes;privacy=off
  20.  
  21. v=0
  22. o=Obit-GW-L 2866427133 2866427134 IN IP4 85.114.2.202
  23. s=Obit-GW-L
  24. c=IN IP4 85.114.2.202
  25. t=0 0
  26. m=audio 26370 RTP/AVP 8 0 18 101
  27. a=rtpmap:8 PCMA/8000
  28. a=rtpmap:0 PCMU/8000
  29. a=rtpmap:18 G729/8000
  30. a=fmtp:18 annexb=no
  31. a=rtpmap:101 telephone-event/8000
  32. a=fmtp:101 0-15
  33. a=ptime:20
  34. <------------->
  35. --- (18 headers 13 lines) ---
  36. Sending to 85.114.2.202:5060 (no NAT)
  37. Sending to 85.114.2.202:5060 (no NAT)
  38. Using INVITE request as basis request - 2a9c7627-8731-1235-cb94-005056b55727
  39. Found peer 'OBIT' for '8129488743' from 85.114.2.202:5060
  40. Found RTP audio format 8
  41. Found RTP audio format 0
  42. Found RTP audio format 18
  43. Found RTP audio format 101
  44. Found audio description format PCMA for ID 8
  45. Found audio description format PCMU for ID 0
  46. Found audio description format G729 for ID 18
  47. Found audio description format telephone-event for ID 101
  48. Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
  49. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  50. Peer audio RTP is at port 85.114.2.202:26370
  51. Looking for 6221762 in from_obit (domain 91.142.85.146)
  52. sip_route_dump: route/path hop: <sip:gateway@85.114.2.202:5060>
  53.  
  54. <--- Transmitting (NAT) to 85.114.2.202:5060 --->
  55. SIP/2.0 100 Trying
  56. Via: SIP/2.0/UDP 85.114.2.202;branch=z9hG4bK1XKem6a18aN8H;received=85.114.2.202;rport=5060
  57. From: "8129488743" <sip:8129488743@85.114.2.202>;tag=XcvQ519Q6aB5N
  58. To: <sip:6221762@91.142.85.146:5060>
  59. Call-ID: 2a9c7627-8731-1235-cb94-005056b55727
  60. CSeq: 104623489 INVITE
  61. Server: Asterisk
  62. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  63. Supported: replaces, timer
  64. Contact: <sip:6221762@91.142.85.146:5060>
  65. Content-Length: 0
  66.  
  67.  
  68. <------------>
  69. Audio is at 11702
  70. Adding codec alaw to SDP
  71. Adding codec ulaw to SDP
  72. Adding non-codec 0x1 (telephone-event) to SDP
  73. Reliably Transmitting (NAT) to 95.213.198.99:5060:
  74. INVITE sip:6221762@ip.b24-5436-1386089498.bitrixphone.com SIP/2.0
  75. Via: SIP/2.0/UDP 91.142.85.146:5060;branch=z9hG4bK53c01b46;rport
  76. Max-Forwards: 70
  77. From: "8129488743" <sip:sip10@ip.b24-5436-1386089498.bitrixphone.com>;tag=as4770a2d6
  78. To: <sip:6221762@ip.b24-5436-1386089498.bitrixphone.com>
  79. Contact: <sip:sip10@91.142.85.146:5060>
  80. Call-ID: 5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com
  81. CSeq: 102 INVITE
  82. User-Agent: Asterisk
  83. Date: Sun, 19 Mar 2017 10:25:07 GMT
  84. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  85. Supported: replaces, timer
  86. Content-Type: application/sdp
  87. Content-Length: 264
  88.  
  89. v=0
  90. o=root 579661157 579661157 IN IP4 91.142.85.146
  91. s=Asterisk
  92. c=IN IP4 91.142.85.146
  93. t=0 0
  94. m=audio 11702 RTP/AVP 8 0 101
  95. a=rtpmap:8 PCMA/8000
  96. a=rtpmap:0 PCMU/8000
  97. a=rtpmap:101 telephone-event/8000
  98. a=fmtp:101 0-16
  99. a=ptime:20
  100. a=maxptime:150
  101. a=sendrecv
  102.  
  103. ---
  104.  
  105. <--- SIP read from UDP:95.213.198.99:5060 --->
  106. SIP/2.0 100 Trying
  107. v: SIP/2.0/UDP 91.142.85.146:5060;rport=5060;received=91.142.85.146;branch=z9hG4bK53c01b46
  108. i: 5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com
  109. f: "8129488743" <sip:sip10@ip.b24-5436-1386089498.bitrixphone.com>;tag=as4770a2d6
  110. t: <sip:6221762@ip.b24-5436-1386089498.bitrixphone.com>
  111. CSeq: 102 INVITE
  112. l: 0
  113.  
  114. <------------->
  115. --- (7 headers 0 lines) ---
  116.  
  117. <--- SIP read from UDP:95.213.198.99:5060 --->
  118. SIP/2.0 407 Proxy Authentication Required
  119. v: SIP/2.0/UDP 91.142.85.146:5060;rport=5060;received=91.142.85.146;branch=z9hG4bK53c01b46
  120. i: 5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com
  121. f: "8129488743" <sip:sip10@ip.b24-5436-1386089498.bitrixphone.com>;tag=as4770a2d6
  122. t: <sip:6221762@ip.b24-5436-1386089498.bitrixphone.com>;tag=z9hG4bK53c01b46
  123. CSeq: 102 INVITE
  124. Proxy-Authenticate: Digest realm="voximplant.com",nonce="569bcccb2843bf6a",opaque="11b5586619d257ba",algorithm=md5
  125. l: 0
  126.  
  127. <------------->
  128. --- (8 headers 0 lines) ---
  129. Transmitting (NAT) to 95.213.198.99:5060:
  130. ACK sip:6221762@ip.b24-5436-1386089498.bitrixphone.com SIP/2.0
  131. Via: SIP/2.0/UDP 91.142.85.146:5060;branch=z9hG4bK53c01b46;rport
  132. Max-Forwards: 70
  133. From: "8129488743" <sip:sip10@ip.b24-5436-1386089498.bitrixphone.com>;tag=as4770a2d6
  134. To: <sip:6221762@ip.b24-5436-1386089498.bitrixphone.com>;tag=z9hG4bK53c01b46
  135. Contact: <sip:sip10@91.142.85.146:5060>
  136. Call-ID: 5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com
  137. CSeq: 102 ACK
  138. User-Agent: Asterisk
  139. Content-Length: 0
  140.  
  141. ---
  142. Audio is at 11702
  143. Adding codec alaw to SDP
  144. Adding codec ulaw to SDP
  145. Adding non-codec 0x1 (telephone-event) to SDP
  146. Reliably Transmitting (NAT) to 95.213.198.99:5060:
  147. INVITE sip:6221762@ip.b24-5436-1386089498.bitrixphone.com SIP/2.0
  148. Via: SIP/2.0/UDP 91.142.85.146:5060;branch=z9hG4bK61dd06e1;rport
  149. Max-Forwards: 70
  150. From: "8129488743" <sip:sip10@ip.b24-5436-1386089498.bitrixphone.com>;tag=as4770a2d6
  151. To: <sip:6221762@ip.b24-5436-1386089498.bitrixphone.com>
  152. Contact: <sip:sip10@91.142.85.146:5060>
  153. Call-ID: 5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com
  154. CSeq: 103 INVITE
  155. User-Agent: Asterisk
  156. Proxy-Authorization: Digest username="sip10", realm="voximplant.com", algorithm=MD5, uri="sip:6221762@ip.b24-5436-1386089498.bitrixphone.com", nonce="569bcccb2843bf6a", response="a8709cbd886cce77f81f80c1b9cafb8c", opaque="11b5586619d257ba"
  157. Date: Sun, 19 Mar 2017 10:25:07 GMT
  158. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  159. Supported: replaces, timer
  160. Content-Type: application/sdp
  161. Content-Length: 264
  162.  
  163. v=0
  164. o=root 579661157 579661158 IN IP4 91.142.85.146
  165. s=Asterisk
  166. c=IN IP4 91.142.85.146
  167. t=0 0
  168. m=audio 11702 RTP/AVP 8 0 101
  169. a=rtpmap:8 PCMA/8000
  170. a=rtpmap:0 PCMU/8000
  171. a=rtpmap:101 telephone-event/8000
  172. a=fmtp:101 0-16
  173. a=ptime:20
  174. a=maxptime:150
  175. a=sendrecv
  176.  
  177. ---
  178.  
  179. <--- SIP read from UDP:95.213.198.99:5060 --->
  180. SIP/2.0 100 Trying
  181. v: SIP/2.0/UDP 91.142.85.146:5060;rport=5060;received=91.142.85.146;branch=z9hG4bK61dd06e1
  182. i: 5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com
  183. f: "8129488743" <sip:sip10@ip.b24-5436-1386089498.bitrixphone.com>;tag=as4770a2d6
  184. t: <sip:6221762@ip.b24-5436-1386089498.bitrixphone.com>
  185. CSeq: 103 INVITE
  186. l: 0
  187.  
  188. <------------->
  189. --- (7 headers 0 lines) ---
  190.  
  191. <--- SIP read from UDP:95.213.198.99:5060 --->
  192. SIP/2.0 100 Trying
  193. v: SIP/2.0/UDP 91.142.85.146:5060;rport=5060;received=91.142.85.146;branch=z9hG4bK61dd06e1
  194. Record-Route: <sip:95.213.198.99:5060;transport=UDP;lr>
  195. i: 5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com
  196. f: "8129488743" <sip:sip10@ip.b24-5436-1386089498.bitrixphone.com>;tag=as4770a2d6
  197. t: <sip:6221762@ip.b24-5436-1386089498.bitrixphone.com>
  198. CSeq: 103 INVITE
  199. l: 0
  200.  
  201. <------------->
  202. --- (8 headers 0 lines) ---
  203.  
  204. <--- SIP read from UDP:95.213.198.99:5060 --->
  205. SIP/2.0 183 Session Progress
  206. v: SIP/2.0/UDP 91.142.85.146:5060;rport=5060;received=91.142.85.146;branch=z9hG4bK61dd06e1
  207. Record-Route: <sip:95.213.198.99:5060;transport=UDP;lr>
  208. i: 5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com
  209. f: "8129488743" <sip:sip10@ip.b24-5436-1386089498.bitrixphone.com>;tag=as4770a2d6
  210. t: <sip:6221762@ip.b24-5436-1386089498.bitrixphone.com>;tag=VVVVV95.213.221.99_5090_46c5c7ff74078ad2a380dee1c6b95e0a
  211. CSeq: 103 INVITE
  212. m: <sip:95.213.221.99:5090>
  213. Allow: INFO, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER
  214. c: application/sdp
  215. l: 251
  216.  
  217. v=0
  218. o=VIMS 234 3 IN IP4 95.213.221.99
  219. s=VIMS
  220. c=IN IP4 95.213.221.99
  221. b=TIAS:13952000
  222. t=0 0
  223. m=audio 14092 RTP/AVP 8 101
  224. b=TIAS:64000
  225. a=rtpmap:8 PCMA/8000
  226. a=rtpmap:101 telephone-event/8000
  227. a=fmtp:101 0-15
  228. a=sendrecv
  229. a=rtcp:14093
  230. a=ptime:20
  231. <------------->
  232. --- (11 headers 14 lines) ---
  233. sip_route_dump: route/path hop: <sip:95.213.198.99:5060;transport=UDP;lr>
  234. Found RTP audio format 8
  235. Found RTP audio format 101
  236. Found audio description format PCMA for ID 8
  237. Found audio description format telephone-event for ID 101
  238. Capabilities: us - (alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
  239. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  240. Peer audio RTP is at port 95.213.221.99:14092
  241.  
  242. <--- Transmitting (NAT) to 85.114.2.202:5060 --->
  243. SIP/2.0 180 Ringing
  244. Via: SIP/2.0/UDP 85.114.2.202;branch=z9hG4bK1XKem6a18aN8H;received=85.114.2.202;rport=5060
  245. From: "8129488743" <sip:8129488743@85.114.2.202>;tag=XcvQ519Q6aB5N
  246. To: <sip:6221762@91.142.85.146:5060>;tag=as74b6d22a
  247. Call-ID: 2a9c7627-8731-1235-cb94-005056b55727
  248. CSeq: 104623489 INVITE
  249. Server: Asterisk
  250. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  251. Supported: replaces, timer
  252. Contact: <sip:6221762@91.142.85.146:5060>
  253. Content-Length: 0
  254.  
  255.  
  256. <------------>
  257.  
  258. <--- SIP read from UDP:95.213.198.99:5060 --->
  259. SIP/2.0 200 OK
  260. v: SIP/2.0/UDP 91.142.85.146:5060;rport=5060;received=91.142.85.146;branch=z9hG4bK61dd06e1
  261. Record-Route: <sip:95.213.198.99:5060;transport=UDP;lr>
  262. i: 5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com
  263. f: "8129488743" <sip:sip10@ip.b24-5436-1386089498.bitrixphone.com>;tag=as4770a2d6
  264. t: <sip:6221762@ip.b24-5436-1386089498.bitrixphone.com>;tag=VVVVV95.213.221.99_5090_46c5c7ff74078ad2a380dee1c6b95e0a
  265. CSeq: 103 INVITE
  266. Allow: INFO, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER
  267. m: <sip:95.213.221.99:5090>
  268. k: replaces
  269. c: application/sdp
  270. l: 251
  271. v=0
  272. o=VIMS 234 3 IN IP4 95.213.221.99
  273. s=VIMS
  274. c=IN IP4 95.213.221.99
  275. b=TIAS:13952000
  276. t=0 0
  277. m=audio 14092 RTP/AVP 8 101
  278. b=TIAS:64000
  279. a=rtpmap:8 PCMA/8000
  280. a=rtpmap:101 telephone-event/8000
  281. a=fmtp:101 0-15
  282. a=sendrecv
  283. a=rtcp:14093
  284. a=ptime:20
  285. <------------->
  286. --- (12 headers 14 lines) ---
  287. sip_route_dump: route/path hop: <sip:95.213.198.99:5060;transport=UDP;lr>
  288. Transmitting (NAT) to 95.213.198.99:5060:
  289. ACK sip:95.213.221.99:5090 SIP/2.0
  290. Via: SIP/2.0/UDP 91.142.85.146:5060;branch=z9hG4bK749a64f5;rport
  291. Route: <sip:95.213.198.99:5060;transport=UDP;lr>
  292. Max-Forwards: 70
  293. From: "8129488743" <sip:sip10@ip.b24-5436-1386089498.bitrixphone.com>;tag=as4770a2d6
  294. To: <sip:6221762@ip.b24-5436-1386089498.bitrixphone.com>;tag=VVVVV95.213.221.99_5090_46c5c7ff74078ad2a380dee1c6b95e0a
  295. Contact: <sip:sip10@91.142.85.146:5060>
  296. Call-ID: 5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com
  297. CSeq: 103 ACK
  298. User-Agent: Asterisk
  299. Content-Length: 0
  300.  
  301.  
  302. ---
  303. Audio is at 17806
  304. Adding codec alaw to SDP
  305. Adding codec ulaw to SDP
  306. Adding non-codec 0x1 (telephone-event) to SDP
  307.  
  308. <--- Reliably Transmitting (NAT) to 85.114.2.202:5060 --->
  309. SIP/2.0 200 OK
  310. Via: SIP/2.0/UDP 85.114.2.202;branch=z9hG4bK1XKem6a18aN8H;received=85.114.2.202;rport=5060
  311. From: "8129488743" <sip:8129488743@85.114.2.202>;tag=XcvQ519Q6aB5N
  312. To: <sip:6221762@91.142.85.146:5060>;tag=as74b6d22a
  313. Call-ID: 2a9c7627-8731-1235-cb94-005056b55727
  314. CSeq: 104623489 INVITE
  315. Server: Asterisk
  316. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  317. Supported: replaces, timer
  318. Contact: <sip:6221762@91.142.85.146:5060>
  319. Content-Type: application/sdp
  320. Content-Length: 266
  321.  
  322. v=0
  323. o=root 1666991734 1666991734 IN IP4 91.142.85.146
  324. s=Asterisk
  325. c=IN IP4 91.142.85.146
  326. t=0 0
  327. m=audio 17806 RTP/AVP 8 0 101
  328. a=rtpmap:8 PCMA/8000
  329. a=rtpmap:0 PCMU/8000
  330. a=rtpmap:101 telephone-event/8000
  331. a=fmtp:101 0-16
  332. a=ptime:20
  333. a=maxptime:150
  334. a=sendrecv
  335.  
  336. <------------>
  337. <--- SIP read from UDP:85.114.2.202:5060 --->
  338. ACK sip:6221762@91.142.85.146:5060 SIP/2.0
  339. Via: SIP/2.0/UDP 85.114.2.202;rport;branch=z9hG4bK26c7N1U45KBUD
  340. Max-Forwards: 70
  341. From: "8129488743" <sip:8129488743@85.114.2.202>;tag=XcvQ519Q6aB5N
  342. To: <sip:6221762@91.142.85.146:5060>;tag=as74b6d22a
  343. Call-ID: 2a9c7627-8731-1235-cb94-005056b55727
  344. CSeq: 104623489 ACK
  345. Contact: <sip:gateway@85.114.2.202:5060>
  346. Content-Length: 0
  347.  
  348. <------------->
  349. --- (9 headers 0 lines) ---
  350.  
  351. <--- SIP read from UDP:85.114.2.202:5060 --->
  352. BYE sip:6221762@91.142.85.146:5060 SIP/2.0
  353. Via: SIP/2.0/UDP 85.114.2.202;rport;branch=z9hG4bK6ajaXDZjtQ45B
  354. Max-Forwards: 70
  355. From: "8129488743" <sip:8129488743@85.114.2.202>;tag=XcvQ519Q6aB5N
  356. To: <sip:6221762@91.142.85.146:5060>;tag=as74b6d22a
  357. Call-ID: 2a9c7627-8731-1235-cb94-005056b55727
  358. CSeq: 104623490 BYE
  359. User-Agent: Obit-GW-L
  360. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
  361. Supported: path, replaces
  362. Reason: Q.850;cause=16;text="Normal call clearing"
  363. Content-Length: 0
  364.  
  365. <------------->
  366. --- (12 headers 0 lines) ---
  367. Sending to 85.114.2.202:5060 (NAT)
  368. Scheduling destruction of SIP dialog '2a9c7627-8731-1235-cb94-005056b55727' in 32000 ms (Method: BYE)
  369.  
  370. <--- Transmitting (NAT) to 85.114.2.202:5060 --->
  371. SIP/2.0 200 OK
  372. Via: SIP/2.0/UDP 85.114.2.202;branch=z9hG4bK6ajaXDZjtQ45B;received=85.114.2.202;rport=5060
  373. From: "8129488743" <sip:8129488743@85.114.2.202>;tag=XcvQ519Q6aB5N
  374. To: <sip:6221762@91.142.85.146:5060>;tag=as74b6d22a
  375. Call-ID: 2a9c7627-8731-1235-cb94-005056b55727
  376. CSeq: 104623490 BYE
  377. Server: Asterisk
  378. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  379. Supported: replaces, timer
  380. Content-Length: 0
  381.  
  382.  
  383. <------------>
  384. Scheduling destruction of SIP dialog '5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com' in 32000 ms (Method: INVITE)
  385. Reliably Transmitting (NAT) to 95.213.198.99:5060:
  386. BYE sip:95.213.221.99:5090 SIP/2.0
  387. Via: SIP/2.0/UDP 91.142.85.146:5060;branch=z9hG4bK05f33c7e;rport
  388. Route: <sip:95.213.198.99:5060;transport=UDP;lr>
  389. Max-Forwards: 70
  390. From: "8129488743" <sip:sip10@ip.b24-5436-1386089498.bitrixphone.com>;tag=as4770a2d6
  391. To: <sip:6221762@ip.b24-5436-1386089498.bitrixphone.com>;tag=VVVVV95.213.221.99_5090_46c5c7ff74078ad2a380dee1c6b95e0a
  392. Call-ID: 5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com
  393. CSeq: 104 BYE
  394. User-Agent: Asterisk
  395. Proxy-Authorization: Digest username="sip10", realm="voximplant.com", algorithm=MD5, uri="sip:95.213.221.99:5090", nonce="569bcccb2843bf6a", response="3155ca5577828fe90e10ff72509344b9", opaque="11b5586619d257ba"
  396. X-Asterisk-HangupCause: Normal Clearing
  397. X-Asterisk-HangupCauseCode: 16
  398. Content-Length: 0
  399.  
  400.  
  401. ---
  402.  
  403. <--- SIP read from UDP:95.213.198.99:5060 --->
  404. SIP/2.0 200 OK
  405. v: SIP/2.0/UDP 91.142.85.146:5060;rport=5060;received=91.142.85.146;branch=z9hG4bK05f33c7e
  406. i: 5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com
  407. f: "8129488743" <sip:sip10@ip.b24-5436-1386089498.bitrixphone.com>;tag=as4770a2d6
  408. t: <sip:6221762@ip.b24-5436-1386089498.bitrixphone.com>;tag=VVVVV95.213.221.99_5090_46c5c7ff74078ad2a380dee1c6b95e0a
  409. CSeq: 104 BYE
  410. m: <sip:95.213.221.99:5090>
  411. l: 0
  412.  
  413. <------------->
  414. --- (8 headers 0 lines) ---
  415. Really destroying SIP dialog '5792beef377ca36447c3ecdc03adefc5@ip.b24-5436-1386089498.bitrixphone.com' Method: INVITE
  416. Really destroying SIP dialog '2a9c7627-8731-1235-cb94-005056b55727' Method: BYE
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