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Aug 30th, 2018
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  1. <--- SIP read from UDP:333.333.333.333:49904 --->
  2. INVITE sip:85555555555@111.111.111.111 SIP/2.0
  3. Via: SIP/2.0/UDP 333.333.333.333:49904;rport;branch=z9hG4bKPj246719f13412480198a8a2b2a41e8d33
  4. Max-Forwards: 70
  5. From: <sip:front4@111.111.111.111>;tag=b6efb37fc42b46869da2ca9410e27a49
  6. To: <sip:85555555555@111.111.111.111>
  7. Contact: <sip:front4@333.333.333.333:49904;ob>
  8. Call-ID: 92691abb058c4044886dcf8b793c3c7d
  9. CSeq: 31552 INVITE
  10. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  11. Supported: replaces, 100rel, timer, norefersub
  12. Session-Expires: 1800
  13. Min-SE: 90
  14. User-Agent: MicroSIP/3.16.9
  15. Content-Type: application/sdp
  16. Content-Length: 344
  17.  
  18. v=0
  19. o=- 3744628894 3744628894 IN IP4 333.333.333.333
  20. s=pjmedia
  21. b=AS:84
  22. t=0 0
  23. a=X-nat:0
  24. m=audio 4004 RTP/AVP 18 8 0 101
  25. c=IN IP4 333.333.333.333
  26. b=TIAS:64000
  27. a=rtcp:4005 IN IP4 192.168.2.9
  28. a=sendrecv
  29. a=rtpmap:18 G729/8000
  30. a=fmtp:18 annexb=no
  31. a=rtpmap:8 PCMA/8000
  32. a=rtpmap:0 PCMU/8000
  33. a=rtpmap:101 telephone-event/8000
  34. a=fmtp:101 0-16
  35. <------------->
  36. --- (15 headers 17 lines) ---
  37. Sending to 333.333.333.333:49904 (NAT)
  38. Using INVITE request as basis request - 92691abb058c4044886dcf8b793c3c7d
  39. Found peer 'front4' for 'front4' from 333.333.333.333:49904
  40.  
  41. <--- Reliably Transmitting (NAT) to 333.333.333.333:49904 --->
  42. SIP/2.0 401 Unauthorized
  43. Via: SIP/2.0/UDP 333.333.333.333:49904;branch=z9hG4bKPj246719f13412480198a8a2b2a41e8d33;received=333.333.333.333;rport=49904
  44. From: <sip:front4@111.111.111.111>;tag=b6efb37fc42b46869da2ca9410e27a49
  45. To: <sip:85555555555@111.111.111.111>;tag=as61bf7440
  46. Call-ID: 92691abb058c4044886dcf8b793c3c7d
  47. CSeq: 31552 INVITE
  48. Server: Asterisk PBX 1.8.32.3
  49. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  50. Supported: replaces, timer
  51. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58c4bfcc"
  52. Content-Length: 0
  53.  
  54.  
  55. <------------>
  56. Scheduling destruction of SIP dialog '92691abb058c4044886dcf8b793c3c7d' in 6400 ms (Method: INVITE)
  57.  
  58. <--- SIP read from UDP:333.333.333.333:49904 --->
  59. ACK sip:85555555555@111.111.111.111 SIP/2.0
  60. Via: SIP/2.0/UDP 333.333.333.333:49904;rport;branch=z9hG4bKPj246719f13412480198a8a2b2a41e8d33
  61. Max-Forwards: 70
  62. From: <sip:front4@111.111.111.111>;tag=b6efb37fc42b46869da2ca9410e27a49
  63. To: <sip:85555555555@111.111.111.111>;tag=as61bf7440
  64. Call-ID: 92691abb058c4044886dcf8b793c3c7d
  65. CSeq: 31552 ACK
  66. Content-Length: 0
  67.  
  68. <------------->
  69. --- (8 headers 0 lines) ---
  70.  
  71. <--- SIP read from UDP:333.333.333.333:49904 --->
  72. INVITE sip:85555555555@111.111.111.111 SIP/2.0
  73. Via: SIP/2.0/UDP 333.333.333.333:49904;rport;branch=z9hG4bKPj1ad095402ea04ab497b0d5db060ce078
  74. Max-Forwards: 70
  75. From: <sip:front4@111.111.111.111>;tag=b6efb37fc42b46869da2ca9410e27a49
  76. To: <sip:85555555555@111.111.111.111>
  77. Contact: <sip:front4@333.333.333.333:49904;ob>
  78. Call-ID: 92691abb058c4044886dcf8b793c3c7d
  79. CSeq: 31553 INVITE
  80. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  81. Supported: replaces, 100rel, timer, norefersub
  82. Session-Expires: 1800
  83. Min-SE: 90
  84. User-Agent: MicroSIP/3.16.9
  85. Authorization: Digest username="front4", realm="asterisk", nonce="58c4bfcc", uri="sip:85555555555@111.111.111.111", response="fac71c9a48a85119f30170293dd24ba1", algorithm=MD5
  86. Content-Type: application/sdp
  87. Content-Length: 344
  88.  
  89. v=0
  90. o=- 3744628894 3744628894 IN IP4 333.333.333.333
  91. s=pjmedia
  92. b=AS:84
  93. t=0 0
  94. a=X-nat:0
  95. m=audio 4004 RTP/AVP 18 8 0 101
  96. c=IN IP4 333.333.333.333
  97. b=TIAS:64000
  98. a=rtcp:4005 IN IP4 192.168.2.9
  99. a=sendrecv
  100. a=rtpmap:18 G729/8000
  101. a=fmtp:18 annexb=no
  102. a=rtpmap:8 PCMA/8000
  103. a=rtpmap:0 PCMU/8000
  104. a=rtpmap:101 telephone-event/8000
  105. a=fmtp:101 0-16
  106. <------------->
  107. --- (16 headers 17 lines) ---
  108. Sending to 333.333.333.333:49904 (NAT)
  109. Using INVITE request as basis request - 92691abb058c4044886dcf8b793c3c7d
  110. Found peer 'front4' for 'front4' from 333.333.333.333:49904
  111. == Using SIP RTP CoS mark 5
  112. Found RTP audio format 18
  113. Found RTP audio format 8
  114. Found RTP audio format 0
  115. Found RTP audio format 101
  116. Found audio description format G729 for ID 18
  117. Found audio description format PCMA for ID 8
  118. Found audio description format PCMU for ID 0
  119. Found audio description format telephone-event for ID 101
  120. Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  121. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  122. Peer audio RTP is at port 333.333.333.333:4004
  123. Looking for 85555555555 in outgoing-calls (domain 111.111.111.111)
  124. list_route: hop: <sip:front4@333.333.333.333:49904;ob>
  125.  
  126. <--- Transmitting (NAT) to 333.333.333.333:49904 --->
  127. SIP/2.0 100 Trying
  128. Via: SIP/2.0/UDP 333.333.333.333:49904;branch=z9hG4bKPj1ad095402ea04ab497b0d5db060ce078;received=333.333.333.333;rport=49904
  129. From: <sip:front4@111.111.111.111>;tag=b6efb37fc42b46869da2ca9410e27a49
  130. To: <sip:85555555555@111.111.111.111>
  131. Call-ID: 92691abb058c4044886dcf8b793c3c7d
  132. CSeq: 31553 INVITE
  133. Server: Asterisk PBX 1.8.32.3
  134. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  135. Supported: replaces, timer
  136. Session-Expires: 1800;refresher=uas
  137. Contact: <sip:85555555555@111.111.111.111:5060>
  138. Content-Length: 0
  139.  
  140.  
  141. <------------>
  142. -- Executing [85555555555@outgoing-calls:1] SetCallerPres("SIP/front4-000083ed", "allowed") in new stack
  143. -- Executing [85555555555@outgoing-calls:2] Set("SIP/front4-000083ed", "CALLERID(ANI)=4444444444") in new stack
  144. -- Executing [85555555555@outgoing-calls:3] Set("SIP/front4-000083ed", "CALLERID(name)=4444444444") in new stack
  145. -- Executing [85555555555@outgoing-calls:4] Set("SIP/front4-000083ed", "CALLERID(number)=4444444444") in new stack
  146. -- Executing [85555555555@outgoing-calls:5] Originate("SIP/front4-000083ed", "SIP/5555555555@vitel-outbound,app,Playback,pinlessivr_success2") in new stack
  147. == Using SIP RTP CoS mark 5
  148. We think we can do text
  149. Audio is at 10570
  150. Adding codec 0x40 (slin) to SDP
  151. Adding codec 0x2 (gsm) to SDP
  152. Adding codec 0x4 (ulaw) to SDP
  153. Adding codec 0x8 (alaw) to SDP
  154. Adding codec 0x20 (adpcm) to SDP
  155. Adding codec 0x80 (lpc10) to SDP
  156. Adding codec 0x200 (speex) to SDP
  157. Adding codec 0x1000 (g722) to SDP
  158. Adding codec 0x8000 (slin16) to SDP
  159. Adding codec 0x200000000 (speex16) to SDP
  160. Adding codec 0x800000000000 (testlaw) to SDP
  161. Adding non-codec 0x1 (telephone-event) to SDP
  162. Reliably Transmitting (NAT) to 222.222.222.222:5060:
  163. INVITE sip:5555555555@222.222.222.222 SIP/2.0
  164. Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK214d55da;rport
  165. Max-Forwards: 70
  166. From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as5ffc73c9
  167. To: <sip:5555555555@222.222.222.222>
  168. Contact: <sip:anonymous@111.111.111.111:5060>
  169. Call-ID: 554599221c719afc735b6d900405c971@111.111.111.111:5060
  170. CSeq: 102 INVITE
  171. User-Agent: Asterisk PBX 1.8.32.3
  172. Date: Thu, 30 Aug 2018 18:41:35 GMT
  173. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  174. Supported: replaces, timer
  175. Content-Type: application/sdp
  176. Content-Length: 467
  177.  
  178. v=0
  179. o=root 1887068487 1887068487 IN IP4 111.111.111.111
  180. s=Asterisk PBX 1.8.32.3
  181. c=IN IP4 111.111.111.111
  182. t=0 0
  183. m=audio 10570 RTP/AVP 10 3 0 8 5 7 110 9 118 117 101
  184. a=rtpmap:10 L16/8000
  185. a=rtpmap:3 GSM/8000
  186. a=rtpmap:0 PCMU/8000
  187. a=rtpmap:8 PCMA/8000
  188. a=rtpmap:5 DVI4/8000
  189. a=rtpmap:7 LPC/8000
  190. a=rtpmap:110 speex/8000
  191. a=rtpmap:9 G722/8000
  192. a=rtpmap:118 L16/16000
  193. a=rtpmap:117 speex/16000
  194. a=rtpmap:101 telephone-event/8000
  195. a=fmtp:101 0-16
  196. a=ptime:20
  197. a=sendrecv
  198.  
  199. ---
  200.  
  201. <--- SIP read from UDP:222.222.222.222:5060 --->
  202. SIP/2.0 100 Trying
  203. Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK214d55da;rport=5060
  204. From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as5ffc73c9
  205. To: <sip:5555555555@222.222.222.222>
  206. Call-ID: 554599221c719afc735b6d900405c971@111.111.111.111:5060
  207. CSeq: 102 INVITE
  208. Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
  209. Content-Length: 0
  210.  
  211. <------------->
  212. --- (8 headers 0 lines) ---
  213.  
  214. <--- SIP read from UDP:222.222.222.222:5060 --->
  215. SIP/2.0 180 Ringing
  216. Via: SIP/2.0/UDP 111.111.111.111:5060;received=111.111.111.111;branch=z9hG4bK214d55da;rport=5060
  217. Record-Route: <sip:222.222.222.222;lr=on>
  218. From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as5ffc73c9
  219. To: <sip:5555555555@222.222.222.222>;tag=as2f3adb42
  220. Call-ID: 554599221c719afc735b6d900405c971@111.111.111.111:5060
  221. CSeq: 102 INVITE
  222. User-Agent: packetrino
  223. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  224. Supported: replaces
  225. Contact: <sip:15555555555@66.241.97.161>
  226. Content-Length: 0
  227.  
  228. <------------->
  229. --- (12 headers 0 lines) ---
  230. list_route: hop: <sip:222.222.222.222;lr=on>
  231.  
  232. <--- SIP read from UDP:222.222.222.222:5060 --->
  233. SIP/2.0 183 Session Progress
  234. Via: SIP/2.0/UDP 111.111.111.111:5060;received=111.111.111.111;branch=z9hG4bK214d55da;rport=5060
  235. Record-Route: <sip:222.222.222.222;lr=on>
  236. From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as5ffc73c9
  237. To: <sip:5555555555@222.222.222.222>;tag=as2f3adb42
  238. Call-ID: 554599221c719afc735b6d900405c971@111.111.111.111:5060
  239. CSeq: 102 INVITE
  240. User-Agent: packetrino
  241. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  242. Supported: replaces
  243. Contact: <sip:15555555555@66.241.97.161>
  244. Content-Type: application/sdp
  245. Content-Length: 265
  246.  
  247. v=0
  248. o=root 32006 32006 IN IP4 66.241.97.161
  249. s=session
  250. c=IN IP4 66.241.97.161
  251. t=0 0
  252. m=audio 19984 RTP/AVP 0 3 101
  253. a=rtpmap:0 PCMU/8000
  254. a=rtpmap:3 GSM/8000
  255. a=rtpmap:101 telephone-event/8000
  256. a=fmtp:101 0-16
  257. a=silenceSupp:off - - - -
  258. a=ptime:20
  259. a=sendrecv
  260. <------------->
  261. --- (13 headers 13 lines) ---
  262. list_route: hop: <sip:222.222.222.222;lr=on>
  263. Found RTP audio format 0
  264. Found RTP audio format 3
  265. Found RTP audio format 101
  266. Found audio description format PCMU for ID 0
  267. Found audio description format GSM for ID 3
  268. Found audio description format telephone-event for ID 101
  269. Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x6 (gsm|ulaw)
  270. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  271. Peer audio RTP is at port 66.241.97.161:19984
  272.  
  273.  
  274. <--- SIP read from UDP:333.333.333.333:49904 --->
  275.  
  276. <------------->
  277. Reliably Transmitting (NAT) to 333.333.333.333:49904:
  278. OPTIONS sip:front4@333.333.333.333:49904;ob SIP/2.0
  279. Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK7c0b8d5d;rport
  280. Max-Forwards: 70
  281. From: "asterisk" <sip:asterisk@111.111.111.111>;tag=as1109b5b6
  282. To: <sip:front4@333.333.333.333:49904;ob>
  283. Contact: <sip:asterisk@111.111.111.111:5060>
  284. Call-ID: 52d8d04d498d9f3d618fa4440e8cb797@111.111.111.111:5060
  285. CSeq: 102 OPTIONS
  286. User-Agent: Asterisk PBX 1.8.32.3
  287. Date: Thu, 30 Aug 2018 18:42:01 GMT
  288. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  289. Supported: replaces, timer
  290. Content-Length: 0
  291.  
  292.  
  293. ---
  294.  
  295. <--- SIP read from UDP:333.333.333.333:49904 --->
  296. SIP/2.0 200 OK
  297. Via: SIP/2.0/UDP 111.111.111.111:5060;rport=5060;received=111.111.111.111;branch=z9hG4bK7c0b8d5d
  298. Call-ID: 52d8d04d498d9f3d618fa4440e8cb797@111.111.111.111:5060
  299. From: "asterisk" <sip:asterisk@111.111.111.111>;tag=as1109b5b6
  300. To: <sip:front4@192.168.2.9;ob>;tag=z9hG4bK7c0b8d5d
  301. CSeq: 102 OPTIONS
  302. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  303. Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
  304. Supported: replaces, 100rel, timer, norefersub
  305. Allow-Events: presence, message-summary, refer
  306. User-Agent: MicroSIP/3.16.9
  307. Content-Length: 0
  308.  
  309. <------------->
  310. --- (12 headers 0 lines) ---
  311. Really destroying SIP dialog '52d8d04d498d9f3d618fa4440e8cb797@111.111.111.111:5060' Method: OPTIONS
  312.  
  313.  
  314. Scheduling destruction of SIP dialog '554599221c719afc735b6d900405c971@111.111.111.111:5060' in 32000 ms (Method: INVITE)
  315. Reliably Transmitting (NAT) to 222.222.222.222:5060:
  316. CANCEL sip:5555555555@222.222.222.222 SIP/2.0
  317. Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK214d55da;rport
  318. Max-Forwards: 70
  319. From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as5ffc73c9
  320. To: <sip:5555555555@222.222.222.222>
  321. Call-ID: 554599221c719afc735b6d900405c971@111.111.111.111:5060
  322. CSeq: 102 CANCEL
  323. User-Agent: Asterisk PBX 1.8.32.3
  324. Content-Length: 0
  325.  
  326.  
  327. ---
  328. Scheduling destruction of SIP dialog '554599221c719afc735b6d900405c971@111.111.111.111:5060' in 32000 ms (Method: INVITE)
  329. -- Executing [85555555555@outgoing-calls:6] Hangup("SIP/front4-000083ed", "") in new stack
  330. == Spawn extension (outgoing-calls, 85555555555, 6) exited non-zero on 'SIP/front4-000083ed'
  331. Scheduling destruction of SIP dialog '92691abb058c4044886dcf8b793c3c7d' in 6400 ms (Method: INVITE)
  332.  
  333. <--- Reliably Transmitting (NAT) to 333.333.333.333:49904 --->
  334. SIP/2.0 603 Declined
  335. Via: SIP/2.0/UDP 333.333.333.333:49904;branch=z9hG4bKPj1ad095402ea04ab497b0d5db060ce078;received=333.333.333.333;rport=49904
  336. From: <sip:front4@111.111.111.111>;tag=b6efb37fc42b46869da2ca9410e27a49
  337. To: <sip:85555555555@111.111.111.111>;tag=as259341aa
  338. Call-ID: 92691abb058c4044886dcf8b793c3c7d
  339. CSeq: 31553 INVITE
  340. Server: Asterisk PBX 1.8.32.3
  341. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  342. Supported: replaces, timer
  343. Session-Expires: 1800;refresher=uas
  344. Content-Length: 0
  345.  
  346.  
  347. <------------>
  348.  
  349. <--- SIP read from UDP:333.333.333.333:49904 --->
  350. ACK sip:85555555555@111.111.111.111 SIP/2.0
  351. Via: SIP/2.0/UDP 333.333.333.333:49904;rport;branch=z9hG4bKPj1ad095402ea04ab497b0d5db060ce078
  352. Max-Forwards: 70
  353. From: <sip:front4@111.111.111.111>;tag=b6efb37fc42b46869da2ca9410e27a49
  354. To: <sip:85555555555@111.111.111.111>;tag=as259341aa
  355. Call-ID: 92691abb058c4044886dcf8b793c3c7d
  356. CSeq: 31553 ACK
  357. Content-Length: 0
  358.  
  359. <------------->
  360. --- (8 headers 0 lines) ---
  361.  
  362. <--- SIP read from UDP:222.222.222.222:5060 --->
  363. SIP/2.0 200 canceling
  364. Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK214d55da;rport=5060
  365. From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as5ffc73c9
  366. To: <sip:5555555555@222.222.222.222>;tag=d8ec4bef69e0ea2b5541b4b24b7b56ef-7dc2
  367. Call-ID: 554599221c719afc735b6d900405c971@111.111.111.111:5060
  368. CSeq: 102 CANCEL
  369. Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
  370. Content-Length: 0
  371.  
  372. <------------->
  373. --- (8 headers 0 lines) ---
  374.  
  375. <--- SIP read from UDP:222.222.222.222:5060 --->
  376. SIP/2.0 487 Request Terminated
  377. Via: SIP/2.0/UDP 111.111.111.111:5060;received=111.111.111.111;branch=z9hG4bK214d55da;rport=5060
  378. From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as5ffc73c9
  379. To: <sip:5555555555@222.222.222.222>;tag=as2f3adb42
  380. Call-ID: 554599221c719afc735b6d900405c971@111.111.111.111:5060
  381. CSeq: 102 INVITE
  382. User-Agent: packetrino
  383. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  384. Supported: replaces
  385. Content-Length: 0
  386.  
  387. <------------->
  388. --- (10 headers 0 lines) ---
  389. Transmitting (NAT) to 222.222.222.222:5060:
  390. ACK sip:15555555555@66.241.97.161 SIP/2.0
  391. Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK214d55da;rport
  392. Max-Forwards: 70
  393. From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as5ffc73c9
  394. To: <sip:5555555555@222.222.222.222>;tag=as2f3adb42
  395. Contact: <sip:anonymous@111.111.111.111:5060>
  396. Call-ID: 554599221c719afc735b6d900405c971@111.111.111.111:5060
  397. CSeq: 102 ACK
  398. User-Agent: Asterisk PBX 1.8.32.3
  399. Content-Length: 0
  400.  
  401.  
  402. ---
  403. Scheduling destruction of SIP dialog '554599221c719afc735b6d900405c971@111.111.111.111:5060' in 32000 ms (Method: INVITE)
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