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Avaya-Asterisk Call Drop SIP Error Log

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  1. <--- SIP read from TCP:10.10.2.220:20823 --->
  2. INVITE sip:8929@10.10.2.12 SIP/2.0
  3. From: "Miguel Del Castillo Soft" <sip:anonymous@anonymous.invalid>;tag=806687127df8e216d7c517f69400
  4. To: "8929" <sip:8929@10.10.2.12>
  5. Call-ID: 806687127df8e216e7c517f69400
  6. CSeq: 1 INVITE
  7. Max-Forwards: 69
  8. Route: <sip:10.10.2.12;lr;phase=terminating;transport=tcp>
  9. Record-Route: <sip:10.10.2.220;lr;transport=tcp>
  10. Via: SIP/2.0/TCP 10.10.2.220;branch=z9hG4bK806687127df8e216f7c517f69400
  11. User-Agent: Avaya CM/R015x.01.1.415.1
  12. Supported: 100rel, timer, replaces, join, histinfo
  13. Allow: INVITE, CANCEL, BYE, ACK, PRACK, SUBSCRIBE, NOTIFY, REFER, OPTIONS, INFO, PUBLISH
  14. Contact: "Miguel Del Castillo Soft" <sip:10.10.2.220;transport=tcp>
  15. Session-Expires: 240;refresher=uac
  16. Min-SE: 240
  17. P-Asserted-Identity: "Miguel Del Castillo Soft" <sip:invalid.unknown.domain>
  18. Accept-Language: en
  19. Privacy: id
  20. Content-Type: application/sdp
  21. History-Info: <sip:8929@10.10.2.12>;index=1
  22. History-Info: "8929" <sip:8929@10.10.2.12>;index=1.1
  23. Alert-Info: <cid:internal@invalid.unknown.domain>;avaya-cm-alert-type=internal
  24. Content-Length: 209
  25.  
  26. v=0
  27. o=- 1 1 IN IP4 10.10.2.220
  28. s=-
  29. c=IN IP4 10.10.2.213
  30. b=AS:64
  31. t=0 0
  32. m=audio 2844 RTP/AVP 0 18 127
  33. a=rtpmap:0 PCMU/8000
  34. a=rtpmap:18 G729/8000
  35. a=fmtp:18 annexb=yes
  36. a=rtpmap:127 telephone-event/8000
  37. <------------->
  38. --- (23 headers 11 lines) ---
  39. Sending to 10.10.2.220:20823 (NAT)
  40. Sending to 10.10.2.220:20823 (NAT)
  41. Using INVITE request as basis request - 806687127df8e216e7c517f69400
  42. No matching peer for 'anonymous' from '10.10.2.220:20823'
  43. [2013-07-25 18:31:19] ERROR[10732][C-0000008e]: sip/reqresp_parser.c:830 get_name_and_number: can not parse name and number from sip header.
  44. Found RTP audio format 0
  45. Found RTP audio format 18
  46. Found RTP audio format 127
  47. Found audio description format PCMU for ID 0
  48. Found audio description format G729 for ID 18
  49. Found audio description format telephone-event for ID 127
  50. Capabilities: us - (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|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|g729)
  51. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  52. Peer audio RTP is at port 10.10.2.213:2844
  53. Looking for 8929 in default (domain 10.10.2.12)
  54. list_route: hop: <sip:10.10.2.220;lr;transport=tcp>
  55.  
  56. <--- Transmitting (NAT) to 10.10.2.220:20823 --->
  57. SIP/2.0 100 Trying
  58. Via: SIP/2.0/TCP 10.10.2.220;branch=z9hG4bK806687127df8e216f7c517f69400;received=10.10.2.220;rport=20823
  59. Record-Route: <sip:10.10.2.220;lr;transport=tcp>
  60. From: "Miguel Del Castillo Soft" <sip:anonymous@anonymous.invalid>;tag=806687127df8e216d7c517f69400
  61. To: "8929" <sip:8929@10.10.2.12>
  62. Call-ID: 806687127df8e216e7c517f69400
  63. CSeq: 1 INVITE
  64. Server: Asterisk PBX 11.5.0
  65. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  66. Supported: replaces, timer
  67. Session-Expires: 240;refresher=uac
  68. Contact: <sip:8929@10.10.2.12:5060;transport=TCP>
  69. Content-Length: 0
  70.  
  71.  
  72. <------------>
  73. We think we can do text
  74. Audio is at 15828
  75. Adding codec 100003 (ulaw) to SDP
  76. Adding codec 100001 (g723) to SDP
  77. Adding codec 100001 (g723) to SDP
  78. Adding codec 100002 (gsm) to SDP
  79. Adding codec 100002 (gsm) to SDP
  80. Adding codec 100002 (gsm) to SDP
  81. Adding codec 100004 (alaw) to SDP
  82. Adding codec 100004 (alaw) to SDP
  83. Adding codec 100004 (alaw) to SDP
  84. Adding codec 100005 (g726aal2) to SDP
  85. Adding codec 100005 (g726aal2) to SDP
  86. Adding codec 100006 (adpcm) to SDP
  87. Adding codec 100006 (adpcm) to SDP
  88. Adding codec 100007 (lpc10) to SDP
  89. Adding codec 100007 (lpc10) to SDP
  90. Adding codec 100008 (g729) to SDP
  91. Adding codec 100008 (g729) to SDP
  92. Adding codec 100009 (speex) to SDP
  93. Adding codec 100009 (speex) to SDP
  94. Adding codec 100010 (ilbc) to SDP
  95. Adding codec 100010 (ilbc) to SDP
  96. Adding codec 100011 (g726) to SDP
  97. Adding codec 100011 (g726) to SDP
  98. Adding codec 100012 (g722) to SDP
  99. Adding codec 100012 (g722) to SDP
  100. Adding codec 100013 (siren7) to SDP
  101. Adding codec 100013 (siren7) to SDP
  102. Adding codec 100014 (siren14) to SDP
  103. Adding codec 100014 (siren14) to SDP
  104. Adding codec 100015 (g719) to SDP
  105. Adding codec 100015 (g719) to SDP
  106. Adding codec 100016 (speex16) to SDP
  107. Adding codec 100016 (speex16) to SDP
  108. Adding codec 100017 (testlaw) to SDP
  109. Adding codec 100017 (testlaw) to SDP
  110. Adding codec 100017 (testlaw) to SDP
  111. Adding codec 100019 (slin) to SDP
  112. Adding codec 100019 (slin) to SDP
  113. Adding codec 100020 (slin12) to SDP
  114. Adding codec 100020 (slin12) to SDP
  115. Adding codec 100021 (slin16) to SDP
  116. Adding codec 100021 (slin16) to SDP
  117. Adding codec 100022 (slin24) to SDP
  118. Adding codec 100022 (slin24) to SDP
  119. Adding codec 100023 (slin32) to SDP
  120. Adding codec 100023 (slin32) to SDP
  121. Adding codec 100024 (slin44) to SDP
  122. Adding codec 100024 (slin44) to SDP
  123. Adding codec 100025 (slin48) to SDP
  124. Adding codec 100025 (slin48) to SDP
  125. Adding codec 100026 (slin96) to SDP
  126. Adding codec 100026 (slin96) to SDP
  127. Adding codec 100027 (slin192) to SDP
  128. Adding codec 100027 (slin192) to SDP
  129. Adding codec 100028 (speex32) to SDP
  130. Adding codec 100028 (speex32) to SDP
  131. Adding non-codec 0x1 (telephone-event) to SDP
  132. Reliably Transmitting (no NAT) to 192.100.180.69:24434:
  133. INVITE sip:8929@192.100.180.69:24434;rinstance=c0ebc96a5675f30a SIP/2.0
  134. Via: SIP/2.0/UDP 10.10.2.12:5060;branch=z9hG4bK73e083a9
  135. Max-Forwards: 70
  136. From: "Miguel Del Castillo Soft" <sip:anonymous@10.10.2.12>;tag=as676e2a02
  137. To: <sip:8929@192.100.180.69:24434;rinstance=c0ebc96a5675f30a>
  138. Contact: <sip:anonymous@10.10.2.12:5060>
  139. Call-ID: 041baf5e219cacc148f8d3e178926940@10.10.2.12:5060
  140. CSeq: 102 INVITE
  141. User-Agent: Asterisk PBX 11.5.0
  142. Date: Thu, 25 Jul 2013 23:31:19 GMT
  143. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  144. Supported: replaces, timer
  145. Remote-Party-ID: "Miguel Del Castillo Soft" <sip:anonymous@10.10.2.12>;party=calling;privacy=off;screen=no
  146. Content-Type: application/sdp
  147. Content-Length: 1536
  148.  
  149. v=0
  150. o=root 530297565 530297565 IN IP4 10.10.2.12
  151. s=Asterisk PBX 11.5.0
  152. c=IN IP4 10.10.2.12
  153. t=0 0
  154. m=audio 15828 RTP/AVP 0 4 4 3 3 3 8 8 8 112 112 5 5 7 7 18 18 110 110 97 97 111 111 9 9 102 102 115 115 116 116 117 117 10 10 118 118 119 119 101
  155. a=rtpmap:0 PCMU/8000
  156. a=rtpmap:4 G723/8000
  157. a=fmtp:4 annexa=no
  158. a=rtpmap:4 G723/8000
  159. a=fmtp:4 annexa=no
  160. a=rtpmap:3 GSM/8000
  161. a=rtpmap:3 GSM/8000
  162. a=rtpmap:3 GSM/8000
  163. a=rtpmap:8 PCMA/8000
  164. a=rtpmap:8 PCMA/8000
  165. a=rtpmap:8 PCMA/8000
  166. a=rtpmap:112 AAL2-G726-32/8000
  167. a=rtpmap:112 AAL2-G726-32/8000
  168. a=rtpmap:5 DVI4/8000
  169. a=rtpmap:5 DVI4/8000
  170. a=rtpmap:7 LPC/8000
  171. a=rtpmap:7 LPC/8000
  172. a=rtpmap:18 G729/8000
  173. a=fmtp:18 annexb=no
  174. a=rtpmap:18 G729/8000
  175. a=fmtp:18 annexb=no
  176. a=rtpmap:110 speex/8000
  177. a=rtpmap:110 speex/8000
  178. a=rtpmap:97 iLBC/8000
  179. a=fmtp:97 mode=30
  180. a=rtpmap:97 iLBC/8000
  181. a=fmtp:97 mode=30
  182. a=rtpmap:111 G726-32/8000
  183. a=rtpmap:111 G726-32/8000
  184. a=rtpmap:9 G722/8000
  185. a=rtpmap:9 G722/8000
  186. a=rtpmap:102 G7221/16000
  187. a=fmtp:102 bitrate=32000
  188. a=rtpmap:102 G7221/16000
  189. a=fmtp:102 bitrate=32000
  190. a=rtpmap:115 G7221/32000
  191. a=fmtp:115 bitrate=48000
  192. a=rtpmap:115 G7221/32000
  193. a=fmtp:115 bitrate=48000
  194. a=rtpmap:116 G719/48000
  195. a=fmtp:116 bitrate=64000
  196. a=rtpmap:116 G719/48000
  197. a=fmtp:116 bitrate=64000
  198. a=rtpmap:117 speex/16000
  199. a=rtpmap:117 speex/16000
  200. a=rtpmap:10 L16/8000
  201. a=rtpmap:10 L16/8000
  202. a=rtpmap:118 L16/16000
  203. a=rtpmap:118 L16/16000
  204. a=rtpmap:119 speex/32000
  205. a=rtpmap:119 speex/32000
  206. a=rtpmap:101 telephone-event/8000
  207. a=fmtp:101 0-16
  208. a=ptime:20
  209. a=sendrecv
  210.  
  211. ---
  212.  
  213. <--- SIP read from UDP:192.100.180.69:24434 --->
  214. SIP/2.0 180 Ringing
  215. Via: SIP/2.0/UDP 10.10.2.12:5060;branch=z9hG4bK73e083a9
  216. Contact: <sip:8929@192.100.180.69:24434>
  217. To: <sip:8929@192.100.180.69:24434;rinstance=c0ebc96a5675f30a>;tag=24c4ff0d
  218. From: "Miguel Del Castillo Soft"<sip:anonymous@10.10.2.12>;tag=as676e2a02
  219. Call-ID: 041baf5e219cacc148f8d3e178926940@10.10.2.12:5060
  220. CSeq: 102 INVITE
  221. User-Agent: X-Lite release 4.5.3 stamp 70569
  222. Allow-Events: hold, talk
  223. Content-Length: 0
  224.  
  225. <------------->
  226. --- (10 headers 0 lines) ---
  227. list_route: hop: <sip:8929@192.100.180.69:24434>
  228.  
  229. <--- Transmitting (NAT) to 10.10.2.220:20823 --->
  230. SIP/2.0 180 Ringing
  231. Via: SIP/2.0/TCP 10.10.2.220;branch=z9hG4bK806687127df8e216f7c517f69400;received=10.10.2.220;rport=20823
  232. Record-Route: <sip:10.10.2.220;lr;transport=tcp>
  233. From: "Miguel Del Castillo Soft" <sip:anonymous@anonymous.invalid>;tag=806687127df8e216d7c517f69400
  234. To: "8929" <sip:8929@10.10.2.12>;tag=as76ffc716
  235. Call-ID: 806687127df8e216e7c517f69400
  236. CSeq: 1 INVITE
  237. Server: Asterisk PBX 11.5.0
  238. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  239. Supported: replaces, timer
  240. Session-Expires: 240;refresher=uac
  241. Contact: <sip:8929@10.10.2.12:5060;transport=TCP>
  242. Remote-Party-ID: "User2" <sip:8929@anonymous.invalid>;party=called;privacy=off;screen=no
  243. Content-Length: 0
  244.  
  245.  
  246. <------------>
  247.  
  248. <--- SIP read from UDP:192.100.180.69:24434 --->
  249. SIP/2.0 200 OK
  250. Via: SIP/2.0/UDP 10.10.2.12:5060;branch=z9hG4bK73e083a9
  251. Contact: <sip:8929@192.100.180.69:24434>
  252. To: <sip:8929@192.100.180.69:24434;rinstance=c0ebc96a5675f30a>;tag=24c4ff0d
  253. From: "Miguel Del Castillo Soft"<sip:anonymous@10.10.2.12>;tag=as676e2a02
  254. Call-ID: 041baf5e219cacc148f8d3e178926940@10.10.2.12:5060
  255. CSeq: 102 INVITE
  256. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  257. Content-Type: application/sdp
  258. Supported: replaces, eventlist
  259. User-Agent: X-Lite release 4.5.3 stamp 70569
  260. Content-Length: 396
  261.  
  262. v=0
  263. o=- 13019268690666034 3 IN IP4 192.100.180.69
  264. s=X-Lite 4 release 4.5.3 stamp 70569
  265. c=IN IP4 192.100.180.69
  266. t=0 0
  267. m=audio 59124 RTP/AVP 0 8 8 8 110 110 97 97 9 9 117 117 101
  268. a=rtpmap:110 speex/8000
  269. a=rtpmap:110 speex/8000
  270. a=rtpmap:97 ILBC/8000
  271. a=rtpmap:97 ILBC/8000
  272. a=rtpmap:117 speex/16000
  273. a=rtpmap:117 speex/16000
  274. a=rtpmap:101 telephone-event/8000
  275. a=fmtp:101 0-15
  276. a=sendrecv
  277. <------------->
  278. --- (12 headers 15 lines) ---
  279. Found RTP audio format 0
  280. Found RTP audio format 8
  281. Found RTP audio format 8
  282. Found RTP audio format 8
  283. Found RTP audio format 110
  284. Found RTP audio format 110
  285. Found RTP audio format 97
  286. Found RTP audio format 97
  287. Found RTP audio format 9
  288. Found RTP audio format 9
  289. Found RTP audio format 117
  290. Found RTP audio format 117
  291. Found RTP audio format 101
  292. Found audio description format speex for ID 110
  293. Found audio description format speex for ID 110
  294. Found audio description format ILBC for ID 97
  295. Found audio description format ILBC for ID 97
  296. Found audio description format speex for ID 117
  297. Found audio description format speex for ID 117
  298. Found audio description format telephone-event for ID 101
  299. Capabilities: us - (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|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192), peer - audio=(ulaw|alaw|speex|speex16|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|speex|speex16|ilbc|g722)
  300. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  301. Peer audio RTP is at port 192.100.180.69:59124
  302. list_route: hop: <sip:8929@192.100.180.69:24434>
  303. set_destination: Parsing <sip:8929@192.100.180.69:24434> for address/port to send to
  304. set_destination: set destination to 192.100.180.69:24434
  305. Transmitting (no NAT) to 192.100.180.69:24434:
  306. ACK sip:8929@192.100.180.69:24434 SIP/2.0
  307. Via: SIP/2.0/UDP 10.10.2.12:5060;branch=z9hG4bK0b94fd79
  308. Max-Forwards: 70
  309. From: "Miguel Del Castillo Soft" <sip:anonymous@10.10.2.12>;tag=as676e2a02
  310. To: <sip:8929@192.100.180.69:24434;rinstance=c0ebc96a5675f30a>;tag=24c4ff0d
  311. Contact: <sip:anonymous@10.10.2.12:5060>
  312. Call-ID: 041baf5e219cacc148f8d3e178926940@10.10.2.12:5060
  313. CSeq: 102 ACK
  314. User-Agent: Asterisk PBX 11.5.0
  315. Content-Length: 0
  316.  
  317.  
  318. ---
  319. Audio is at 17820
  320. Adding codec 100003 (ulaw) to SDP
  321. Adding codec 100003 (ulaw) to SDP
  322. Adding codec 100008 (g729) to SDP
  323. Adding non-codec 0x1 (telephone-event) to SDP
  324.  
  325. <--- Reliably Transmitting (NAT) to 10.10.2.220:20823 --->
  326. SIP/2.0 200 OK
  327. Via: SIP/2.0/TCP 10.10.2.220;branch=z9hG4bK806687127df8e216f7c517f69400;received=10.10.2.220;rport=20823
  328. Record-Route: <sip:10.10.2.220;lr;transport=tcp>
  329. From: "Miguel Del Castillo Soft" <sip:anonymous@anonymous.invalid>;tag=806687127df8e216d7c517f69400
  330. To: "8929" <sip:8929@10.10.2.12>;tag=as76ffc716
  331. Call-ID: 806687127df8e216e7c517f69400
  332. CSeq: 1 INVITE
  333. Server: Asterisk PBX 11.5.0
  334. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  335. Supported: replaces, timer
  336. Session-Expires: 240;refresher=uac
  337. Contact: <sip:8929@10.10.2.12:5060;transport=TCP>
  338. Remote-Party-ID: "User2" <sip:8929@anonymous.invalid>;party=called;privacy=off;screen=no
  339. Content-Type: application/sdp
  340. Require: timer
  341. Content-Length: 300
  342.  
  343. v=0
  344. o=root 584840798 584840798 IN IP4 10.10.2.12
  345. s=Asterisk PBX 11.5.0
  346. c=IN IP4 10.10.2.12
  347. t=0 0
  348. m=audio 17820 RTP/AVP 0 0 18 101
  349. a=rtpmap:0 PCMU/8000
  350. a=rtpmap:0 PCMU/8000
  351. a=rtpmap:18 G729/8000
  352. a=fmtp:18 annexb=no
  353. a=rtpmap:101 telephone-event/8000
  354. a=fmtp:101 0-16
  355. a=ptime:20
  356. a=sendrecv
  357.  
  358. <------------>
  359.  
  360. <--- SIP read from TCP:10.10.2.220:20823 --->
  361. ACK sip:8929@10.10.2.12:5060;transport=TCP SIP/2.0
  362. From: "Miguel Del Castillo Soft" <sip:anonymous@anonymous.invalid>;tag=806687127df8e216d7c517f69400
  363. To: "8929" <sip:8929@10.10.2.12>;tag=as76ffc716
  364. Call-ID: 806687127df8e216e7c517f69400
  365. CSeq: 1 ACK
  366. Max-Forwards: 70
  367. Via: SIP/2.0/TCP 10.10.2.220;branch=z9hG4bK801a4c177df8e21727c517f69400
  368. User-Agent: Avaya CM/R015x.01.1.415.1
  369. Content-Length: 0
  370.  
  371. <------------->
  372. --- (9 headers 0 lines) ---
  373.  
  374. <--- SIP read from TCP:10.10.2.220:20823 --->
  375. BYE sip:8929@10.10.2.12:5060;transport=TCP SIP/2.0
  376. From: "Miguel Del Castillo Soft" <sip:anonymous@anonymous.invalid>;tag=806687127df8e216d7c517f69400
  377. To: "8929" <sip:8929@10.10.2.12>;tag=as76ffc716
  378. Call-ID: 806687127df8e216e7c517f69400
  379. CSeq: 2 BYE
  380. Max-Forwards: 70
  381. Via: SIP/2.0/TCP 10.10.2.220;branch=z9hG4bK801a4c177df8e21737c517f69400
  382. User-Agent: Avaya CM/R015x.01.1.415.1
  383. Content-Length: 0
  384.  
  385. <------------->
  386. --- (9 headers 0 lines) ---
  387. Sending to 10.10.2.220:20823 (NAT)
  388. Scheduling destruction of SIP dialog '806687127df8e216e7c517f69400' in 32000 ms (Method: BYE)
  389.  
  390. <--- Transmitting (NAT) to 10.10.2.220:20823 --->
  391. SIP/2.0 200 OK
  392. Via: SIP/2.0/TCP 10.10.2.220;branch=z9hG4bK801a4c177df8e21737c517f69400;received=10.10.2.220;rport=20823
  393. From: "Miguel Del Castillo Soft" <sip:anonymous@anonymous.invalid>;tag=806687127df8e216d7c517f69400
  394. To: "8929" <sip:8929@10.10.2.12>;tag=as76ffc716
  395. Call-ID: 806687127df8e216e7c517f69400
  396. CSeq: 2 BYE
  397. Server: Asterisk PBX 11.5.0
  398. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  399. Supported: replaces, timer
  400. Content-Length: 0
  401.  
  402.  
  403. <------------>
  404. Scheduling destruction of SIP dialog '041baf5e219cacc148f8d3e178926940@10.10.2.12:5060' in 32000 ms (Method: INVITE)
  405. set_destination: Parsing <sip:8929@192.100.180.69:24434> for address/port to send to
  406. set_destination: set destination to 192.100.180.69:24434
  407. Reliably Transmitting (no NAT) to 192.100.180.69:24434:
  408. BYE sip:8929@192.100.180.69:24434 SIP/2.0
  409. Via: SIP/2.0/UDP 10.10.2.12:5060;branch=z9hG4bK1ccdedaf
  410. Max-Forwards: 70
  411. From: "Miguel Del Castillo Soft" <sip:anonymous@10.10.2.12>;tag=as676e2a02
  412. To: <sip:8929@192.100.180.69:24434;rinstance=c0ebc96a5675f30a>;tag=24c4ff0d
  413. Call-ID: 041baf5e219cacc148f8d3e178926940@10.10.2.12:5060
  414. CSeq: 103 BYE
  415. User-Agent: Asterisk PBX 11.5.0
  416. X-Asterisk-HangupCause: Normal Clearing
  417. X-Asterisk-HangupCauseCode: 16
  418. Content-Length: 0
  419.  
  420.  
  421. ---
  422.  
  423. <--- SIP read from UDP:192.100.180.69:24434 --->
  424. SIP/2.0 200 OK
  425. Via: SIP/2.0/UDP 10.10.2.12:5060;branch=z9hG4bK1ccdedaf
  426. Contact: <sip:8929@192.100.180.69:24434>
  427. To: <sip:8929@192.100.180.69:24434;rinstance=c0ebc96a5675f30a>;tag=24c4ff0d
  428. From: "Miguel Del Castillo Soft"<sip:anonymous@10.10.2.12>;tag=as676e2a02
  429. Call-ID: 041baf5e219cacc148f8d3e178926940@10.10.2.12:5060
  430. CSeq: 103 BYE
  431. User-Agent: X-Lite release 4.5.3 stamp 70569
  432. Content-Length: 0
  433.  
  434. <------------->
  435. --- (9 headers 0 lines) ---
  436. Really destroying SIP dialog '041baf5e219cacc148f8d3e178926940@10.10.2.12:5060' Method: INVITE
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