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Jun 14th, 2011
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  1. Bad call from PSTN to 9012XX1XX1
  2.  
  3.  
  4. Call is in Queue waiting for agent to pick up
  5. Asterisk18*CLI> core show channels
  6. Channel Location State Application(Data)
  7. SIP/9013XX9XX8-00000 [email protected] Up Queue(irock.com,t)
  8. 1 active channel
  9. 1 active call
  10. 32 calls processed
  11.  
  12.  
  13. Asterisk18*CLI> sip show peers
  14. Name/username Host Dyn Forcerport ACL Port Status Realtime
  15. 173.xx.xx.63 173.xx.xx.63 5060 Unmonitored
  16. 9012xx2xx4/9012xx2xx4 173.xx.xx.107 N 5060 Unmonitored Cached RT
  17. 9013XX9XX8/9013XX9XX8 173.xx.xx.107 N 5060 Unmonitored Cached RT
  18. 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]
  19. -- Told SIP/9013XX9XX8-0000002b in irock.com their queue position (which was 1)
  20. -- <SIP/9013XX9XX8-0000002b> Playing 'queue-thankyou.gsm' (language 'en')
  21.  
  22.  
  23.  
  24.  
  25.  
  26. <--- SIP read from UDP:173.xx.xx.107:5060 --->
  27. INVITE sip:[email protected] SIP/2.0
  28. Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
  29. Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.0
  30. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0
  31. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b
  32. To: <sip:[email protected]>
  33. CSeq: 103 INVITE
  34. Call-ID: B2B.365.2114915
  35. Content-Length: 332
  36. User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux))
  37. Content-Type: application/sdp
  38. Supported: replaces
  39. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  40. Max-Forwards: 68
  41. Contact: <sip:173.203.82.88:5060;transport=udp>
  42. P-hint: outbound->inbound
  43. P-hint: Route[6]: mediaproxy
  44.  
  45. v=0
  46. o=root 15753 15753 IN IP4 64.2.142.15
  47. s=session
  48. c=IN IP4 64.2.142.15
  49. t=0 0
  50. m=audio 16632 RTP/AVP 0 8 3 18 101
  51. a=rtpmap:0 PCMU/8000
  52. a=rtpmap:8 PCMA/8000
  53. a=rtpmap:3 GSM/8000
  54. a=rtpmap:18 G729/8000
  55. a=fmtp:18 annexb=no
  56. a=rtpmap:101 telephone-event/8000
  57. a=fmtp:101 0-16
  58. a=silenceSupp:off - - - -
  59. a=ptime:20
  60. a=sendrecv
  61. <------------->
  62. --- (17 headers 16 lines) ---
  63. Sending to 173.xx.xx.107:5060 (no NAT)
  64. Using INVITE request as basis request - B2B.365.2114915
  65. Found peer '9013XX9XX8' for '9014817029' from 173.xx.xx.107:5060
  66. == Using SIP RTP CoS mark 5
  67. Found RTP audio format 0
  68. Found RTP audio format 8
  69. Found RTP audio format 3
  70. Found RTP audio format 18
  71. Found RTP audio format 101
  72. Found audio description format PCMU for ID 0
  73. Found audio description format PCMA for ID 8
  74. Found audio description format GSM for ID 3
  75. Found audio description format G729 for ID 18
  76. Found audio description format telephone-event for ID 101
  77. Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
  78. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  79. Peer audio RTP is at port 64.2.142.15:16632
  80. Looking for 9012XX1XX1 in irock.com (domain irock.com)
  81. list_route: hop: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
  82.  
  83. <--- Transmitting (NAT) to 173.xx.xx.107:5060 --->
  84. SIP/2.0 100 Trying
  85. Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.0;received=173.xx.xx.107;rport=5060
  86. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0
  87. Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
  88. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b
  89. To: <sip:[email protected]>
  90. Call-ID: B2B.365.2114915
  91. CSeq: 103 INVITE
  92. Server: Asterisk PBX 1.8.4.2
  93. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  94. Supported: replaces, timer
  95. Contact: <sip:[email protected]:5060>
  96. Content-Length: 0
  97.  
  98.  
  99. <------------>
  100. -- Executing [[email protected]:1] Answer("SIP/9013XX9XX8-0000002c", "") in new stack
  101. Audio is at 5060
  102. Adding codec 0x2 (gsm) to SDP
  103. Adding codec 0x4 (ulaw) to SDP
  104. Adding codec 0x8 (alaw) to SDP
  105. Adding non-codec 0x1 (telephone-event) to SDP
  106.  
  107. <--- Reliably Transmitting (NAT) to 173.xx.xx.107:5060 --->
  108. SIP/2.0 200 OK
  109. Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.0;received=173.xx.xx.107;rport=5060
  110. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0
  111. Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
  112. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b
  113. To: <sip:[email protected]>;tag=as588736bf
  114. Call-ID: B2B.365.2114915
  115. CSeq: 103 INVITE
  116. Server: Asterisk PBX 1.8.4.2
  117. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  118. Supported: replaces, timer
  119. Contact: <sip:[email protected]:5060>
  120. Content-Type: application/sdp
  121. Content-Length: 312
  122.  
  123. v=0
  124. o=root 1750099629 1750099629 IN IP4 173.xx.xx.63
  125. s=Asterisk PBX 1.8.4.2
  126. c=IN IP4 173.xx.xx.63
  127. t=0 0
  128. m=audio 11978 RTP/AVP 3 0 8 101
  129. a=rtpmap:3 GSM/8000
  130. a=rtpmap:0 PCMU/8000
  131. a=rtpmap:8 PCMA/8000
  132. a=rtpmap:101 telephone-event/8000
  133. a=fmtp:101 0-16
  134. a=silenceSupp:off - - - -
  135. a=ptime:20
  136. a=sendrecv
  137.  
  138. <------------>
  139.  
  140. <--- SIP read from UDP:173.xx.xx.107:5060 --->
  141. ACK sip:[email protected]:5060 SIP/2.0
  142. Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b>
  143. Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.2
  144. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.28049323.0
  145. To: <sip:[email protected]>;tag=as588736bf
  146. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b
  147. CSeq: 103 ACK
  148. Call-ID: B2B.365.2114915
  149. Content-Length: 0
  150. User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux))
  151. Max-Forwards: 69
  152. Contact: <sip:173.203.82.88:5060;transport=udp>
  153.  
  154. <------------->
  155. --- (12 headers 0 lines) ---
  156. -- Executing [[email protected]:2] Set("SIP/9013XX9XX8-0000002c", "QUEUE_MAX_PENALTY=10") in new stack
  157. -- Executing [[email protected]:3] Queue("SIP/9013XX9XX8-0000002c", "irock.com,t") in new stack
  158. -- Started music on hold, class 'default', on SIP/9013XX9XX8-0000002c
  159. -- Stopped music on hold on SIP/9013XX9XX8-0000002c
  160. -- <SIP/9013XX9XX8-0000002c> Playing 'queue-youarenext.gsm' (language 'en')
  161.  
  162.  
  163.  
  164.  
  165.  
  166.  
  167.  
  168.  
  169.  
  170.  
  171.  
  172.  
  173.  
  174.  
  175.  
  176.  
  177.  
  178.  
  179.  
  180.  
  181.  
  182.  
  183.  
  184.  
  185.  
  186.  
  187.  
  188.  
  189.  
  190. U 2011/06/14 16:03:48.757522 173.203.82.88:5060 -> 173.xx.xx.107:5060
  191. INVITE sip:[email protected]:5060 SIP/2.0.
  192. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
  193. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
  194. To: <[email protected]>.
  195. CSeq: 103 INVITE.
  196. Call-ID: B2B.365.2114915.
  197. Content-Length: 332.
  198. User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
  199. Content-Type: application/sdp.
  200. Supported: replaces.
  201. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
  202. Max-Forwards: 69.
  203. Contact: <sip:173.203.82.88:5060;transport=udp>.
  204. .
  205. v=0.
  206. o=root 15753 15753 IN IP4 64.2.142.15.
  207. s=session.
  208. c=IN IP4 64.2.142.15.
  209. t=0 0.
  210. m=audio 16632 RTP/AVP 0 8 3 18 101.
  211. a=rtpmap:0 PCMU/8000.
  212. a=rtpmap:8 PCMA/8000.
  213. a=rtpmap:3 GSM/8000.
  214. a=rtpmap:18 G729/8000.
  215. a=fmtp:18 annexb=no.
  216. a=rtpmap:101 telephone-event/8000.
  217. a=fmtp:101 0-16.
  218. a=silenceSupp:off - - - -.
  219. a=ptime:20.
  220. a=sendrecv.
  221.  
  222.  
  223. .
  224.  
  225. #
  226. U 2011/06/14 16:03:49.758654 173.203.82.88:5060 -> 173.xx.xx.107:5060
  227. INVITE sip:[email protected]:5060 SIP/2.0.
  228. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
  229. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
  230. To: <[email protected]>.
  231. CSeq: 103 INVITE.
  232. Call-ID: B2B.365.2114915.
  233. Content-Length: 332.
  234. User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
  235. Content-Type: application/sdp.
  236. Supported: replaces.
  237. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
  238. Max-Forwards: 69.
  239. Contact: <sip:173.203.82.88:5060;transport=udp>.
  240. .
  241. v=0.
  242. o=root 15753 15753 IN IP4 64.2.142.15.
  243. s=session.
  244. c=IN IP4 64.2.142.15.
  245. t=0 0.
  246. m=audio 16632 RTP/AVP 0 8 3 18 101.
  247. a=rtpmap:0 PCMU/8000.
  248. a=rtpmap:8 PCMA/8000.
  249. a=rtpmap:3 GSM/8000.
  250. a=rtpmap:18 G729/8000.
  251. a=fmtp:18 annexb=no.
  252. a=rtpmap:101 telephone-event/8000.
  253. a=fmtp:101 0-16.
  254. a=silenceSupp:off - - - -.
  255. a=ptime:20.
  256. a=sendrecv.
  257.  
  258.  
  259. .
  260.  
  261. #
  262. U 2011/06/14 16:03:50.315548 173.xx.xx.107:5060 -> 173.203.82.88:5060
  263. SIP/2.0 100 Giving a try.
  264. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
  265. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
  266. To: <[email protected]>.
  267. CSeq: 103 INVITE.
  268. Call-ID: B2B.365.2114915.
  269. Server: ae SIP Proxy.
  270. Content-Length: 0.
  271. .
  272.  
  273. #
  274. U 2011/06/14 16:03:50.315709 173.xx.xx.107:5060 -> 173.xx.xx.63:5060
  275. INVITE sip:[email protected] SIP/2.0.
  276. Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
  277. .
  278. Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.0.
  279. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
  280. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
  281. To: <sip:[email protected]>.
  282. CSeq: 103 INVITE.
  283. Call-ID: B2B.365.2114915.
  284. Content-Length: 332.
  285. User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
  286. Content-Type: application/sdp.
  287. Supported: replaces.
  288. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
  289. Max-Forwards: 68.
  290. Contact: <sip:173.203.82.88:5060;transport=udp>.
  291. P-hint: outbound->inbound .
  292. P-hint: Route[6]: mediaproxy .
  293. .
  294. v=0.
  295. o=root 15753 15753 IN IP4 64.2.142.15.
  296. s=session.
  297. c=IN IP4 64.2.142.15.
  298. t=0 0.
  299. m=audio 16632 RTP/AVP 0 8 3 18 101.
  300. a=rtpmap:0 PCMU/8000.
  301. a=rtpmap:8 PCMA/8000.
  302. a=rtpmap:3 GSM/8000.
  303. a=rtpmap:18 G729/8000.
  304. a=fmtp:18 annexb=no.
  305. a=rtpmap:101 telephone-event/8000.
  306. a=fmtp:101 0-16.
  307. a=silenceSupp:off - - - -.
  308. a=ptime:20.
  309. a=sendrecv.
  310.  
  311.  
  312.  
  313. #
  314. U 2011/06/14 16:03:50.316504 173.xx.xx.107:5060 -> 173.203.82.88:5060
  315. SIP/2.0 100 Giving a try.
  316. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
  317. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
  318. To: <[email protected]>.
  319. CSeq: 103 INVITE.
  320. Call-ID: B2B.365.2114915.
  321. Server: ae SIP Proxy.
  322. Content-Length: 0.
  323. .
  324.  
  325.  
  326.  
  327. #
  328. U 2011/06/14 16:03:50.318765 173.xx.xx.63:5060 -> 173.xx.xx.107:5060
  329. SIP/2.0 100 Trying.
  330. Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.0;received=173.xx.xx.107;rport=5060.
  331. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
  332. Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
  333. .
  334. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
  335. To: <sip:[email protected]>.
  336. Call-ID: B2B.365.2114915.
  337. CSeq: 103 INVITE.
  338. Server: Asterisk PBX 1.8.4.2.
  339. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  340. Supported: replaces, timer.
  341. Contact: <sip:[email protected]:5060>.
  342. Content-Length: 0.
  343. .
  344.  
  345. #
  346. U 2011/06/14 16:03:50.319325 173.xx.xx.63:5060 -> 173.xx.xx.107:5060
  347. SIP/2.0 200 OK.
  348. Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.0;received=173.xx.xx.107;rport=5060.
  349. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
  350. Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
  351. .
  352. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
  353. To: <sip:[email protected]>;tag=as588736bf.
  354. Call-ID: B2B.365.2114915.
  355. CSeq: 103 INVITE.
  356. Server: Asterisk PBX 1.8.4.2.
  357. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  358. Supported: replaces, timer.
  359. Contact: <sip:[email protected]:5060>.
  360. Content-Type: application/sdp.
  361. Content-Length: 312.
  362. .
  363. v=0.
  364. o=root 1750099629 1750099629 IN IP4 173.xx.xx.63.
  365. s=Asterisk PBX 1.8.4.2.
  366. c=IN IP4 173.xx.xx.63.
  367. t=0 0.
  368. m=audio 11978 RTP/AVP 3 0 8 101.
  369. a=rtpmap:3 GSM/8000.
  370. a=rtpmap:0 PCMU/8000.
  371. a=rtpmap:8 PCMA/8000.
  372. a=rtpmap:101 telephone-event/8000.
  373. a=fmtp:101 0-16.
  374. a=silenceSupp:off - - - -.
  375. a=ptime:20.
  376. a=sendrecv.
  377.  
  378.  
  379. #
  380. U 2011/06/14 16:03:50.551048 173.xx.xx.107:5060 -> 173.203.82.88:5060
  381. SIP/2.0 200 OK.
  382. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.18049323.0.
  383. Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->
  384. .
  385. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
  386. To: <[email protected]>;tag=as588736bf.
  387. Call-ID: B2B.365.2114915.
  388. CSeq: 103 INVITE.
  389. Server: Asterisk PBX 1.8.4.2.
  390. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  391. Supported: replaces, timer.
  392. Contact: <sip:[email protected]:5060>.
  393. Content-Type: application/sdp.
  394. Content-Length: 312.
  395. .
  396. v=0.
  397. o=root 1750099629 1750099629 IN IP4 173.xx.xx.63.
  398. s=Asterisk PBX 1.8.4.2.
  399. c=IN IP4 173.xx.xx.63.
  400. t=0 0.
  401. m=audio 11978 RTP/AVP 3 0 8 101.
  402. a=rtpmap:3 GSM/8000.
  403. a=rtpmap:0 PCMU/8000.
  404. a=rtpmap:8 PCMA/8000.
  405. a=rtpmap:101 telephone-event/8000.
  406. a=fmtp:101 0-16.
  407. a=silenceSupp:off - - - -.
  408. a=ptime:20.
  409. a=sendrecv.
  410.  
  411.  
  412.  
  413. #
  414. U 2011/06/14 16:03:50.586584 173.203.82.88:5060 -> 173.xx.xx.107:5060
  415. ACK sip:[email protected]:5060 SIP/2.0.
  416. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.28049323.0.
  417. To: <[email protected]>;tag=as588736bf.
  418. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
  419. CSeq: 103 ACK.
  420. Call-ID: B2B.365.2114915.
  421. Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b;did=99.fcd5a0b6;vst=SllBCAsBAAQDAHEHBgJuW0JcTVIdTV5dNw-->.
  422. Content-Length: 0.
  423. User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
  424. Max-Forwards: 70.
  425. Contact: <sip:173.203.82.88:5060;transport=udp>.
  426. .
  427.  
  428.  
  429. .
  430.  
  431. #
  432. U 2011/06/14 16:03:50.635278 173.xx.xx.107:5060 -> 173.xx.xx.63:5060
  433. ACK sip:[email protected]:5060 SIP/2.0.
  434. Record-Route: <sip:173.xx.xx.107;lr=on;ftag=35f20411ce58ce710915b044a6bfe2c1-5e8b>.
  435. Via: SIP/2.0/UDP 173.xx.xx.107;branch=z9hG4bKb6f7.4b79c394.2.
  436. Via: SIP/2.0/UDP 173.203.82.88;branch=z9hG4bKb6f7.28049323.0.
  437. To: <sip:[email protected]>;tag=as588736bf.
  438. From: <sip:[email protected]>;tag=35f20411ce58ce710915b044a6bfe2c1-5e8b.
  439. CSeq: 103 ACK.
  440. Call-ID: B2B.365.2114915.
  441. Content-Length: 0.
  442. User-Agent: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
  443. Max-Forwards: 69.
  444. Contact: <sip:173.203.82.88:5060;transport=udp>.
  445. .
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