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  1. [Jun 17 16:09:17] VERBOSE[9636] chan_sip.c: Really destroying SIP dialog '1146327453@xxx.xxx.xxx.xxx' Method: OPTIONS
  2. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c:
  3. <--- SIP read from UDP://208.94.157.10:5060 --->
  4. INVITE sip:17168980077@xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0
  5.  
  6. From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
  7.  
  8. To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>
  9.  
  10. Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
  11.  
  12. CSeq: 1 INVITE
  13.  
  14. Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f239-4c1a80ee-2435a30b-24414b39
  15.  
  16. Max-Forwards: 68
  17.  
  18. P-Asserted-Identity: <sip:7169074915@cxc.dashcs.com:5060>
  19.  
  20. Supported: 100rel
  21.  
  22. Content-Disposition: session;handling=optional
  23.  
  24. Contact: <sip:7169074915@208.94.157.10:5060;transport=udp>
  25.  
  26. Min-SE: 900
  27.  
  28. Session-Expires: 1800
  29.  
  30. Content-Type: application/sdp
  31.  
  32. Content-Length: 238
  33.  
  34.  
  35.  
  36. v=0
  37.  
  38. o=Acme_UAS 0 1 IN IP4 208.94.157.10
  39.  
  40. s=SIP Media Capabilities
  41.  
  42. c=IN IP4 208.94.157.10
  43.  
  44. t=0 0
  45.  
  46. m=audio 22126 RTP/AVP 0 18 101
  47.  
  48. a=rtpmap:0 PCMU/8000
  49.  
  50. a=rtpmap:18 G729/8000
  51.  
  52. a=rtpmap:101 telephone-event/8000
  53.  
  54. a=maxptime:20
  55.  
  56. a=sendrecv
  57.  
  58.  
  59. <------------->
  60. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: --- (15 headers 11 lines) ---
  61. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Sending to 208.94.157.10 : 5060 (NAT)
  62. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Using INVITE request as basis request - CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
  63. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found peer '4-208.94.157.10' for '7169074915' from 208.94.157.10:5060
  64. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found RTP audio format 0
  65. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found RTP audio format 18
  66. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found RTP audio format 101
  67. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Peer audio RTP is at port 208.94.157.10:22126
  68. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found audio description format PCMU for ID 0
  69. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found audio description format G729 for ID 18
  70. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Found audio description format telephone-event for ID 101
  71. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Capabilities: us - 0xec6 (gsm|ulaw|g726|slin|lpc10|speex|ilbc), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  72. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  73. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Peer audio RTP is at port 208.94.157.10:22126
  74. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: Looking for 17168980077 in from-external (domain xxx.xxx.xxx.xxx)
  75. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: list_route: hop: <sip:7169074915@208.94.157.10:5060;transport=udp>
  76. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c:
  77. <--- Transmitting (no NAT) to 208.94.157.10:5060 --->
  78. SIP/2.0 100 Trying
  79.  
  80. Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f239-4c1a80ee-2435a30b-24414b39;received=208.94.157.10
  81.  
  82. From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
  83.  
  84. To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>
  85.  
  86. Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
  87.  
  88. CSeq: 1 INVITE
  89.  
  90. Server: i-Communicate
  91.  
  92. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  93.  
  94. Supported: replaces, timer
  95.  
  96. Contact: <sip:17168980077@xxx.xxx.xxx.xxx>
  97.  
  98. Content-Length: 0
  99.  
  100.  
  101.  
  102.  
  103. <------------>
  104. [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Audio is at xxx.xxx.xxx.xxx port 16414
  105. [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  106. [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x1000 (g722) to SDP
  107. [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x8 (alaw) to SDP
  108. [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x40 (slin) to SDP
  109. [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x2 (gsm) to SDP
  110. [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x800 (g726) to SDP
  111. [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x20 (adpcm) to SDP
  112. [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding codec 0x80 (lpc10) to SDP
  113. [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  114. [Jun 17 16:09:19] VERBOSE[6654] chan_sip.c: Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:5060:
  115. INVITE sip:01163956@xxx.xxx.xxx.xxx SIP/2.0
  116.  
  117. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b2cd08d;rport
  118.  
  119. Max-Forwards: 70
  120.  
  121. From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
  122.  
  123. To: <sip:01163956@xxx.xxx.xxx.xxx>
  124.  
  125. Contact: <sip:7169074915@xxx.xxx.xxx.xxx>
  126.  
  127. Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
  128.  
  129. CSeq: 102 INVITE
  130.  
  131. User-Agent: i-Communicate
  132.  
  133. Date: Thu, 17 Jun 2010 20:09:19 GMT
  134.  
  135. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  136.  
  137. Supported: replaces, timer
  138.  
  139. Diversion: <sip:7168980077@betapbx.i-evolve.com>
  140.  
  141. Content-Type: application/sdp
  142.  
  143. Content-Length: 406
  144.  
  145.  
  146.  
  147. v=0
  148.  
  149. o=root 801668698 801668698 IN IP4 xxx.xxx.xxx.xxx
  150.  
  151. s=Asterisk PBX 1.6.1.6
  152.  
  153. c=IN IP4 xxx.xxx.xxx.xxx
  154.  
  155. t=0 0
  156.  
  157. m=audio 16414 RTP/AVP 0 9 8 10 3 111 5 7 101
  158.  
  159. a=rtpmap:0 PCMU/8000
  160.  
  161. a=rtpmap:9 G722/8000
  162.  
  163. a=rtpmap:8 PCMA/8000
  164.  
  165. a=rtpmap:10 L16/8000
  166.  
  167. a=rtpmap:3 GSM/8000
  168.  
  169. a=rtpmap:111 G726-32/8000
  170.  
  171. a=rtpmap:5 DVI4/8000
  172.  
  173. a=rtpmap:7 LPC/8000
  174.  
  175. a=rtpmap:101 telephone-event/8000
  176.  
  177. a=fmtp:101 0-16
  178.  
  179. a=ptime:20
  180.  
  181. a=sendrecv
  182.  
  183.  
  184. ---
  185. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c:
  186. <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
  187. SIP/2.0 100 Giving a try
  188.  
  189. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b2cd08d;rport=5060
  190.  
  191. From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
  192.  
  193. To: <sip:01163956@xxx.xxx.xxx.xxx>
  194.  
  195. Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
  196.  
  197. CSeq: 102 INVITE
  198.  
  199. Server: Enswitch SIP proxy
  200.  
  201. Content-Length: 0
  202.  
  203. Warning: 392 xxx.xxx.xxx.xxx:5060 "Noisy feedback tells:  pid=7889 req_src_ip=xxx.xxx.xxx.xxx req_src_port=5060 in_uri=sip:01163956@xxx.xxx.xxx.xxx out_uri=sip:01163956@72.237.213.162:5060 via_cnt==1"
  204.  
  205.  
  206.  
  207.  
  208. <------------->
  209. [Jun 17 16:09:19] VERBOSE[9636] chan_sip.c: --- (9 headers 0 lines) ---
  210. [Jun 17 16:09:19] VERBOSE[6650] chan_sip.c:
  211. <--- Transmitting (no NAT) to 208.94.157.10:5060 --->
  212. SIP/2.0 180 Ringing
  213.  
  214. Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f239-4c1a80ee-2435a30b-24414b39;received=208.94.157.10
  215.  
  216. From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
  217.  
  218. To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>;tag=as25aa0af6
  219.  
  220. Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
  221.  
  222. CSeq: 1 INVITE
  223.  
  224. Server: i-Communicate
  225.  
  226. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  227.  
  228. Supported: replaces, timer
  229.  
  230. Contact: <sip:17168980077@xxx.xxx.xxx.xxx>
  231.  
  232. Content-Length: 0
  233.  
  234.  
  235.  
  236.  
  237. <------------>
  238. [Jun 17 16:09:20] VERBOSE[9636] chan_sip.c:
  239. <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
  240. SIP/2.0 180 Ringing
  241.  
  242. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bK0b2cd08d;rport=5060
  243.  
  244. From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
  245.  
  246. To: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
  247.  
  248. CSeq: 102 INVITE
  249.  
  250. Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
  251.  
  252. Contact: <sip:01163956@72.237.213.162:5060>
  253.  
  254. Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
  255.  
  256. User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.3.1734
  257.  
  258. Allow-Events: talk,hold,conference
  259.  
  260. Accept-Language: en
  261.  
  262. Content-Length: 0
  263.  
  264.  
  265.  
  266.  
  267. <------------->
  268. [Jun 17 16:09:20] VERBOSE[9636] chan_sip.c: --- (12 headers 0 lines) ---
  269. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c:
  270. <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
  271. SIP/2.0 200 OK
  272.  
  273. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bK0b2cd08d;rport=5060
  274.  
  275. From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
  276.  
  277. To: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
  278.  
  279. CSeq: 102 INVITE
  280.  
  281. Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
  282.  
  283. Contact: <sip:01163956@72.237.213.162:5060>
  284.  
  285. Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
  286.  
  287. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  288.  
  289. Supported: 100rel,replaces
  290.  
  291. User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.3.1734
  292.  
  293. Accept-Language: en
  294.  
  295. Content-Type: application/sdp
  296.  
  297. Content-Length: 223
  298.  
  299.  
  300.  
  301. v=0
  302.  
  303. o=- 1276804589 1276804589 IN IP4 72.237.213.162
  304.  
  305. s=Polycom IP Phone
  306.  
  307. c=IN IP4 72.237.213.162
  308.  
  309. t=0 0
  310.  
  311. m=audio 2224 RTP/AVP 0 127
  312.  
  313. a=sendrecv
  314.  
  315. a=rtpmap:0 PCMU/8000
  316.  
  317. a=rtpmap:127 telephone-event/8000
  318.  
  319. a=direction:active
  320.  
  321.  
  322. <------------->
  323. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: --- (14 headers 10 lines) ---
  324. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Found RTP audio format 0
  325. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Found RTP audio format 127
  326. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Peer audio RTP is at port 72.237.213.162:2224
  327. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Found audio description format PCMU for ID 0
  328. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Found audio description format telephone-event for ID 127
  329. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Capabilities: us - 0x1eee (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|speex|ilbc|g722), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  330. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  331. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Peer audio RTP is at port 72.237.213.162:2224
  332. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: list_route: hop: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
  333. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: set_destination: Parsing <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748> for address/port to send to
  334. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: set_destination: set destination to xxx.xxx.xxx.xxx, port 5060
  335. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: Transmitting (NAT) to xxx.xxx.xxx.xxx:5060:
  336. ACK sip:01163956@72.237.213.162:5060 SIP/2.0
  337.  
  338. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK1b713980;rport
  339.  
  340. Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
  341.  
  342. Max-Forwards: 70
  343.  
  344. From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
  345.  
  346. To: <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
  347.  
  348. Contact: <sip:7169074915@xxx.xxx.xxx.xxx>
  349.  
  350. Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
  351.  
  352. CSeq: 102 ACK
  353.  
  354. User-Agent: i-Communicate
  355.  
  356. Content-Length: 0
  357.  
  358.  
  359.  
  360.  
  361. ---
  362. [Jun 17 16:09:21] VERBOSE[6650] chan_sip.c: Audio is at xxx.xxx.xxx.xxx port 15146
  363. [Jun 17 16:09:21] VERBOSE[6650] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  364. [Jun 17 16:09:21] VERBOSE[6650] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  365. [Jun 17 16:09:21] VERBOSE[6650] chan_sip.c:
  366. <--- Reliably Transmitting (no NAT) to 208.94.157.10:5060 --->
  367. SIP/2.0 200 OK
  368.  
  369. Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f239-4c1a80ee-2435a30b-24414b39;received=208.94.157.10
  370.  
  371. From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
  372.  
  373. To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>;tag=as25aa0af6
  374.  
  375. Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
  376.  
  377. CSeq: 1 INVITE
  378.  
  379. Server: i-Communicate
  380.  
  381. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  382.  
  383. Supported: replaces, timer
  384.  
  385. Contact: <sip:17168980077@xxx.xxx.xxx.xxx>
  386.  
  387. Content-Type: application/sdp
  388.  
  389. Content-Length: 234
  390.  
  391.  
  392.  
  393. v=0
  394.  
  395. o=root 1777021475 1777021475 IN IP4 xxx.xxx.xxx.xxx
  396.  
  397. s=Asterisk PBX 1.6.1.6
  398.  
  399. c=IN IP4 xxx.xxx.xxx.xxx
  400.  
  401. t=0 0
  402.  
  403. m=audio 15146 RTP/AVP 0 101
  404.  
  405. a=rtpmap:0 PCMU/8000
  406.  
  407. a=rtpmap:101 telephone-event/8000
  408.  
  409. a=fmtp:101 0-16
  410.  
  411. a=ptime:20
  412.  
  413. a=sendrecv
  414.  
  415.  
  416. <------------>
  417. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c:
  418. <--- SIP read from UDP://208.94.157.10:5060 --->
  419. ACK sip:17168980077@xxx.xxx.xxx.xxx SIP/2.0
  420.  
  421. From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
  422.  
  423. To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>;tag=as25aa0af6
  424.  
  425. Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
  426.  
  427. CSeq: 1 ACK
  428.  
  429. Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f26e-4c1a80f0-2435ab2a-5efdfff
  430.  
  431. Max-Forwards: 69
  432.  
  433. Contact: <sip:7169074915@208.94.157.10:5060;transport=udp>
  434.  
  435. Content-Length: 0
  436.  
  437.  
  438.  
  439.  
  440. <------------->
  441. [Jun 17 16:09:21] VERBOSE[9636] chan_sip.c: --- (9 headers 0 lines) ---
  442. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c:
  443. <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
  444. ACK sip:7169074915@xxx.xxx.xxx.xxx SIP/2.0
  445.  
  446. Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=2BB89B89-76CD01D4>
  447.  
  448. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bK316d00ef28E9F3DA
  449.  
  450. Via: SIP/2.0/UDP 72.237.213.162:5060;branch=z9hG4bK316d00ef28E9F3DA
  451.  
  452. From: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
  453.  
  454. To: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
  455.  
  456. Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
  457.  
  458. CSeq: 1 ACK
  459.  
  460. Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
  461.  
  462. Contact: <sip:01163956@72.237.213.162:5060>
  463.  
  464. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  465.  
  466. User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.3.1734
  467.  
  468. Accept-Language: en
  469.  
  470. Max-Forwards: 69
  471.  
  472. Content-Length: 0
  473.  
  474. X-Enswitch-RURI: sip:7169074915@xxx.xxx.xxx.xxx
  475.  
  476. X-Enswitch-Source: 72.237.213.162:5060
  477.  
  478.  
  479.  
  480.  
  481. <------------->
  482. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: --- (17 headers 0 lines) ---
  483. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c:
  484. <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
  485. INVITE sip:7169074915@xxx.xxx.xxx.xxx SIP/2.0
  486.  
  487. Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=2BB89B89-76CD01D4>
  488.  
  489. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKa495.f3cd42b.0
  490.  
  491. Via: SIP/2.0/UDP 72.237.213.162:5060;branch=z9hG4bKcd850595F1DA3120
  492.  
  493. From: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
  494.  
  495. To: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
  496.  
  497. Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
  498.  
  499. CSeq: 2 INVITE
  500.  
  501. Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
  502.  
  503. Contact: <sip:01163956@72.237.213.162:5060>
  504.  
  505. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  506.  
  507. User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.3.1734
  508.  
  509. Accept-Language: en
  510.  
  511. Supported: 100rel,replaces
  512.  
  513. Allow-Events: talk,hold,conference
  514.  
  515. Proxy-Authorization: Digest username="01163956", realm="xxx.xxx.xxx.xxx", nonce="4c1a81100000aab2b38bba0eb354d50ebda8f8920e08a560", uri="sip:7169074915@xxx.xxx.xxx.xxx", response="eeba5533d886d1d79c9d25983b1ad637", algorithm=MD5
  516.  
  517. Max-Forwards: 69
  518.  
  519. Content-Type: application/sdp
  520.  
  521. Content-Length: 215
  522.  
  523. X-Enswitch-RURI: sip:7169074915@xxx.xxx.xxx.xxx
  524.  
  525. X-Enswitch-Source: 72.237.213.162:5060
  526.  
  527.  
  528.  
  529. v=0
  530.  
  531. o=- 1276804589 1276804590 IN IP4 72.237.213.162
  532.  
  533. s=Polycom IP Phone
  534.  
  535. c=IN IP4 72.237.213.162
  536.  
  537. t=0 0
  538.  
  539. a=sendonly
  540.  
  541. m=audio 2224 RTP/AVP 0 101
  542.  
  543. a=sendonly
  544.  
  545. a=rtpmap:0 PCMU/8000
  546.  
  547. a=rtpmap:101 telephone-event/8000
  548.  
  549.  
  550. <------------->
  551. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: --- (21 headers 10 lines) ---
  552. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
  553. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Found RTP audio format 0
  554. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Found RTP audio format 101
  555. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Peer audio RTP is at port 72.237.213.162:2224
  556. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Found audio description format PCMU for ID 0
  557. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Found audio description format telephone-event for ID 101
  558. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Capabilities: us - 0x1eee (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|speex|ilbc|g722), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  559. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  560. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Peer audio RTP is at port 72.237.213.162:2224
  561. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c:
  562. <--- Transmitting (NAT) to xxx.xxx.xxx.xxx:5060 --->
  563. SIP/2.0 100 Trying
  564.  
  565. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKa495.f3cd42b.0;received=xxx.xxx.xxx.xxx
  566.  
  567. Via: SIP/2.0/UDP 72.237.213.162:5060;branch=z9hG4bKcd850595F1DA3120
  568.  
  569. Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=2BB89B89-76CD01D4>
  570.  
  571. From: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
  572.  
  573. To: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
  574.  
  575. Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
  576.  
  577. CSeq: 2 INVITE
  578.  
  579. Server: i-Communicate
  580.  
  581. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  582.  
  583. Supported: replaces, timer
  584.  
  585. Contact: <sip:7169074915@xxx.xxx.xxx.xxx>
  586.  
  587. Content-Length: 0
  588.  
  589.  
  590.  
  591.  
  592. <------------>
  593. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Audio is at xxx.xxx.xxx.xxx port 16414
  594. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
  595. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  596. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c:
  597. <--- Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:5060 --->
  598. SIP/2.0 200 OK
  599.  
  600. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKa495.f3cd42b.0;received=xxx.xxx.xxx.xxx
  601.  
  602. Via: SIP/2.0/UDP 72.237.213.162:5060;branch=z9hG4bKcd850595F1DA3120
  603.  
  604. Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=2BB89B89-76CD01D4>
  605.  
  606. From: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
  607.  
  608. To: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
  609.  
  610. Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
  611.  
  612. CSeq: 2 INVITE
  613.  
  614. Server: i-Communicate
  615.  
  616. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  617.  
  618. Supported: replaces, timer
  619.  
  620. Contact: <sip:7169074915@xxx.xxx.xxx.xxx>
  621.  
  622. Content-Type: application/sdp
  623.  
  624. Content-Length: 232
  625.  
  626.  
  627.  
  628. v=0
  629.  
  630. o=root 801668698 801668699 IN IP4 xxx.xxx.xxx.xxx
  631.  
  632. s=Asterisk PBX 1.6.1.6
  633.  
  634. c=IN IP4 xxx.xxx.xxx.xxx
  635.  
  636. t=0 0
  637.  
  638. m=audio 16414 RTP/AVP 0 101
  639.  
  640. a=rtpmap:0 PCMU/8000
  641.  
  642. a=rtpmap:101 telephone-event/8000
  643.  
  644. a=fmtp:101 0-16
  645.  
  646. a=ptime:20
  647.  
  648. a=recvonly
  649.  
  650.  
  651. <------------>
  652. [Jun 17 16:09:22] VERBOSE[9636] chan_sip.c:
  653. <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
  654. ACK sip:7169074915@xxx.xxx.xxx.xxx SIP/2.0
  655.  
  656. Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=2BB89B89-76CD01D4>
  657.  
  658. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKa495.f3cd42b.2
  659.  
  660. Via: SIP/2.0/UDP 72.237.213.162:5060;branch=z9hG4bK35274ca15EB5ED6C
  661.  
  662. From: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
  663.  
  664. To: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
  665.  
  666. Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
  667.  
  668. CSeq: 2 ACK
  669.  
  670. Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
  671.  
  672. Contact: <sip:01163956@72.237.213.162:5060>
  673.  
  674. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  675.  
  676. User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.3.1734
  677.  
  678. Accept-Language: en
  679.  
  680. Max-Forwards: 69
  681.  
  682. Content-Length: 0
  683.  
  684. X-Enswitch-RURI: sip:7169074915@xxx.xxx.xxx.xxx
  685.  
  686. X-Enswitch-Source: 72.237.213.162:5060
  687.  
  688.  
  689.  
  690.  
  691.  
  692.  
  693.  
  694. <--- SIP read from UDP://208.94.157.10:5060 --->
  695. BYE sip:17168980077@xxx.xxx.xxx.xxx SIP/2.0
  696.  
  697. From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
  698.  
  699. To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>;tag=as25aa0af6
  700.  
  701. Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
  702.  
  703. CSeq: 2 BYE
  704.  
  705. Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f334-4c1a80fa-2435d106-bde9e0e
  706.  
  707. Max-Forwards: 69
  708.  
  709. Content-Length: 0
  710.  
  711.  
  712.  
  713.  
  714. <------------->
  715. [Jun 17 16:09:31] VERBOSE[9636] chan_sip.c: --- (8 headers 0 lines) ---
  716. [Jun 17 16:09:31] VERBOSE[9636] chan_sip.c: Sending to 208.94.157.10 : 5060 (no NAT)
  717. [Jun 17 16:09:31] VERBOSE[9636] chan_sip.c:
  718. <--- Transmitting (no NAT) to 208.94.157.10:5060 --->
  719. SIP/2.0 200 OK
  720.  
  721. Via: SIP/2.0/UDP 208.94.157.10:5060;branch=z9hG4bK-8f334-4c1a80fa-2435d106-bde9e0e;received=208.94.157.10
  722.  
  723. From: <sip:7169074915@208.94.157.10:5060>;tag=a9d5ed0-13c4-4c1a80ee-2435a30b-1478a700
  724.  
  725. To: <sip:17168980077@xxx.xxx.xxx.xxx:5060>;tag=as25aa0af6
  726.  
  727. Call-ID: CXC-436-741b1390-a9d5ed0-13c4-4c1a80ee-2435a30b-13aa2832@208.94.157.10
  728.  
  729. CSeq: 2 BYE
  730.  
  731. Server: i-Communicate
  732.  
  733. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  734.  
  735. Supported: replaces, timer
  736.  
  737. Content-Length: 0
  738.  
  739.  
  740.  
  741.  
  742. <------------>
  743. [Jun 17 16:09:32] VERBOSE[6654] chan_sip.c: Scheduling destruction of SIP dialog '496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx' in 32000 ms (Method: ACK)
  744. [Jun 17 16:09:32] VERBOSE[6654] chan_sip.c: set_destination: Parsing <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748> for address/port to send to
  745. [Jun 17 16:09:32] VERBOSE[6654] chan_sip.c: set_destination: set destination to xxx.xxx.xxx.xxx, port 5060
  746. [Jun 17 16:09:32] VERBOSE[6654] chan_sip.c: Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:5060:
  747. BYE sip:01163956@72.237.213.162:5060 SIP/2.0
  748.  
  749. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK07359977;rport
  750.  
  751. Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
  752.  
  753. Max-Forwards: 70
  754.  
  755. From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
  756.  
  757. To: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
  758.  
  759. Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
  760.  
  761. CSeq: 103 BYE
  762.  
  763. User-Agent: i-Communicate
  764.  
  765. X-Asterisk-HangupCause: Normal Clearing
  766.  
  767. X-Asterisk-HangupCauseCode: 16
  768.  
  769. Content-Length: 0
  770.  
  771.  
  772.  
  773.  
  774. ---
  775. [Jun 17 16:09:32] VERBOSE[9636] chan_sip.c:
  776. <--- SIP read from UDP://xxx.xxx.xxx.xxx:5060 --->
  777. SIP/2.0 200 OK
  778.  
  779. Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received=xxx.xxx.xxx.xxx;branch=z9hG4bK07359977;rport=5060
  780.  
  781. From: "7169074915" <sip:7169074915@xxx.xxx.xxx.xxx>;tag=as30d2a748
  782.  
  783. To: "E. Schmid." <sip:01163956@xxx.xxx.xxx.xxx>;tag=2BB89B89-76CD01D4
  784.  
  785. CSeq: 103 BYE
  786.  
  787. Call-ID: 496a8b710f48c3174620fcbf39628265@xxx.xxx.xxx.xxx
  788.  
  789. Contact: <sip:01163956@72.237.213.162:5060>
  790.  
  791. Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as30d2a748>
  792.  
  793. User-Agent: PolycomSoundPointIP-SPIP_330-UA/3.2.3.1734
  794.  
  795. Accept-Language: en
  796.  
  797. Content-Length: 0
  798.  
  799.  
  800.  
  801.  
  802.  
  803.  
  804.  
  805.  
  806. <------------>
  807. [Jun 17 16:10:15] VERBOSE[9636] chan_sip.c: Scheduling destruction of SIP dialog '1769911092@xxx.xxx.xxx.xxx' in 32000 ms (Method: OPTIONS)
  808. [Jun 17 16:10:17] VERBOSE[9636] chan_sip.c: Really destroying SIP dialog '349366721@xxx.xxx.xxx.xxx' Method: OPTIONS
RAW Paste Data
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