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  1. Reliably Transmitting (no NAT) to 213.160.242.71:5060:
  2. OPTIONS sip:gsmgw1.briiz.no SIP/2.0
  3. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK42974b78;rport
  4. From: "asterisk" <sip:asterisk@85.196.86.82>;tag=as5a27ffff
  5. To: <sip:gsmgw1.briiz.no>
  6. Contact: <sip:asterisk@85.196.86.82>
  7. Call-ID: 0fd99a954eea8d2f1ddfec6c318490a4@85.196.86.82
  8. CSeq: 102 OPTIONS
  9. User-Agent: Asterisk PBX
  10. Max-Forwards: 70
  11. Date: Tue, 09 Oct 2007 23:13:59 GMT
  12. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  13. Content-Length: 0
  14.  
  15.  
  16. ---
  17. 12 headers, 0 lines
  18. Reliably Transmitting (NAT) to *ASTSERVERIP*:50210:
  19. OPTIONS sip:ftj_00@*ASTSERVERIP*:50210;rinstance=4101f7ab1ac8e09f SIP/2.0
  20. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK7d6ab167;rport
  21. From: "asterisk" <sip:asterisk@85.196.86.82>;tag=as070a07a2
  22. To: <sip:ftj_00@*ASTSERVERIP*:50210;rinstance=4101f7ab1ac8e09f>
  23. Contact: <sip:asterisk@85.196.86.82>
  24. Call-ID: 5752ed6e497dafd21e17b4a75c19d902@85.196.86.82
  25. CSeq: 102 OPTIONS
  26. User-Agent: Asterisk PBX
  27. Max-Forwards: 70
  28. Date: Tue, 09 Oct 2007 23:13:59 GMT
  29. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  30. Content-Length: 0
  31.  
  32.  
  33. ---
  34. 12 headers, 0 lines
  35. Reliably Transmitting (no NAT) to 213.160.242.71:5060:
  36. OPTIONS sip:gsmgw1.briiz.no SIP/2.0
  37. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK4947df58;rport
  38. From: "asterisk" <sip:asterisk@85.196.86.82>;tag=as2ba77d42
  39. To: <sip:gsmgw1.briiz.no>
  40. Contact: <sip:asterisk@85.196.86.82>
  41. Call-ID: 430aec7838afd29f760a0ee80a1a8d93@85.196.86.82
  42. CSeq: 102 OPTIONS
  43. User-Agent: Asterisk PBX
  44. Max-Forwards: 70
  45. Date: Tue, 09 Oct 2007 23:13:59 GMT
  46. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  47. Content-Length: 0
  48.  
  49.  
  50. ---
  51.  
  52. <-- SIP read from 213.160.242.71:5060:
  53. SIP/2.0 404 Not Found
  54. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK42974b78;rport;received=85.196.86.82
  55. From: "asterisk" <sip:asterisk@85.196.86.82>;tag=as5a27ffff
  56. To: <sip:gsmgw1.briiz.no>;tag=as6df884d8
  57. Call-ID: 0fd99a954eea8d2f1ddfec6c318490a4@85.196.86.82
  58. CSeq: 102 OPTIONS
  59. User-Agent: Asterisk PBX
  60. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  61. Accept: application/sdp
  62. Content-Length: 0
  63.  
  64.  
  65. --- (10 headers 0 lines) ---
  66. Destroying call '0fd99a954eea8d2f1ddfec6c318490a4@85.196.86.82'
  67.  
  68. <-- SIP read from 213.160.242.71:5060:
  69. SIP/2.0 404 Not Found
  70. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK4947df58;rport;received=85.196.86.82
  71. From: "asterisk" <sip:asterisk@85.196.86.82>;tag=as2ba77d42
  72. To: <sip:gsmgw1.briiz.no>;tag=as2dd503d8
  73. Call-ID: 430aec7838afd29f760a0ee80a1a8d93@85.196.86.82
  74. CSeq: 102 OPTIONS
  75. User-Agent: Asterisk PBX
  76. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  77. Accept: application/sdp
  78. Content-Length: 0
  79.  
  80.  
  81. --- (10 headers 0 lines) ---
  82. Destroying call '430aec7838afd29f760a0ee80a1a8d93@85.196.86.82'
  83.  
  84. <-- SIP read from *ASTSERVERIP*:50210:
  85. SIP/2.0 200 OK
  86. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK7d6ab167;rport=5060
  87. Contact: <sip:192.168.1.110:22258>
  88. To: <sip:ftj_00@*ASTSERVERIP*:50210;rinstance=4101f7ab1ac8e09f>;tag=526e9f22
  89. From: "asterisk"<sip:asterisk@85.196.86.82>;tag=as070a07a2
  90. Call-ID: 5752ed6e497dafd21e17b4a75c19d902@85.196.86.82
  91. CSeq: 102 OPTIONS
  92. Accept: application/sdp
  93. Accept-Language: en
  94. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  95. User-Agent: X-Lite release 1011b stamp 39984
  96. Content-Length: 0
  97.  
  98.  
  99. --- (12 headers 0 lines) ---
  100. Destroying call '5752ed6e497dafd21e17b4a75c19d902@85.196.86.82'
  101. Oct 9 19:14:00 NOTICE[11738]: chan_sip.c:5401 sip_reregister: -- Re-registration for 1400007@gsmgw1.briiz.no
  102. REGISTER 12 headers, 0 lines
  103. Reliably Transmitting (no NAT) to 213.160.242.71:5060:
  104. REGISTER sip:gsmgw1.briiz.no SIP/2.0
  105. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK265f2a44;rport
  106. From: <sip:1400007@gsmgw1.briiz.no>;tag=as585338a2
  107. To: <sip:1400007@gsmgw1.briiz.no>
  108. Call-ID: 198b896637afd69860611ab92f8f9c10@85.196.86.82
  109. CSeq: 102 REGISTER
  110. User-Agent: Asterisk PBX
  111. Max-Forwards: 70
  112. Expires: 120
  113. Contact: <sip:s@85.196.86.82>
  114. Event: registration
  115. Content-Length: 0
  116.  
  117.  
  118. ---
  119.  
  120. <-- SIP read from 213.160.242.71:5060:
  121. SIP/2.0 100 Trying
  122. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK265f2a44;received=85.196.86.82;rport=5060
  123. From: <sip:1400007@gsmgw1.briiz.no>;tag=as585338a2
  124. To: <sip:1400007@gsmgw1.briiz.no>
  125. Call-ID: 198b896637afd69860611ab92f8f9c10@85.196.86.82
  126. CSeq: 102 REGISTER
  127. User-Agent: Asterisk PBX
  128. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  129. Contact: <sip:1400007@213.160.242.71>
  130. Content-Length: 0
  131.  
  132.  
  133. --- (10 headers 0 lines) ---
  134.  
  135. <-- SIP read from 213.160.242.71:5060:
  136. SIP/2.0 401 Unauthorized
  137. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK265f2a44;received=85.196.86.82;rport=5060
  138. From: <sip:1400007@gsmgw1.briiz.no>;tag=as585338a2
  139. To: <sip:1400007@gsmgw1.briiz.no>;tag=as4ee7cc7d
  140. Call-ID: 198b896637afd69860611ab92f8f9c10@85.196.86.82
  141. CSeq: 102 REGISTER
  142. User-Agent: Asterisk PBX
  143. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  144. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a6ce1dc"
  145. Content-Length: 0
  146.  
  147.  
  148. --- (10 headers 0 lines) ---
  149. Responding to challenge, registration to domain/host name gsmgw1.briiz.no
  150. REGISTER 13 headers, 0 lines
  151. Reliably Transmitting (no NAT) to 213.160.242.71:5060:
  152. REGISTER sip:gsmgw1.briiz.no SIP/2.0
  153. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK1475f06f;rport
  154. From: <sip:1400007@gsmgw1.briiz.no>;tag=as101f84da
  155. To: <sip:1400007@gsmgw1.briiz.no>
  156. Call-ID: 198b896637afd69860611ab92f8f9c10@85.196.86.82
  157. CSeq: 103 REGISTER
  158. User-Agent: Asterisk PBX
  159. Max-Forwards: 70
  160. Authorization: Digest username="1400007", realm="asterisk", algorithm=MD5, uri="sip:gsmgw1.briiz.no", nonce="0a6ce1dc", response="04a44966bb605812969d4ebf99427b38", opaque=""
  161. Expires: 120
  162. Contact: <sip:s@85.196.86.82>
  163. Event: registration
  164. Content-Length: 0
  165.  
  166.  
  167. ---
  168.  
  169. <-- SIP read from 213.160.242.71:5060:
  170. SIP/2.0 100 Trying
  171. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK1475f06f;received=85.196.86.82;rport=5060
  172. From: <sip:1400007@gsmgw1.briiz.no>;tag=as101f84da
  173. To: <sip:1400007@gsmgw1.briiz.no>
  174. Call-ID: 198b896637afd69860611ab92f8f9c10@85.196.86.82
  175. CSeq: 103 REGISTER
  176. User-Agent: Asterisk PBX
  177. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  178. Contact: <sip:1400007@213.160.242.71>
  179. Content-Length: 0
  180.  
  181.  
  182. --- (10 headers 0 lines) ---
  183.  
  184. <-- SIP read from 213.160.242.71:5060:
  185. SIP/2.0 200 OK
  186. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK1475f06f;received=85.196.86.82;rport=5060
  187. From: <sip:1400007@gsmgw1.briiz.no>;tag=as101f84da
  188. To: <sip:1400007@gsmgw1.briiz.no>;tag=as4ee7cc7d
  189. Call-ID: 198b896637afd69860611ab92f8f9c10@85.196.86.82
  190. CSeq: 103 REGISTER
  191. User-Agent: Asterisk PBX
  192. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  193. Expires: 120
  194. Contact: <sip:s@85.196.86.82>;expires=120
  195. Date: Tue, 09 Oct 2007 15:40:50 GMT
  196. Content-Length: 0
  197.  
  198.  
  199. --- (12 headers 0 lines) ---
  200. Scheduling destruction of call '198b896637afd69860611ab92f8f9c10@85.196.86.82' in 32000 ms
  201. Oct 9 19:14:00 NOTICE[11738]: chan_sip.c:9897 handle_response_register: Outbound Registration: Expiry for gsmgw1.briiz.no is 120 sec (Scheduling reregistration in 105 s)
  202.  
  203. <-- SIP read from *ASTSERVERIP*:50210:
  204. INVITE sip:41242597@85.196.86.82 SIP/2.0
  205. Via: SIP/2.0/UDP 192.168.1.110:22258;branch=z9hG4bK-d87543-eb98f25a6467365a-1--d87543-;rport
  206. Max-Forwards: 70
  207. Contact: <sip:ftj_00@*ASTSERVERIP*:50210>
  208. To: "41242597"<sip:41242597@85.196.86.82>
  209. From: "Sindre 1400007"<sip:ftj_00@85.196.86.82>;tag=146a8848
  210. Call-ID: OTI3N2M3NDQ2MTBhZWRhYzg2OTRlNWRmODNiYjc4MTI.
  211. CSeq: 1 INVITE
  212. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  213. Content-Type: application/sdp
  214. User-Agent: X-Lite release 1011b stamp 39984
  215. Content-Length: 455
  216.  
  217. v=0
  218. o=- 5 2 IN IP4 192.168.1.110
  219. s=CounterPath X-Lite 3.0
  220. c=IN IP4 192.168.1.110
  221. t=0 0
  222. m=audio 7216 RTP/AVP 100 106 0 105 98 8 101
  223. a=fmtp:101 0-15
  224. a=rtpmap:100 SPEEX/16000
  225. a=rtpmap:106 SPEEX-FEC/16000
  226. a=rtpmap:105 SPEEX-FEC/8000
  227. a=rtpmap:98 iLBC/8000
  228. a=rtpmap:101 telephone-event/8000
  229. a=alt:1 3 : PPXzxzof aeNBVR+P 192.168.1.110 7216
  230. a=alt:2 2 : cmplSET9 uM3WF01V 10.37.129.3 7216
  231. a=alt:3 1 : vTacRxJN rDn6b9ra 10.211.55.3 7216
  232. a=sendrecv
  233.  
  234. --- (12 headers 16 lines) ---
  235. Using INVITE request as basis request - OTI3N2M3NDQ2MTBhZWRhYzg2OTRlNWRmODNiYjc4MTI.
  236. Sending to 192.168.1.110 : 22258 (NAT)
  237. Reliably Transmitting (NAT) to *ASTSERVERIP*:50210:
  238. SIP/2.0 407 Proxy Authentication Required
  239. Via: SIP/2.0/UDP 192.168.1.110:22258;branch=z9hG4bK-d87543-eb98f25a6467365a-1--d87543-;received=*ASTSERVERIP*;rport=50210
  240. From: "Sindre 1400007"<sip:ftj_00@85.196.86.82>;tag=146a8848
  241. To: "41242597"<sip:41242597@85.196.86.82>;tag=as1bf4dffc
  242. Call-ID: OTI3N2M3NDQ2MTBhZWRhYzg2OTRlNWRmODNiYjc4MTI.
  243. CSeq: 1 INVITE
  244. User-Agent: Asterisk PBX
  245. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  246. Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4fa5b636"
  247. Content-Length: 0
  248.  
  249.  
  250. ---
  251. Scheduling destruction of call 'OTI3N2M3NDQ2MTBhZWRhYzg2OTRlNWRmODNiYjc4MTI.' in 15000 ms
  252. Found user 'ftj_00'
  253.  
  254. <-- SIP read from *ASTSERVERIP*:50210:
  255. ACK sip:41242597@85.196.86.82 SIP/2.0
  256. Via: SIP/2.0/UDP 192.168.1.110:22258;branch=z9hG4bK-d87543-eb98f25a6467365a-1--d87543-;rport
  257. To: "41242597"<sip:41242597@85.196.86.82>;tag=as1bf4dffc
  258. From: "Sindre 1400007"<sip:ftj_00@85.196.86.82>;tag=146a8848
  259. Call-ID: OTI3N2M3NDQ2MTBhZWRhYzg2OTRlNWRmODNiYjc4MTI.
  260. CSeq: 1 ACK
  261. Content-Length: 0
  262.  
  263.  
  264. --- (7 headers 0 lines) ---
  265.  
  266. <-- SIP read from *ASTSERVERIP*:50210:
  267. INVITE sip:41242597@85.196.86.82 SIP/2.0
  268. Via: SIP/2.0/UDP 192.168.1.110:22258;branch=z9hG4bK-d87543-10818c2fbbceea0d-1--d87543-;rport
  269. Max-Forwards: 70
  270. Contact: <sip:ftj_00@*ASTSERVERIP*:50210>
  271. To: "41242597"<sip:41242597@85.196.86.82>
  272. From: "Sindre 1400007"<sip:ftj_00@85.196.86.82>;tag=146a8848
  273. Call-ID: OTI3N2M3NDQ2MTBhZWRhYzg2OTRlNWRmODNiYjc4MTI.
  274. CSeq: 2 INVITE
  275. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  276. Content-Type: application/sdp
  277. Proxy-Authorization: Digest username="ftj_00",realm="asterisk",nonce="4fa5b636",uri="sip:41242597@85.196.86.82",response="c12ef933ee21ccfa085db8328e1f49d5",algorithm=MD5
  278. User-Agent: X-Lite release 1011b stamp 39984
  279. Content-Length: 455
  280.  
  281. v=0
  282. o=- 5 2 IN IP4 192.168.1.110
  283. s=CounterPath X-Lite 3.0
  284. c=IN IP4 192.168.1.110
  285. t=0 0
  286. m=audio 7216 RTP/AVP 100 106 0 105 98 8 101
  287. a=fmtp:101 0-15
  288. a=rtpmap:100 SPEEX/16000
  289. a=rtpmap:106 SPEEX-FEC/16000
  290. a=rtpmap:105 SPEEX-FEC/8000
  291. a=rtpmap:98 iLBC/8000
  292. a=rtpmap:101 telephone-event/8000
  293. a=alt:1 3 : PPXzxzof aeNBVR+P 192.168.1.110 7216
  294. a=alt:2 2 : cmplSET9 uM3WF01V 10.37.129.3 7216
  295. a=alt:3 1 : vTacRxJN rDn6b9ra 10.211.55.3 7216
  296. a=sendrecv
  297.  
  298. --- (13 headers 16 lines) ---
  299. Using INVITE request as basis request - OTI3N2M3NDQ2MTBhZWRhYzg2OTRlNWRmODNiYjc4MTI.
  300. Sending to 192.168.1.110 : 22258 (NAT)
  301. Found user 'ftj_00'
  302. Found RTP audio format 100
  303. Found RTP audio format 106
  304. Found RTP audio format 0
  305. Found RTP audio format 105
  306. Found RTP audio format 98
  307. Found RTP audio format 8
  308. Found RTP audio format 101
  309. Peer audio RTP is at port 192.168.1.110:7216
  310. Found description format SPEEX
  311. Found description format SPEEX-FEC
  312. Found description format SPEEX-FEC
  313. Found description format iLBC
  314. Found description format telephone-event
  315. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x60c (ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
  316. Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  317. Looking for 41242597 in ftj_outgoing (domain 85.196.86.82)
  318. list_route: hop: <sip:ftj_00@*ASTSERVERIP*:50210>
  319. Transmitting (NAT) to *ASTSERVERIP*:50210:
  320. SIP/2.0 100 Trying
  321. Via: SIP/2.0/UDP 192.168.1.110:22258;branch=z9hG4bK-d87543-10818c2fbbceea0d-1--d87543-;received=*ASTSERVERIP*;rport=50210
  322. From: "Sindre 1400007"<sip:ftj_00@85.196.86.82>;tag=146a8848
  323. To: "41242597"<sip:41242597@85.196.86.82>
  324. Call-ID: OTI3N2M3NDQ2MTBhZWRhYzg2OTRlNWRmODNiYjc4MTI.
  325. CSeq: 2 INVITE
  326. User-Agent: Asterisk PBX
  327. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  328. Contact: <sip:41242597@85.196.86.82>
  329. Content-Length: 0
  330.  
  331.  
  332. ---
  333. -- Executing Dial("SIP/ftj_00-08197dd0", "SIP/41242597@out_21971502") in new stack
  334. We're at 85.196.86.82 port 19490
  335. Adding codec 0x4 (ulaw) to SDP
  336. Adding non-codec 0x1 (telephone-event) to SDP
  337. 13 headers, 10 lines
  338. Reliably Transmitting (no NAT) to 213.160.242.71:5060:
  339. INVITE sip:41242597@gsmgw1.briiz.no SIP/2.0
  340. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK425c20d6;rport
  341. From: "Sindre 1400007" <sip:21971502@gsmgw1.briiz.no>;tag=as79dadd32
  342. To: <sip:41242597@gsmgw1.briiz.no>
  343. Contact: <sip:21971502@85.196.86.82>
  344. Call-ID: 39adf743595564121223915b67bbbf8c@gsmgw1.briiz.no
  345. CSeq: 102 INVITE
  346. User-Agent: Asterisk PBX
  347. Max-Forwards: 70
  348. Date: Tue, 09 Oct 2007 23:14:08 GMT
  349. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  350. Content-Type: application/sdp
  351. Content-Length: 216
  352.  
  353. v=0
  354. o=root 11724 11724 IN IP4 85.196.86.82
  355. s=session
  356. c=IN IP4 85.196.86.82
  357. t=0 0
  358. m=audio 19490 RTP/AVP 0 101
  359. a=rtpmap:0 PCMU/8000
  360. a=rtpmap:101 telephone-event/8000
  361. a=fmtp:101 0-16
  362. a=silenceSupp:off - - - -
  363.  
  364. ---
  365. -- Called 41242597@out_21971502
  366.  
  367. <-- SIP read from 213.160.242.71:5060:
  368. SIP/2.0 407 Proxy Authentication Required
  369. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK425c20d6;received=85.196.86.82;rport=5060
  370. From: "Sindre 1400007" <sip:21971502@gsmgw1.briiz.no>;tag=as79dadd32
  371. To: <sip:41242597@gsmgw1.briiz.no>;tag=as3a0ff233
  372. Call-ID: 39adf743595564121223915b67bbbf8c@gsmgw1.briiz.no
  373. CSeq: 102 INVITE
  374. User-Agent: Asterisk PBX
  375. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  376. Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="18d3e67c"
  377. Content-Length: 0
  378.  
  379.  
  380. --- (10 headers 0 lines) ---
  381. Transmitting (no NAT) to 213.160.242.71:5060:
  382. ACK sip:41242597@gsmgw1.briiz.no SIP/2.0
  383. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK425c20d6;rport
  384. From: "Sindre 1400007" <sip:21971502@gsmgw1.briiz.no>;tag=as79dadd32
  385. To: <sip:41242597@gsmgw1.briiz.no>;tag=as3a0ff233
  386. Contact: <sip:21971502@85.196.86.82>
  387. Call-ID: 39adf743595564121223915b67bbbf8c@gsmgw1.briiz.no
  388. CSeq: 102 ACK
  389. User-Agent: Asterisk PBX
  390. Max-Forwards: 70
  391. Content-Length: 0
  392.  
  393.  
  394. ---
  395. We're at 85.196.86.82 port 19490
  396. Adding codec 0x4 (ulaw) to SDP
  397. Adding non-codec 0x1 (telephone-event) to SDP
  398. Reliably Transmitting (no NAT) to 213.160.242.71:5060:
  399. INVITE sip:41242597@gsmgw1.briiz.no SIP/2.0
  400. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK77b6197f;rport
  401. From: "Sindre 1400007" <sip:21971502@gsmgw1.briiz.no>;tag=as79dadd32
  402. To: <sip:41242597@gsmgw1.briiz.no>
  403. Contact: <sip:21971502@85.196.86.82>
  404. Call-ID: 39adf743595564121223915b67bbbf8c@gsmgw1.briiz.no
  405. CSeq: 103 INVITE
  406. User-Agent: Asterisk PBX
  407. Max-Forwards: 70
  408. Proxy-Authorization: Digest username="1400007", realm="asterisk", algorithm=MD5, uri="sip:41242597@gsmgw1.briiz.no", nonce="18d3e67c", response="c2bf7aac5aa279e77bdcbe78bf7ab69e", opaque=""
  409. Date: Tue, 09 Oct 2007 23:14:08 GMT
  410. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  411. Content-Type: application/sdp
  412. Content-Length: 216
  413.  
  414. v=0
  415. o=root 11724 11725 IN IP4 85.196.86.82
  416. s=session
  417. c=IN IP4 85.196.86.82
  418. t=0 0
  419. m=audio 19490 RTP/AVP 0 101
  420. a=rtpmap:0 PCMU/8000
  421. a=rtpmap:101 telephone-event/8000
  422. a=fmtp:101 0-16
  423. a=silenceSupp:off - - - -
  424.  
  425. ---
  426. Retransmitting #1 (no NAT) to 213.160.242.71:5060:
  427. INVITE sip:41242597@gsmgw1.briiz.no SIP/2.0
  428. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK77b6197f;rport
  429. From: "Sindre 1400007" <sip:21971502@gsmgw1.briiz.no>;tag=as79dadd32
  430. To: <sip:41242597@gsmgw1.briiz.no>
  431. Contact: <sip:21971502@85.196.86.82>
  432. Call-ID: 39adf743595564121223915b67bbbf8c@gsmgw1.briiz.no
  433. CSeq: 103 INVITE
  434. User-Agent: Asterisk PBX
  435. Max-Forwards: 70
  436. Proxy-Authorization: Digest username="1400007", realm="asterisk", algorithm=MD5, uri="sip:41242597@gsmgw1.briiz.no", nonce="18d3e67c", response="c2bf7aac5aa279e77bdcbe78bf7ab69e", opaque=""
  437. Date: Tue, 09 Oct 2007 23:14:08 GMT
  438. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  439. Content-Type: application/sdp
  440. Content-Length: 216
  441.  
  442. v=0
  443. o=root 11724 11725 IN IP4 85.196.86.82
  444. s=session
  445. c=IN IP4 85.196.86.82
  446. t=0 0
  447. m=audio 19490 RTP/AVP 0 101
  448. a=rtpmap:0 PCMU/8000
  449. a=rtpmap:101 telephone-event/8000
  450. a=fmtp:101 0-16
  451. a=silenceSupp:off - - - -
  452.  
  453. ---
  454.  
  455. <-- SIP read from 213.160.242.71:5060:
  456. SIP/2.0 488 Not acceptable here
  457. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK77b6197f;received=85.196.86.82;rport=5060
  458. From: "Sindre 1400007" <sip:21971502@gsmgw1.briiz.no>;tag=as79dadd32
  459. To: <sip:41242597@gsmgw1.briiz.no>;tag=as3a0ff233
  460. Call-ID: 39adf743595564121223915b67bbbf8c@gsmgw1.briiz.no
  461. CSeq: 103 INVITE
  462. User-Agent: Asterisk PBX
  463. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  464. Content-Length: 0
  465.  
  466.  
  467. --- (9 headers 0 lines) ---
  468. -- Got SIP response 488 "Not acceptable here" back from 213.160.242.71
  469. Transmitting (no NAT) to 213.160.242.71:5060:
  470. ACK sip:41242597@gsmgw1.briiz.no SIP/2.0
  471. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK77b6197f;rport
  472. From: "Sindre 1400007" <sip:21971502@gsmgw1.briiz.no>;tag=as79dadd32
  473. To: <sip:41242597@gsmgw1.briiz.no>;tag=as3a0ff233
  474. Contact: <sip:21971502@85.196.86.82>
  475. Call-ID: 39adf743595564121223915b67bbbf8c@gsmgw1.briiz.no
  476. CSeq: 103 ACK
  477. User-Agent: Asterisk PBX
  478. Max-Forwards: 70
  479. Content-Length: 0
  480.  
  481.  
  482. ---
  483. -- SIP/out_21971502-0819d420 is circuit-busy
  484. == Everyone is busy/congested at this time (1:0/1/0)
  485.  
  486. <-- SIP read from 213.160.242.71:5060:
  487. SIP/2.0 407 Proxy Authentication Required
  488. Via: SIP/2.0/UDP 85.196.86.82:5060;branch=z9hG4bK77b6197f;received=85.196.86.82;rport=5060
  489. From: "Sindre 1400007" <sip:21971502@gsmgw1.briiz.no>;tag=as79dadd32
  490. To: <sip:41242597@gsmgw1.briiz.no>;tag=as6279f765
  491. Call-ID: 39adf743595564121223915b67bbbf8c@gsmgw1.briiz.no
  492. CSeq: 103 INVITE
  493. User-Agent: Asterisk PBX
  494. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  495. Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5457f7cf"
  496. Content-Length: 0
  497.  
  498.  
  499. --- (10 headers 0 lines) ---
  500. Destroying call '39adf743595564121223915b67bbbf8c@gsmgw1.briiz.no'
  501.  
  502. <-- SIP read from *ASTSERVERIP*:50210:
  503. CANCEL sip:41242597@85.196.86.82 SIP/2.0
  504. Via: SIP/2.0/UDP 192.168.1.110:22258;branch=z9hG4bK-d87543-10818c2fbbceea0d-1--d87543-;rport
  505. To: "41242597"<sip:41242597@85.196.86.82>
  506. From: "Sindre 1400007"<sip:ftj_00@85.196.86.82>;tag=146a8848
  507. Call-ID: OTI3N2M3NDQ2MTBhZWRhYzg2OTRlNWRmODNiYjc4MTI.
  508. CSeq: 2 CANCEL
  509. Proxy-Authorization: Digest username="ftj_00",realm="asterisk",nonce="4fa5b636",uri="sip:41242597@85.196.86.82",response="150467c5c799dbfe81dbd5a169bde101",algorithm=MD5
  510. User-Agent: X-Lite release 1011b stamp 39984
  511. Content-Length: 0
  512.  
  513.  
  514. --- (9 headers 0 lines) ---
  515. Sending to 192.168.1.110 : 22258 (NAT)
  516. Reliably Transmitting (NAT) to *ASTSERVERIP*:50210:
  517. SIP/2.0 487 Request Terminated
  518. Via: SIP/2.0/UDP 192.168.1.110:22258;branch=z9hG4bK-d87543-10818c2fbbceea0d-1--d87543-;received=*ASTSERVERIP*;rport=50210
  519. From: "Sindre 1400007"<sip:ftj_00@85.196.86.82>;tag=146a8848
  520. To: "41242597"<sip:41242597@85.196.86.82>;tag=as6e0b0303
  521. Call-ID: OTI3N2M3NDQ2MTBhZWRhYzg2OTRlNWRmODNiYjc4MTI.
  522. CSeq: 2 INVITE
  523. User-Agent: Asterisk PBX
  524. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  525. Content-Length: 0
  526. X-Asterisk-HangupCause: Bearer capability not available
  527.  
  528.  
  529. ---
  530. Transmitting (NAT) to *ASTSERVERIP*:50210:
  531. SIP/2.0 200 OK
  532. Via: SIP/2.0/UDP 192.168.1.110:22258;branch=z9hG4bK-d87543-10818c2fbbceea0d-1--d87543-;received=*ASTSERVERIP*;rport=50210
  533. From: "Sindre 1400007"<sip:ftj_00@85.196.86.82>;tag=146a8848
  534. To: "41242597"<sip:41242597@85.196.86.82>;tag=as6e0b0303
  535. Call-ID: OTI3N2M3NDQ2MTBhZWRhYzg2OTRlNWRmODNiYjc4MTI.
  536. CSeq: 2 CANCEL
  537. User-Agent: Asterisk PBX
  538. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  539. Contact: <sip:41242597@85.196.86.82>
  540. Content-Length: 0
  541. X-Asterisk-HangupCause: Bearer capability not available
  542.  
  543.  
  544. ---
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