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sip debug

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Jan 4th, 2012
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  1. ast18*CLI> originate SIP/test02/9221536790 application Dial
  2. <--- SIP read from UDP:192.168.1.55:5060 --->
  3. OPTIONS sip:192.168.1.109:5060 SIP/2.0
  4. Max-Forwards: 10
  5. Record-Route: <sip:192.168.1.55;r2=on;lr=on;ftag=2d3174b1>
  6. Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=2d3174b1>
  7. Via: SIP/2.0/UDP 192.168.1.55;branch=z9hG4bK457c.70d0d6d1.0
  8. Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0
  9. From: sip:pinger@sipwise.local;tag=2d3174b1
  10. To: sip:192.168.1.109:5060
  11. Call-ID: 3091dd03-f37e4983-855bc42@127.0.0.1
  12. CSeq: 1 OPTIONS
  13. Content-Length: 0
  14.  
  15. <------------->
  16. --- (11 headers 0 lines) ---
  17. Looking for s in default (domain 192.168.1.109:5060)
  18.  
  19. <--- Transmitting (NAT) to 192.168.1.55:5060 --->
  20. SIP/2.0 404 Not Found
  21. Via: SIP/2.0/UDP 192.168.1.55;branch=z9hG4bK457c.70d0d6d1.0;received=192.168.1.55;rport=5060
  22. Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0
  23. From: sip:pinger@sipwise.local;tag=2d3174b1
  24. To: sip:192.168.1.109:5060;tag=as1ae382e5
  25. Call-ID: 3091dd03-f37e4983-855bc42@127.0.0.1
  26. CSeq: 1 OPTIONS
  27. Server: Asterisk PBX 1.8.8.1
  28. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  29. Supported: replaces, timer
  30. Accept: application/sdp
  31. Content-Length: 0
  32.  
  33.  
  34. <------------>
  35. Scheduling destruction of SIP dialog '3091dd03-f37e4983-855bc42@127.0.0.1' in 32000 ms (Method: OPTIONS)
  36. Really destroying SIP dialog '3091dd03-e37e4983-a35bc42@127.0.0.1' Method: OPTIONS
  37. Really destroying SIP dialog '0ffc9e704f27c7b544742b0b10937baa@192.168.1.109' Method: REGISTER
  38. ast18*CLI> originate SIP/test02/9221536790 application Dial
  39. Audio is at 5060
  40. Adding codec 0x2 (gsm) to SDP
  41. Adding codec 0x4 (ulaw) to SDP
  42. Adding codec 0x8 (alaw) to SDP
  43. Adding codec 0x800000000000 (testlaw) to SDP
  44. Adding non-codec 0x1 (telephone-event) to SDP
  45. Reliably Transmitting (no NAT) to 192.168.1.55:5060:
  46. INVITE sip:9221536790@192.168.1.55 SIP/2.0
  47. Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK31b2fdfd
  48. Max-Forwards: 70
  49. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  50. To: <sip:9221536790@192.168.1.55>
  51. Contact: <sip:test02@192.168.1.109:5060>
  52. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  53. CSeq: 102 INVITE
  54. User-Agent: Asterisk PBX 1.8.8.1
  55. Date: Wed, 04 Jan 2012 07:02:40 GMT
  56. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  57. Supported: replaces, timer
  58. Content-Type: application/sdp
  59. Content-Length: 283
  60.  
  61. v=0
  62. o=root 316975547 316975547 IN IP4 192.168.1.109
  63. s=Asterisk PBX 1.8.8.1
  64. c=IN IP4 192.168.1.109
  65. t=0 0
  66. m=audio 17764 RTP/AVP 3 0 8 101
  67. a=rtpmap:3 GSM/8000
  68. a=rtpmap:0 PCMU/8000
  69. a=rtpmap:8 PCMA/8000
  70. a=rtpmap:101 telephone-event/8000
  71. a=fmtp:101 0-16
  72. a=ptime:20
  73. a=sendrecv
  74.  
  75. ---
  76.  
  77. <--- SIP read from UDP:192.168.1.55:5060 --->
  78. SIP/2.0 100 Trying
  79. Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK31b2fdfd;rport=5060
  80. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  81. To: <sip:9221536790@192.168.1.55>
  82. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  83. CSeq: 102 INVITE
  84. Server: Sipwise NGCP LB 2.X
  85. Content-Length: 0
  86.  
  87. <------------->
  88. --- (8 headers 0 lines) ---
  89.  
  90. <--- SIP read from UDP:192.168.1.55:5060 --->
  91. SIP/2.0 407 Proxy Authentication Required
  92. Via: SIP/2.0/UDP 192.168.1.109:5060;rport=5060;branch=z9hG4bK31b2fdfd
  93. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  94. To: <sip:9221536790@192.168.1.55>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.cd9c
  95. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  96. CSeq: 102 INVITE
  97. Proxy-Authenticate: Digest realm="anonymous.invalid", nonce="TwP6vE8D+ZCxFT8cujmaiPxNP3F7TJR4"
  98. Server: Sipwise NGCP Proxy 2.X
  99. Content-Length: 0
  100.  
  101. <------------->
  102. --- (9 headers 0 lines) ---
  103. Transmitting (no NAT) to 192.168.1.55:5060:
  104. ACK sip:9221536790@192.168.1.55 SIP/2.0
  105. Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK31b2fdfd
  106. Max-Forwards: 70
  107. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  108. To: <sip:9221536790@192.168.1.55>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.cd9c
  109. Contact: <sip:test02@192.168.1.109:5060>
  110. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  111. CSeq: 102 ACK
  112. User-Agent: Asterisk PBX 1.8.8.1
  113. Content-Length: 0
  114.  
  115.  
  116. ---
  117. Audio is at 5060
  118. Adding codec 0x2 (gsm) to SDP
  119. Adding codec 0x4 (ulaw) to SDP
  120. Adding codec 0x8 (alaw) to SDP
  121. Adding codec 0x800000000000 (testlaw) to SDP
  122. Adding non-codec 0x1 (telephone-event) to SDP
  123. Reliably Transmitting (no NAT) to 192.168.1.55:5060:
  124. INVITE sip:9221536790@192.168.1.55 SIP/2.0
  125. Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK1efe7d0c
  126. Max-Forwards: 70
  127. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  128. To: <sip:9221536790@192.168.1.55>
  129. Contact: <sip:test02@192.168.1.109:5060>
  130. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  131. CSeq: 103 INVITE
  132. User-Agent: Asterisk PBX 1.8.8.1
  133. Proxy-Authorization: Digest username="test02", realm="anonymous.invalid", algorithm=MD5, uri="sip:9221536790@192.168.1.55", nonce="TwP6vE8D+ZCxFT8cujmaiPxNP3F7TJR4", response="c56f3b0172e4bd15a03fc032ba1dec3e"
  134. Date: Wed, 04 Jan 2012 07:02:40 GMT
  135. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  136. Supported: replaces, timer
  137. Content-Type: application/sdp
  138. Content-Length: 283
  139.  
  140. v=0
  141. o=root 316975547 316975548 IN IP4 192.168.1.109
  142. s=Asterisk PBX 1.8.8.1
  143. c=IN IP4 192.168.1.109
  144. t=0 0
  145. m=audio 17764 RTP/AVP 3 0 8 101
  146. a=rtpmap:3 GSM/8000
  147. a=rtpmap:0 PCMU/8000
  148. a=rtpmap:8 PCMA/8000
  149. a=rtpmap:101 telephone-event/8000
  150. a=fmtp:101 0-16
  151. a=ptime:20
  152. a=sendrecv
  153.  
  154. ---
  155.  
  156. <--- SIP read from UDP:192.168.1.55:5060 --->
  157. SIP/2.0 100 Trying
  158. Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK1efe7d0c;rport=5060
  159. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  160. To: <sip:9221536790@192.168.1.55>
  161. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  162. CSeq: 103 INVITE
  163. Server: Sipwise NGCP LB 2.X
  164. Content-Length: 0
  165.  
  166. <------------->
  167. --- (8 headers 0 lines) ---
  168.  
  169. <--- SIP read from UDP:192.168.1.55:5060 --->
  170. SIP/2.0 407 Proxy Authentication Required
  171. Via: SIP/2.0/UDP 192.168.1.109:5060;rport=5060;branch=z9hG4bK1efe7d0c
  172. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  173. To: <sip:9221536790@192.168.1.55>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.6705
  174. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  175. CSeq: 103 INVITE
  176. Proxy-Authenticate: Digest realm="anonymous.invalid", nonce="TwP6vE8D+ZCxFT8cujmaiPxNP3F7TJR4"
  177. Server: Sipwise NGCP Proxy 2.X
  178. Content-Length: 0
  179.  
  180. <------------->
  181. --- (9 headers 0 lines) ---
  182. Transmitting (no NAT) to 192.168.1.55:5060:
  183. ACK sip:9221536790@192.168.1.55 SIP/2.0
  184. Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK1efe7d0c
  185. Max-Forwards: 70
  186. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  187. To: <sip:9221536790@192.168.1.55>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.6705
  188. Contact: <sip:test02@192.168.1.109:5060>
  189. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  190. CSeq: 103 ACK
  191. User-Agent: Asterisk PBX 1.8.8.1
  192. Content-Length: 0
  193.  
  194.  
  195. ---
  196. Audio is at 5060
  197. Adding codec 0x2 (gsm) to SDP
  198. Adding codec 0x4 (ulaw) to SDP
  199. Adding codec 0x8 (alaw) to SDP
  200. Adding codec 0x800000000000 (testlaw) to SDP
  201. Adding non-codec 0x1 (telephone-event) to SDP
  202. Reliably Transmitting (no NAT) to 192.168.1.55:5060:
  203. INVITE sip:9221536790@192.168.1.55 SIP/2.0
  204. Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK1b310208
  205. Max-Forwards: 70
  206. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  207. To: <sip:9221536790@192.168.1.55>
  208. Contact: <sip:test02@192.168.1.109:5060>
  209. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  210. CSeq: 104 INVITE
  211. User-Agent: Asterisk PBX 1.8.8.1
  212. Proxy-Authorization: Digest username="test02", realm="anonymous.invalid", algorithm=MD5, uri="sip:9221536790@192.168.1.55", nonce="TwP6vE8D+ZCxFT8cujmaiPxNP3F7TJR4", response="c56f3b0172e4bd15a03fc032ba1dec3e"
  213. Date: Wed, 04 Jan 2012 07:02:40 GMT
  214. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  215. Supported: replaces, timer
  216. Content-Type: application/sdp
  217. Content-Length: 283
  218.  
  219. v=0
  220. o=root 316975547 316975549 IN IP4 192.168.1.109
  221. s=Asterisk PBX 1.8.8.1
  222. c=IN IP4 192.168.1.109
  223. t=0 0
  224. m=audio 17764 RTP/AVP 3 0 8 101
  225. a=rtpmap:3 GSM/8000
  226. a=rtpmap:0 PCMU/8000
  227. a=rtpmap:8 PCMA/8000
  228. a=rtpmap:101 telephone-event/8000
  229. a=fmtp:101 0-16
  230. a=ptime:20
  231. a=sendrecv
  232.  
  233. ---
  234.  
  235. <--- SIP read from UDP:192.168.1.55:5060 --->
  236. SIP/2.0 100 Trying
  237. Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK1b310208;rport=5060
  238. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  239. To: <sip:9221536790@192.168.1.55>
  240. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  241. CSeq: 104 INVITE
  242. Server: Sipwise NGCP LB 2.X
  243. Content-Length: 0
  244.  
  245. <------------->
  246. --- (8 headers 0 lines) ---
  247.  
  248. <--- SIP read from UDP:192.168.1.55:5060 --->
  249. SIP/2.0 407 Proxy Authentication Required
  250. Via: SIP/2.0/UDP 192.168.1.109:5060;rport=5060;branch=z9hG4bK1b310208
  251. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  252. To: <sip:9221536790@192.168.1.55>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.6e24
  253. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  254. CSeq: 104 INVITE
  255. Proxy-Authenticate: Digest realm="anonymous.invalid", nonce="TwP6vE8D+ZCxFT8cujmaiPxNP3F7TJR4"
  256. Server: Sipwise NGCP Proxy 2.X
  257. Content-Length: 0
  258.  
  259. <------------->
  260. --- (9 headers 0 lines) ---
  261. Transmitting (no NAT) to 192.168.1.55:5060:
  262. ACK sip:9221536790@192.168.1.55 SIP/2.0
  263. Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK1b310208
  264. Max-Forwards: 70
  265. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  266. To: <sip:9221536790@192.168.1.55>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.6e24
  267. Contact: <sip:test02@192.168.1.109:5060>
  268. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  269. CSeq: 104 ACK
  270. User-Agent: Asterisk PBX 1.8.8.1
  271. Content-Length: 0
  272.  
  273.  
  274. ---
  275. Audio is at 5060
  276. Adding codec 0x2 (gsm) to SDP
  277. Adding codec 0x4 (ulaw) to SDP
  278. Adding codec 0x8 (alaw) to SDP
  279. Adding codec 0x800000000000 (testlaw) to SDP
  280. Adding non-codec 0x1 (telephone-event) to SDP
  281. Reliably Transmitting (no NAT) to 192.168.1.55:5060:
  282. INVITE sip:9221536790@192.168.1.55 SIP/2.0
  283. Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK30a7c983
  284. Max-Forwards: 70
  285. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  286. To: <sip:9221536790@192.168.1.55>
  287. Contact: <sip:test02@192.168.1.109:5060>
  288. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  289. CSeq: 105 INVITE
  290. User-Agent: Asterisk PBX 1.8.8.1
  291. Proxy-Authorization: Digest username="test02", realm="anonymous.invalid", algorithm=MD5, uri="sip:9221536790@192.168.1.55", nonce="TwP6vE8D+ZCxFT8cujmaiPxNP3F7TJR4", response="c56f3b0172e4bd15a03fc032ba1dec3e"
  292. Date: Wed, 04 Jan 2012 07:02:40 GMT
  293. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  294. Supported: replaces, timer
  295. Content-Type: application/sdp
  296. Content-Length: 283
  297.  
  298. v=0
  299. o=root 316975547 316975550 IN IP4 192.168.1.109
  300. s=Asterisk PBX 1.8.8.1
  301. c=IN IP4 192.168.1.109
  302. t=0 0
  303. m=audio 17764 RTP/AVP 3 0 8 101
  304. a=rtpmap:3 GSM/8000
  305. a=rtpmap:0 PCMU/8000
  306. a=rtpmap:8 PCMA/8000
  307. a=rtpmap:101 telephone-event/8000
  308. a=fmtp:101 0-16
  309. a=ptime:20
  310. a=sendrecv
  311.  
  312. ---
  313.  
  314. <--- SIP read from UDP:192.168.1.55:5060 --->
  315. SIP/2.0 100 Trying
  316. Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK30a7c983;rport=5060
  317. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  318. To: <sip:9221536790@192.168.1.55>
  319. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  320. CSeq: 105 INVITE
  321. Server: Sipwise NGCP LB 2.X
  322. Content-Length: 0
  323.  
  324. <------------->
  325. --- (8 headers 0 lines) ---
  326.  
  327. <--- SIP read from UDP:192.168.1.55:5060 --->
  328. SIP/2.0 407 Proxy Authentication Required
  329. Via: SIP/2.0/UDP 192.168.1.109:5060;rport=5060;branch=z9hG4bK30a7c983
  330. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  331. To: <sip:9221536790@192.168.1.55>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.bbf7
  332. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  333. CSeq: 105 INVITE
  334. Proxy-Authenticate: Digest realm="anonymous.invalid", nonce="TwP6vE8D+ZCxFT8cujmaiPxNP3F7TJR4"
  335. Server: Sipwise NGCP Proxy 2.X
  336. Content-Length: 0
  337.  
  338. <------------->
  339. --- (9 headers 0 lines) ---
  340. Transmitting (no NAT) to 192.168.1.55:5060:
  341. ACK sip:9221536790@192.168.1.55 SIP/2.0
  342. Via: SIP/2.0/UDP 192.168.1.109:5060;branch=z9hG4bK30a7c983
  343. Max-Forwards: 70
  344. From: "Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289
  345. To: <sip:9221536790@192.168.1.55>;tag=1d24a28a0bded6c40d31e6db8aab9ac6.bbf7
  346. Contact: <sip:test02@192.168.1.109:5060>
  347. Call-ID: 04e6052323d0123c465ad274103bd93b@192.168.1.109:5060
  348. CSeq: 105 ACK
  349. User-Agent: Asterisk PBX 1.8.8.1
  350. Content-Length: 0
  351.  
  352.  
  353. ---
  354. [Jan 4 10:02:40] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to '"Anonymous" <sip:test02@anonymous.invalid>;tag=as4c200289'
  355. ast18*CLI> quit
  356. root@ast18:~#
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