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  1.  
  2. s-tel*CLI> sip set debug ip 90.188.9.102
  3. SIP Debugging Enabled for IP: 90.188.9.102
  4. -- SIP/as5350-2-000010b4 answered SIP/7500025-000010b3
  5.  
  6. <--- SIP read from 90.188.9.102:5060 --->
  7. REGISTER sip:212.109.194.3 SIP/2.0
  8. Via: SIP/2.0/UDP 90.188.9.102:5060;branch=z9hG4bK3733e33d
  9. Max-Forwards: 70
  10. From: <sip:7500004@212.109.194.3>;tag=as2c666496
  11. To: <sip:7500004@212.109.194.3>
  12. Call-ID: 61a2fdd35abedeca326497547c6a2476@212.109.194.3
  13. CSeq: 121 REGISTER
  14. User-Agent: Asterisk PBX 1.8.10.1
  15. Authorization: Digest username="7500004", realm="s-tel.ru", algorithm=MD5, uri="sip:212.109.194.3", nonce="7537ac4d", response="a33577c12c133d7f51e3e15689c0d815"
  16. Expires: 120
  17. Contact: <sip:7500004@90.188.9.102:5060>
  18. Content-Length: 0
  19.  
  20.  
  21. <------------->
  22. --- (12 headers 0 lines) ---
  23. Using latest REGISTER request as basis request
  24. Sending to 90.188.9.102 : 5060 (no NAT)
  25.  
  26. <--- Transmitting (NAT) to 90.188.9.102:5060 --->
  27. SIP/2.0 100 Trying
  28. v: SIP/2.0/UDP 90.188.9.102:5060;branch=z9hG4bK3733e33d;received=90.188.9.102
  29. f: <sip:7500004@212.109.194.3>;tag=as2c666496
  30. t: <sip:7500004@212.109.194.3>
  31. i: 61a2fdd35abedeca326497547c6a2476@212.109.194.3
  32. CSeq: 121 REGISTER
  33. User-Agent: Asterisk PBX
  34. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  35. k: replaces
  36. l: 0
  37.  
  38.  
  39. <------------>
  40.  
  41. <--- Transmitting (NAT) to 90.188.9.102:5060 --->
  42. SIP/2.0 401 Unauthorized
  43. v: SIP/2.0/UDP 90.188.9.102:5060;branch=z9hG4bK3733e33d;received=90.188.9.102
  44. f: <sip:7500004@212.109.194.3>;tag=as2c666496
  45. t: <sip:7500004@212.109.194.3>;tag=as1fc5db73
  46. i: 61a2fdd35abedeca326497547c6a2476@212.109.194.3
  47. CSeq: 121 REGISTER
  48. User-Agent: Asterisk PBX
  49. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  50. k: replaces
  51. WWW-Authenticate: Digest algorithm=MD5, realm="s-tel.ru", nonce="5d2b4bb8"
  52. l: 0
  53.  
  54.  
  55. <------------>
  56. Scheduling destruction of SIP dialog '61a2fdd35abedeca326497547c6a2476@212.109.194.3' in 32000 ms (Method: REGISTER)
  57.  
  58. <--- SIP read from 90.188.9.102:5060 --->
  59. REGISTER sip:212.109.194.3 SIP/2.0
  60. Via: SIP/2.0/UDP 90.188.9.102:5060;branch=z9hG4bK2014686a
  61. Max-Forwards: 70
  62. From: <sip:7500004@212.109.194.3>;tag=as6ea44388
  63. To: <sip:7500004@212.109.194.3>
  64. Call-ID: 61a2fdd35abedeca326497547c6a2476@212.109.194.3
  65. CSeq: 122 REGISTER
  66. User-Agent: Asterisk PBX 1.8.10.1
  67. Authorization: Digest username="7500004", realm="s-tel.ru", algorithm=MD5, uri="sip:212.109.194.3", nonce="5d2b4bb8", response="94cbf174ce7c81be3c68a084746a2272"
  68. Expires: 120
  69. Contact: <sip:7500004@90.188.9.102:5060>
  70. Content-Length: 0
  71.  
  72.  
  73. <------------->
  74. --- (12 headers 0 lines) ---
  75. Using latest REGISTER request as basis request
  76. Sending to 90.188.9.102 : 5060 (NAT)
  77.  
  78. <--- Transmitting (NAT) to 90.188.9.102:5060 --->
  79. SIP/2.0 100 Trying
  80. v: SIP/2.0/UDP 90.188.9.102:5060;branch=z9hG4bK2014686a;received=90.188.9.102
  81. f: <sip:7500004@212.109.194.3>;tag=as6ea44388
  82. t: <sip:7500004@212.109.194.3>
  83. i: 61a2fdd35abedeca326497547c6a2476@212.109.194.3
  84. CSeq: 122 REGISTER
  85. User-Agent: Asterisk PBX
  86. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  87. k: replaces
  88. l: 0
  89.  
  90.  
  91. <------------>
  92. Reliably Transmitting (NAT) to 90.188.9.102:5060:
  93. OPTIONS sip:7500004@90.188.9.102:5060 SIP/2.0
  94. v: SIP/2.0/UDP 212.109.194.3:5060;branch=z9hG4bK2c4bd415;rport
  95. f: "Unknown" <sip:Unknown@212.109.194.3>;tag=as47f7528d
  96. t: <sip:7500004@90.188.9.102:5060>
  97. m: <sip:Unknown@212.109.194.3>
  98. i: 32db2a483822842d109a719669943082@212.109.194.3
  99. CSeq: 102 OPTIONS
  100. User-Agent: Asterisk PBX
  101. Max-Forwards: 70
  102. Date: Wed, 22 Aug 2012 02:54:59 GMT
  103. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  104. k: replaces
  105. l: 0
  106.  
  107.  
  108. ---
  109.  
  110. <--- Transmitting (NAT) to 90.188.9.102:5060 --->
  111. SIP/2.0 200 OK
  112. v: SIP/2.0/UDP 90.188.9.102:5060;branch=z9hG4bK2014686a;received=90.188.9.102
  113. f: <sip:7500004@212.109.194.3>;tag=as6ea44388
  114. t: <sip:7500004@212.109.194.3>;tag=as1fc5db73
  115. i: 61a2fdd35abedeca326497547c6a2476@212.109.194.3
  116. CSeq: 122 REGISTER
  117. User-Agent: Asterisk PBX
  118. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  119. k: replaces
  120. Expires: 120
  121. m: <sip:7500004@90.188.9.102:5060>;expires=120
  122. Date: Wed, 22 Aug 2012 02:54:59 GMT
  123. l: 0
  124.  
  125.  
  126. <------------>
  127. Scheduling destruction of SIP dialog '61a2fdd35abedeca326497547c6a2476@212.109.194.3' in 32000 ms (Method: REGISTER)
  128.  
  129. <--- SIP read from 90.188.9.102:5060 --->
  130. SIP/2.0 200 OK
  131. Via: SIP/2.0/UDP 212.109.194.3:5060;branch=z9hG4bK2c4bd415;rport;received=212.109.194.3
  132. From: "Unknown" <sip:Unknown@212.109.194.3>;tag=as47f7528d
  133. To: <sip:7500004@90.188.9.102:5060>;tag=as7da60e80
  134. Call-ID: 32db2a483822842d109a719669943082@212.109.194.3
  135. CSeq: 102 OPTIONS
  136. Server: Asterisk PBX 1.8.10.1
  137. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  138. Supported: replaces, timer
  139. Contact: <sip:90.188.9.102:5060>
  140. Accept: application/sdp
  141. Content-Length: 0
  142.  
  143.  
  144. <------------->
  145. --- (12 headers 0 lines) ---
  146. Really destroying SIP dialog '32db2a483822842d109a719669943082@212.109.194.3' Method: OPTIONS
  147. -- Executing [7500004@default:1] AGI("SIP/7580101-000010b5", "agi-rad-auth.agi|Mode=Account&CLID=7580101&DNID=7500004") in new stack
  148. -- Launched AGI Script /usr/local/share/asterisk/agi-bin/agi-rad-auth.agi
  149. -- agi-rad-auth.agi|Mode=Account&CLID=7580101&DNID=7500004: CLLR2 = , CLID2 = , DNID2 =
  150. -- agi-rad-auth.agi|Mode=Account&CLID=7580101&DNID=7500004: not Pool + not clid2
  151. -- agi-rad-auth.agi|Mode=Account&CLID=7580101&DNID=7500004: CLLR = 7580101, CLID = 7580101, DNID = 7500004, Password = 58d14sd31
  152. -- agi-rad-auth.agi|Mode=Account&CLID=7580101&DNID=7500004: params{'CLID'} = 7580101, UserName = 7580101
  153. -- agi-rad-auth.agi|Mode=Account&CLID=7580101&DNID=7500004: RADIUS server response type = 2
  154. -- agi-rad-auth.agi|Mode=Account&CLID=7580101&DNID=7500004: attr: name=Service-Type value=Framed-User
  155. -- agi-rad-auth.agi|Mode=Account&CLID=7580101&DNID=7500004: attr: name=Framed-Protocol value=PPP
  156. -- agi-rad-auth.agi|Mode=Account&CLID=7580101&DNID=7500004: attr: name=Acct-Interim-Interval value=100
  157. -- agi-rad-auth.agi|Mode=Account&CLID=7580101&DNID=7500004: attr: name=h323-credit-amount value=21000.00
  158. -- agi-rad-auth.agi|Mode=Account&CLID=7580101&DNID=7500004: attr: name=h323-credit-time value=210909
  159. -- agi-rad-auth.agi|Mode=Account&CLID=7580101&DNID=7500004: attr: name=h323-return-code value=0
  160. -- agi-rad-auth.agi|Mode=Account&CLID=7580101&DNID=7500004: attr: name=h323-currency value=USD
  161. -- AGI Script Executing Application: (UserEvent) Options: (_SIP_Auth|User-Name: 7580101|CLID: 7580101|DNID: 7500004|Channel: SIP/7580101-000010b5)
  162. -- AGI Script agi-rad-auth.agi completed, returning 0
  163. -- Executing [7500004@default:2] Goto("SIP/7580101-000010b5", "dial:default|7500004|1") in new stack
  164. -- Goto (dial:default,7500004,1)
  165. -- Executing [7500004@dial:default:1] GotoIf("SIP/7580101-000010b5", "1?5:2") in new stack
  166. -- Goto (dial:default,7500004,5)
  167. -- Executing [7500004@dial:default:5] Goto("SIP/7580101-000010b5", "dial:default-2|7500004|1") in new stack
  168. -- Goto (dial:default-2,7500004,1)
  169. -- Executing [7500004@dial:default-2:1] Dial("SIP/7580101-000010b5", "SIP/7500004|25|tT") in new stack
  170. Video is at 212.109.194.3 port 14796
  171. Audio is at 212.109.194.3 port 13840
  172. Adding codec 0x8 (alaw) to SDP
  173. Adding codec 0x800 (g726) to SDP
  174. Adding codec 0x400 (ilbc) to SDP
  175. Adding codec 0x100 (g729) to SDP
  176. Adding codec 0x2 (gsm) to SDP
  177. Adding codec 0x80000 (h263) to SDP
  178. Adding non-codec 0x1 (telephone-event) to SDP
  179. Reliably Transmitting (NAT) to 90.188.9.102:5060:
  180. INVITE sip:7500004@90.188.9.102:5060 SIP/2.0
  181. v: SIP/2.0/UDP 212.109.194.3:5060;branch=z9hG4bK0a097a6b;rport
  182. f: "7580101" <sip:73832580101@212.109.194.3>;tag=as3f795fc0
  183. t: <sip:7500004@90.188.9.102:5060>
  184. m: <sip:73832580101@212.109.194.3>
  185. i: 008502dc42aa6f1816f8d98b7c06eefd@212.109.194.3
  186. CSeq: 102 INVITE
  187. User-Agent: Asterisk PBX
  188. Max-Forwards: 70
  189. Date: Wed, 22 Aug 2012 02:55:01 GMT
  190. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  191. k: replaces
  192. c: application/sdp
  193. l: 457
  194.  
  195. v=0
  196. o=root 765 765 IN IP4 212.109.194.3
  197. s=session
  198. c=IN IP4 212.109.194.3
  199. b=CT:2048
  200. t=0 0
  201. m=audio 13840 RTP/AVP 8 111 97 18 3 101
  202. a=rtpmap:8 PCMA/8000
  203. a=rtpmap:111 G726-32/8000
  204. a=rtpmap:97 iLBC/8000
  205. a=fmtp:97 mode=30
  206. a=rtpmap:18 G729/8000
  207. a=fmtp:18 annexb=no
  208. a=rtpmap:3 GSM/8000
  209. a=rtpmap:101 telephone-event/8000
  210. a=fmtp:101 0-16
  211. a=silenceSupp:off - - - -
  212. a=ptime:20
  213. a=sendrecv
  214. m=video 14796 RTP/AVP 34
  215. a=rtpmap:34 H263/90000
  216. a=sendrecv
  217.  
  218. ---
  219. -- Called 7500004
  220.  
  221. <--- SIP read from 90.188.9.102:5060 --->
  222. SIP/2.0 488 Not acceptable here
  223. Via: SIP/2.0/UDP 212.109.194.3:5060;branch=z9hG4bK0a097a6b;received=212.109.194.3;rport=5060
  224. From: "7580101" <sip:73832580101@212.109.194.3>;tag=as3f795fc0
  225. To: <sip:7500004@90.188.9.102:5060>;tag=as43b1c572
  226. Call-ID: 008502dc42aa6f1816f8d98b7c06eefd@212.109.194.3
  227. CSeq: 102 INVITE
  228. Server: Asterisk PBX 1.8.10.1
  229. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  230. Supported: replaces, timer
  231. Content-Length: 0
  232.  
  233.  
  234. <------------->
  235. --- (10 headers 0 lines) ---
  236. Transmitting (NAT) to 90.188.9.102:5060:
  237. ACK sip:7500004@90.188.9.102:5060 SIP/2.0
  238. v: SIP/2.0/UDP 212.109.194.3:5060;branch=z9hG4bK0a097a6b;rport
  239. f: "7580101" <sip:73832580101@212.109.194.3>;tag=as3f795fc0
  240. t: <sip:7500004@90.188.9.102:5060>;tag=as43b1c572
  241. m: <sip:73832580101@212.109.194.3>
  242. i: 008502dc42aa6f1816f8d98b7c06eefd@212.109.194.3
  243. CSeq: 102 ACK
  244. User-Agent: Asterisk PBX
  245. Max-Forwards: 70
  246. l: 0
  247.  
  248.  
  249. ---
  250. -- SIP/7500004-000010b6 is circuit-busy
  251. == Everyone is busy/congested at this time (1:0/1/0)
  252. -- Executing [7500004@dial:default-2:2] Playback("SIP/7580101-000010b5", "/usr/local/share/asterisk/sounds/ru/nomer&/usr/local/share/asterisk/sounds/ru/ne-otvechaet") in new stack
  253. -- <SIP/7580101-000010b5> Playing '/usr/local/share/asterisk/sounds/ru/nomer' (language 'ru')
  254. Really destroying SIP dialog '008502dc42aa6f1816f8d98b7c06eefd@212.109.194.3' Method: INVITE
  255. -- <SIP/7580101-000010b5> Playing '/usr/local/share/asterisk/sounds/ru/ne-otvechaet' (language 'ru')
  256. -- Executing [7500004@dial:default-2:3] VoiceMail("SIP/7580101-000010b5", "7500004@voice_mailbox") in new stack
  257. -- <SIP/7580101-000010b5> Playing 'vm-intro' (language 'ru')
  258. Scheduling destruction of SIP dialog '1605251615757e747632af826ef1d695@212.109.194.3' in 6400 ms (Method: NOTIFY)
  259. Reliably Transmitting (NAT) to 90.188.9.102:5060:
  260. NOTIFY sip:7500004@90.188.9.102:5060 SIP/2.0
  261. v: SIP/2.0/UDP 212.109.194.3:5060;branch=z9hG4bK096213ef;rport
  262. f: "Unknown" <sip:Unknown@212.109.194.3>;tag=as2d6c2bd4
  263. t: <sip:7500004@90.188.9.102:5060>
  264. m: <sip:Unknown@212.109.194.3>
  265. i: 1605251615757e747632af826ef1d695@212.109.194.3
  266. CSeq: 102 NOTIFY
  267. User-Agent: Asterisk PBX
  268. Max-Forwards: 70
  269. o: message-summary
  270. c: application/simple-message-summary
  271. l: 96
  272.  
  273. Messages-Waiting: yes
  274. Message-Account: sip:asterisk@212.109.194.3
  275. Voice-Message: 100/0 (0/0)
  276.  
  277. ---
  278.  
  279. <--- SIP read from 90.188.9.102:5060 --->
  280. SIP/2.0 489 Bad event
  281. Via: SIP/2.0/UDP 212.109.194.3:5060;branch=z9hG4bK096213ef;rport;received=212.109.194.3
  282. From: "Unknown" <sip:Unknown@212.109.194.3>;tag=as2d6c2bd4
  283. To: <sip:7500004@90.188.9.102:5060>;tag=as2a902b5a
  284. Call-ID: 1605251615757e747632af826ef1d695@212.109.194.3
  285. CSeq: 102 NOTIFY
  286. Server: Asterisk PBX 1.8.10.1
  287. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  288. Supported: replaces, timer
  289. Content-Length: 0
  290.  
  291.  
  292. <------------->
  293. --- (10 headers 0 lines) ---
  294. -- Got SIP response 489 "Bad event" back from 90.188.9.102
  295. Really destroying SIP dialog '1605251615757e747632af826ef1d695@212.109.194.3' Method: NOTIFY
  296. == Spawn extension (dial:default-2, 7500004, 3) exited non-zero on 'SIP/7580101-000010b5'
  297. s-tel*CLI> sip set debug off
  298. SIP Debugging Disabled
  299. s-tel*CLI>
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