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  1. <--- SIP read from UDP:192.168.2.5:5060 --->
  2. INVITE sip:15173386543@vancouver.example.com:5060;user=phone SIP/2.0
  3. Record-Route: <sip:toronto.example.com;lr;did=238.982885e6>
  4. Via: SIP/2.0/UDP toronto.example.com;branch=z9hG4bK9686.20e87cf4.0
  5. Via: SIP/2.0/UDP 192.168.2.11;rport=5060;received=192.168.2.11;branch=z9hG4bKd16566c393A5426A
  6. From: "1001" <sip:1001@toronto.example.com>;tag=92E5E5E5-E154756
  7. To: <sip:15173386543@toronto.example.com;user=phone>
  8. CSeq: 1 INVITE
  9. Call-ID: 31535a59-40a0580f-ee00c050@192.168.2.11
  10. Contact: <sip:1001@192.168.2.11;nat=yes>
  11. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  12. User-Agent: PolycomSoundPointIP-SPIP_301-UA/3.1.8.0070
  13. Accept-Language: en
  14. Supported: 100rel,replaces
  15. Allow-Events: talk,hold,conference
  16. Max-Forwards: 69
  17. Content-Type: application/sdp
  18. Content-Length: 288
  19.  
  20. v=0
  21. o=- 1357502784 1357502784 IN IP4 192.168.2.11
  22. s=Polycom IP Phone
  23. c=IN IP4 192.168.2.5
  24. t=0 0
  25. a=sendrecv
  26. m=audio 18252 RTP/AVP 0 8 18 101
  27. a=rtpmap:0 PCMU/8000
  28. a=rtpmap:8 PCMA/8000
  29. a=rtpmap:18 G729/8000
  30. a=fmtp:18 annexb=no
  31. a=rtpmap:101 telephone-event/8000
  32. a=nortpproxy:yes
  33. <------------->
  34. --- (17 headers 13 lines) ---
  35. Sending to 192.168.2.5:5060 (no NAT)
  36. Using INVITE request as basis request - 31535a59-40a0580f-ee00c050@192.168.2.11
  37. Found peer '1001' for '1001' from 192.168.2.5:5060
  38. == Using SIP RTP CoS mark 5
  39. Found RTP audio format 0
  40. Found RTP audio format 8
  41. Found RTP audio format 18
  42. Found RTP audio format 101
  43. Found audio description format PCMU for ID 0
  44. Found audio description format PCMA for ID 8
  45. Found audio description format G729 for ID 18
  46. Found audio description format telephone-event for ID 101
  47. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
  48. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  49. Peer audio RTP is at port 192.168.2.5:18252
  50. Looking for 15173386543 in context-from-toronto (domain vancouver.example.com:5060)
  51. list_route: hop: <sip:toronto.example.com;lr;did=238.982885e6>
  52.  
  53. <--- Transmitting (NAT) to 192.168.2.5:5060 --->
  54. SIP/2.0 100 Trying
  55. Via: SIP/2.0/UDP toronto.example.com;branch=z9hG4bK9686.20e87cf4.0;received=192.168.2.5;rport=5060
  56. Via: SIP/2.0/UDP 192.168.2.11;rport=5060;received=192.168.2.11;branch=z9hG4bKd16566c393A5426A
  57. Record-Route: <sip:toronto.example.com;lr;did=238.982885e6>
  58. From: "1001" <sip:1001@toronto.example.com>;tag=92E5E5E5-E154756
  59. To: <sip:15173386543@toronto.example.com;user=phone>
  60. Call-ID: 31535a59-40a0580f-ee00c050@192.168.2.11
  61. CSeq: 1 INVITE
  62. Server: Asterisk PBX UNKNOWN__and_probably_unsupported
  63. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  64. Supported: replaces, timer
  65. Contact: <sip:15173386543@192.168.2.10:5060>
  66. Content-Length: 0
  67.  
  68.  
  69. <------------>
  70. -- Executing [15173386543@context-from-toronto:1] Dial("SIP/1001-00000002", "SIP/001110315173386543@sbc.sipcarrier.com")
  71. == Using SIP RTP CoS mark 5
  72. Audio is at 5060
  73. Adding codec 0x8 (alaw) to SDP
  74. Adding codec 0x2 (gsm) to SDP
  75. Adding codec 0x4 (ulaw) to SDP
  76. Adding codec 0x800000000000 (testlaw) to SDP
  77. Adding non-codec 0x1 (telephone-event) to SDP
  78. Reliably Transmitting (no NAT) to 108.59.2.133:5060:
  79. INVITE sip:001110315173386543@sbc.sipcarrier.com SIP/2.0
  80. Via: SIP/2.0/UDP 23.120.227.82:5060;branch=z9hG4bK6d33a6e1
  81. Max-Forwards: 70
  82. From: "Nick Camisos" <sip:5173386543@23.120.227.82>;tag=as023b7a00
  83. To: <sip:001110315173386543@sbc.sipcarrier.com>
  84. Contact: <sip:5173386543@23.120.227.82:5060>
  85. Call-ID: 666ec4863b6bca2e13d6ec6b521b39ba@23.120.227.82:5060
  86. CSeq: 102 INVITE
  87. User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported
  88. Date: Wed, 15 Dec 1999 20:37:26 GMT
  89. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  90. Supported: replaces, timer
  91. Content-Type: application/sdp
  92. Content-Length: 336
  93.  
  94. v=0
  95. o=root 728580775 728580775 IN IP4 23.120.227.82
  96. s=Asterisk PBX UNKNOWN__and_probably_unsupported
  97. c=IN IP4 23.120.227.82
  98. t=0 0
  99. m=audio 63592 RTP/AVP 8 3 0 101
  100. a=rtpmap:8 PCMA/8000
  101. a=rtpmap:3 GSM/8000
  102. a=rtpmap:0 PCMU/8000
  103. a=rtpmap:101 telephone-event/8000
  104. a=fmtp:101 0-16
  105. a=silenceSupp:off - - - -
  106. a=ptime:20
  107. a=sendrecv
  108.  
  109. ---
  110. -- Called SIP/001110315173386543@sbc.sipcarrier.com
  111. Retransmitting #1 (no NAT) to 108.59.2.133:5060:
  112. INVITE sip:001110315173386543@sbc.sipcarrier.com SIP/2.0
  113. Via: SIP/2.0/UDP 23.120.227.82:5060;branch=z9hG4bK6d33a6e1
  114. Max-Forwards: 70
  115. From: "Nick Camisos" <sip:5173386543@23.120.227.82>;tag=as023b7a00
  116. To: <sip:001110315173386543@sbc.sipcarrier.com>
  117. Contact: <sip:5173386543@23.120.227.82:5060>
  118. Call-ID: 666ec4863b6bca2e13d6ec6b521b39ba@23.120.227.82:5060
  119. CSeq: 102 INVITE
  120. User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported
  121. Date: Wed, 15 Dec 1999 20:37:26 GMT
  122. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  123. Supported: replaces, timer
  124. Content-Type: application/sdp
  125. Content-Length: 336
  126.  
  127. v=0
  128. o=root 728580775 728580775 IN IP4 23.120.227.82
  129. s=Asterisk PBX UNKNOWN__and_probably_unsupported
  130. c=IN IP4 23.120.227.82
  131. t=0 0
  132. m=audio 63592 RTP/AVP 8 3 0 101
  133. a=rtpmap:8 PCMA/8000
  134. a=rtpmap:3 GSM/8000
  135. a=rtpmap:0 PCMU/8000
  136. a=rtpmap:101 telephone-event/8000
  137. a=fmtp:101 0-16
  138. a=silenceSupp:off - - - -
  139. a=ptime:20
  140. a=sendrecv
  141.  
  142. ---
  143. Retransmitting #2 (no NAT) to 108.59.2.133:5060:
  144. INVITE sip:001110315173386543@sbc.sipcarrier.com SIP/2.0
  145. Via: SIP/2.0/UDP 23.120.227.82:5060;branch=z9hG4bK6d33a6e1
  146. Max-Forwards: 70
  147. From: "Nick Camisos" <sip:5173386543@23.120.227.82>;tag=as023b7a00
  148. To: <sip:001110315173386543@sbc.sipcarrier.com>
  149. Contact: <sip:5173386543@23.120.227.82:5060>
  150. Call-ID: 666ec4863b6bca2e13d6ec6b521b39ba@23.120.227.82:5060
  151. CSeq: 102 INVITE
  152. User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported
  153. Date: Wed, 15 Dec 1999 20:37:26 GMT
  154. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  155. Supported: replaces, timer
  156. Content-Type: application/sdp
  157. Content-Length: 336
  158.  
  159. v=0
  160. o=root 728580775 728580775 IN IP4 23.120.227.82
  161. s=Asterisk PBX UNKNOWN__and_probably_unsupported
  162. c=IN IP4 23.120.227.82
  163. t=0 0
  164. m=audio 63592 RTP/AVP 8 3 0 101
  165. a=rtpmap:8 PCMA/8000
  166. a=rtpmap:3 GSM/8000
  167. a=rtpmap:0 PCMU/8000
  168. a=rtpmap:101 telephone-event/8000
  169. a=fmtp:101 0-16
  170. a=silenceSupp:off - - - -
  171. a=ptime:20
  172. a=sendrecv
  173.  
  174. ---
  175. Retransmitting #3 (no NAT) to 108.59.2.133:5060:
  176. INVITE sip:001110315173386543@sbc.sipcarrier.com SIP/2.0
  177. Via: SIP/2.0/UDP 23.120.227.82:5060;branch=z9hG4bK6d33a6e1
  178. Max-Forwards: 70
  179. From: "Nick Camisos" <sip:5173386543@23.120.227.82>;tag=as023b7a00
  180. To: <sip:001110315173386543@sbc.sipcarrier.com>
  181. Contact: <sip:5173386543@23.120.227.82:5060>
  182. Call-ID: 666ec4863b6bca2e13d6ec6b521b39ba@23.120.227.82:5060
  183. CSeq: 102 INVITE
  184. User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported
  185. Date: Wed, 15 Dec 1999 20:37:26 GMT
  186. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  187. Supported: replaces, timer
  188. Content-Type: application/sdp
  189. Content-Length: 336
  190.  
  191. v=0
  192. o=root 728580775 728580775 IN IP4 23.120.227.82
  193. s=Asterisk PBX UNKNOWN__and_probably_unsupported
  194. c=IN IP4 23.120.227.82
  195. t=0 0
  196. m=audio 63592 RTP/AVP 8 3 0 101
  197. a=rtpmap:8 PCMA/8000
  198. a=rtpmap:3 GSM/8000
  199. a=rtpmap:0 PCMU/8000
  200. a=rtpmap:101 telephone-event/8000
  201. a=fmtp:101 0-16
  202. a=silenceSupp:off - - - -
  203. a=ptime:20
  204. a=sendrecv
  205.  
  206. ---
  207. Retransmitting #4 (no NAT) to 108.59.2.133:5060:
  208. INVITE sip:001110315173386543@sbc.sipcarrier.com SIP/2.0
  209. Via: SIP/2.0/UDP 23.120.227.82:5060;branch=z9hG4bK6d33a6e1
  210. Max-Forwards: 70
  211. From: "Nick Camisos" <sip:5173386543@23.120.227.82>;tag=as023b7a00
  212. To: <sip:001110315173386543@sbc.sipcarrier.com>
  213. Contact: <sip:5173386543@23.120.227.82:5060>
  214. Call-ID: 666ec4863b6bca2e13d6ec6b521b39ba@23.120.227.82:5060
  215. CSeq: 102 INVITE
  216. User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported
  217. Date: Wed, 15 Dec 1999 20:37:26 GMT
  218. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  219. Supported: replaces, timer
  220. Content-Type: application/sdp
  221. Content-Length: 336
  222.  
  223. v=0
  224. o=root 728580775 728580775 IN IP4 23.120.227.82
  225. s=Asterisk PBX UNKNOWN__and_probably_unsupported
  226. c=IN IP4 23.120.227.82
  227. t=0 0
  228. m=audio 63592 RTP/AVP 8 3 0 101
  229. a=rtpmap:8 PCMA/8000
  230. a=rtpmap:3 GSM/8000
  231. a=rtpmap:0 PCMU/8000
  232. a=rtpmap:101 telephone-event/8000
  233. a=fmtp:101 0-16
  234. a=silenceSupp:off - - - -
  235. a=ptime:20
  236. a=sendrecv
  237.  
  238. ---
  239. Retransmitting #5 (no NAT) to 108.59.2.133:5060:
  240. INVITE sip:001110315173386543@sbc.sipcarrier.com SIP/2.0
  241. Via: SIP/2.0/UDP 23.120.227.82:5060;branch=z9hG4bK6d33a6e1
  242. Max-Forwards: 70
  243. From: "Nick Camisos" <sip:5173386543@23.120.227.82>;tag=as023b7a00
  244. To: <sip:001110315173386543@sbc.sipcarrier.com>
  245. Contact: <sip:5173386543@23.120.227.82:5060>
  246. Call-ID: 666ec4863b6bca2e13d6ec6b521b39ba@23.120.227.82:5060
  247. CSeq: 102 INVITE
  248. User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported
  249. Date: Wed, 15 Dec 1999 20:37:26 GMT
  250. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  251. Supported: replaces, timer
  252. Content-Type: application/sdp
  253. Content-Length: 336
  254.  
  255. v=0
  256. o=root 728580775 728580775 IN IP4 23.120.227.82
  257. s=Asterisk PBX UNKNOWN__and_probably_unsupported
  258. c=IN IP4 23.120.227.82
  259. t=0 0
  260. m=audio 63592 RTP/AVP 8 3 0 101
  261. a=rtpmap:8 PCMA/8000
  262. a=rtpmap:3 GSM/8000
  263. a=rtpmap:0 PCMU/8000
  264. a=rtpmap:101 telephone-event/8000
  265. a=fmtp:101 0-16
  266. a=silenceSupp:off - - - -
  267. a=ptime:20
  268. a=sendrecv
  269.  
  270. ---
  271.  
  272. <--- SIP read from UDP:192.168.2.5:5060 --->
  273. CANCEL sip:15173386543@vancouver.example.com:5060;user=phone SIP/2.0
  274. Via: SIP/2.0/UDP toronto.example.com;branch=z9hG4bK9686.20e87cf4.0
  275. From: "1001" <sip:1001@toronto.example.com>;tag=92E5E5E5-E154756
  276. Call-ID: 31535a59-40a0580f-ee00c050@192.168.2.11
  277. To: <sip:15173386543@toronto.example.com;user=phone>
  278. CSeq: 1 CANCEL
  279. Max-Forwards: 70
  280. Reason: SIP;cause=487;text="ORIGINATOR_CANCEL"
  281. toronto.example.com
  282. Content-Length: 0
  283.  
  284. <------------->
  285. --- (10 headers 0 lines) ---
  286. Sending to 192.168.2.5:5060 (NAT)
  287.  
  288. <--- Reliably Transmitting (NAT) to 192.168.2.5:5060 --->
  289. SIP/2.0 487 Request Terminated
  290. Via: SIP/2.0/UDP toronto.example.com;branch=z9hG4bK9686.20e87cf4.0;received=192.168.2.5;rport=5060
  291. Via: SIP/2.0/UDP 192.168.2.11;rport=5060;received=192.168.2.11;branch=z9hG4bKd16566c393A5426A
  292. From: "1001" <sip:1001@toronto.example.com>;tag=92E5E5E5-E154756
  293. To: <sip:15173386543@toronto.example.com;user=phone>;tag=as09104933
  294. Call-ID: 31535a59-40a0580f-ee00c050@192.168.2.11
  295. CSeq: 1 INVITE
  296. Server: Asterisk PBX UNKNOWN__and_probably_unsupported
  297. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  298. Supported: replaces, timer
  299. Content-Length: 0
  300.  
  301.  
  302. <------------>
  303.  
  304. <--- Transmitting (NAT) to 192.168.2.5:5060 --->
  305. SIP/2.0 200 OK
  306. Via: SIP/2.0/UDP toronto.example.com;branch=z9hG4bK9686.20e87cf4.0;received=192.168.2.5;rport=5060
  307. From: "1001" <sip:1001@toronto.example.com>;tag=92E5E5E5-E154756
  308. To: <sip:15173386543@toronto.example.com;user=phone>;tag=as09104933
  309. Call-ID: 31535a59-40a0580f-ee00c050@192.168.2.11
  310. CSeq: 1 CANCEL
  311. Server: Asterisk PBX UNKNOWN__and_probably_unsupported
  312. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  313. Supported: replaces, timer
  314. Content-Length: 0
  315.  
  316.  
  317. <------------>
  318. Scheduling destruction of SIP dialog '666ec4863b6bca2e13d6ec6b521b39ba@23.120.227.82:5060' in 32000 ms (Method: INVITE)
  319. == Spawn extension (context-from-toronto, 15173386543, 1) exited non-zero on 'SIP/1001-00000002'
  320.  
  321. <--- SIP read from UDP:192.168.2.5:5060 --->
  322. ACK sip:15173386543@vancouver.example.com:5060;user=phone SIP/2.0
  323. Via: SIP/2.0/UDP toronto.example.com;branch=z9hG4bK9686.20e87cf4.0
  324. From: "1001" <sip:1001@toronto.example.com>;tag=92E5E5E5-E154756
  325. Call-ID: 31535a59-40a0580f-ee00c050@192.168.2.11
  326. To: <sip:15173386543@toronto.example.com;user=phone>;tag=as09104933
  327. CSeq: 1 ACK
  328. Max-Forwards: 70
  329. toronto.example.com
  330. Content-Length: 0
  331.  
  332. <------------->
  333. --- (9 headers 0 lines) ---
  334. Really destroying SIP dialog '31535a59-40a0580f-ee00c050@192.168.2.11' Method: ACK
  335. Retransmitting #6 (no NAT) to 108.59.2.133:5060:
  336. INVITE sip:001110315173386543@sbc.sipcarrier.com SIP/2.0
  337. Via: SIP/2.0/UDP 23.120.227.82:5060;branch=z9hG4bK6d33a6e1
  338. Max-Forwards: 70
  339. From: "Nick Camisos" <sip:5173386543@23.120.227.82>;tag=as023b7a00
  340. To: <sip:001110315173386543@sbc.sipcarrier.com>
  341. Contact: <sip:5173386543@23.120.227.82:5060>
  342. Call-ID: 666ec4863b6bca2e13d6ec6b521b39ba@23.120.227.82:5060
  343. CSeq: 102 INVITE
  344. User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported
  345. Date: Wed, 15 Dec 1999 20:37:26 GMT
  346. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  347. Supported: replaces, timer
  348. Content-Type: application/sdp
  349. Content-Length: 336
  350.  
  351. v=0
  352. o=root 728580775 728580775 IN IP4 23.120.227.82
  353. s=Asterisk PBX UNKNOWN__and_probably_unsupported
  354. c=IN IP4 23.120.227.82
  355. t=0 0
  356. m=audio 63592 RTP/AVP 8 3 0 101
  357. a=rtpmap:8 PCMA/8000
  358. a=rtpmap:3 GSM/8000
  359. a=rtpmap:0 PCMU/8000
  360. a=rtpmap:101 telephone-event/8000
  361. a=fmtp:101 0-16
  362. a=silenceSupp:off - - - -
  363. a=ptime:20
  364. a=sendrecv
  365.  
  366. ---
  367. [Dec 15 15:37:58] WARNING[2553]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission 666ec4863b6bca2e13d6ec6b521b39ba@23.120.227.82:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  368. Packet timed out after 32000ms with no response
  369. Really destroying SIP dialog '666ec4863b6bca2e13d6ec6b521b39ba@23.120.227.82:5060' Method: INVITE
  370. vancouver*CLI> sip set debug off
  371. SIP Debugging Disabled
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