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