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Asterisk SIP Debug Capture

amartin
May 11th, 2015
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  1. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] pbx.c: -- Timeout on SIP/113-00002b7d, continuing...
  2. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] pbx.c: -- Executing [s@asterisk-test:2] Queue("SIP/113-00002b7d", "test,rtnC,18") in new stack
  3. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] netsock2.c: == Using SIP RTP CoS mark 5
  4. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] netsock2.c: == Using SIP RTP CoS mark 5
  5. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] netsock2.c: == Using SIP RTP CoS mark 5
  6. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] chan_sip.c: Audio is at 18882
  7. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] chan_sip.c: Adding codec 100003 (ulaw) to SDP
  8. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] chan_sip.c: Adding codec 100004 (alaw) to SDP
  9. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] chan_sip.c: Adding codec 100002 (gsm) to SDP
  10. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  11. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.32.96:5062:
  12. INVITE sip:146@10.10.32.96:5062 SIP/2.0
  13. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK3a984081
  14. Max-Forwards: 70
  15. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  16. To: <sip:146@10.10.32.96:5062>
  17. Contact: <sip:113@10.10.32.251:5060>
  18. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  19. CSeq: 102 INVITE
  20. User-Agent: Asterisk PBX 11.12.0
  21. Date: Mon, 11 May 2015 15:30:28 GMT
  22. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  23. Supported: replaces, timer
  24. Content-Type: application/sdp
  25. Content-Length: 283
  26.  
  27. v=0
  28. o=root 2012033363 2012033363 IN IP4 10.10.32.251
  29. s=Asterisk PBX 11.12.0
  30. c=IN IP4 10.10.32.251
  31. t=0 0
  32. m=audio 18882 RTP/AVP 0 8 3 101
  33. a=rtpmap:0 PCMU/8000
  34. a=rtpmap:8 PCMA/8000
  35. a=rtpmap:3 GSM/8000
  36. a=rtpmap:101 telephone-event/8000
  37. a=fmtp:101 0-16
  38. a=ptime:20
  39. a=sendrecv
  40.  
  41. ---
  42. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] app_queue.c: -- SIP/169-00002b7e is ringing
  43. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] app_queue.c: -- SIP/169-00002b7e is ringing
  44. [2015-05-11 10:30:28] VERBOSE[21037] chan_sip.c:
  45. <--- SIP read from UDP:10.10.32.96:5062 --->
  46. SIP/2.0 180 Ringing
  47. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK3a984081
  48. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  49. To: <sip:146@10.10.32.96:5062>;tag=2114358324
  50. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  51. CSeq: 102 INVITE
  52. Contact: <sip:146@10.10.32.96:5062>
  53. Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
  54. User-Agent: Yealink SIP-T38G 38.70.1.33
  55. Allow-Events: talk,hold,conference,refer,check-sync
  56. Content-Length: 0
  57.  
  58. <------------->
  59. [2015-05-11 10:30:28] VERBOSE[21037] chan_sip.c: --- (11 headers 0 lines) ---
  60. [2015-05-11 10:30:28] VERBOSE[21037][C-00001d5f] chan_sip.c: list_route: hop: <sip:146@10.10.32.96:5062>
  61. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] app_queue.c: -- SIP/146-00002b80 is ringing
  62. [2015-05-11 10:30:28] VERBOSE[21037] chan_sip.c:
  63. <--- SIP read from UDP:10.10.32.96:5062 --->
  64. SIP/2.0 180 Ringing
  65. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK3a984081
  66. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  67. To: <sip:146@10.10.32.96:5062>;tag=2114358324
  68. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  69. CSeq: 102 INVITE
  70. Contact: <sip:146@10.10.32.96:5062>
  71. Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
  72. User-Agent: Yealink SIP-T38G 38.70.1.33
  73. Allow-Events: talk,hold,conference,refer,check-sync
  74. Content-Length: 0
  75.  
  76. <------------->
  77. [2015-05-11 10:30:28] VERBOSE[21037] chan_sip.c: --- (11 headers 0 lines) ---
  78. [2015-05-11 10:30:28] VERBOSE[21037][C-00001d5f] chan_sip.c: list_route: hop: <sip:146@10.10.32.96:5062>
  79. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] app_queue.c: -- SIP/146-00002b80 is ringing
  80. [2015-05-11 10:30:28] VERBOSE[3649][C-00001d5f] app_queue.c: -- SIP/178-00002b7f is ringing
  81. [2015-05-11 10:30:31] VERBOSE[21037] chan_sip.c:
  82. <--- SIP read from UDP:10.10.32.96:5062 --->
  83. SIP/2.0 200 OK
  84. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK3a984081
  85. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  86. To: <sip:146@10.10.32.96:5062>;tag=2114358324
  87. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  88. CSeq: 102 INVITE
  89. Contact: <sip:146@10.10.32.96:5062>
  90. Content-Type: application/sdp
  91. Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
  92. User-Agent: Yealink SIP-T38G 38.70.1.33
  93. Content-Length: 209
  94.  
  95. v=0
  96. o=- 20222 20222 IN IP4 10.10.32.96
  97. s=SDP data
  98. c=IN IP4 10.10.32.96
  99. t=0 0
  100. m=audio 11798 RTP/AVP 0 101
  101. a=rtpmap:0 PCMU/8000
  102. a=sendrecv
  103. a=ptime:20
  104. a=fmtp:101 0-15
  105. a=rtpmap:101 telephone-event/8000
  106. <------------->
  107. [2015-05-11 10:30:31] VERBOSE[21037] chan_sip.c: --- (11 headers 11 lines) ---
  108. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: Found RTP audio format 0
  109. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: Found RTP audio format 101
  110. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: Found audio description format PCMU for ID 0
  111. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: Found audio description format telephone-event for ID 101
  112. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
  113. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  114. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: Peer audio RTP is at port 10.10.32.96:11798
  115. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: list_route: hop: <sip:146@10.10.32.96:5062>
  116. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: set_destination: Parsing <sip:146@10.10.32.96:5062> for address/port to send to
  117. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: set_destination: set destination to 10.10.32.96:5062
  118. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: Transmitting (no NAT) to 10.10.32.96:5062:
  119. ACK sip:146@10.10.32.96:5062 SIP/2.0
  120. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK357f89e8
  121. Max-Forwards: 70
  122. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  123. To: <sip:146@10.10.32.96:5062>;tag=2114358324
  124. Contact: <sip:113@10.10.32.251:5060>
  125. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  126. CSeq: 102 ACK
  127. User-Agent: Asterisk PBX 11.12.0
  128. Content-Length: 0
  129.  
  130.  
  131. ---
  132. [2015-05-11 10:30:31] VERBOSE[3649][C-00001d5f] app_queue.c: -- SIP/146-00002b80 answered SIP/113-00002b7d
  133. [2015-05-11 10:30:31] VERBOSE[3649][C-00001d5f] chan_sip.c: set_destination: Parsing <sip:146@10.10.32.96:5062> for address/port to send to
  134. [2015-05-11 10:30:31] VERBOSE[3649][C-00001d5f] chan_sip.c: set_destination: set destination to 10.10.32.96:5062
  135. [2015-05-11 10:30:31] VERBOSE[3649][C-00001d5f] chan_sip.c: Audio is at 18882
  136. [2015-05-11 10:30:31] VERBOSE[3649][C-00001d5f] chan_sip.c: Adding codec 100003 (ulaw) to SDP
  137. [2015-05-11 10:30:31] VERBOSE[3649][C-00001d5f] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
  138. [2015-05-11 10:30:31] VERBOSE[3649][C-00001d5f] chan_sip.c: Reliably Transmitting (no NAT) to 10.10.32.96:5062:
  139. INVITE sip:146@10.10.32.96:5062 SIP/2.0
  140. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK53433712
  141. Max-Forwards: 70
  142. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  143. To: <sip:146@10.10.32.96:5062>;tag=2114358324
  144. Contact: <sip:113@10.10.32.251:5060>
  145. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  146. CSeq: 103 INVITE
  147. User-Agent: Asterisk PBX 11.12.0
  148. Access-URL: <18>;mode=active
  149. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  150. Supported: replaces, timer
  151. X-asterisk-Info: SIP re-invite (External RTP bridge)
  152. Content-Type: application/sdp
  153. Content-Length: 236
  154.  
  155. v=0
  156. o=root 2012033363 2012033364 IN IP4 10.10.32.251
  157. s=Asterisk PBX 11.12.0
  158. c=IN IP4 10.10.32.251
  159. t=0 0
  160. m=audio 18882 RTP/AVP 0 101
  161. a=rtpmap:0 PCMU/8000
  162. a=rtpmap:101 telephone-event/8000
  163. a=fmtp:101 0-16
  164. a=ptime:20
  165. a=sendrecv
  166.  
  167. ---
  168. [2015-05-11 10:30:31] VERBOSE[21037] chan_sip.c:
  169. <--- SIP read from UDP:10.10.32.96:5062 --->
  170. SIP/2.0 200 OK
  171. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK3a984081
  172. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  173. To: <sip:146@10.10.32.96:5062>;tag=2114358324
  174. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  175. CSeq: 102 INVITE
  176. Contact: <sip:146@10.10.32.96:5062>
  177. Content-Type: application/sdp
  178. Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
  179. User-Agent: Yealink SIP-T38G 38.70.1.33
  180. Content-Length: 209
  181.  
  182. v=0
  183. o=- 20222 20222 IN IP4 10.10.32.96
  184. s=SDP data
  185. c=IN IP4 10.10.32.96
  186. t=0 0
  187. m=audio 11798 RTP/AVP 0 101
  188. a=rtpmap:0 PCMU/8000
  189. a=sendrecv
  190. a=ptime:20
  191. a=fmtp:101 0-15
  192. a=rtpmap:101 telephone-event/8000
  193. <------------->
  194. [2015-05-11 10:30:31] VERBOSE[21037] chan_sip.c: --- (11 headers 11 lines) ---
  195. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: set_destination: Parsing <sip:146@10.10.32.96:5062> for address/port to send to
  196. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: set_destination: set destination to 10.10.32.96:5062
  197. [2015-05-11 10:30:31] VERBOSE[21037][C-00001d5f] chan_sip.c: Transmitting (no NAT) to 10.10.32.96:5062:
  198. ACK sip:146@10.10.32.96:5062 SIP/2.0
  199. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK3e577126
  200. Max-Forwards: 70
  201. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  202. To: <sip:146@10.10.32.96:5062>;tag=2114358324
  203. Contact: <sip:113@10.10.32.251:5060>
  204. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  205. CSeq: 102 ACK
  206. User-Agent: Asterisk PBX 11.12.0
  207. Content-Length: 0
  208.  
  209.  
  210. ---
  211. [2015-05-11 10:30:31] VERBOSE[3649][C-00001d5f] res_rtp_asterisk.c: > 0x7f29b00e7a80 -- Probation passed - setting RTP source address to 10.10.32.96:11798
  212. [2015-05-11 10:30:31] VERBOSE[21037] chan_sip.c: Retransmitting #1 (no NAT) to 10.10.32.96:5062:
  213. INVITE sip:146@10.10.32.96:5062 SIP/2.0
  214. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK53433712
  215. Max-Forwards: 70
  216. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  217. To: <sip:146@10.10.32.96:5062>;tag=2114358324
  218. Contact: <sip:113@10.10.32.251:5060>
  219. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  220. CSeq: 103 INVITE
  221. User-Agent: Asterisk PBX 11.12.0
  222. Access-URL: <18>;mode=active
  223. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  224. Supported: replaces, timer
  225. X-asterisk-Info: SIP re-invite (External RTP bridge)
  226. Content-Type: application/sdp
  227. Content-Length: 236
  228.  
  229. v=0
  230. o=root 2012033363 2012033364 IN IP4 10.10.32.251
  231. s=Asterisk PBX 11.12.0
  232. c=IN IP4 10.10.32.251
  233. t=0 0
  234. m=audio 18882 RTP/AVP 0 101
  235. a=rtpmap:0 PCMU/8000
  236. a=rtpmap:101 telephone-event/8000
  237. a=fmtp:101 0-16
  238. a=ptime:20
  239. a=sendrecv
  240.  
  241. ---
  242. [2015-05-11 10:30:32] VERBOSE[21037] chan_sip.c: Retransmitting #2 (no NAT) to 10.10.32.96:5062:
  243. INVITE sip:146@10.10.32.96:5062 SIP/2.0
  244. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK53433712
  245. Max-Forwards: 70
  246. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  247. To: <sip:146@10.10.32.96:5062>;tag=2114358324
  248. Contact: <sip:113@10.10.32.251:5060>
  249. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  250. CSeq: 103 INVITE
  251. User-Agent: Asterisk PBX 11.12.0
  252. Access-URL: <18>;mode=active
  253. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  254. Supported: replaces, timer
  255. X-asterisk-Info: SIP re-invite (External RTP bridge)
  256. Content-Type: application/sdp
  257. Content-Length: 236
  258.  
  259. v=0
  260. o=root 2012033363 2012033364 IN IP4 10.10.32.251
  261. s=Asterisk PBX 11.12.0
  262. c=IN IP4 10.10.32.251
  263. t=0 0
  264. m=audio 18882 RTP/AVP 0 101
  265. a=rtpmap:0 PCMU/8000
  266. a=rtpmap:101 telephone-event/8000
  267. a=fmtp:101 0-16
  268. a=ptime:20
  269. a=sendrecv
  270.  
  271. ---
  272. [2015-05-11 10:30:34] VERBOSE[21037] chan_sip.c: Retransmitting #3 (no NAT) to 10.10.32.96:5062:
  273. INVITE sip:146@10.10.32.96:5062 SIP/2.0
  274. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK53433712
  275. Max-Forwards: 70
  276. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  277. To: <sip:146@10.10.32.96:5062>;tag=2114358324
  278. Contact: <sip:113@10.10.32.251:5060>
  279. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  280. CSeq: 103 INVITE
  281. User-Agent: Asterisk PBX 11.12.0
  282. Access-URL: <18>;mode=active
  283. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  284. Supported: replaces, timer
  285. X-asterisk-Info: SIP re-invite (External RTP bridge)
  286. Content-Type: application/sdp
  287. Content-Length: 236
  288.  
  289. v=0
  290. o=root 2012033363 2012033364 IN IP4 10.10.32.251
  291. s=Asterisk PBX 11.12.0
  292. c=IN IP4 10.10.32.251
  293. t=0 0
  294. m=audio 18882 RTP/AVP 0 101
  295. a=rtpmap:0 PCMU/8000
  296. a=rtpmap:101 telephone-event/8000
  297. a=fmtp:101 0-16
  298. a=ptime:20
  299. a=sendrecv
  300.  
  301. ---
  302. [2015-05-11 10:30:38] VERBOSE[21037] chan_sip.c: Retransmitting #4 (no NAT) to 10.10.32.96:5062:
  303. INVITE sip:146@10.10.32.96:5062 SIP/2.0
  304. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK53433712
  305. Max-Forwards: 70
  306. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  307. To: <sip:146@10.10.32.96:5062>;tag=2114358324
  308. Contact: <sip:113@10.10.32.251:5060>
  309. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  310. CSeq: 103 INVITE
  311. User-Agent: Asterisk PBX 11.12.0
  312. Access-URL: <18>;mode=active
  313. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  314. Supported: replaces, timer
  315. X-asterisk-Info: SIP re-invite (External RTP bridge)
  316. Content-Type: application/sdp
  317. Content-Length: 236
  318.  
  319. v=0
  320. o=root 2012033363 2012033364 IN IP4 10.10.32.251
  321. s=Asterisk PBX 11.12.0
  322. c=IN IP4 10.10.32.251
  323. t=0 0
  324. m=audio 18882 RTP/AVP 0 101
  325. a=rtpmap:0 PCMU/8000
  326. a=rtpmap:101 telephone-event/8000
  327. a=fmtp:101 0-16
  328. a=ptime:20
  329. a=sendrecv
  330.  
  331. ---
  332. [2015-05-11 10:30:46] VERBOSE[21037] chan_sip.c: Retransmitting #5 (no NAT) to 10.10.32.96:5062:
  333. INVITE sip:146@10.10.32.96:5062 SIP/2.0
  334. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK53433712
  335. Max-Forwards: 70
  336. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  337. To: <sip:146@10.10.32.96:5062>;tag=2114358324
  338. Contact: <sip:113@10.10.32.251:5060>
  339. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  340. CSeq: 103 INVITE
  341. User-Agent: Asterisk PBX 11.12.0
  342. Access-URL: <18>;mode=active
  343. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  344. Supported: replaces, timer
  345. X-asterisk-Info: SIP re-invite (External RTP bridge)
  346. Content-Type: application/sdp
  347. Content-Length: 236
  348.  
  349. v=0
  350. o=root 2012033363 2012033364 IN IP4 10.10.32.251
  351. s=Asterisk PBX 11.12.0
  352. c=IN IP4 10.10.32.251
  353. t=0 0
  354. m=audio 18882 RTP/AVP 0 101
  355. a=rtpmap:0 PCMU/8000
  356. a=rtpmap:101 telephone-event/8000
  357. a=fmtp:101 0-16
  358. a=ptime:20
  359. a=sendrecv
  360.  
  361. ---
  362. [2015-05-11 10:30:49] VERBOSE[21037] chan_sip.c: Really destroying SIP dialog '194030494@10.10.32.96' Method: REGISTER
  363. [2015-05-11 10:30:49] VERBOSE[20088] asterisk.c: -- Remote UNIX connection disconnected
  364. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c:
  365. <--- SIP read from UDP:10.10.32.96:5062 --->
  366. SUBSCRIBE sip:102@10.10.32.251 SIP/2.0
  367. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK942434642
  368. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=880813742
  369. To: <sip:102@10.10.32.251>
  370. Call-ID: 11630500@10.10.32.96
  371. CSeq: 1 SUBSCRIBE
  372. Contact: <sip:146@10.10.32.96:5062>
  373. Accept: application/dialog-info+xml
  374. Max-Forwards: 70
  375. User-Agent: Yealink SIP-T38G 38.70.1.33
  376. Expires: 1800
  377. Event: dialog
  378. Content-Length: 0
  379.  
  380. <------------->
  381. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c: --- (13 headers 0 lines) ---
  382. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c: Sending to 10.10.32.96:5062 (no NAT)
  383. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c:
  384. <--- Transmitting (no NAT) to 10.10.32.96:5062 --->
  385. SIP/2.0 403 Forbidden (policy)
  386. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK942434642;received=10.10.32.96
  387. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=880813742
  388. To: <sip:102@10.10.32.251>;tag=as5254bb18
  389. Call-ID: 11630500@10.10.32.96
  390. CSeq: 1 SUBSCRIBE
  391. Server: Asterisk PBX 11.12.0
  392. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  393. Supported: replaces, timer
  394. Content-Length: 0
  395.  
  396.  
  397. <------------>
  398. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c: Really destroying SIP dialog '11630500@10.10.32.96' Method: SUBSCRIBE
  399. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c:
  400. <--- SIP read from UDP:10.10.32.96:5062 --->
  401. SUBSCRIBE sip:102@10.10.32.251 SIP/2.0
  402. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK942434642
  403. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=880813742
  404. To: <sip:102@10.10.32.251>
  405. Call-ID: 11630500@10.10.32.96
  406. CSeq: 1 SUBSCRIBE
  407. Contact: <sip:146@10.10.32.96:5062>
  408. Accept: application/dialog-info+xml
  409. Max-Forwards: 70
  410. User-Agent: Yealink SIP-T38G 38.70.1.33
  411. Expires: 1800
  412. Event: dialog
  413. Content-Length: 0
  414.  
  415. <------------->
  416. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c: --- (13 headers 0 lines) ---
  417. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c: Sending to 10.10.32.96:5062 (no NAT)
  418. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c:
  419. <--- Transmitting (no NAT) to 10.10.32.96:5062 --->
  420. SIP/2.0 403 Forbidden (policy)
  421. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK942434642;received=10.10.32.96
  422. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=880813742
  423. To: <sip:102@10.10.32.251>;tag=as51201365
  424. Call-ID: 11630500@10.10.32.96
  425. CSeq: 1 SUBSCRIBE
  426. Server: Asterisk PBX 11.12.0
  427. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  428. Supported: replaces, timer
  429. Content-Length: 0
  430.  
  431.  
  432. <------------>
  433. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c: Really destroying SIP dialog '11630500@10.10.32.96' Method: SUBSCRIBE
  434. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c:
  435. <--- SIP read from UDP:10.10.32.96:5062 --->
  436. SUBSCRIBE sip:430@10.10.32.251 SIP/2.0
  437. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK426004668
  438. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=2071236564
  439. To: <sip:430@10.10.32.251>
  440. Call-ID: 900664969@10.10.32.96
  441. CSeq: 1 SUBSCRIBE
  442. Contact: <sip:146@10.10.32.96:5062>
  443. Accept: application/dialog-info+xml
  444. Max-Forwards: 70
  445. User-Agent: Yealink SIP-T38G 38.70.1.33
  446. Expires: 1800
  447. Event: dialog
  448. Content-Length: 0
  449.  
  450. <------------->
  451. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c: --- (13 headers 0 lines) ---
  452. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c: Sending to 10.10.32.96:5062 (no NAT)
  453. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c:
  454. <--- Transmitting (no NAT) to 10.10.32.96:5062 --->
  455. SIP/2.0 403 Forbidden (policy)
  456. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK426004668;received=10.10.32.96
  457. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=2071236564
  458. To: <sip:430@10.10.32.251>;tag=as6c765d62
  459. Call-ID: 900664969@10.10.32.96
  460. CSeq: 1 SUBSCRIBE
  461. Server: Asterisk PBX 11.12.0
  462. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  463. Supported: replaces, timer
  464. Content-Length: 0
  465.  
  466.  
  467. <------------>
  468. [2015-05-11 10:31:01] VERBOSE[21037] chan_sip.c: Really destroying SIP dialog '900664969@10.10.32.96' Method: SUBSCRIBE
  469. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: Retransmitting #6 (no NAT) to 10.10.32.96:5062:
  470. INVITE sip:146@10.10.32.96:5062 SIP/2.0
  471. Via: SIP/2.0/UDP 10.10.32.251:5060;branch=z9hG4bK53433712
  472. Max-Forwards: 70
  473. From: <sip:113@10.10.32.251>;tag=as5f4bb25b
  474. To: <sip:146@10.10.32.96:5062>;tag=2114358324
  475. Contact: <sip:113@10.10.32.251:5060>
  476. Call-ID: 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060
  477. CSeq: 103 INVITE
  478. User-Agent: Asterisk PBX 11.12.0
  479. Access-URL: <18>;mode=active
  480. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  481. Supported: replaces, timer
  482. X-asterisk-Info: SIP re-invite (External RTP bridge)
  483. Content-Type: application/sdp
  484. Content-Length: 236
  485.  
  486. v=0
  487. o=root 2012033363 2012033364 IN IP4 10.10.32.251
  488. s=Asterisk PBX 11.12.0
  489. c=IN IP4 10.10.32.251
  490. t=0 0
  491. m=audio 18882 RTP/AVP 0 101
  492. a=rtpmap:0 PCMU/8000
  493. a=rtpmap:101 telephone-event/8000
  494. a=fmtp:101 0-16
  495. a=ptime:20
  496. a=sendrecv
  497.  
  498. ---
  499. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c:
  500. <--- SIP read from UDP:10.10.32.96:5062 --->
  501. SUBSCRIBE sip:103@10.10.32.251 SIP/2.0
  502. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK2019802991
  503. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1387953645
  504. To: <sip:103@10.10.32.251>
  505. Call-ID: 1281210855@10.10.32.96
  506. CSeq: 1 SUBSCRIBE
  507. Contact: <sip:146@10.10.32.96:5062>
  508. Accept: application/dialog-info+xml
  509. Max-Forwards: 70
  510. User-Agent: Yealink SIP-T38G 38.70.1.33
  511. Expires: 1800
  512. Event: dialog
  513. Content-Length: 0
  514.  
  515. <------------->
  516. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: --- (13 headers 0 lines) ---
  517. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: Sending to 10.10.32.96:5062 (no NAT)
  518. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c:
  519. <--- Transmitting (no NAT) to 10.10.32.96:5062 --->
  520. SIP/2.0 403 Forbidden (policy)
  521. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK2019802991;received=10.10.32.96
  522. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1387953645
  523. To: <sip:103@10.10.32.251>;tag=as7e1b5c96
  524. Call-ID: 1281210855@10.10.32.96
  525. CSeq: 1 SUBSCRIBE
  526. Server: Asterisk PBX 11.12.0
  527. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  528. Supported: replaces, timer
  529. Content-Length: 0
  530.  
  531.  
  532. <------------>
  533. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: Really destroying SIP dialog '1281210855@10.10.32.96' Method: SUBSCRIBE
  534. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c:
  535. <--- SIP read from UDP:10.10.32.96:5062 --->
  536. SUBSCRIBE sip:103@10.10.32.251 SIP/2.0
  537. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK2019802991
  538. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1387953645
  539. To: <sip:103@10.10.32.251>
  540. Call-ID: 1281210855@10.10.32.96
  541. CSeq: 1 SUBSCRIBE
  542. Contact: <sip:146@10.10.32.96:5062>
  543. Accept: application/dialog-info+xml
  544. Max-Forwards: 70
  545. User-Agent: Yealink SIP-T38G 38.70.1.33
  546. Expires: 1800
  547. Event: dialog
  548. Content-Length: 0
  549.  
  550. <------------->
  551. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: --- (13 headers 0 lines) ---
  552. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: Sending to 10.10.32.96:5062 (no NAT)
  553. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c:
  554. <--- Transmitting (no NAT) to 10.10.32.96:5062 --->
  555. SIP/2.0 403 Forbidden (policy)
  556. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK2019802991;received=10.10.32.96
  557. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1387953645
  558. To: <sip:103@10.10.32.251>;tag=as5df8318a
  559. Call-ID: 1281210855@10.10.32.96
  560. CSeq: 1 SUBSCRIBE
  561. Server: Asterisk PBX 11.12.0
  562. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  563. Supported: replaces, timer
  564. Content-Length: 0
  565.  
  566.  
  567. <------------>
  568. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: Really destroying SIP dialog '1281210855@10.10.32.96' Method: SUBSCRIBE
  569. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c:
  570. <--- SIP read from UDP:10.10.32.96:5062 --->
  571. SUBSCRIBE sip:104@10.10.32.251 SIP/2.0
  572. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK158331888
  573. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1926732360
  574. To: <sip:104@10.10.32.251>
  575. Call-ID: 1474351627@10.10.32.96
  576. CSeq: 1 SUBSCRIBE
  577. Contact: <sip:146@10.10.32.96:5062>
  578. Accept: application/dialog-info+xml
  579. Max-Forwards: 70
  580. User-Agent: Yealink SIP-T38G 38.70.1.33
  581. Expires: 1800
  582. Event: dialog
  583. Content-Length: 0
  584.  
  585. <------------->
  586. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: --- (13 headers 0 lines) ---
  587. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: Sending to 10.10.32.96:5062 (no NAT)
  588. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c:
  589. <--- Transmitting (no NAT) to 10.10.32.96:5062 --->
  590. SIP/2.0 403 Forbidden (policy)
  591. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK158331888;received=10.10.32.96
  592. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1926732360
  593. To: <sip:104@10.10.32.251>;tag=as01c34f54
  594. Call-ID: 1474351627@10.10.32.96
  595. CSeq: 1 SUBSCRIBE
  596. Server: Asterisk PBX 11.12.0
  597. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  598. Supported: replaces, timer
  599. Content-Length: 0
  600.  
  601.  
  602. <------------>
  603. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: Really destroying SIP dialog '1474351627@10.10.32.96' Method: SUBSCRIBE
  604. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c:
  605. <--- SIP read from UDP:10.10.32.96:5062 --->
  606. SUBSCRIBE sip:104@10.10.32.251 SIP/2.0
  607. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK158331888
  608. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1926732360
  609. To: <sip:104@10.10.32.251>
  610. Call-ID: 1474351627@10.10.32.96
  611. CSeq: 1 SUBSCRIBE
  612. Contact: <sip:146@10.10.32.96:5062>
  613. Accept: application/dialog-info+xml
  614. Max-Forwards: 70
  615. User-Agent: Yealink SIP-T38G 38.70.1.33
  616. Expires: 1800
  617. Event: dialog
  618. Content-Length: 0
  619.  
  620. <------------->
  621. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: --- (13 headers 0 lines) ---
  622. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: Sending to 10.10.32.96:5062 (no NAT)
  623. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c:
  624. <--- Transmitting (no NAT) to 10.10.32.96:5062 --->
  625. SIP/2.0 403 Forbidden (policy)
  626. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK158331888;received=10.10.32.96
  627. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1926732360
  628. To: <sip:104@10.10.32.251>;tag=as33ce207b
  629. Call-ID: 1474351627@10.10.32.96
  630. CSeq: 1 SUBSCRIBE
  631. Server: Asterisk PBX 11.12.0
  632. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  633. Supported: replaces, timer
  634. Content-Length: 0
  635.  
  636.  
  637. <------------>
  638. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: Really destroying SIP dialog '1474351627@10.10.32.96' Method: SUBSCRIBE
  639. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c:
  640. <--- SIP read from UDP:10.10.32.96:5062 --->
  641. SUBSCRIBE sip:105@10.10.32.251 SIP/2.0
  642. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK1778366124
  643. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1399201939
  644. To: <sip:105@10.10.32.251>
  645. Call-ID: 1279736949@10.10.32.96
  646. CSeq: 1 SUBSCRIBE
  647. Contact: <sip:146@10.10.32.96:5062>
  648. Accept: application/dialog-info+xml
  649. Max-Forwards: 70
  650. User-Agent: Yealink SIP-T38G 38.70.1.33
  651. Expires: 1800
  652. Event: dialog
  653. Content-Length: 0
  654.  
  655. <------------->
  656. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: --- (13 headers 0 lines) ---
  657. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: Sending to 10.10.32.96:5062 (no NAT)
  658. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c:
  659. <--- Transmitting (no NAT) to 10.10.32.96:5062 --->
  660. SIP/2.0 403 Forbidden (policy)
  661. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK1778366124;received=10.10.32.96
  662. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1399201939
  663. To: <sip:105@10.10.32.251>;tag=as3d724237
  664. Call-ID: 1279736949@10.10.32.96
  665. CSeq: 1 SUBSCRIBE
  666. Server: Asterisk PBX 11.12.0
  667. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  668. Supported: replaces, timer
  669. Content-Length: 0
  670.  
  671.  
  672. <------------>
  673. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: Really destroying SIP dialog '1279736949@10.10.32.96' Method: SUBSCRIBE
  674. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c:
  675. <--- SIP read from UDP:10.10.32.96:5062 --->
  676. SUBSCRIBE sip:105@10.10.32.251 SIP/2.0
  677. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK1778366124
  678. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1399201939
  679. To: <sip:105@10.10.32.251>
  680. Call-ID: 1279736949@10.10.32.96
  681. CSeq: 1 SUBSCRIBE
  682. Contact: <sip:146@10.10.32.96:5062>
  683. Accept: application/dialog-info+xml
  684. Max-Forwards: 70
  685. User-Agent: Yealink SIP-T38G 38.70.1.33
  686. Expires: 1800
  687. Event: dialog
  688. Content-Length: 0
  689.  
  690. <------------->
  691. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: --- (13 headers 0 lines) ---
  692. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: Sending to 10.10.32.96:5062 (no NAT)
  693. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c:
  694. <--- Transmitting (no NAT) to 10.10.32.96:5062 --->
  695. SIP/2.0 403 Forbidden (policy)
  696. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK1778366124;received=10.10.32.96
  697. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1399201939
  698. To: <sip:105@10.10.32.251>;tag=as547bc4cd
  699. Call-ID: 1279736949@10.10.32.96
  700. CSeq: 1 SUBSCRIBE
  701. Server: Asterisk PBX 11.12.0
  702. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  703. Supported: replaces, timer
  704. Content-Length: 0
  705.  
  706.  
  707. <------------>
  708. [2015-05-11 10:31:02] VERBOSE[21037] chan_sip.c: Really destroying SIP dialog '1279736949@10.10.32.96' Method: SUBSCRIBE
  709. [2015-05-11 10:31:03] VERBOSE[21037] chan_sip.c:
  710. <--- SIP read from UDP:10.10.32.96:5062 --->
  711. SUBSCRIBE sip:178@10.10.32.251 SIP/2.0
  712. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK944104374
  713. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1315261870
  714. To: <sip:178@10.10.32.251>
  715. Call-ID: 454544405@10.10.32.96
  716. CSeq: 1 SUBSCRIBE
  717. Contact: <sip:146@10.10.32.96:5062>
  718. Accept: application/dialog-info+xml
  719. Max-Forwards: 70
  720. User-Agent: Yealink SIP-T38G 38.70.1.33
  721. Expires: 1800
  722. Event: dialog
  723. Content-Length: 0
  724.  
  725. <------------->
  726. [2015-05-11 10:31:03] VERBOSE[21037] chan_sip.c: --- (13 headers 0 lines) ---
  727. [2015-05-11 10:31:03] VERBOSE[21037] chan_sip.c: Sending to 10.10.32.96:5062 (no NAT)
  728. [2015-05-11 10:31:03] VERBOSE[21037] chan_sip.c:
  729. <--- Transmitting (no NAT) to 10.10.32.96:5062 --->
  730. SIP/2.0 403 Forbidden (policy)
  731. Via: SIP/2.0/UDP 10.10.32.96:5062;branch=z9hG4bK944104374;received=10.10.32.96
  732. From: "Andrew Martin" <sip:146@10.10.32.251>;tag=1315261870
  733. To: <sip:178@10.10.32.251>;tag=as70dd2a17
  734. Call-ID: 454544405@10.10.32.96
  735. CSeq: 1 SUBSCRIBE
  736. Server: Asterisk PBX 11.12.0
  737. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  738. Supported: replaces, timer
  739. Content-Length: 0
  740.  
  741.  
  742. <------------>
  743. [2015-05-11 10:31:03] VERBOSE[21037] chan_sip.c: Really destroying SIP dialog '454544405@10.10.32.96' Method: SUBSCRIBE
  744. [2015-05-11 10:31:03] WARNING[21037] chan_sip.c: Retransmission timeout reached on transmission 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  745. Packet timed out after 32000ms with no response
  746. [2015-05-11 10:31:03] WARNING[21037] chan_sip.c: Hanging up call 5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  747. [2015-05-11 10:31:03] VERBOSE[3649][C-00001d5f] pbx.c: == Spawn extension (asterisk-test, s, 2) exited non-zero on 'SIP/113-00002b7d'
  748. [2015-05-11 10:31:03] VERBOSE[21037] chan_sip.c: Really destroying SIP dialog '5f3e73454952c906363ae26614aae5d8@10.10.32.251:5060' Method: INVITE
  749. [2015-05-11 10:31:09] VERBOSE[21037] chan_sip.c:
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