API tools faq
paste
thufir

Untitled

thufir
Jul 4th, 2016
122
0
Never
Add comment
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
text 28.49 KB | None | 0 0
raw download clone embed print report
  1. mordor*CLI>
  2. mordor*CLI>
  3. mordor*CLI> sip set debug on
  4. SIP Debugging enabled
  5. mordor*CLI>
  6.  
  7. <--- SIP read from UDP:192.168.1.5:5062 --->
  8. NOTIFY sip:192.168.1.8 SIP/2.0
  9. Via: SIP/2.0/UDP 192.168.1.5:5062;branch=z9hG4bK-32cc6186
  10. From: "thufir" <sip:[email protected]>;tag=a2ca115049f9e611o2
  11. To: <sip:192.168.1.8>
  12. Call-ID: [email protected]
  13. CSeq: 290 NOTIFY
  14. Max-Forwards: 70
  15. Contact: "thufir" <sip:[email protected]:5062>
  16. Event: keep-alive
  17. User-Agent: Linksys/SPA942-6.1.5(a)
  18. Content-Length: 0
  19.  
  20. <------------->
  21. --- (11 headers 0 lines) ---
  22.  
  23. <--- Transmitting (no NAT) to 192.168.1.5:5062 --->
  24. SIP/2.0 200 OK
  25. Via: SIP/2.0/UDP 192.168.1.5:5062;branch=z9hG4bK-32cc6186;received=192.168.1.5
  26. From: "thufir" <sip:[email protected]>;tag=a2ca115049f9e611o2
  27. To: <sip:192.168.1.8>;tag=as74e592c7
  28. Call-ID: [email protected]
  29. CSeq: 290 NOTIFY
  30. Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  31. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  32. Supported: replaces, timer
  33. Content-Length: 0
  34.  
  35.  
  36. <------------>
  37. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: NOTIFY)
  38.  
  39. <--- SIP read from UDP:192.168.1.5:5063 --->
  40. NOTIFY sip:192.168.1.8 SIP/2.0
  41. Via: SIP/2.0/UDP 192.168.1.5:5063;branch=z9hG4bK-8f948873
  42. From: "piter" <sip:[email protected]>;tag=b81adf58e68c2fa9o3
  43. To: <sip:192.168.1.8>
  44. Call-ID: [email protected]
  45. CSeq: 290 NOTIFY
  46. Max-Forwards: 70
  47. Contact: "piter" <sip:[email protected]:5063>
  48. Event: keep-alive
  49. User-Agent: Linksys/SPA942-6.1.5(a)
  50. Content-Length: 0
  51.  
  52. <------------->
  53. --- (11 headers 0 lines) ---
  54.  
  55. <--- Transmitting (no NAT) to 192.168.1.5:5063 --->
  56. SIP/2.0 200 OK
  57. Via: SIP/2.0/UDP 192.168.1.5:5063;branch=z9hG4bK-8f948873;received=192.168.1.5
  58. From: "piter" <sip:[email protected]>;tag=b81adf58e68c2fa9o3
  59. To: <sip:192.168.1.8>;tag=as5f760c15
  60. Call-ID: [email protected]
  61. CSeq: 290 NOTIFY
  62. Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  63. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  64. Supported: replaces, timer
  65. Content-Length: 0
  66.  
  67.  
  68. <------------>
  69. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: NOTIFY)
  70. Reliably Transmitting (NAT) to 192.76.120.10:5060:
  71. OPTIONS sip:sip.telnyx.com SIP/2.0
  72. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK4bfdeae9;rport
  73. Max-Forwards: 70
  74. From: "asterisk" <sip:[email protected]>;tag=as6ec43654
  75. To: <sip:sip.telnyx.com>
  76. Contact: <sip:[email protected]:5060>
  77. Call-ID: [email protected]:5060
  78. CSeq: 102 OPTIONS
  79. User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  80. Date: Mon, 04 Jul 2016 12:52:49 GMT
  81. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  82. Supported: replaces
  83. Content-Length: 0
  84.  
  85.  
  86. ---
  87. Reliably Transmitting (no NAT) to 192.168.1.5:5062:
  88. OPTIONS sip:[email protected]:5062 SIP/2.0
  89. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK5312285a
  90. Max-Forwards: 70
  91. From: "asterisk" <sip:[email protected]>;tag=as33fb9676
  92. To: <sip:[email protected]:5062>
  93. Contact: <sip:[email protected]:5060>
  94. Call-ID: [email protected]:5060
  95. CSeq: 102 OPTIONS
  96. User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  97. Date: Mon, 04 Jul 2016 12:52:49 GMT
  98. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  99. Supported: replaces, timer
  100. Content-Length: 0
  101.  
  102.  
  103. ---
  104. Reliably Transmitting (no NAT) to 192.168.1.5:5063:
  105. OPTIONS sip:[email protected]:5063 SIP/2.0
  106. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK11592e76
  107. Max-Forwards: 70
  108. From: "asterisk" <sip:[email protected]>;tag=as24d0fcf6
  109. To: <sip:[email protected]:5063>
  110. Contact: <sip:[email protected]:5060>
  111. Call-ID: [email protected]:5060
  112. CSeq: 102 OPTIONS
  113. User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  114. Date: Mon, 04 Jul 2016 12:52:49 GMT
  115. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  116. Supported: replaces, timer
  117. Content-Length: 0
  118.  
  119.  
  120. ---
  121.  
  122. <--- SIP read from UDP:192.168.1.5:5062 --->
  123. SIP/2.0 200 OK
  124. To: <sip:[email protected]:5062>;tag=cc9bf57cb72f6fd5i2
  125. From: "asterisk" <sip:[email protected]>;tag=as33fb9676
  126. Call-ID: [email protected]:5060
  127. CSeq: 102 OPTIONS
  128. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK5312285a
  129. Server: Linksys/SPA942-6.1.5(a)
  130. Content-Length: 0
  131. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  132. Supported: replaces
  133.  
  134. <------------->
  135. --- (10 headers 0 lines) ---
  136. Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
  137.  
  138. <--- SIP read from UDP:192.168.1.5:5063 --->
  139. SIP/2.0 200 OK
  140. To: <sip:[email protected]:5063>;tag=12c9b044695bb3adi3
  141. From: "asterisk" <sip:[email protected]>;tag=as24d0fcf6
  142. Call-ID: [email protected]:5060
  143. CSeq: 102 OPTIONS
  144. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK11592e76
  145. Server: Linksys/SPA942-6.1.5(a)
  146. Content-Length: 0
  147. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  148. Supported: replaces
  149.  
  150. <------------->
  151. --- (10 headers 0 lines) ---
  152. Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
  153.  
  154. <--- SIP read from UDP:192.76.120.10:5060 --->
  155. SIP/2.0 200 Ok
  156. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK4bfdeae9;rport=5060;received=192.157.119.39
  157. From: "asterisk" <sip:[email protected]>;tag=as6ec43654
  158. To: <sip:sip.telnyx.com>;tag=40bff79dde18996cbd83e35337950d77.b08f
  159. Call-ID: [email protected]:5060
  160. CSeq: 102 OPTIONS
  161. Server: kamailio (4.4.0 (x86_64/linux))
  162. Content-Length: 0
  163.  
  164. <------------->
  165. --- (8 headers 0 lines) ---
  166. Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
  167.  
  168. <--- SIP read from UDP:192.76.120.10:5060 --->
  169. INVITE sip:[email protected]:5060 SIP/2.0
  170. Record-Route: <sip:192.76.120.10;lr;ftag=3K1e6Sp9t04FD>
  171. Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bK8a36.dbac970f381d34ef8e7e41a62cd2eedd.0
  172. v:SIP/2.0/UDP 64.16.240.36:5082;received=64.16.240.36;rport=5082;branch=z9hG4bKej1aDjgUc2egj
  173. Max-Forwards:67
  174. f:"saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  175. t:<sip:[email protected]:5060>
  176. i:0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  177. CSeq:93482321 INVITE
  178. m:<sip:[email protected]:5082>
  179. User-Agent:FreeSWITCH-mod_sofia/1.6.8-15-99de0ad~64bit
  180. Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,NOTIFY
  181. k:timer,path,replaces
  182. u:talk,hold,conference,refer
  183. Privacy:none
  184. c:application/sdp
  185. Content-Disposition:session
  186. l:294
  187. P-Asserted-Identity:"saunders.nichol"<sip:[email protected]>
  188.  
  189. v=0
  190. o=FreeSWITCH 1467607133 1467607134 IN IP4 64.16.240.36
  191. s=FreeSWITCH
  192. c=IN IP4 64.16.240.36
  193. t=0 0
  194. m=audio 29638 RTP/AVP 9 0 8 18 101
  195. a=rtpmap:9 G722/8000
  196. a=rtpmap:0 PCMU/8000
  197. a=rtpmap:8 PCMA/8000
  198. a=rtpmap:18 G729/8000
  199. a=rtpmap:101 telephone-event/8000
  200. a=fmtp:101 0-16
  201. a=ptime:20
  202. <------------->
  203. --- (19 headers 13 lines) ---
  204. Sending to 192.76.120.10:5060 (no NAT)
  205. Sending to 192.76.120.10:5060 (no NAT)
  206. Using INVITE request as basis request - 0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  207. Found peer 'TELNYX' for '1234567809' from 192.76.120.10:5060
  208. == Using SIP RTP CoS mark 5
  209. Found RTP audio format 9
  210. Found RTP audio format 0
  211. Found RTP audio format 8
  212. Found RTP audio format 18
  213. Found RTP audio format 101
  214. Found audio description format G722 for ID 9
  215. Found audio description format PCMU for ID 0
  216. Found audio description format PCMA for ID 8
  217. Found audio description format G729 for ID 18
  218. Found audio description format telephone-event for ID 101
  219. Capabilities: us - (ulaw|gsm), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (ulaw)
  220. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  221. Peer audio RTP is at port 64.16.240.36:29638
  222. Looking for +16044494243 in inbound (domain 192.157.119.39)
  223. sip_route_dump: route/path hop: <sip:192.76.120.10;lr;ftag=3K1e6Sp9t04FD>
  224.  
  225. <--- Transmitting (NAT) to 192.76.120.10:5060 --->
  226. SIP/2.0 100 Trying
  227. Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bK8a36.dbac970f381d34ef8e7e41a62cd2eedd.0;received=192.76.120.10;rport=5060
  228. Via: SIP/2.0/UDP 64.16.240.36:5082;received=64.16.240.36;rport=5082;branch=z9hG4bKej1aDjgUc2egj
  229. Record-Route: <sip:192.76.120.10;lr;ftag=3K1e6Sp9t04FD>
  230. From: "saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  231. To: <sip:[email protected]:5060>
  232. Call-ID: 0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  233. CSeq: 93482321 INVITE
  234. Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  235. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  236. Supported: replaces
  237. Contact: <sip:[email protected]:5060>
  238. Content-Length: 0
  239.  
  240.  
  241. <------------>
  242. -- Executing [+16044494243@inbound:1] NoOp("SIP/TELNYX-00000000", "") in new stack
  243. -- Executing [+16044494243@inbound:2] Dial("SIP/TELNYX-00000000", "SIP/thufir,60") in new stack
  244. == Using SIP RTP CoS mark 5
  245. Audio is at 17834
  246. Adding codec ulaw to SDP
  247. Adding codec alaw to SDP
  248. Adding codec gsm to SDP
  249. Adding non-codec 0x1 (telephone-event) to SDP
  250. Reliably Transmitting (no NAT) to 192.168.1.5:5062:
  251. INVITE sip:[email protected]:5062 SIP/2.0
  252. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK3d95f5f4
  253. Max-Forwards: 70
  254. From: "saunders.nichol" <sip:[email protected]>;tag=as0f32224c
  255. To: <sip:[email protected]:5062>
  256. Contact: <sip:[email protected]:5060>
  257. Call-ID: [email protected]:5060
  258. CSeq: 102 INVITE
  259. User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  260. Date: Mon, 04 Jul 2016 12:52:51 GMT
  261. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  262. Supported: replaces, timer
  263. Content-Type: application/sdp
  264. Content-Length: 300
  265.  
  266. v=0
  267. o=root 1451220070 1451220070 IN IP4 192.168.1.8
  268. s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  269. c=IN IP4 192.168.1.8
  270. t=0 0
  271. m=audio 17834 RTP/AVP 0 8 3 101
  272. a=rtpmap:0 PCMU/8000
  273. a=rtpmap:8 PCMA/8000
  274. a=rtpmap:3 GSM/8000
  275. a=rtpmap:101 telephone-event/8000
  276. a=fmtp:101 0-16
  277. a=maxptime:150
  278. a=sendrecv
  279.  
  280. ---
  281. -- Called SIP/thufir
  282.  
  283. <--- SIP read from UDP:192.168.1.5:5062 --->
  284. SIP/2.0 100 Trying
  285. To: <sip:[email protected]:5062>
  286. From: "saunders.nichol" <sip:[email protected]>;tag=as0f32224c
  287. Call-ID: [email protected]:5060
  288. CSeq: 102 INVITE
  289. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK3d95f5f4
  290. Server: Linksys/SPA942-6.1.5(a)
  291. Content-Length: 0
  292.  
  293. <------------->
  294. --- (8 headers 0 lines) ---
  295.  
  296. <--- SIP read from UDP:192.168.1.5:5062 --->
  297. SIP/2.0 180 Ringing
  298. To: <sip:[email protected]:5062>;tag=fd8ee681751d28bi2
  299. From: "saunders.nichol" <sip:[email protected]>;tag=as0f32224c
  300. Call-ID: [email protected]:5060
  301. CSeq: 102 INVITE
  302. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK3d95f5f4
  303. Contact: "thufir" <sip:[email protected]:5062>
  304. Server: Linksys/SPA942-6.1.5(a)
  305. Content-Length: 0
  306.  
  307. <------------->
  308. --- (9 headers 0 lines) ---
  309. sip_route_dump: route/path hop: <sip:[email protected]:5062>
  310. -- SIP/thufir-00000001 is ringing
  311.  
  312. <--- Transmitting (NAT) to 192.76.120.10:5060 --->
  313. SIP/2.0 180 Ringing
  314. Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bK8a36.dbac970f381d34ef8e7e41a62cd2eedd.0;received=192.76.120.10;rport=5060
  315. Via: SIP/2.0/UDP 64.16.240.36:5082;received=64.16.240.36;rport=5082;branch=z9hG4bKej1aDjgUc2egj
  316. Record-Route: <sip:192.76.120.10;lr;ftag=3K1e6Sp9t04FD>
  317. From: "saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  318. To: <sip:[email protected]:5060>;tag=as07fb4b6b
  319. Call-ID: 0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  320. CSeq: 93482321 INVITE
  321. Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  322. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  323. Supported: replaces
  324. Contact: <sip:[email protected]:5060>
  325. Content-Length: 0
  326.  
  327.  
  328. <------------>
  329.  
  330. <--- SIP read from UDP:192.168.1.5:5062 --->
  331. SIP/2.0 200 OK
  332. To: <sip:[email protected]:5062>;tag=fd8ee681751d28bi2
  333. From: "saunders.nichol" <sip:[email protected]>;tag=as0f32224c
  334. Call-ID: [email protected]:5060
  335. CSeq: 102 INVITE
  336. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK3d95f5f4
  337. Contact: "thufir" <sip:[email protected]:5062>
  338. Server: Linksys/SPA942-6.1.5(a)
  339. Content-Length: 204
  340. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  341. Supported: replaces
  342. Content-Type: application/sdp
  343.  
  344. v=0
  345. o=- 441497 441497 IN IP4 192.168.1.5
  346. s=-
  347. c=IN IP4 192.168.1.5
  348. t=0 0
  349. m=audio 16408 RTP/AVP 0 101
  350. a=rtpmap:0 PCMU/8000
  351. a=rtpmap:101 telephone-event/8000
  352. a=fmtp:101 0-15
  353. a=ptime:20
  354. a=sendrecv
  355. <------------->
  356. --- (12 headers 11 lines) ---
  357. Found RTP audio format 0
  358. Found RTP audio format 101
  359. Found audio description format PCMU for ID 0
  360. Found audio description format telephone-event for ID 101
  361. Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
  362. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  363. Peer audio RTP is at port 192.168.1.5:16408
  364. sip_route_dump: route/path hop: <sip:[email protected]:5062>
  365. set_destination: Parsing <sip:[email protected]:5062> for address/port to send to
  366. set_destination: set destination to 192.168.1.5:5062
  367. Transmitting (no NAT) to 192.168.1.5:5062:
  368. ACK sip:[email protected]:5062 SIP/2.0
  369. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK76b9833e
  370. Max-Forwards: 70
  371. From: "saunders.nichol" <sip:[email protected]>;tag=as0f32224c
  372. To: <sip:[email protected]:5062>;tag=fd8ee681751d28bi2
  373. Contact: <sip:[email protected]:5060>
  374. Call-ID: [email protected]:5060
  375. CSeq: 102 ACK
  376. User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  377. Content-Length: 0
  378.  
  379.  
  380. ---
  381. -- SIP/thufir-00000001 answered SIP/TELNYX-00000000
  382. Audio is at 14126
  383. Adding codec ulaw to SDP
  384. Adding codec gsm to SDP
  385. Adding non-codec 0x1 (telephone-event) to SDP
  386.  
  387. <--- Reliably Transmitting (NAT) to 192.76.120.10:5060 --->
  388. SIP/2.0 200 OK
  389. Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bK8a36.dbac970f381d34ef8e7e41a62cd2eedd.0;received=192.76.120.10;rport=5060
  390. Via: SIP/2.0/UDP 64.16.240.36:5082;received=64.16.240.36;rport=5082;branch=z9hG4bKej1aDjgUc2egj
  391. Record-Route: <sip:192.76.120.10;lr;ftag=3K1e6Sp9t04FD>
  392. From: "saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  393. To: <sip:[email protected]:5060>;tag=as07fb4b6b
  394. Call-ID: 0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  395. CSeq: 93482321 INVITE
  396. Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  397. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  398. Supported: replaces
  399. Contact: <sip:[email protected]:5060>
  400. Content-Type: application/sdp
  401. Content-Length: 276
  402.  
  403. v=0
  404. o=root 2125564869 2125564869 IN IP4 192.168.1.8
  405. s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  406. c=IN IP4 192.168.1.8
  407. t=0 0
  408. m=audio 14126 RTP/AVP 0 3 101
  409. a=rtpmap:0 PCMU/8000
  410. a=rtpmap:3 GSM/8000
  411. a=rtpmap:101 telephone-event/8000
  412. a=fmtp:101 0-16
  413. a=maxptime:150
  414. a=sendrecv
  415.  
  416. <------------>
  417. -- Channel SIP/TELNYX-00000000 joined 'simple_bridge' basic-bridge <10bf4a3b-0365-4f06-abc3-e7e56b77e427>
  418. -- Channel SIP/thufir-00000001 joined 'simple_bridge' basic-bridge <10bf4a3b-0365-4f06-abc3-e7e56b77e427>
  419. > Bridge 10bf4a3b-0365-4f06-abc3-e7e56b77e427: switching from simple_bridge technology to native_rtp
  420. > 0x7f725000d1c0 -- Probation passed - setting RTP source address to 192.168.1.5:16408
  421. Retransmitting #1 (NAT) to 192.76.120.10:5060:
  422. SIP/2.0 200 OK
  423. Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bK8a36.dbac970f381d34ef8e7e41a62cd2eedd.0;received=192.76.120.10;rport=5060
  424. Via: SIP/2.0/UDP 64.16.240.36:5082;received=64.16.240.36;rport=5082;branch=z9hG4bKej1aDjgUc2egj
  425. Record-Route: <sip:192.76.120.10;lr;ftag=3K1e6Sp9t04FD>
  426. From: "saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  427. To: <sip:[email protected]:5060>;tag=as07fb4b6b
  428. Call-ID: 0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  429. CSeq: 93482321 INVITE
  430. Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  431. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  432. Supported: replaces
  433. Contact: <sip:[email protected]:5060>
  434. Content-Type: application/sdp
  435. Content-Length: 276
  436.  
  437. v=0
  438. o=root 2125564869 2125564869 IN IP4 192.168.1.8
  439. s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  440. c=IN IP4 192.168.1.8
  441. t=0 0
  442. m=audio 14126 RTP/AVP 0 3 101
  443. a=rtpmap:0 PCMU/8000
  444. a=rtpmap:3 GSM/8000
  445. a=rtpmap:101 telephone-event/8000
  446. a=fmtp:101 0-16
  447. a=maxptime:150
  448. a=sendrecv
  449.  
  450. ---
  451. Retransmitting #2 (NAT) to 192.76.120.10:5060:
  452. SIP/2.0 200 OK
  453. Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bK8a36.dbac970f381d34ef8e7e41a62cd2eedd.0;received=192.76.120.10;rport=5060
  454. Via: SIP/2.0/UDP 64.16.240.36:5082;received=64.16.240.36;rport=5082;branch=z9hG4bKej1aDjgUc2egj
  455. Record-Route: <sip:192.76.120.10;lr;ftag=3K1e6Sp9t04FD>
  456. From: "saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  457. To: <sip:[email protected]:5060>;tag=as07fb4b6b
  458. Call-ID: 0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  459. CSeq: 93482321 INVITE
  460. Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  461. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  462. Supported: replaces
  463. Contact: <sip:[email protected]:5060>
  464. Content-Type: application/sdp
  465. Content-Length: 276
  466.  
  467. v=0
  468. o=root 2125564869 2125564869 IN IP4 192.168.1.8
  469. s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  470. c=IN IP4 192.168.1.8
  471. t=0 0
  472. m=audio 14126 RTP/AVP 0 3 101
  473. a=rtpmap:0 PCMU/8000
  474. a=rtpmap:3 GSM/8000
  475. a=rtpmap:101 telephone-event/8000
  476. a=fmtp:101 0-16
  477. a=maxptime:150
  478. a=sendrecv
  479.  
  480. ---
  481. > 0x7f728c00d7c0 -- Probation passed - setting RTP source address to 64.16.240.36:29638
  482. Retransmitting #3 (NAT) to 192.76.120.10:5060:
  483. SIP/2.0 200 OK
  484. Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bK8a36.dbac970f381d34ef8e7e41a62cd2eedd.0;received=192.76.120.10;rport=5060
  485. Via: SIP/2.0/UDP 64.16.240.36:5082;received=64.16.240.36;rport=5082;branch=z9hG4bKej1aDjgUc2egj
  486. Record-Route: <sip:192.76.120.10;lr;ftag=3K1e6Sp9t04FD>
  487. From: "saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  488. To: <sip:[email protected]:5060>;tag=as07fb4b6b
  489. Call-ID: 0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  490. CSeq: 93482321 INVITE
  491. Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  492. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  493. Supported: replaces
  494. Contact: <sip:[email protected]:5060>
  495. Content-Type: application/sdp
  496. Content-Length: 276
  497.  
  498. v=0
  499. o=root 2125564869 2125564869 IN IP4 192.168.1.8
  500. s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  501. c=IN IP4 192.168.1.8
  502. t=0 0
  503. m=audio 14126 RTP/AVP 0 3 101
  504. a=rtpmap:0 PCMU/8000
  505. a=rtpmap:3 GSM/8000
  506. a=rtpmap:101 telephone-event/8000
  507. a=fmtp:101 0-16
  508. a=maxptime:150
  509. a=sendrecv
  510.  
  511. ---
  512. Retransmitting #4 (NAT) to 192.76.120.10:5060:
  513. SIP/2.0 200 OK
  514. Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bK8a36.dbac970f381d34ef8e7e41a62cd2eedd.0;received=192.76.120.10;rport=5060
  515. Via: SIP/2.0/UDP 64.16.240.36:5082;received=64.16.240.36;rport=5082;branch=z9hG4bKej1aDjgUc2egj
  516. Record-Route: <sip:192.76.120.10;lr;ftag=3K1e6Sp9t04FD>
  517. From: "saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  518. To: <sip:[email protected]:5060>;tag=as07fb4b6b
  519. Call-ID: 0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  520. CSeq: 93482321 INVITE
  521. Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  522. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  523. Supported: replaces
  524. Contact: <sip:[email protected]:5060>
  525. Content-Type: application/sdp
  526. Content-Length: 276
  527.  
  528. v=0
  529. o=root 2125564869 2125564869 IN IP4 192.168.1.8
  530. s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  531. c=IN IP4 192.168.1.8
  532. t=0 0
  533. m=audio 14126 RTP/AVP 0 3 101
  534. a=rtpmap:0 PCMU/8000
  535. a=rtpmap:3 GSM/8000
  536. a=rtpmap:101 telephone-event/8000
  537. a=fmtp:101 0-16
  538. a=maxptime:150
  539. a=sendrecv
  540.  
  541. ---
  542. Retransmitting #5 (NAT) to 192.76.120.10:5060:
  543. SIP/2.0 200 OK
  544. Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bK8a36.dbac970f381d34ef8e7e41a62cd2eedd.0;received=192.76.120.10;rport=5060
  545. Via: SIP/2.0/UDP 64.16.240.36:5082;received=64.16.240.36;rport=5082;branch=z9hG4bKej1aDjgUc2egj
  546. Record-Route: <sip:192.76.120.10;lr;ftag=3K1e6Sp9t04FD>
  547. From: "saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  548. To: <sip:[email protected]:5060>;tag=as07fb4b6b
  549. Call-ID: 0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  550. CSeq: 93482321 INVITE
  551. Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  552. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  553. Supported: replaces
  554. Contact: <sip:[email protected]:5060>
  555. Content-Type: application/sdp
  556. Content-Length: 276
  557.  
  558. v=0
  559. o=root 2125564869 2125564869 IN IP4 192.168.1.8
  560. s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  561. c=IN IP4 192.168.1.8
  562. t=0 0
  563. m=audio 14126 RTP/AVP 0 3 101
  564. a=rtpmap:0 PCMU/8000
  565. a=rtpmap:3 GSM/8000
  566. a=rtpmap:101 telephone-event/8000
  567. a=fmtp:101 0-16
  568. a=maxptime:150
  569. a=sendrecv
  570.  
  571. ---
  572.  
  573. <--- SIP read from UDP:192.168.1.5:5062 --->
  574. NOTIFY sip:192.168.1.8 SIP/2.0
  575. Via: SIP/2.0/UDP 192.168.1.5:5062;branch=z9hG4bK-63a52174
  576. From: "thufir" <sip:[email protected]>;tag=a2ca115049f9e611o2
  577. To: <sip:192.168.1.8>
  578. Call-ID: [email protected]
  579. CSeq: 291 NOTIFY
  580. Max-Forwards: 70
  581. Contact: "thufir" <sip:[email protected]:5062>
  582. Event: keep-alive
  583. User-Agent: Linksys/SPA942-6.1.5(a)
  584. Content-Length: 0
  585.  
  586. <------------->
  587. --- (11 headers 0 lines) ---
  588.  
  589. <--- Transmitting (no NAT) to 192.168.1.5:5062 --->
  590. SIP/2.0 200 OK
  591. Via: SIP/2.0/UDP 192.168.1.5:5062;branch=z9hG4bK-63a52174;received=192.168.1.5
  592. From: "thufir" <sip:[email protected]>;tag=a2ca115049f9e611o2
  593. To: <sip:192.168.1.8>;tag=as74e592c7
  594. Call-ID: [email protected]
  595. CSeq: 291 NOTIFY
  596. Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  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. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: NOTIFY)
  604.  
  605. <--- SIP read from UDP:192.168.1.5:5063 --->
  606. NOTIFY sip:192.168.1.8 SIP/2.0
  607. Via: SIP/2.0/UDP 192.168.1.5:5063;branch=z9hG4bK-c4821f3a
  608. From: "piter" <sip:[email protected]>;tag=b81adf58e68c2fa9o3
  609. To: <sip:192.168.1.8>
  610. Call-ID: [email protected]
  611. CSeq: 291 NOTIFY
  612. Max-Forwards: 70
  613. Contact: "piter" <sip:[email protected]:5063>
  614. Event: keep-alive
  615. User-Agent: Linksys/SPA942-6.1.5(a)
  616. Content-Length: 0
  617.  
  618. <------------->
  619. --- (11 headers 0 lines) ---
  620.  
  621. <--- Transmitting (no NAT) to 192.168.1.5:5063 --->
  622. SIP/2.0 200 OK
  623. Via: SIP/2.0/UDP 192.168.1.5:5063;branch=z9hG4bK-c4821f3a;received=192.168.1.5
  624. From: "piter" <sip:[email protected]>;tag=b81adf58e68c2fa9o3
  625. To: <sip:192.168.1.8>;tag=as5f760c15
  626. Call-ID: [email protected]
  627. CSeq: 291 NOTIFY
  628. Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  629. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  630. Supported: replaces, timer
  631. Content-Length: 0
  632.  
  633.  
  634. <------------>
  635. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: NOTIFY)
  636. Retransmitting #6 (NAT) to 192.76.120.10:5060:
  637. SIP/2.0 200 OK
  638. Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bK8a36.dbac970f381d34ef8e7e41a62cd2eedd.0;received=192.76.120.10;rport=5060
  639. Via: SIP/2.0/UDP 64.16.240.36:5082;received=64.16.240.36;rport=5082;branch=z9hG4bKej1aDjgUc2egj
  640. Record-Route: <sip:192.76.120.10;lr;ftag=3K1e6Sp9t04FD>
  641. From: "saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  642. To: <sip:[email protected]:5060>;tag=as07fb4b6b
  643. Call-ID: 0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  644. CSeq: 93482321 INVITE
  645. Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  646. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  647. Supported: replaces
  648. Contact: <sip:[email protected]:5060>
  649. Content-Type: application/sdp
  650. Content-Length: 276
  651.  
  652. v=0
  653. o=root 2125564869 2125564869 IN IP4 192.168.1.8
  654. s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  655. c=IN IP4 192.168.1.8
  656. t=0 0
  657. m=audio 14126 RTP/AVP 0 3 101
  658. a=rtpmap:0 PCMU/8000
  659. a=rtpmap:3 GSM/8000
  660. a=rtpmap:101 telephone-event/8000
  661. a=fmtp:101 0-16
  662. a=maxptime:150
  663. a=sendrecv
  664.  
  665. ---
  666. [Jul 4 05:53:01] WARNING[18275]: chan_sip.c:4047 retrans_pkt: Retransmission timeout reached on transmission 0f0fdbf5-bc89-1234-13bd-002590fd3b6c for seqno 93482321 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  667. Packet timed out after 6399ms with no response
  668. [Jul 4 05:53:01] WARNING[18275]: chan_sip.c:4076 retrans_pkt: Hanging up call 0f0fdbf5-bc89-1234-13bd-002590fd3b6c - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  669. -- Channel SIP/TELNYX-00000000 left 'native_rtp' basic-bridge <10bf4a3b-0365-4f06-abc3-e7e56b77e427>
  670. -- Channel SIP/thufir-00000001 left 'native_rtp' basic-bridge <10bf4a3b-0365-4f06-abc3-e7e56b77e427>
  671. Scheduling destruction of SIP dialog '[email protected]:5060' in 6400 ms (Method: INVITE)
  672. set_destination: Parsing <sip:[email protected]:5062> for address/port to send to
  673. set_destination: set destination to 192.168.1.5:5062
  674. Reliably Transmitting (no NAT) to 192.168.1.5:5062:
  675. BYE sip:[email protected]:5062 SIP/2.0
  676. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK46bd5606
  677. Max-Forwards: 70
  678. From: "saunders.nichol" <sip:[email protected]>;tag=as0f32224c
  679. To: <sip:[email protected]:5062>;tag=fd8ee681751d28bi2
  680. Call-ID: [email protected]:5060
  681. CSeq: 103 BYE
  682. User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  683. X-Asterisk-HangupCause: No user responding
  684. X-Asterisk-HangupCauseCode: 18
  685. Content-Length: 0
  686.  
  687.  
  688. ---
  689. == Spawn extension (inbound, +16044494243, 2) exited non-zero on 'SIP/TELNYX-00000000'
  690. Scheduling destruction of SIP dialog '0f0fdbf5-bc89-1234-13bd-002590fd3b6c' in 6400 ms (Method: INVITE)
  691. Reliably Transmitting (NAT) to 192.76.120.10:5060:
  692. BYE sip:[email protected]:5082 SIP/2.0
  693. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK5e6376e2;rport
  694. Route: <sip:192.76.120.10;lr;ftag=3K1e6Sp9t04FD>
  695. Max-Forwards: 70
  696. From: <sip:[email protected]:5060>;tag=as07fb4b6b
  697. To: "saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  698. Call-ID: 0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  699. CSeq: 102 BYE
  700. User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  701. X-Asterisk-HangupCause: No user responding
  702. X-Asterisk-HangupCauseCode: 18
  703. Content-Length: 0
  704.  
  705.  
  706. ---
  707.  
  708. <--- SIP read from UDP:192.168.1.5:5062 --->
  709. SIP/2.0 200 OK
  710. To: <sip:[email protected]:5062>;tag=fd8ee681751d28bi2
  711. From: "saunders.nichol" <sip:[email protected]>;tag=as0f32224c
  712. Call-ID: [email protected]:5060
  713. CSeq: 103 BYE
  714. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK46bd5606
  715. Server: Linksys/SPA942-6.1.5(a)
  716. Content-Length: 0
  717.  
  718. <------------->
  719. --- (8 headers 0 lines) ---
  720. Really destroying SIP dialog '[email protected]:5060' Method: INVITE
  721. Retransmitting #1 (NAT) to 192.76.120.10:5060:
  722. BYE sip:[email protected]:5082 SIP/2.0
  723. Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK5e6376e2;rport
  724. Route: <sip:192.76.120.10;lr;ftag=3K1e6Sp9t04FD>
  725. Max-Forwards: 70
  726. From: <sip:[email protected]:5060>;tag=as07fb4b6b
  727. To: "saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  728. Call-ID: 0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  729. CSeq: 102 BYE
  730. User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
  731. X-Asterisk-HangupCause: No user responding
  732. X-Asterisk-HangupCauseCode: 18
  733. Content-Length: 0
  734.  
  735.  
  736. ---
  737.  
  738. <--- SIP read from UDP:192.76.120.10:5060 --->
  739. SIP/2.0 200 OK
  740. v:SIP/2.0/UDP 192.168.1.8:5060;received=192.157.119.39;branch=z9hG4bK5e6376e2;rport=5060
  741. f:<sip:[email protected]:5060>;tag=as07fb4b6b
  742. t:"saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  743. i:0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  744. CSeq:102 BYE
  745. User-Agent:FreeSWITCH-mod_sofia/1.6.8-15-99de0ad~64bit
  746. Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,NOTIFY
  747. k:timer,path,replaces
  748. l:0
  749.  
  750. <------------->
  751. --- (10 headers 0 lines) ---
  752. SIP Response message for INCOMING dialog BYE arrived
  753. Really destroying SIP dialog '0f0fdbf5-bc89-1234-13bd-002590fd3b6c' Method: INVITE
  754.  
  755. <--- SIP read from UDP:192.76.120.10:5060 --->
  756. SIP/2.0 200 OK
  757. v:SIP/2.0/UDP 192.168.1.8:5060;received=192.157.119.39;branch=z9hG4bK5e6376e2;rport=5060
  758. f:<sip:[email protected]:5060>;tag=as07fb4b6b
  759. t:"saunders.nichol"<sip:[email protected]>;tag=3K1e6Sp9t04FD
  760. i:0f0fdbf5-bc89-1234-13bd-002590fd3b6c
  761. CSeq:102 BYE
  762. User-Agent:FreeSWITCH-mod_sofia/1.6.8-15-99de0ad~64bit
  763. Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,NOTIFY
  764. k:timer,path,replaces
  765. l:0
  766.  
  767. <------------->
  768. --- (10 headers 0 lines) ---
  769. mordor*CLI>
  770. mordor*CLI>
  771. mordor*CLI>
  772. mordor*CLI> sip set debug off
  773. SIP Debugging Disabled
  774. mordor*CLI>
Advertisement
Add Comment
Please, Sign In to add comment
Public Pastes
We use cookies for various purposes including analytics. By continuing to use Pastebin, you agree to our use of cookies as described in the Cookies Policy.  OK, I Understand
Not a member of Pastebin yet?
Sign Up, it unlocks many cool features!