API tools faq
paste
tuxmartin

asterisk sip log

tuxmartin
Nov 12th, 2015
200
0
Never
Add comment
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
text 28.19 KB | None | 0 0
raw download clone embed print report
  1. root@voip:~# asterisk -r
  2. Asterisk 13.6.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  3. Created by Mark Spencer <[email protected]>
  4. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  5. This is free software, with components licensed under the GNU General Public
  6. License version 2 and other licenses; you are welcome to redistribute it under
  7. certain conditions. Type 'core show license' for details.
  8. =========================================================================
  9. Connected to Asterisk 13.6.0 currently running on voip (pid = 32018)
  10. voip*CLI> sip set debug on
  11. SIP Debugging enabled
  12. Retransmitting #6 (NAT) to 155.94.64.34:5089:
  13. SIP/2.0 401 Unauthorized
  14. Via: SIP/2.0/UDP 155.94.64.34:5089;branch=z9hG4bK-0575c362977ea6340ca6fcbe6260b446;received=155.94.64.34;rport=5089
  15. From: 101<sip:[email protected]>;tag=8a359551
  16. To: 991130972597723173<sip:[email protected]>;tag=as1e4415a6
  17. Call-ID: 0575c362977ea6340ca6fcbe6260b446
  18. CSeq: 1 INVITE
  19. Server: Asterisk PBX 13.6.0
  20. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  21. Supported: replaces, timer
  22. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54467743"
  23. Content-Length: 0
  24.  
  25.  
  26. ---
  27.  
  28. <--- SIP read from UDP:89.190.50.140:39059 --->
  29. INVITE sip:[email protected] SIP/2.0
  30. Via: SIP/2.0/UDP 192.168.1.112:39059;rport;branch=z9hG4bK1595020550
  31. From: <sip:[email protected]>;tag=726823752
  32. To: <sip:[email protected]>
  33. Call-ID: 1937419984
  34. CSeq: 20 INVITE
  35. Contact: <sip:[email protected]:39059>
  36. Content-Type: application/sdp
  37. Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
  38. Max-Forwards: 70
  39. User-Agent: antisip/5.1.0-549-g2028176-Oct-15-2015 amdroid/4.2.5 HM 1SW/4.4.4
  40. Subject: Talk
  41. Supported: 100rel, replaces
  42. Content-Length: 744
  43.  
  44. v=0
  45. o=amsip 1989349040 0 IN IP4 192.168.1.112
  46. s=talk
  47. c=IN IP4 192.168.1.112
  48. t=0 0
  49. m=audio 10000 RTP/AVP 98 105 100 97 99 103 0 8 101
  50. a=rtpmap:98 SILK/16000
  51. a=rtpmap:105 OPUS/48000/2
  52. a=fmtp:105 useinbandfec=1
  53. a=rtpmap:100 speex/16000
  54. a=fmtp:100 mode="6,any"
  55. a=rtpmap:97 SILK/8000
  56. a=rtpmap:99 speex/8000
  57. a=fmtp:99 mode="6,any"
  58. a=rtpmap:103 iLBC/8000
  59. a=fmtp:103 mode=30
  60. a=rtpmap:0 PCMU/8000
  61. a=rtpmap:8 PCMA/8000
  62. a=rtpmap:101 telephone-event/8000
  63. a=fmtp:101 0-15
  64. a=rtcp-mux
  65. m=video 10020 RTP/AVP 118 117 116 115 34
  66. b=AS:128
  67. a=rtpmap:118 VP8/90000
  68. a=rtpmap:117 H264/90000
  69. a=fmtp:117 profile-level-id=42800c; packetization-mode=1
  70. a=rtpmap:116 MP4V-ES/90000
  71. a=rtpmap:115 H263-1998/90000
  72. a=rtpmap:34 H263/90000
  73. a=rtcp-mux
  74. <------------->
  75. --- (14 headers 30 lines) ---
  76. Sending to 89.190.50.140:39059 (NAT)
  77. Sending to 89.190.50.140:39059 (NAT)
  78. Using INVITE request as basis request - 1937419984
  79. Found peer '100' for '100' from 89.190.50.140:39059
  80.  
  81. <--- Reliably Transmitting (NAT) to 89.190.50.140:39059 --->
  82. SIP/2.0 401 Unauthorized
  83. Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK1595020550;received=89.190.50.140;rport=39059
  84. From: <sip:[email protected]>;tag=726823752
  85. To: <sip:[email protected]>;tag=as1d1f47bb
  86. Call-ID: 1937419984
  87. CSeq: 20 INVITE
  88. Server: Asterisk PBX 13.6.0
  89. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  90. Supported: replaces, timer
  91. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="34225d72"
  92. Content-Length: 0
  93.  
  94.  
  95. <------------>
  96. Scheduling destruction of SIP dialog '1937419984' in 32000 ms (Method: INVITE)
  97.  
  98. <--- SIP read from UDP:89.190.50.140:39059 --->
  99. ACK sip:[email protected] SIP/2.0
  100. Via: SIP/2.0/UDP 192.168.1.112:39059;rport;branch=z9hG4bK1595020550
  101. From: <sip:[email protected]>;tag=726823752
  102. To: <sip:[email protected]>;tag=as1d1f47bb
  103. Call-ID: 1937419984
  104. CSeq: 20 ACK
  105. Content-Length: 0
  106.  
  107. <------------->
  108. --- (7 headers 0 lines) ---
  109.  
  110. <--- SIP read from UDP:89.190.50.140:39059 --->
  111. INVITE sip:[email protected] SIP/2.0
  112. Via: SIP/2.0/UDP 192.168.1.112:39059;rport;branch=z9hG4bK938901029
  113. From: <sip:[email protected]>;tag=726823752
  114. To: <sip:[email protected]>
  115. Call-ID: 1937419984
  116. CSeq: 21 INVITE
  117. Contact: <sip:[email protected]:39059>
  118. Authorization: Digest username="100", realm="asterisk", nonce="34225d72", uri="sip:[email protected]", response="a36f5b293a0667c06d76504da7b5d857", algorithm=MD5
  119. Content-Type: application/sdp
  120. Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, REFER, NOTIFY, MESSAGE
  121. Max-Forwards: 70
  122. User-Agent: antisip/5.1.0-549-g2028176-Oct-15-2015 amdroid/4.2.5 HM 1SW/4.4.4
  123. Subject: Talk
  124. Supported: 100rel, replaces
  125. Content-Length: 744
  126.  
  127. v=0
  128. o=amsip 1989349040 0 IN IP4 192.168.1.112
  129. s=talk
  130. c=IN IP4 192.168.1.112
  131. t=0 0
  132. m=audio 10000 RTP/AVP 98 105 100 97 99 103 0 8 101
  133. a=rtpmap:98 SILK/16000
  134. a=rtpmap:105 OPUS/48000/2
  135. a=fmtp:105 useinbandfec=1
  136. a=rtpmap:100 speex/16000
  137. a=fmtp:100 mode="6,any"
  138. a=rtpmap:97 SILK/8000
  139. a=rtpmap:99 speex/8000
  140. a=fmtp:99 mode="6,any"
  141. a=rtpmap:103 iLBC/8000
  142. a=fmtp:103 mode=30
  143. a=rtpmap:0 PCMU/8000
  144. a=rtpmap:8 PCMA/8000
  145. a=rtpmap:101 telephone-event/8000
  146. a=fmtp:101 0-15
  147. a=rtcp-mux
  148. m=video 10020 RTP/AVP 118 117 116 115 34
  149. b=AS:128
  150. a=rtpmap:118 VP8/90000
  151. a=rtpmap:117 H264/90000
  152. a=fmtp:117 profile-level-id=42800c; packetization-mode=1
  153. a=rtpmap:116 MP4V-ES/90000
  154. a=rtpmap:115 H263-1998/90000
  155. a=rtpmap:34 H263/90000
  156. a=rtcp-mux
  157. <------------->
  158. --- (15 headers 30 lines) ---
  159. Sending to 89.190.50.140:39059 (NAT)
  160. Using INVITE request as basis request - 1937419984
  161. Found peer '100' for '100' from 89.190.50.140:39059
  162. Found RTP audio format 98
  163. Found RTP audio format 105
  164. Found RTP audio format 100
  165. Found RTP audio format 97
  166. Found RTP audio format 99
  167. Found RTP audio format 103
  168. Found RTP audio format 0
  169. Found RTP audio format 8
  170. Found RTP audio format 101
  171. Found unknown media description format SILK for ID 98
  172. Found audio description format OPUS for ID 105
  173. Found audio description format speex for ID 100
  174. Found unknown media description format SILK for ID 97
  175. Found audio description format speex for ID 99
  176. Found audio description format iLBC for ID 103
  177. Found audio description format PCMU for ID 0
  178. Found audio description format PCMA for ID 8
  179. Found audio description format telephone-event for ID 101
  180. Found RTP video format 118
  181. Found RTP video format 117
  182. Found RTP video format 116
  183. Found RTP video format 115
  184. Found RTP video format 34
  185. Found video description format VP8 for ID 118
  186. Found video description format H264 for ID 117
  187. Found video description format MP4V-ES for ID 116
  188. Found video description format H263-1998 for ID 115
  189. Found video description format H263 for ID 34
  190. Capabilities: us - (ulaw|h263|gsm), peer - audio=(ulaw|alaw|speex|speex16|ilbc|opus)/video=(h263|h263p|mpeg4|h264|vp8)/text=(nothing), combined - (ulaw|h263)
  191. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  192. Peer audio RTP is at port 192.168.1.112:10000
  193. Peer video RTP is at port 192.168.1.112:10020
  194. Looking for 101 in pokus (domain eu.vancl.eu)
  195. sip_route_dump: route/path hop: <sip:[email protected]:39059>
  196.  
  197. <--- Transmitting (NAT) to 89.190.50.140:39059 --->
  198. SIP/2.0 100 Trying
  199. Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
  200. From: <sip:[email protected]>;tag=726823752
  201. To: <sip:[email protected]>
  202. Call-ID: 1937419984
  203. CSeq: 21 INVITE
  204. Server: Asterisk PBX 13.6.0
  205. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  206. Supported: replaces, timer
  207. Contact: <sip:[email protected]:5060>
  208. Content-Length: 0
  209.  
  210.  
  211. <------------>
  212. Audio is at 10964
  213. Video is at 192.168.1.7:10272
  214. Adding codec ulaw to SDP
  215. Adding video codec h263 to SDP
  216. Adding codec gsm to SDP
  217. Adding non-codec 0x1 (telephone-event) to SDP
  218. Reliably Transmitting (NAT) to 89.190.50.140:5060:
  219. INVITE sip:[email protected]:5060 SIP/2.0
  220. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK3a997588;rport
  221. Max-Forwards: 70
  222. From: "(Testovaci uzivatel" <sip:[email protected]>;tag=as7a28997b
  223. To: <sip:[email protected]:5060>
  224. Contact: <sip:[email protected]:5060>
  225. Call-ID: [email protected]:5060
  226. CSeq: 102 INVITE
  227. User-Agent: Asterisk PBX 13.6.0
  228. Date: Thu, 12 Nov 2015 16:11:33 GMT
  229. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  230. Supported: replaces, timer
  231. Content-Type: application/sdp
  232. Content-Length: 332
  233.  
  234. v=0
  235. o=root 2143136553 2143136553 IN IP4 192.168.1.7
  236. s=Asterisk PBX 13.6.0
  237. c=IN IP4 192.168.1.7
  238. b=CT:384
  239. t=0 0
  240. m=audio 10964 RTP/AVP 0 3 101
  241. a=rtpmap:0 PCMU/8000
  242. a=rtpmap:3 GSM/8000
  243. a=rtpmap:101 telephone-event/8000
  244. a=fmtp:101 0-16
  245. a=maxptime:150
  246. a=sendrecv
  247. m=video 10272 RTP/AVP 34
  248. a=rtpmap:34 H263/90000
  249. a=sendrecv
  250.  
  251. ---
  252.  
  253. <--- SIP read from UDP:89.190.50.140:5060 --->
  254. SIP/2.0 100 Trying
  255. CSeq: 102 INVITE
  256. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK3a997588;rport=5060;received=93.89.146.5
  257. From: "(Testovaci uzivatel" <sip:[email protected]>;tag=as7a28997b
  258. Call-ID: [email protected]:5060
  259. To: <sip:[email protected]:5060>
  260. Content-Length: 0
  261.  
  262. <------------->
  263. --- (7 headers 0 lines) ---
  264.  
  265. <--- SIP read from UDP:89.190.50.140:5060 --->
  266. SIP/2.0 180 Ringing
  267. CSeq: 102 INVITE
  268. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK3a997588;rport=5060;received=93.89.146.5
  269. User-Agent: Ekiga/4.0.1
  270. From: "(Testovaci uzivatel" <sip:[email protected]>;tag=as7a28997b
  271. Call-ID: [email protected]:5060
  272. To: "101" <sip:[email protected]>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
  273. Contact: "101" <sip:[email protected]>
  274. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
  275. Content-Length: 0
  276.  
  277. <------------->
  278. --- (10 headers 0 lines) ---
  279. sip_route_dump: route/path hop: <sip:[email protected]>
  280.  
  281. <--- Transmitting (NAT) to 89.190.50.140:39059 --->
  282. SIP/2.0 180 Ringing
  283. Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
  284. From: <sip:[email protected]>;tag=726823752
  285. To: <sip:[email protected]>;tag=as75b76746
  286. Call-ID: 1937419984
  287. CSeq: 21 INVITE
  288. Server: Asterisk PBX 13.6.0
  289. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  290. Supported: replaces, timer
  291. Contact: <sip:[email protected]:5060>
  292. Content-Length: 0
  293.  
  294.  
  295. <------------>
  296. Retransmitting #7 (NAT) to 155.94.64.34:5089:
  297. SIP/2.0 401 Unauthorized
  298. Via: SIP/2.0/UDP 155.94.64.34:5089;branch=z9hG4bK-0575c362977ea6340ca6fcbe6260b446;received=155.94.64.34;rport=5089
  299. From: 101<sip:[email protected]>;tag=8a359551
  300. To: 991130972597723173<sip:[email protected]>;tag=as1e4415a6
  301. Call-ID: 0575c362977ea6340ca6fcbe6260b446
  302. CSeq: 1 INVITE
  303. Server: Asterisk PBX 13.6.0
  304. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  305. Supported: replaces, timer
  306. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54467743"
  307. Content-Length: 0
  308.  
  309.  
  310. ---
  311.  
  312. <--- SIP read from UDP:89.190.50.140:5060 --->
  313. SIP/2.0 200 OK
  314. CSeq: 102 INVITE
  315. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK3a997588;rport=5060;received=93.89.146.5
  316. User-Agent: Ekiga/4.0.1
  317. From: "(Testovaci uzivatel" <sip:[email protected]>;tag=as7a28997b
  318. Call-ID: [email protected]:5060
  319. To: "101" <sip:[email protected]>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
  320. Contact: "101" <sip:[email protected]>
  321. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
  322. Content-Length: 328
  323. Content-Type: application/sdp
  324.  
  325. v=0
  326. o=- 1447344693 1 IN IP4 89.190.50.140
  327. s=Ekiga/4.0.1
  328. c=IN IP4 89.190.50.140
  329. t=0 0
  330. m=audio 5066 RTP/AVP 0 101
  331. a=sendrecv
  332. a=rtpmap:0 PCMU/8000/1
  333. a=rtpmap:101 telephone-event/8000
  334. a=fmtp:101 0-16
  335. a=maxptime:240
  336. m=video 5068 RTP/AVP 34
  337. b=AS:4096
  338. b=TIAS:4096000
  339. a=recvonly
  340. a=rtpmap:34 H263/90000
  341. a=fmtp:34 QCIF=3
  342. <------------->
  343. --- (11 headers 17 lines) ---
  344. Found RTP audio format 0
  345. Found RTP audio format 101
  346. Found audio description format PCMU for ID 0
  347. Found audio description format telephone-event for ID 101
  348. Found RTP video format 34
  349. Found video description format H263 for ID 34
  350. Capabilities: us - (ulaw|h263|gsm), peer - audio=(ulaw)/video=(h263)/text=(nothing), combined - (ulaw|h263)
  351. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  352. Peer audio RTP is at port 89.190.50.140:5066
  353. Peer video RTP is at port 89.190.50.140:5068
  354. sip_route_dump: route/path hop: <sip:[email protected]>
  355. Transmitting (NAT) to 89.190.50.140:5060:
  356. ACK sip:[email protected] SIP/2.0
  357. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK75e4afb8;rport
  358. Max-Forwards: 70
  359. From: "(Testovaci uzivatel" <sip:[email protected]>;tag=as7a28997b
  360. To: <sip:[email protected]:5060>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
  361. Contact: <sip:[email protected]:5060>
  362. Call-ID: [email protected]:5060
  363. CSeq: 102 ACK
  364. User-Agent: Asterisk PBX 13.6.0
  365. Content-Length: 0
  366.  
  367.  
  368. ---
  369. Audio is at 10466
  370. Video is at 192.168.1.7:10800
  371. Adding codec ulaw to SDP
  372. Adding video codec h263 to SDP
  373. Adding codec gsm to SDP
  374. Adding non-codec 0x1 (telephone-event) to SDP
  375.  
  376. <--- Reliably Transmitting (NAT) to 89.190.50.140:39059 --->
  377. SIP/2.0 200 OK
  378. Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
  379. From: <sip:[email protected]>;tag=726823752
  380. To: <sip:[email protected]>;tag=as75b76746
  381. Call-ID: 1937419984
  382. CSeq: 21 INVITE
  383. Server: Asterisk PBX 13.6.0
  384. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  385. Supported: replaces, timer
  386. Contact: <sip:[email protected]:5060>
  387. Content-Type: application/sdp
  388. Content-Length: 332
  389.  
  390. v=0
  391. o=root 1498995089 1498995089 IN IP4 192.168.1.7
  392. s=Asterisk PBX 13.6.0
  393. c=IN IP4 192.168.1.7
  394. b=CT:384
  395. t=0 0
  396. m=audio 10466 RTP/AVP 0 3 101
  397. a=rtpmap:0 PCMU/8000
  398. a=rtpmap:3 GSM/8000
  399. a=rtpmap:101 telephone-event/8000
  400. a=fmtp:101 0-16
  401. a=maxptime:150
  402. a=sendrecv
  403. m=video 10800 RTP/AVP 34
  404. a=rtpmap:34 H263/90000
  405. a=sendrecv
  406.  
  407. <------------>
  408. Audio is at 10964
  409. Video is at 192.168.1.112:10020
  410. Adding codec ulaw to SDP
  411. Adding video codec h263 to SDP
  412. Adding codec gsm to SDP
  413. Adding non-codec 0x1 (telephone-event) to SDP
  414. Reliably Transmitting (NAT) to 89.190.50.140:5060:
  415. INVITE sip:[email protected] SIP/2.0
  416. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK212a6db9;rport
  417. Max-Forwards: 70
  418. From: "(Testovaci uzivatel" <sip:[email protected]>;tag=as7a28997b
  419. To: <sip:[email protected]:5060>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
  420. Contact: <sip:[email protected]:5060>
  421. Call-ID: [email protected]:5060
  422. CSeq: 103 INVITE
  423. User-Agent: Asterisk PBX 13.6.0
  424. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  425. Supported: replaces, timer
  426. X-asterisk-Info: SIP re-invite (External RTP bridge)
  427. Content-Type: application/sdp
  428. Content-Length: 425
  429.  
  430. v=0
  431. o=root 2143136553 2143136554 IN IP4 192.168.1.112
  432. s=Asterisk PBX 13.6.0
  433. c=IN IP4 192.168.1.112
  434. b=CT:384
  435. t=0 0
  436. m=audio 10000 RTP/AVP 0 3 101
  437. a=rtpmap:0 PCMU/8000
  438. a=rtpmap:3 GSM/8000
  439. a=rtpmap:101 telephone-event/8000
  440. a=fmtp:101 0-16
  441. a=maxptime:150
  442. a=sendrecv
  443. m=video 10020 RTP/AVP 34
  444. a=rtpmap:34 H263/90000
  445. a=fmtp:34 SQCIF=0;QCIF=3;CIF=0;CIF4=0;CIF16=0;VGA=0;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
  446. a=sendrecv
  447.  
  448. ---
  449.  
  450. <--- SIP read from UDP:89.190.50.140:5060 --->
  451. SIP/2.0 100 Trying
  452. CSeq: 103 INVITE
  453. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK212a6db9;rport=5060;received=93.89.146.5
  454. From: "(Testovaci uzivatel" <sip:[email protected]>;tag=as7a28997b
  455. Call-ID: [email protected]:5060
  456. To: <sip:[email protected]:5060>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
  457. Content-Length: 0
  458.  
  459. <------------->
  460. --- (7 headers 0 lines) ---
  461.  
  462. <--- SIP read from UDP:89.190.50.140:5060 --->
  463. SIP/2.0 200 OK
  464. CSeq: 103 INVITE
  465. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK212a6db9;rport=5060;received=93.89.146.5
  466. User-Agent: Ekiga/4.0.1
  467. From: "(Testovaci uzivatel" <sip:[email protected]>;tag=as7a28997b
  468. Call-ID: [email protected]:5060
  469. To: "101" <sip:[email protected]:5060>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
  470. Contact: "101" <sip:[email protected]>
  471. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
  472. Content-Length: 328
  473. Content-Type: application/sdp
  474.  
  475. v=0
  476. o=- 1447344693 2 IN IP4 89.190.50.140
  477. s=Ekiga/4.0.1
  478. c=IN IP4 89.190.50.140
  479. t=0 0
  480. m=audio 5066 RTP/AVP 0 101
  481. a=sendrecv
  482. a=rtpmap:0 PCMU/8000/1
  483. a=rtpmap:101 telephone-event/8000
  484. a=fmtp:101 0-16
  485. a=maxptime:240
  486. m=video 5068 RTP/AVP 34
  487. b=AS:4096
  488. b=TIAS:4096000
  489. a=recvonly
  490. a=rtpmap:34 H263/90000
  491. a=fmtp:34 QCIF=3
  492. <------------->
  493. --- (11 headers 17 lines) ---
  494. Found RTP audio format 0
  495. Found RTP audio format 101
  496. Found audio description format PCMU for ID 0
  497. Found audio description format telephone-event for ID 101
  498. Found RTP video format 34
  499. Found video description format H263 for ID 34
  500. Capabilities: us - (ulaw|h263|gsm), peer - audio=(ulaw)/video=(h263)/text=(nothing), combined - (ulaw|h263)
  501. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  502. Peer audio RTP is at port 89.190.50.140:5066
  503. Peer video RTP is at port 89.190.50.140:5068
  504. Transmitting (NAT) to 89.190.50.140:5060:
  505. ACK sip:[email protected] SIP/2.0
  506. Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK51ea1e97;rport
  507. Max-Forwards: 70
  508. From: "(Testovaci uzivatel" <sip:[email protected]>;tag=as7a28997b
  509. To: <sip:[email protected]:5060>;tag=22dba4b5-c587-e511-9b83-fcaa148f322d
  510. Contact: <sip:[email protected]:5060>
  511. Call-ID: [email protected]:5060
  512. CSeq: 103 ACK
  513. User-Agent: Asterisk PBX 13.6.0
  514. Content-Length: 0
  515.  
  516.  
  517. ---
  518. Retransmitting #1 (NAT) to 89.190.50.140:39059:
  519. SIP/2.0 200 OK
  520. Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
  521. From: <sip:[email protected]>;tag=726823752
  522. To: <sip:[email protected]>;tag=as75b76746
  523. Call-ID: 1937419984
  524. CSeq: 21 INVITE
  525. Server: Asterisk PBX 13.6.0
  526. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  527. Supported: replaces, timer
  528. Contact: <sip:[email protected]:5060>
  529. Content-Type: application/sdp
  530. Content-Length: 332
  531.  
  532. v=0
  533. o=root 1498995089 1498995089 IN IP4 192.168.1.7
  534. s=Asterisk PBX 13.6.0
  535. c=IN IP4 192.168.1.7
  536. b=CT:384
  537. t=0 0
  538. m=audio 10466 RTP/AVP 0 3 101
  539. a=rtpmap:0 PCMU/8000
  540. a=rtpmap:3 GSM/8000
  541. a=rtpmap:101 telephone-event/8000
  542. a=fmtp:101 0-16
  543. a=maxptime:150
  544. a=sendrecv
  545. m=video 10800 RTP/AVP 34
  546. a=rtpmap:34 H263/90000
  547. a=sendrecv
  548.  
  549. ---
  550. Retransmitting #2 (NAT) to 89.190.50.140:39059:
  551. SIP/2.0 200 OK
  552. Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
  553. From: <sip:[email protected]>;tag=726823752
  554. To: <sip:[email protected]>;tag=as75b76746
  555. Call-ID: 1937419984
  556. CSeq: 21 INVITE
  557. Server: Asterisk PBX 13.6.0
  558. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  559. Supported: replaces, timer
  560. Contact: <sip:[email protected]:5060>
  561. Content-Type: application/sdp
  562. Content-Length: 332
  563.  
  564. v=0
  565. o=root 1498995089 1498995089 IN IP4 192.168.1.7
  566. s=Asterisk PBX 13.6.0
  567. c=IN IP4 192.168.1.7
  568. b=CT:384
  569. t=0 0
  570. m=audio 10466 RTP/AVP 0 3 101
  571. a=rtpmap:0 PCMU/8000
  572. a=rtpmap:3 GSM/8000
  573. a=rtpmap:101 telephone-event/8000
  574. a=fmtp:101 0-16
  575. a=maxptime:150
  576. a=sendrecv
  577. m=video 10800 RTP/AVP 34
  578. a=rtpmap:34 H263/90000
  579. a=sendrecv
  580.  
  581. ---
  582.  
  583. <--- SIP read from UDP:89.190.50.140:5060 --->
  584. OPTIONS sip:[email protected] SIP/2.0
  585. Route: <sip:93.89.146.5:5060;lr>
  586. CSeq: 67 OPTIONS
  587. Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK8ccb37b9-c587-e511-9b83-fcaa148f322d;rport
  588. User-Agent: Ekiga/4.0.1
  589. From: <sip:[email protected]>;tag=70c937b9-c587-e511-9b83-fcaa148f322d
  590. Call-ID: 0cbf37b9-c587-e511-9b83-fcaa148f322d@meo2
  591. To: <sip:[email protected]>
  592. Accept: application/sdp, application/media_control+xml, application/dtmf, application/dtmf-relay
  593. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
  594. Expires: 0
  595. Content-Length: 0
  596. Max-Forwards: 70
  597.  
  598. <------------->
  599. --- (13 headers 0 lines) ---
  600. Sending to 89.190.50.140:5060 (NAT)
  601. Looking for 102 in public (domain eu.vancl.eu)
  602.  
  603. <--- Transmitting (NAT) to 89.190.50.140:5060 --->
  604. SIP/2.0 404 Not Found
  605. Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK8ccb37b9-c587-e511-9b83-fcaa148f322d;received=89.190.50.140;rport=5060
  606. From: <sip:[email protected]>;tag=70c937b9-c587-e511-9b83-fcaa148f322d
  607. To: <sip:[email protected]>;tag=as6661812b
  608. Call-ID: 0cbf37b9-c587-e511-9b83-fcaa148f322d@meo2
  609. CSeq: 67 OPTIONS
  610. Server: Asterisk PBX 13.6.0
  611. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  612. Supported: replaces, timer
  613. Accept: application/sdp
  614. Content-Length: 0
  615.  
  616.  
  617. <------------>
  618. Scheduling destruction of SIP dialog '0cbf37b9-c587-e511-9b83-fcaa148f322d@meo2' in 32000 ms (Method: OPTIONS)
  619.  
  620. <--- SIP read from UDP:89.190.50.140:5060 --->
  621. OPTIONS sip:[email protected] SIP/2.0
  622. Route: <sip:93.89.146.5:5060;lr>
  623. CSeq: 68 OPTIONS
  624. Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK38e837b9-c587-e511-9b83-fcaa148f322d;rport
  625. User-Agent: Ekiga/4.0.1
  626. From: <sip:[email protected]>;tag=76e637b9-c587-e511-9b83-fcaa148f322d
  627. Call-ID: a6de37b9-c587-e511-9b83-fcaa148f322d@meo2
  628. To: <sip:[email protected]>
  629. Accept: application/sdp, application/media_control+xml, application/dtmf, application/dtmf-relay
  630. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
  631. Expires: 0
  632. Content-Length: 0
  633. Max-Forwards: 70
  634.  
  635. <------------->
  636. --- (13 headers 0 lines) ---
  637. Sending to 89.190.50.140:5060 (NAT)
  638. Looking for 101 in public (domain eu.vancl.eu)
  639.  
  640. <--- Transmitting (NAT) to 89.190.50.140:5060 --->
  641. SIP/2.0 404 Not Found
  642. Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK38e837b9-c587-e511-9b83-fcaa148f322d;received=89.190.50.140;rport=5060
  643. From: <sip:[email protected]>;tag=76e637b9-c587-e511-9b83-fcaa148f322d
  644. To: <sip:[email protected]>;tag=as46ba0e06
  645. Call-ID: a6de37b9-c587-e511-9b83-fcaa148f322d@meo2
  646. CSeq: 68 OPTIONS
  647. Server: Asterisk PBX 13.6.0
  648. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  649. Supported: replaces, timer
  650. Accept: application/sdp
  651. Content-Length: 0
  652.  
  653.  
  654. <------------>
  655. Scheduling destruction of SIP dialog 'a6de37b9-c587-e511-9b83-fcaa148f322d@meo2' in 32000 ms (Method: OPTIONS)
  656.  
  657. <--- SIP read from UDP:89.190.50.140:5060 --->
  658. PUBLISH sip:[email protected] SIP/2.0
  659. CSeq: 69 PUBLISH
  660. Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK2616e4b9-c587-e511-9b83-fcaa148f322d;rport
  661. User-Agent: Ekiga/4.0.1
  662. From: <sip:[email protected]>
  663. Call-ID: 96496414-c587-e511-9b83-fcaa148f322d@meo2
  664. To: <sip:[email protected]>
  665. Expires: 300
  666. Event: presence
  667. Content-Length: 482
  668. Content-Type: application/pidf+xml
  669. Max-Forwards: 70
  670.  
  671. <?xml version="1.0" encoding="UTF-8"?>
  672. <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="pres:[email protected]"> <tuple id="TB628D577"> <status> <basic>open</basic> </status> <contact priority="1">sip:[email protected]</contact> <note>I&apos;m available using Ekiga</note> <timestamp>2015-11-12T17:07:21+01:00</timestamp> </tuple>
  673. </presence>
  674. <------------->
  675. --- (12 headers 3 lines) ---
  676. Retransmitting #3 (NAT) to 89.190.50.140:39059:
  677. SIP/2.0 200 OK
  678. Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
  679. From: <sip:[email protected]>;tag=726823752
  680. To: <sip:[email protected]>;tag=as75b76746
  681. Call-ID: 1937419984
  682. CSeq: 21 INVITE
  683. Server: Asterisk PBX 13.6.0
  684. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  685. Supported: replaces, timer
  686. Contact: <sip:[email protected]:5060>
  687. Content-Type: application/sdp
  688. Content-Length: 332
  689.  
  690. v=0
  691. o=root 1498995089 1498995089 IN IP4 192.168.1.7
  692. s=Asterisk PBX 13.6.0
  693. c=IN IP4 192.168.1.7
  694. b=CT:384
  695. t=0 0
  696. m=audio 10466 RTP/AVP 0 3 101
  697. a=rtpmap:0 PCMU/8000
  698. a=rtpmap:3 GSM/8000
  699. a=rtpmap:101 telephone-event/8000
  700. a=fmtp:101 0-16
  701. a=maxptime:150
  702. a=sendrecv
  703. m=video 10800 RTP/AVP 34
  704. a=rtpmap:34 H263/90000
  705. a=sendrecv
  706.  
  707. ---
  708.  
  709. <--- SIP read from UDP:89.190.50.140:5060 --->
  710. PUBLISH sip:[email protected] SIP/2.0
  711. CSeq: 69 PUBLISH
  712. Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK2616e4b9-c587-e511-9b83-fcaa148f322d;rport
  713. User-Agent: Ekiga/4.0.1
  714. From: <sip:[email protected]>
  715. Call-ID: 96496414-c587-e511-9b83-fcaa148f322d@meo2
  716. To: <sip:[email protected]>
  717. Expires: 300
  718. Event: presence
  719. Content-Length: 482
  720. Content-Type: application/pidf+xml
  721. Max-Forwards: 70
  722.  
  723. <?xml version="1.0" encoding="UTF-8"?>
  724. <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="pres:[email protected]"> <tuple id="TB628D577"> <status> <basic>open</basic> </status> <contact priority="1">sip:[email protected]</contact> <note>I&apos;m available using Ekiga</note> <timestamp>2015-11-12T17:07:21+01:00</timestamp> </tuple>
  725. </presence>
  726. <------------->
  727. --- (12 headers 3 lines) ---
  728. Retransmitting #8 (NAT) to 155.94.64.34:5089:
  729. SIP/2.0 401 Unauthorized
  730. Via: SIP/2.0/UDP 155.94.64.34:5089;branch=z9hG4bK-0575c362977ea6340ca6fcbe6260b446;received=155.94.64.34;rport=5089
  731. From: 101<sip:[email protected]>;tag=8a359551
  732. To: 991130972597723173<sip:[email protected]>;tag=as1e4415a6
  733. Call-ID: 0575c362977ea6340ca6fcbe6260b446
  734. CSeq: 1 INVITE
  735. Server: Asterisk PBX 13.6.0
  736. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  737. Supported: replaces, timer
  738. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54467743"
  739. Content-Length: 0
  740.  
  741.  
  742. ---
  743. Really destroying SIP dialog '562256a7-c587-e511-9b83-fcaa148f322d@meo2' Method: OPTIONS
  744. Really destroying SIP dialog 'b44156a7-c587-e511-9b83-fcaa148f322d@meo2' Method: OPTIONS
  745.  
  746. <--- SIP read from UDP:89.190.50.140:5060 --->
  747. PUBLISH sip:[email protected] SIP/2.0
  748. CSeq: 69 PUBLISH
  749. Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK2616e4b9-c587-e511-9b83-fcaa148f322d;rport
  750. User-Agent: Ekiga/4.0.1
  751. From: <sip:[email protected]>
  752. Call-ID: 96496414-c587-e511-9b83-fcaa148f322d@meo2
  753. To: <sip:[email protected]>
  754. Expires: 300
  755. Event: presence
  756. Content-Length: 482
  757. Content-Type: application/pidf+xml
  758. Max-Forwards: 70
  759.  
  760. <?xml version="1.0" encoding="UTF-8"?>
  761. <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="pres:[email protected]"> <tuple id="TB628D577"> <status> <basic>open</basic> </status> <contact priority="1">sip:[email protected]</contact> <note>I&apos;m available using Ekiga</note> <timestamp>2015-11-12T17:07:21+01:00</timestamp> </tuple>
  762. </presence>
  763. <------------->
  764. --- (12 headers 3 lines) ---
  765.  
  766. <--- SIP read from UDP:89.190.50.140:5060 --->
  767. PUBLISH sip:[email protected] SIP/2.0
  768. CSeq: 69 PUBLISH
  769. Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bK2616e4b9-c587-e511-9b83-fcaa148f322d;rport
  770. User-Agent: Ekiga/4.0.1
  771. From: <sip:[email protected]>
  772. Call-ID: 96496414-c587-e511-9b83-fcaa148f322d@meo2
  773. To: <sip:[email protected]>
  774. Expires: 300
  775. Event: presence
  776. Content-Length: 482
  777. Content-Type: application/pidf+xml
  778. Max-Forwards: 70
  779.  
  780. <?xml version="1.0" encoding="UTF-8"?>
  781. <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="pres:[email protected]"> <tuple id="TB628D577"> <status> <basic>open</basic> </status> <contact priority="1">sip:[email protected]</contact> <note>I&apos;m available using Ekiga</note> <timestamp>2015-11-12T17:07:21+01:00</timestamp> </tuple>
  782. </presence>
  783. <------------->
  784. --- (12 headers 3 lines) ---
  785. Retransmitting #4 (NAT) to 89.190.50.140:39059:
  786. SIP/2.0 200 OK
  787. Via: SIP/2.0/UDP 192.168.1.112:39059;branch=z9hG4bK938901029;received=89.190.50.140;rport=39059
  788. From: <sip:[email protected]>;tag=726823752
  789. To: <sip:[email protected]>;tag=as75b76746
  790. Call-ID: 1937419984
  791. CSeq: 21 INVITE
  792. Server: Asterisk PBX 13.6.0
  793. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  794. Supported: replaces, timer
  795. Contact: <sip:[email protected]:5060>
  796. Content-Type: application/sdp
  797. Content-Length: 332
  798.  
  799. v=0
  800. o=root 1498995089 1498995089 IN IP4 192.168.1.7
  801. s=Asterisk PBX 13.6.0
  802. c=IN IP4 192.168.1.7
  803. b=CT:384
  804. t=0 0
  805. m=audio 10466 RTP/AVP 0 3 101
  806. a=rtpmap:0 PCMU/8000
  807. a=rtpmap:3 GSM/8000
  808. a=rtpmap:101 telephone-event/8000
  809. a=fmtp:101 0-16
  810. a=maxptime:150
  811. a=sendrecv
  812. m=video 10800 RTP/AVP 34
  813. a=rtpmap:34 H263/90000
  814. a=sendrecv
  815.  
  816. ---
  817. voip*CLI>
  818. Disconnected from Asterisk server
  819. Asterisk cleanly ending (0).
  820. Executing last minute cleanups
  821. root@voip:~#
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!