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  1. <--- SIP read from UDP:95.26.67.247:5062 --->
  2. INVITE sip:[email protected] SIP/2.0
  3. Via: SIP/2.0/UDP 95.26.67.247:5062;rport;branch=z9hG4bK1796111490
  4. From: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  5. To: <sip:[email protected]>
  6. Call-ID: [email protected]
  7. CSeq: 20 INVITE
  8. Contact: <sip:[email protected]:5062>
  9. Max-Forwards: 70
  10. User-Agent: qutecom/rev-g-trunk
  11. Expires: 120
  12. Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
  13. Content-Type: application/sdp
  14. Content-Length: 370
  15.  
  16. v=0
  17. o=userX 20000001 20000001 IN IP4 95.26.67.247
  18. s=A call
  19. c=IN IP4 95.26.67.247
  20. t=1335471407 1335475007
  21. m=audio 10600 RTP/AVP 8 0 3 9 101
  22. a=rtpmap:8 PCMA/8000/1
  23. a=rtpmap:0 PCMU/8000/1
  24. a=rtpmap:3 GSM/8000/1
  25. a=rtpmap:9 G722/8000/1
  26. a=rtpmap:101 telephone-event/8000/1
  27. a=ptime:20
  28. m=video 10702 RTP/AVP 34 31
  29. a=rtpmap:34 H263/90000/1
  30. a=rtpmap:31 H261/90000/1
  31. <------------->
  32. --- (13 headers 15 lines) ---
  33. Sending to 95.26.67.247:5062 (NAT)
  34. Using INVITE request as basis request - [email protected]
  35. Found peer 'root' for 'root' from 95.26.67.247:5062
  36.  
  37. <--- Reliably Transmitting (NAT) to 95.26.67.247:5062 --->
  38. SIP/2.0 401 Unauthorized
  39. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK1796111490;received=95.26.67.247;rport=5062
  40. From: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  41. To: <sip:[email protected]>;tag=as50b20f6c
  42. Call-ID: [email protected]
  43. CSeq: 20 INVITE
  44. Server: SipPhone
  45. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  46. Supported: replaces, timer
  47. WWW-Authenticate: Digest algorithm=MD5, realm="ster", nonce="71a32703"
  48. Content-Length: 0
  49.  
  50.  
  51. <------------>
  52. Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
  53.  
  54. <--- SIP read from UDP:95.26.67.247:5062 --->
  55. ACK sip:[email protected] SIP/2.0
  56. Via: SIP/2.0/UDP 95.26.67.247:5062;rport;branch=z9hG4bK1796111490
  57. From: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  58. To: <sip:[email protected]>;tag=as50b20f6c
  59. Call-ID: [email protected]
  60. CSeq: 20 ACK
  61. Content-Length: 0
  62.  
  63. <------------->
  64. --- (7 headers 0 lines) ---
  65.  
  66. <--- SIP read from UDP:95.26.67.247:5062 --->
  67. INVITE sip:[email protected] SIP/2.0
  68. Via: SIP/2.0/UDP 95.26.67.247:5062;rport;branch=z9hG4bK1134124904
  69. From: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  70. To: <sip:[email protected]>
  71. Call-ID: [email protected]
  72. CSeq: 21 INVITE
  73. Contact: <sip:[email protected]:5062>
  74. Authorization: Digest username="root", realm="ster", nonce="71a32703", uri="sip:[email protected]", response="814e56bf029c6f3b240be8f8b753d5a3", algorithm=MD5
  75. Max-Forwards: 70
  76. User-Agent: qutecom/rev-g-trunk
  77. Expires: 120
  78. Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
  79. Content-Type: application/sdp
  80. Content-Length: 370
  81.  
  82. v=0
  83. o=userX 20000001 20000001 IN IP4 95.26.67.247
  84. s=A call
  85. c=IN IP4 95.26.67.247
  86. t=1335471407 1335475007
  87. m=audio 10600 RTP/AVP 8 0 3 9 101
  88. a=rtpmap:8 PCMA/8000/1
  89. a=rtpmap:0 PCMU/8000/1
  90. a=rtpmap:3 GSM/8000/1
  91. a=rtpmap:9 G722/8000/1
  92. a=rtpmap:101 telephone-event/8000/1
  93. a=ptime:20
  94. m=video 10702 RTP/AVP 34 31
  95. a=rtpmap:34 H263/90000/1
  96. a=rtpmap:31 H261/90000/1
  97. <------------->
  98. --- (14 headers 15 lines) ---
  99. Sending to 95.26.67.247:5062 (NAT)
  100. Using INVITE request as basis request - [email protected]
  101. Found peer 'root' for 'root' from 95.26.67.247:5062
  102. == Using SIP RTP CoS mark 5
  103. Found RTP audio format 8
  104. Found RTP audio format 0
  105. Found RTP audio format 3
  106. Found RTP audio format 9
  107. Found RTP audio format 101
  108. Found audio description format PCMA for ID 8
  109. Found audio description format PCMU for ID 0
  110. Found audio description format GSM for ID 3
  111. Found audio description format G722 for ID 9
  112. Found audio description format telephone-event for ID 101
  113. Found RTP video format 34
  114. Found RTP video format 31
  115. Found video description format H263 for ID 34
  116. Found video description format H261 for ID 31
  117. Capabilities: us - (ulaw|alaw|g729), peer - audio=(gsm|ulaw|alaw|g722)/video=(h261|h263)/text=(nothing), combined - (ulaw|alaw)
  118. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  119. Peer audio RTP is at port 95.26.67.247:10600
  120. Looking for 400 in sip (domain sip.mydomain.lol)
  121. list_route: hop: <sip:[email protected]:5062>
  122.  
  123. <--- Transmitting (NAT) to 95.26.67.247:5062 --->
  124. SIP/2.0 100 Trying
  125. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK1134124904;received=95.26.67.247;rport=5062
  126. From: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  127. To: <sip:[email protected]>
  128. Call-ID: [email protected]
  129. CSeq: 21 INVITE
  130. Server: SipPhone
  131. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  132. Supported: replaces, timer
  133. Contact: <sip:[email protected]:5060>
  134. Content-Length: 0
  135.  
  136.  
  137. <------------>
  138. -- Executing [400@sip:1] Answer("SIP/root-00000001", "") in new stack
  139. Audio is at 18066
  140. Adding codec 100003 (ulaw) to SDP
  141. Adding codec 100004 (alaw) to SDP
  142. Adding non-codec 0x1 (telephone-event) to SDP
  143.  
  144. <--- Reliably Transmitting (NAT) to 95.26.67.247:5062 --->
  145. SIP/2.0 200 OK
  146. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK1134124904;received=95.26.67.247;rport=5062
  147. From: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  148. To: <sip:[email protected]>;tag=as7ac7ecea
  149. Call-ID: [email protected]
  150. CSeq: 21 INVITE
  151. Server: SipPhone
  152. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  153. Supported: replaces, timer
  154. Contact: <sip:[email protected]:5060>
  155. Content-Type: application/sdp
  156. Content-Length: 290
  157.  
  158. v=0
  159. o=root 2053219929 2053219929 IN IP4 11.11.11.11
  160. s=Asterisk PBX 10.4.0-rc2
  161. c=IN IP4 11.11.11.11
  162. t=0 0
  163. m=audio 18066 RTP/AVP 0 8 101
  164. a=rtpmap:0 PCMU/8000
  165. a=rtpmap:8 PCMA/8000
  166. a=rtpmap:101 telephone-event/8000
  167. a=fmtp:101 0-16
  168. a=ptime:20
  169. a=sendrecv
  170. m=video 0 RTP/AVP 34 31
  171.  
  172. <------------>
  173. Retransmitting #1 (NAT) to 95.26.67.247:5062:
  174. SIP/2.0 200 OK
  175. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK1134124904;received=95.26.67.247;rport=5062
  176. From: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  177. To: <sip:[email protected]>;tag=as7ac7ecea
  178. Call-ID: [email protected]
  179. CSeq: 21 INVITE
  180. Server: SipPhone
  181. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  182. Supported: replaces, timer
  183. Contact: <sip:[email protected]:5060>
  184. Content-Type: application/sdp
  185. Content-Length: 290
  186.  
  187. v=0
  188. o=root 2053219929 2053219929 IN IP4 11.11.11.11
  189. s=Asterisk PBX 10.4.0-rc2
  190. c=IN IP4 11.11.11.11
  191. t=0 0
  192. m=audio 18066 RTP/AVP 0 8 101
  193. a=rtpmap:0 PCMU/8000
  194. a=rtpmap:8 PCMA/8000
  195. a=rtpmap:101 telephone-event/8000
  196. a=fmtp:101 0-16
  197. a=ptime:20
  198. a=sendrecv
  199. m=video 0 RTP/AVP 34 31
  200.  
  201. ---
  202. Retransmitting #2 (NAT) to 95.26.67.247:5062:
  203. SIP/2.0 200 OK
  204. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK1134124904;received=95.26.67.247;rport=5062
  205. From: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  206. To: <sip:[email protected]>;tag=as7ac7ecea
  207. Call-ID: [email protected]
  208. CSeq: 21 INVITE
  209. Server: SipPhone
  210. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  211. Supported: replaces, timer
  212. Contact: <sip:[email protected]:5060>
  213. Content-Type: application/sdp
  214. Content-Length: 290
  215.  
  216. v=0
  217. o=root 2053219929 2053219929 IN IP4 11.11.11.11
  218. s=Asterisk PBX 10.4.0-rc2
  219. c=IN IP4 11.11.11.11
  220. t=0 0
  221. m=audio 18066 RTP/AVP 0 8 101
  222. a=rtpmap:0 PCMU/8000
  223. a=rtpmap:8 PCMA/8000
  224. a=rtpmap:101 telephone-event/8000
  225. a=fmtp:101 0-16
  226. a=ptime:20
  227. a=sendrecv
  228. m=video 0 RTP/AVP 34 31
  229.  
  230. ---
  231. -- Executing [400@sip:2] Playback("SIP/root-00000001", "demo-congrats") in new stack
  232. -- <SIP/root-00000001> Playing 'demo-congrats.slin' (language 'en')
  233. Retransmitting #3 (NAT) to 95.26.67.247:5062:
  234. SIP/2.0 200 OK
  235. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK1134124904;received=95.26.67.247;rport=5062
  236. From: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  237. To: <sip:[email protected]>;tag=as7ac7ecea
  238. Call-ID: [email protected]
  239. CSeq: 21 INVITE
  240. Server: SipPhone
  241. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  242. Supported: replaces, timer
  243. Contact: <sip:[email protected]:5060>
  244. Content-Type: application/sdp
  245. Content-Length: 290
  246.  
  247. v=0
  248. o=root 2053219929 2053219929 IN IP4 11.11.11.11
  249. s=Asterisk PBX 10.4.0-rc2
  250. c=IN IP4 11.11.11.11
  251. t=0 0
  252. m=audio 18066 RTP/AVP 0 8 101
  253. a=rtpmap:0 PCMU/8000
  254. a=rtpmap:8 PCMA/8000
  255. a=rtpmap:101 telephone-event/8000
  256. a=fmtp:101 0-16
  257. a=ptime:20
  258. a=sendrecv
  259. m=video 0 RTP/AVP 34 31
  260.  
  261. ---
  262. Retransmitting #4 (NAT) to 95.26.67.247:5062:
  263. SIP/2.0 200 OK
  264. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK1134124904;received=95.26.67.247;rport=5062
  265. From: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  266. To: <sip:[email protected]>;tag=as7ac7ecea
  267. Call-ID: [email protected]
  268. CSeq: 21 INVITE
  269. Server: SipPhone
  270. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  271. Supported: replaces, timer
  272. Contact: <sip:[email protected]:5060>
  273. Content-Type: application/sdp
  274. Content-Length: 290
  275.  
  276. v=0
  277. o=root 2053219929 2053219929 IN IP4 11.11.11.11
  278. s=Asterisk PBX 10.4.0-rc2
  279. c=IN IP4 11.11.11.11
  280. t=0 0
  281. m=audio 18066 RTP/AVP 0 8 101
  282. a=rtpmap:0 PCMU/8000
  283. a=rtpmap:8 PCMA/8000
  284. a=rtpmap:101 telephone-event/8000
  285. a=fmtp:101 0-16
  286. a=ptime:20
  287. a=sendrecv
  288. m=video 0 RTP/AVP 34 31
  289.  
  290. ---
  291. Retransmitting #5 (NAT) to 95.26.67.247:5062:
  292. SIP/2.0 200 OK
  293. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK1134124904;received=95.26.67.247;rport=5062
  294. From: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  295. To: <sip:[email protected]>;tag=as7ac7ecea
  296. Call-ID: [email protected]
  297. CSeq: 21 INVITE
  298. Server: SipPhone
  299. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  300. Supported: replaces, timer
  301. Contact: <sip:[email protected]:5060>
  302. Content-Type: application/sdp
  303. Content-Length: 290
  304.  
  305. v=0
  306. o=root 2053219929 2053219929 IN IP4 11.11.11.11
  307. s=Asterisk PBX 10.4.0-rc2
  308. c=IN IP4 11.11.11.11
  309. t=0 0
  310. m=audio 18066 RTP/AVP 0 8 101
  311. a=rtpmap:0 PCMU/8000
  312. a=rtpmap:8 PCMA/8000
  313. a=rtpmap:101 telephone-event/8000
  314. a=fmtp:101 0-16
  315. a=ptime:20
  316. a=sendrecv
  317. m=video 0 RTP/AVP 34 31
  318.  
  319. ---
  320. Retransmitting #6 (NAT) to 95.26.67.247:5062:
  321. SIP/2.0 200 OK
  322. Via: SIP/2.0/UDP 95.26.67.247:5062;branch=z9hG4bK1134124904;received=95.26.67.247;rport=5062
  323. From: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  324. To: <sip:[email protected]>;tag=as7ac7ecea
  325. Call-ID: [email protected]
  326. CSeq: 21 INVITE
  327. Server: SipPhone
  328. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  329. Supported: replaces, timer
  330. Contact: <sip:[email protected]:5060>
  331. Content-Type: application/sdp
  332. Content-Length: 290
  333.  
  334. v=0
  335. o=root 2053219929 2053219929 IN IP4 11.11.11.11
  336. s=Asterisk PBX 10.4.0-rc2
  337. c=IN IP4 11.11.11.11
  338. t=0 0
  339. m=audio 18066 RTP/AVP 0 8 101
  340. a=rtpmap:0 PCMU/8000
  341. a=rtpmap:8 PCMA/8000
  342. a=rtpmap:101 telephone-event/8000
  343. a=fmtp:101 0-16
  344. a=ptime:20
  345. a=sendrecv
  346. m=video 0 RTP/AVP 34 31
  347.  
  348. ---
  349. [Apr 27 00:16:52] WARNING[15921]: chan_sip.c:3663 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 21 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  350. Packet timed out after 6399ms with no response
  351. [Apr 27 00:16:52] WARNING[15921]: chan_sip.c:3692 retrans_pkt: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  352. == Spawn extension (sip, 400, 2) exited non-zero on 'SIP/root-00000001'
  353. Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
  354. set_destination: Parsing <sip:[email protected]:5062> for address/port to send to
  355. set_destination: set destination to 95.26.67.247:5062
  356. Reliably Transmitting (NAT) to 95.26.67.247:5062:
  357. BYE sip:[email protected]:5062 SIP/2.0
  358. Via: SIP/2.0/UDP 11.11.11.11:5060;branch=z9hG4bK1ab6c10a;rport
  359. Max-Forwards: 70
  360. From: <sip:[email protected]>;tag=as7ac7ecea
  361. To: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  362. Call-ID: [email protected]
  363. CSeq: 102 BYE
  364. User-Agent: SipPhone
  365. Proxy-Authorization: Digest username="root", realm="ster", algorithm=MD5, uri="sip:sip.mydomain.lol", nonce="", response="1eaf83c6194a3c55299d347b99fe4fa9"
  366. X-Asterisk-HangupCause: Protocol error, unspecified
  367. X-Asterisk-HangupCauseCode: 111
  368. Content-Length: 0
  369.  
  370.  
  371. ---
  372.  
  373. <--- SIP read from UDP:95.26.67.247:5062 --->
  374. SIP/2.0 100 Trying
  375. Via: SIP/2.0/UDP 11.11.11.11:5060;branch=z9hG4bK1ab6c10a;rport=5060
  376. From: <sip:[email protected]>;tag=as7ac7ecea
  377. To: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  378. Call-ID: [email protected]
  379. CSeq: 102 BYE
  380. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
  381. Content-Length: 0
  382.  
  383. <------------->
  384. --- (8 headers 0 lines) ---
  385.  
  386. <--- SIP read from UDP:95.26.67.247:5062 --->
  387. SIP/2.0 200 OK
  388. Via: SIP/2.0/UDP 11.11.11.11:5060;branch=z9hG4bK1ab6c10a;rport=5060
  389. From: <sip:[email protected]>;tag=as7ac7ecea
  390. To: root_sip.mydomain.lol <sip:[email protected]>;tag=566068988
  391. Call-ID: [email protected]
  392. CSeq: 102 BYE
  393. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
  394. Content-Length: 0
  395.  
  396. <------------->
  397. --- (8 headers 0 lines) ---
  398. SIP Response message for INCOMING dialog BYE arrived
  399. Really destroying SIP dialog '[email protected]' Method: INVITE
  400. Reliably Transmitting (NAT) to 95.26.67.247:5062:
  401. OPTIONS sip:[email protected]:5062 SIP/2.0
  402. Via: SIP/2.0/UDP 11.11.11.11:5060;branch=z9hG4bK3c29f994;rport
  403. Max-Forwards: 70
  404. From: "asterisk" <sip:[email protected]>;tag=as747e7700
  405. To: <sip:[email protected]:5062>
  406. Contact: <sip:[email protected]:5060>
  407. Call-ID: [email protected]:5060
  408. CSeq: 102 OPTIONS
  409. User-Agent: SipPhone
  410. Date: Thu, 26 Apr 2012 20:16:57 GMT
  411. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  412. Supported: replaces, timer
  413. Content-Length: 0
  414.  
  415.  
  416. ---
  417.  
  418. <--- SIP read from UDP:95.26.67.247:5062 --->
  419. SIP/2.0 100 Trying
  420. Via: SIP/2.0/UDP 11.11.11.11:5060;branch=z9hG4bK3c29f994;rport=5060
  421. From: "asterisk" <sip:[email protected]:5060>;tag=as747e7700
  422. To: <sip:[email protected]:5062>
  423. Call-ID: [email protected]:5060
  424. CSeq: 102 OPTIONS
  425. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
  426. Content-Length: 0
  427.  
  428. <------------->
  429. --- (8 headers 0 lines) ---
  430.  
  431. <--- SIP read from UDP:95.26.67.247:5062 --->
  432. SIP/2.0 200 OK
  433. Via: SIP/2.0/UDP 11.11.11.11:5060;branch=z9hG4bK3c29f994;rport=5060
  434. From: "asterisk" <sip:[email protected]:5060>;tag=as747e7700
  435. To: <sip:[email protected]:5062>;tag=489205442
  436. Call-ID: [email protected]:5060
  437. CSeq: 102 OPTIONS
  438. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO, REFER
  439. Content-Length: 0
  440.  
  441. <------------->
  442. --- (8 headers 0 lines) ---
  443. Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
  444. ster*CLI> sip set debug off
  445. SIP Debugging Disabled
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