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  1. ast*CLI> sip set debug on
  2. SIP Debugging enabled
  3.  
  4. <--- SIP read from WS:192.168.88.188:49766 --->
  5. INVITE sip:893@192.168.88.251 SIP/2.0
  6. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK4TJwAbhyvycP8FPc8pd2ftH1Jh71laTx;rport
  7. From: "892"<sip:892@192.168.88.251>;tag=3qmJ7kplqHaoNe2PmT96
  8. To: <sip:893@192.168.88.251>
  9. Contact: "892"<sip:892@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
  10. Call-ID: fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5
  11. CSeq: 968 INVITE
  12. Content-Type: application/sdp
  13. Content-Length: 1589
  14. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  15. Max-Forwards: 70
  16. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  17. Organization: Doubango Telecom
  18.  
  19. v=0
  20. o=- 1167517370132397300 2 IN IP4 127.0.0.1
  21. s=Doubango Telecom - chrome
  22. t=0 0
  23. a=group:BUNDLE audio
  24. a=msid-semantic: WMS LlZMU4MJprpSGHd9dvxx4KtwecIfF7DckvJE
  25. m=audio 60358 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
  26. c=IN IP4 192.168.88.188
  27. a=rtcp:60358 IN IP4 192.168.88.188
  28. a=candidate:3254617721 1 udp 2122194687 192.168.88.188 60358 typ host generation 0
  29. a=candidate:3254617721 2 udp 2122194687 192.168.88.188 60358 typ host generation 0
  30. a=candidate:2407430793 1 tcp 1518214911 192.168.88.188 0 typ host tcptype active generation 0
  31. a=candidate:2407430793 2 tcp 1518214911 192.168.88.188 0 typ host tcptype active generation 0
  32. a=ice-ufrag:87X/I7LlWhWwClf2
  33. a=ice-pwd:l1tkNrCq4ExXhkVxH2zFotHX
  34. a=ice-options:google-ice
  35. a=fingerprint:sha-256 51:EF:F4:71:0C:EE:2F:A6:D2:2A:85:60:AC:26:68:C2:D3:57:39:88:55:23:53:0C:40:3A:2D:A5:4E:D9:A1:0F
  36. a=setup:actpass
  37. a=mid:audio
  38. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  39. a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  40. a=sendrecv
  41. a=rtcp-mux
  42. a=rtpmap:111 opus/48000/2
  43. a=fmtp:111 minptime=10
  44. a=rtpmap:103 ISAC/16000
  45. a=rtpmap:104 ISAC/32000
  46. a=rtpmap:9 G722/8000
  47. a=rtpmap:0 PCMU/8000
  48. a=rtpmap:8 PCMA/8000
  49. a=rtpmap:106 CN/32000
  50. a=rtpmap:105 CN/16000
  51. a=rtpmap:13 CN/8000
  52. a=rtpmap:126 telephone-event/8000
  53. a=maxptime:60
  54. a=ssrc:445475247 cname:eimWKxkI4IntBTKg
  55. a=ssrc:445475247 msid:LlZMU4MJprpSGHd9dvxx4KtwecIfF7DckvJE f1141c3f-0b1f-4921-aedb-5d4de42c7143
  56. a=ssrc:445475247 mslabel:LlZMU4MJprpSGHd9dvxx4KtwecIfF7DckvJE
  57. a=ssrc:445475247 label:f1141c3f-0b1f-4921-aedb-5d4de42c7143
  58. <------------->
  59. --- (13 headers 39 lines) ---
  60. Using INVITE request as basis request - fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5
  61. Found peer '892' for '892' from 192.168.88.188:49766
  62.  
  63. <--- Reliably Transmitting (no NAT) to 192.168.88.188:5060 --->
  64. SIP/2.0 401 Unauthorized
  65. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK4TJwAbhyvycP8FPc8pd2ftH1Jh71laTx;rport;received=192.168.88.188
  66. From: "892"<sip:892@192.168.88.251>;tag=3qmJ7kplqHaoNe2PmT96
  67. To: <sip:893@192.168.88.251>;tag=as78a0a681
  68. Call-ID: fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5
  69. CSeq: 968 INVITE
  70. Server: Asterisk PBX 13.2.0
  71. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  72. Supported: replaces, timer
  73. WWW-Authenticate: Digest algorithm=MD5, realm="192.168.88.251", nonce="37c2ba18"
  74. Content-Length: 0
  75.  
  76.  
  77. <------------>
  78. Scheduling destruction of SIP dialog 'fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5' in 32000 ms (Method: INVITE)
  79.  
  80. <--- SIP read from WS:192.168.88.188:49766 --->
  81. ACK sip:893@192.168.88.251 SIP/2.0
  82. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK4TJwAbhyvycP8FPc8pd2ftH1Jh71laTx;rport
  83. From: "892"<sip:892@192.168.88.251>;tag=3qmJ7kplqHaoNe2PmT96
  84. To: <sip:893@192.168.88.251>;tag=as78a0a681
  85. Call-ID: fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5
  86. CSeq: 968 ACK
  87. Content-Length: 0
  88. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  89. Max-Forwards: 70
  90.  
  91. <------------->
  92. --- (9 headers 0 lines) ---
  93.  
  94. <--- SIP read from WS:192.168.88.188:49766 --->
  95. INVITE sip:893@192.168.88.251 SIP/2.0
  96. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKTTiLnzTNhdTum42O23Cq0slLIr6raKO3;rport
  97. From: "892"<sip:892@192.168.88.251>;tag=3qmJ7kplqHaoNe2PmT96
  98. To: <sip:893@192.168.88.251>
  99. Contact: "892"<sip:892@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
  100. Call-ID: fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5
  101. CSeq: 969 INVITE
  102. Content-Type: application/sdp
  103. Content-Length: 1589
  104. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  105. Max-Forwards: 70
  106. Authorization: Digest username="892",realm="192.168.88.251",nonce="37c2ba18",uri="sip:893@192.168.88.251",response="54a8a58fae96cc1583fea91021a16d41",algorithm=MD5
  107. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  108. Organization: Doubango Telecom
  109.  
  110. v=0
  111. o=- 1167517370132397300 2 IN IP4 127.0.0.1
  112. s=Doubango Telecom - chrome
  113. t=0 0
  114. a=group:BUNDLE audio
  115. a=msid-semantic: WMS LlZMU4MJprpSGHd9dvxx4KtwecIfF7DckvJE
  116. m=audio 60358 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
  117. c=IN IP4 192.168.88.188
  118. a=rtcp:60358 IN IP4 192.168.88.188
  119. a=candidate:3254617721 1 udp 2122194687 192.168.88.188 60358 typ host generation 0
  120. a=candidate:3254617721 2 udp 2122194687 192.168.88.188 60358 typ host generation 0
  121. a=candidate:2407430793 1 tcp 1518214911 192.168.88.188 0 typ host tcptype active generation 0
  122. a=candidate:2407430793 2 tcp 1518214911 192.168.88.188 0 typ host tcptype active generation 0
  123. a=ice-ufrag:87X/I7LlWhWwClf2
  124. a=ice-pwd:l1tkNrCq4ExXhkVxH2zFotHX
  125. a=ice-options:google-ice
  126. a=fingerprint:sha-256 51:EF:F4:71:0C:EE:2F:A6:D2:2A:85:60:AC:26:68:C2:D3:57:39:88:55:23:53:0C:40:3A:2D:A5:4E:D9:A1:0F
  127. a=setup:actpass
  128. a=mid:audio
  129. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  130. a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  131. a=sendrecv
  132. a=rtcp-mux
  133. a=rtpmap:111 opus/48000/2
  134. a=fmtp:111 minptime=10
  135. a=rtpmap:103 ISAC/16000
  136. a=rtpmap:104 ISAC/32000
  137. a=rtpmap:9 G722/8000
  138. a=rtpmap:0 PCMU/8000
  139. a=rtpmap:8 PCMA/8000
  140. a=rtpmap:106 CN/32000
  141. a=rtpmap:105 CN/16000
  142. a=rtpmap:13 CN/8000
  143. a=rtpmap:126 telephone-event/8000
  144. a=maxptime:60
  145. a=ssrc:445475247 cname:eimWKxkI4IntBTKg
  146. a=ssrc:445475247 msid:LlZMU4MJprpSGHd9dvxx4KtwecIfF7DckvJE f1141c3f-0b1f-4921-aedb-5d4de42c7143
  147. a=ssrc:445475247 mslabel:LlZMU4MJprpSGHd9dvxx4KtwecIfF7DckvJE
  148. a=ssrc:445475247 label:f1141c3f-0b1f-4921-aedb-5d4de42c7143
  149. <------------->
  150. --- (14 headers 39 lines) ---
  151. Using INVITE request as basis request - fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5
  152. Found peer '892' for '892' from 192.168.88.188:49766
  153. == Using SIP RTP CoS mark 5
  154. Found RTP audio format 111
  155. Found RTP audio format 103
  156. Found RTP audio format 104
  157. Found RTP audio format 9
  158. Found RTP audio format 0
  159. Found RTP audio format 8
  160. Found RTP audio format 106
  161. Found RTP audio format 105
  162. Found RTP audio format 13
  163. Found RTP audio format 126
  164. Found audio description format opus for ID 111
  165. Found unknown media description format ISAC for ID 103
  166. Found unknown media description format ISAC for ID 104
  167. Found audio description format G722 for ID 9
  168. Found audio description format PCMU for ID 0
  169. Found audio description format PCMA for ID 8
  170. Found unknown media description format CN for ID 106
  171. Found unknown media description format CN for ID 105
  172. Found audio description format CN for ID 13
  173. Found audio description format telephone-event for ID 126
  174. Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  175. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
  176. Peer audio RTP is at port 192.168.88.188:60358
  177. Looking for 893 in default (domain 192.168.88.251)
  178. sip_route_dump: route/path hop: <sip:892@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>
  179.  
  180. <--- Transmitting (no NAT) to 192.168.88.188:5060 --->
  181. SIP/2.0 100 Trying
  182. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKTTiLnzTNhdTum42O23Cq0slLIr6raKO3;rport;received=192.168.88.188
  183. From: "892"<sip:892@192.168.88.251>;tag=3qmJ7kplqHaoNe2PmT96
  184. To: <sip:893@192.168.88.251>
  185. Call-ID: fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5
  186. CSeq: 969 INVITE
  187. Server: Asterisk PBX 13.2.0
  188. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  189. Supported: replaces, timer
  190. Contact: <sip:893@192.168.88.251:5060;transport=WS>
  191. Content-Length: 0
  192.  
  193.  
  194. <------------>
  195. -- Executing [893@default:1] Dial("SIP/892-00000094", "SIP/893") in new stack
  196. == Using SIP RTP CoS mark 5
  197. Audio is at 18840
  198. Adding codec ulaw to SDP
  199. Adding codec alaw to SDP
  200. Adding codec gsm to SDP
  201. Adding non-codec 0x1 (telephone-event) to SDP
  202. Reliably Transmitting (no NAT) to 192.168.88.189:54237:
  203. INVITE sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
  204. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK0be4fa34
  205. Max-Forwards: 70
  206. From: "892" <sip:892@192.168.88.251>;tag=as1fafb47c
  207. To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
  208. Contact: <sip:892@192.168.88.251:5060;transport=WS>
  209. Call-ID: 6283efbe61b581826d757f084eb27585@192.168.88.251:5060
  210. CSeq: 102 INVITE
  211. User-Agent: Asterisk PBX 13.2.0
  212. Date: Wed, 18 Feb 2015 01:54:03 GMT
  213. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  214. Supported: replaces, timer
  215. Content-Type: application/sdp
  216. Content-Length: 676
  217.  
  218. v=0
  219. o=root 1029705251 1029705251 IN IP4 192.168.88.251
  220. s=Asterisk PBX 13.2.0
  221. c=IN IP4 192.168.88.251
  222. t=0 0
  223. m=audio 18840 RTP/SAVPF 0 8 3 101
  224. a=rtpmap:0 PCMU/8000
  225. a=rtpmap:8 PCMA/8000
  226. a=rtpmap:3 GSM/8000
  227. a=rtpmap:101 telephone-event/8000
  228. a=fmtp:101 0-16
  229. a=maxptime:150
  230. a=ice-ufrag:1ede86030f4ad541363f29a63a8c0931
  231. a=ice-pwd:2d82634148af312d55cf20030e69ab63
  232. a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 18840 typ host
  233. a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 18841 typ host
  234. a=connection:new
  235. a=setup:actpass
  236. a=fingerprint:SHA-256 0D:6A:59:76:3A:91:CF:86:2D:91:D5:8D:D4:95:CC:06:CB:16:9C:83:4C:C4:34:B1:64:2F:91:10:E8:76:A2:52
  237. a=sendrecv
  238.  
  239. ---
  240. -- Called SIP/893
  241.  
  242. <--- SIP read from WS:192.168.88.189:54237 --->
  243. SIP/2.0 100 Trying (sent from the Transaction Layer)
  244. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK0be4fa34
  245. From: "892"<sip:892@192.168.88.251>;tag=as1fafb47c
  246. To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>
  247. Call-ID: 6283efbe61b581826d757f084eb27585@192.168.88.251:5060
  248. CSeq: 102 INVITE
  249. Content-Length: 0
  250.  
  251. <------------->
  252. --- (7 headers 0 lines) ---
  253.  
  254. <--- SIP read from WS:192.168.88.189:54237 --->
  255. SIP/2.0 180 Ringing
  256. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK0be4fa34
  257. From: "892"<sip:892@192.168.88.251>;tag=as1fafb47c
  258. To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=lN4NH7rAcX5wob9xZ0zX
  259. Contact: <sip:893@df7jal23ls0d.invalid;transport=ws>
  260. Call-ID: 6283efbe61b581826d757f084eb27585@192.168.88.251:5060
  261. CSeq: 102 INVITE
  262. Content-Length: 0
  263. Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
  264.  
  265. <------------->
  266. --- (9 headers 0 lines) ---
  267. sip_route_dump: route/path hop: <sip:893@df7jal23ls0d.invalid;transport=ws>
  268. -- SIP/893-00000095 is ringing
  269.  
  270. <--- Transmitting (no NAT) to 192.168.88.188:5060 --->
  271. SIP/2.0 180 Ringing
  272. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKTTiLnzTNhdTum42O23Cq0slLIr6raKO3;rport;received=192.168.88.188
  273. From: "892"<sip:892@192.168.88.251>;tag=3qmJ7kplqHaoNe2PmT96
  274. To: <sip:893@192.168.88.251>;tag=as46a08706
  275. Call-ID: fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5
  276. CSeq: 969 INVITE
  277. Server: Asterisk PBX 13.2.0
  278. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  279. Supported: replaces, timer
  280. Contact: <sip:893@192.168.88.251:5060;transport=WS>
  281. Content-Length: 0
  282.  
  283.  
  284. <------------>
  285. Really destroying SIP dialog 'dc8a463e-01f5-927f-6d0f-31c24e5c091e' Method: REGISTER
  286.  
  287. <--- Transmitting (no NAT) to 192.168.88.182:5060 --->
  288. SIP/2.0 180 Ringing
  289. Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK2955643;received=192.168.88.182
  290. From: <sip:889@192.168.88.251>;tag=psh84tc644
  291. To: <sip:888@192.168.88.251>;tag=as777e89a5
  292. Call-ID: 32s4vm781dqfjpfs8q4v
  293. CSeq: 5147 INVITE
  294. Server: Asterisk PBX 13.2.0
  295. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  296. Supported: replaces, timer
  297. Contact: <sip:888@192.168.88.251:5060;transport=WS>
  298. Content-Length: 0
  299.  
  300.  
  301. <------------>
  302.  
  303. <--- SIP read from WS:192.168.88.189:54237 --->
  304. SIP/2.0 200 OK
  305. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK0be4fa34
  306. From: "892"<sip:892@192.168.88.251>;tag=as1fafb47c
  307. To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=lN4NH7rAcX5wob9xZ0zX
  308. Contact: <sip:893@df7jal23ls0d.invalid;transport=ws>
  309. Call-ID: 6283efbe61b581826d757f084eb27585@192.168.88.251:5060
  310. CSeq: 102 INVITE
  311. Content-Type: application/sdp
  312. Content-Length: 1169
  313. Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
  314.  
  315. v=0
  316. o=- 7655595058993037000 2 IN IP4 127.0.0.1
  317. s=Doubango Telecom - chrome
  318. t=0 0
  319. a=msid-semantic: WMS WB9Fou7XYMih0LTh6wFWUScNqcwRy6AjNVRM
  320. m=audio 51700 UDP/TLS/RTP/SAVPF 0 8 101
  321. c=IN IP4 192.168.88.189
  322. a=rtcp:51701 IN IP4 192.168.88.189
  323. a=candidate:2034360604 1 udp 2122194687 192.168.88.189 51700 typ host generation 0
  324. a=candidate:2034360604 2 udp 2122194686 192.168.88.189 51701 typ host generation 0
  325. a=candidate:935468524 1 tcp 1518214911 192.168.88.189 0 typ host tcptype active generation 0
  326. a=candidate:935468524 2 tcp 1518214910 192.168.88.189 0 typ host tcptype active generation 0
  327. a=ice-ufrag:vP3ahbSgwUbTXQzm
  328. a=ice-pwd:ci2bsDxZ+AoOZBJlZy7iAx71
  329. a=fingerprint:sha-256 33:7A:5F:05:75:AB:62:A6:20:78:D5:F7:EF:BB:BB:3A:0A:D2:89:0F:DA:4D:22:50:8C:E7:90:A4:51:34:F8:61
  330. a=setup:active
  331. a=mid:audio
  332. a=sendrecv
  333. a=rtpmap:0 PCMU/8000
  334. a=rtpmap:8 PCMA/8000
  335. a=rtpmap:101 telephone-event/8000
  336. a=ssrc:392221519 cname:x+dCQ2RjVZWzZivM
  337. a=ssrc:392221519 msid:WB9Fou7XYMih0LTh6wFWUScNqcwRy6AjNVRM 918c7c04-5378-4f32-835c-e3bbbabf864b
  338. a=ssrc:392221519 mslabel:WB9Fou7XYMih0LTh6wFWUScNqcwRy6AjNVRM
  339. a=ssrc:392221519 label:918c7c04-5378-4f32-835c-e3bbbabf864b
  340. <------------->
  341. --- (10 headers 25 lines) ---
  342. Found RTP audio format 0
  343. Found RTP audio format 8
  344. Found RTP audio format 101
  345. Found audio description format PCMU for ID 0
  346. Found audio description format PCMA for ID 8
  347. Found audio description format telephone-event for ID 101
  348. Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  349. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  350. Peer audio RTP is at port 192.168.88.189:51700
  351. sip_route_dump: route/path hop: <sip:893@df7jal23ls0d.invalid;transport=ws>
  352.  
  353. <--- Transmitting (no NAT) to 192.168.88.182:5060 --->
  354. SIP/2.0 180 Ringing
  355. Via: SIP/2.0/WS 192.0.2.134;branch=z9hG4bK5314555;received=192.168.88.182
  356. From: <sip:889@192.168.88.251>;tag=s380d81cec
  357. To: <sip:888@192.168.88.251>;tag=as2df952cd
  358. Call-ID: j5m6suht7i1aolb9i99h
  359. CSeq: 8678 INVITE
  360. Server: Asterisk PBX 13.2.0
  361. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  362. Supported: replaces, timer
  363. Contact: <sip:888@192.168.88.251:5060;transport=WS>
  364. Content-Length: 0
  365.  
  366.  
  367. <------------>
  368. [Feb 18 03:54:22] ERROR[5857][C-00000068]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
  369. [Feb 18 03:54:22] WARNING[5857][C-00000068]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
  370. set_destination: Parsing <sip:893@df7jal23ls0d.invalid;transport=ws> for address/port to send to
  371. set_destination: URI is for WebSocket, we can't set destination
  372. Transmitting (no NAT) to 192.168.88.189:54237:
  373. ACK sip:893@df7jal23ls0d.invalid;transport=ws SIP/2.0
  374. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK6d915098
  375. Max-Forwards: 70
  376. From: "892" <sip:892@192.168.88.251>;tag=as1fafb47c
  377. To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=lN4NH7rAcX5wob9xZ0zX
  378. Contact: <sip:892@192.168.88.251:5060;transport=WS>
  379. Call-ID: 6283efbe61b581826d757f084eb27585@192.168.88.251:5060
  380. CSeq: 102 ACK
  381. User-Agent: Asterisk PBX 13.2.0
  382. Content-Length: 0
  383.  
  384.  
  385. ---
  386. -- SIP/893-00000095 answered SIP/892-00000094
  387. Audio is at 18542
  388. Adding codec ulaw to SDP
  389. Adding codec alaw to SDP
  390. Adding codec gsm to SDP
  391. Adding non-codec 0x1 (telephone-event) to SDP
  392.  
  393. <--- Reliably Transmitting (no NAT) to 192.168.88.188:5060 --->
  394. SIP/2.0 200 OK
  395. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKTTiLnzTNhdTum42O23Cq0slLIr6raKO3;rport;received=192.168.88.188
  396. From: "892"<sip:892@192.168.88.251>;tag=3qmJ7kplqHaoNe2PmT96
  397. To: <sip:893@192.168.88.251>;tag=as46a08706
  398. Call-ID: fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5
  399. CSeq: 969 INVITE
  400. Server: Asterisk PBX 13.2.0
  401. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  402. Supported: replaces, timer
  403. Contact: <sip:893@192.168.88.251:5060;transport=WS>
  404. Content-Type: application/sdp
  405. Content-Length: 675
  406.  
  407. v=0
  408. o=root 1555078210 1555078210 IN IP4 192.168.88.251
  409. s=Asterisk PBX 13.2.0
  410. c=IN IP4 192.168.88.251
  411. t=0 0
  412. m=audio 18542 RTP/SAVPF 0 8 3 126
  413. a=rtpmap:0 PCMU/8000
  414. a=rtpmap:8 PCMA/8000
  415. a=rtpmap:3 GSM/8000
  416. a=rtpmap:126 telephone-event/8000
  417. a=fmtp:126 0-16
  418. a=maxptime:150
  419. a=ice-ufrag:75c8b30f1ab24b681011e59639adf492
  420. a=ice-pwd:3bac2b2f486b6c926e7e54f479b79040
  421. a=candidate:Hc0a858fb 1 UDP 2130706431 192.168.88.251 18542 typ host
  422. a=candidate:Hc0a858fb 2 UDP 2130706430 192.168.88.251 18543 typ host
  423. a=connection:new
  424. a=setup:active
  425. a=fingerprint:SHA-256 0D:6A:59:76:3A:91:CF:86:2D:91:D5:8D:D4:95:CC:06:CB:16:9C:83:4C:C4:34:B1:64:2F:91:10:E8:76:A2:52
  426. a=sendrecv
  427.  
  428. <------------>
  429. -- Channel SIP/892-00000094 joined 'simple_bridge' basic-bridge <1afd0190-37f4-424f-bc2d-833cfe407d2b>
  430. -- Channel SIP/893-00000095 joined 'simple_bridge' basic-bridge <1afd0190-37f4-424f-bc2d-833cfe407d2b>
  431.  
  432. <--- SIP read from WS:192.168.88.189:54237 --->
  433. SIP/2.0 200 OK
  434. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK0be4fa34
  435. From: "892"<sip:892@192.168.88.251>;tag=as1fafb47c
  436. To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=lN4NH7rAcX5wob9xZ0zX
  437. Contact: <sip:893@df7jal23ls0d.invalid;transport=ws>
  438. Call-ID: 6283efbe61b581826d757f084eb27585@192.168.88.251:5060
  439. CSeq: 102 INVITE
  440. Content-Type: application/sdp
  441. Content-Length: 1169
  442. Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
  443.  
  444. v=0
  445. o=- 7655595058993037000 2 IN IP4 127.0.0.1
  446. s=Doubango Telecom - chrome
  447. t=0 0
  448. a=msid-semantic: WMS WB9Fou7XYMih0LTh6wFWUScNqcwRy6AjNVRM
  449. m=audio 51700 UDP/TLS/RTP/SAVPF 0 8 101
  450. c=IN IP4 192.168.88.189
  451. a=rtcp:51701 IN IP4 192.168.88.189
  452. a=candidate:2034360604 1 udp 2122194687 192.168.88.189 51700 typ host generation 0
  453. a=candidate:2034360604 2 udp 2122194686 192.168.88.189 51701 typ host generation 0
  454. a=candidate:935468524 1 tcp 1518214911 192.168.88.189 0 typ host tcptype active generation 0
  455. a=candidate:935468524 2 tcp 1518214910 192.168.88.189 0 typ host tcptype active generation 0
  456. a=ice-ufrag:vP3ahbSgwUbTXQzm
  457. a=ice-pwd:ci2bsDxZ+AoOZBJlZy7iAx71
  458. a=fingerprint:sha-256 33:7A:5F:05:75:AB:62:A6:20:78:D5:F7:EF:BB:BB:3A:0A:D2:89:0F:DA:4D:22:50:8C:E7:90:A4:51:34:F8:61
  459. a=setup:active
  460. a=mid:audio
  461. a=sendrecv
  462. a=rtpmap:0 PCMU/8000
  463. a=rtpmap:8 PCMA/8000
  464. a=rtpmap:101 telephone-event/8000
  465. a=ssrc:392221519 cname:x+dCQ2RjVZWzZivM
  466. a=ssrc:392221519 msid:WB9Fou7XYMih0LTh6wFWUScNqcwRy6AjNVRM 918c7c04-5378-4f32-835c-e3bbbabf864b
  467. a=ssrc:392221519 mslabel:WB9Fou7XYMih0LTh6wFWUScNqcwRy6AjNVRM
  468. a=ssrc:392221519 label:918c7c04-5378-4f32-835c-e3bbbabf864b
  469. <------------->
  470. --- (10 headers 25 lines) ---
  471.  
  472. <--- SIP read from WS:192.168.88.188:49766 --->
  473. ACK sip:893@192.168.88.251:5060;transport=WS SIP/2.0
  474. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKdOTEecbZXPg7kSW2fyVX;rport
  475. From: "892"<sip:892@192.168.88.251>;tag=3qmJ7kplqHaoNe2PmT96
  476. To: <sip:893@192.168.88.251>;tag=as46a08706
  477. Contact: "892"<sip:892@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
  478. Call-ID: fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5
  479. CSeq: 969 ACK
  480. Content-Length: 0
  481. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  482. Max-Forwards: 70
  483. Authorization: Digest username="892",realm="192.168.88.251",nonce="37c2ba18",uri="sip:893@192.168.88.251:5060;transport=WS",response="b62459cf24c2902bef7454f1564e2a58",algorithm=MD5
  484. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  485. Organization: Doubango Telecom
  486.  
  487. <------------->
  488. --- (13 headers 0 lines) ---
  489. [Feb 18 03:54:23] ERROR[5857][C-00000068]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
  490. [Feb 18 03:54:23] WARNING[5857][C-00000068]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
  491. set_destination: Parsing <sip:893@df7jal23ls0d.invalid;transport=ws> for address/port to send to
  492. set_destination: URI is for WebSocket, we can't set destination
  493. Transmitting (no NAT) to 192.168.88.189:54237:
  494. ACK sip:893@df7jal23ls0d.invalid;transport=ws SIP/2.0
  495. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK35a10347
  496. Max-Forwards: 70
  497. From: "892" <sip:892@192.168.88.251>;tag=as1fafb47c
  498. To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=lN4NH7rAcX5wob9xZ0zX
  499. Contact: <sip:892@192.168.88.251:5060;transport=WS>
  500. Call-ID: 6283efbe61b581826d757f084eb27585@192.168.88.251:5060
  501. CSeq: 102 ACK
  502. User-Agent: Asterisk PBX 13.2.0
  503. Content-Length: 0
  504.  
  505.  
  506. ---
  507. > 0x7fe19c18f6f0 -- Probation passed - setting RTP source address to 192.168.88.189:51700
  508.  
  509. <--- SIP read from WS:192.168.88.189:54237 --->
  510. SIP/2.0 200 OK
  511. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK0be4fa34
  512. From: "892"<sip:892@192.168.88.251>;tag=as1fafb47c
  513. To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=lN4NH7rAcX5wob9xZ0zX
  514. Contact: <sip:893@df7jal23ls0d.invalid;transport=ws>
  515. Call-ID: 6283efbe61b581826d757f084eb27585@192.168.88.251:5060
  516. CSeq: 102 INVITE
  517. Content-Type: application/sdp
  518. Content-Length: 1169
  519. Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
  520.  
  521. v=0
  522. o=- 7655595058993037000 2 IN IP4 127.0.0.1
  523. s=Doubango Telecom - chrome
  524. t=0 0
  525. a=msid-semantic: WMS WB9Fou7XYMih0LTh6wFWUScNqcwRy6AjNVRM
  526. m=audio 51700 UDP/TLS/RTP/SAVPF 0 8 101
  527. c=IN IP4 192.168.88.189
  528. a=rtcp:51701 IN IP4 192.168.88.189
  529. a=candidate:2034360604 1 udp 2122194687 192.168.88.189 51700 typ host generation 0
  530. a=candidate:2034360604 2 udp 2122194686 192.168.88.189 51701 typ host generation 0
  531. a=candidate:935468524 1 tcp 1518214911 192.168.88.189 0 typ host tcptype active generation 0
  532. a=candidate:935468524 2 tcp 1518214910 192.168.88.189 0 typ host tcptype active generation 0
  533. a=ice-ufrag:vP3ahbSgwUbTXQzm
  534. a=ice-pwd:ci2bsDxZ+AoOZBJlZy7iAx71
  535. a=fingerprint:sha-256 33:7A:5F:05:75:AB:62:A6:20:78:D5:F7:EF:BB:BB:3A:0A:D2:89:0F:DA:4D:22:50:8C:E7:90:A4:51:34:F8:61
  536. a=setup:active
  537. a=mid:audio
  538. a=sendrecv
  539. a=rtpmap:0 PCMU/8000
  540. a=rtpmap:8 PCMA/8000
  541. a=rtpmap:101 telephone-event/8000
  542. a=ssrc:392221519 cname:x+dCQ2RjVZWzZivM
  543. a=ssrc:392221519 msid:WB9Fou7XYMih0LTh6wFWUScNqcwRy6AjNVRM 918c7c04-5378-4f32-835c-e3bbbabf864b
  544. a=ssrc:392221519 mslabel:WB9Fou7XYMih0LTh6wFWUScNqcwRy6AjNVRM
  545. a=ssrc:392221519 label:918c7c04-5378-4f32-835c-e3bbbabf864b
  546. <------------->
  547. --- (10 headers 25 lines) ---
  548. [Feb 18 03:54:23] ERROR[5857][C-00000068]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
  549. [Feb 18 03:54:23] WARNING[5857][C-00000068]: chan_sip.c:16158 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
  550. set_destination: Parsing <sip:893@df7jal23ls0d.invalid;transport=ws> for address/port to send to
  551. set_destination: URI is for WebSocket, we can't set destination
  552. Transmitting (no NAT) to 192.168.88.189:54237:
  553. ACK sip:893@df7jal23ls0d.invalid;transport=ws SIP/2.0
  554. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK77e26327
  555. Max-Forwards: 70
  556. From: "892" <sip:892@192.168.88.251>;tag=as1fafb47c
  557. To: <sip:893@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;tag=lN4NH7rAcX5wob9xZ0zX
  558. Contact: <sip:892@192.168.88.251:5060;transport=WS>
  559. Call-ID: 6283efbe61b581826d757f084eb27585@192.168.88.251:5060
  560. CSeq: 102 ACK
  561. User-Agent: Asterisk PBX 13.2.0
  562. Content-Length: 0
  563.  
  564.  
  565. ---
  566. Really destroying SIP dialog '5cdfabc0-6c7e-2ad8-2c8e-6f2fd316f9bc' Method: REGISTER
  567.  
  568. <--- SIP read from WS:192.168.88.189:54237 --->
  569. BYE sip:892@192.168.88.251:5060;transport=WS SIP/2.0
  570. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKsZgzAMrosRSIqkJ2DUPJz0lK6du69ETK;rport
  571. From: <sip:893@df7jal23ls0d.invalid>;tag=lN4NH7rAcX5wob9xZ0zX
  572. To: "892"<sip:892@192.168.88.251>;tag=as1fafb47c
  573. Call-ID: 6283efbe61b581826d757f084eb27585@192.168.88.251:5060
  574. CSeq: 378 BYE
  575. Content-Length: 0
  576. Route: <sip:192.168.88.251:5060;lr;sipml5-outbound;transport=udp>
  577. Max-Forwards: 70
  578. Accept-Contact: *;+g.oma.sip-im
  579. Accept-Contact: *;language="en,fr"
  580. Accept-Contact: *;+g.oma.sip-im
  581. Accept-Contact: *;language="en,fr"
  582. User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
  583. Organization: Doubango Telecom
  584.  
  585. <------------->
  586. --- (15 headers 0 lines) ---
  587. Scheduling destruction of SIP dialog '6283efbe61b581826d757f084eb27585@192.168.88.251:5060' in 32000 ms (Method: BYE)
  588.  
  589. <--- Transmitting (no NAT) to 192.168.88.189:5060 --->
  590. SIP/2.0 200 OK
  591. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKsZgzAMrosRSIqkJ2DUPJz0lK6du69ETK;rport;received=192.168.88.189
  592. From: <sip:893@df7jal23ls0d.invalid>;tag=lN4NH7rAcX5wob9xZ0zX
  593. To: "892"<sip:892@192.168.88.251>;tag=as1fafb47c
  594. Call-ID: 6283efbe61b581826d757f084eb27585@192.168.88.251:5060
  595. CSeq: 378 BYE
  596. Server: Asterisk PBX 13.2.0
  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. -- Channel SIP/893-00000095 left 'simple_bridge' basic-bridge <1afd0190-37f4-424f-bc2d-833cfe407d2b>
  604. -- Channel SIP/892-00000094 left 'simple_bridge' basic-bridge <1afd0190-37f4-424f-bc2d-833cfe407d2b>
  605. == Spawn extension (default, 893, 1) exited non-zero on 'SIP/892-00000094'
  606. Scheduling destruction of SIP dialog 'fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5' in 32000 ms (Method: INVITE)
  607. set_destination: Parsing <sip:892@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws> for address/port to send to
  608. set_destination: URI is for WebSocket, we can't set destination
  609. Reliably Transmitting (no NAT) to 192.168.88.188:5060:
  610. BYE sip:892@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0
  611. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK4653d0b2
  612. Max-Forwards: 70
  613. From: <sip:893@192.168.88.251>;tag=as46a08706
  614. To: "892"<sip:892@192.168.88.251>;tag=3qmJ7kplqHaoNe2PmT96
  615. Call-ID: fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5
  616. CSeq: 102 BYE
  617. User-Agent: Asterisk PBX 13.2.0
  618. Proxy-Authorization: Digest username="889", realm="192.168.88.251", algorithm=MD5, uri="sip:192.168.88.251", nonce="37c2ba18", response="4aa604bdcf5aff5d7ff34e49747b7b4e"
  619. X-Asterisk-HangupCause: Normal Clearing
  620. X-Asterisk-HangupCauseCode: 16
  621. Content-Length: 0
  622.  
  623.  
  624. ---
  625.  
  626. <--- SIP read from WS:192.168.88.188:49766 --->
  627. SIP/2.0 200 OK
  628. Via: SIP/2.0/WS 192.168.88.251:5060;branch=z9hG4bK4653d0b2
  629. From: <sip:893@192.168.88.251>;tag=as46a08706
  630. To: "892"<sip:892@192.168.88.251>;tag=3qmJ7kplqHaoNe2PmT96
  631. Contact: <sip:892@df7jal23ls0d.invalid;transport=ws>
  632. Call-ID: fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5
  633. CSeq: 102 BYE
  634. Content-Length: 0
  635.  
  636. <------------->
  637. --- (8 headers 0 lines) ---
  638. SIP Response message for INCOMING dialog BYE arrived
  639. Really destroying SIP dialog 'fec6211f-46bc-53e6-1b0a-ff4a4f5d38e5' Method: INVITE
  640.  
  641. <--- Transmitting (no NAT) to 192.168.88.182:5060 --->
  642. SIP/2.0 180 Ringing
  643. Via: SIP/2.0/WS 192.0.2.157;branch=z9hG4bK4472428;received=192.168.88.182
  644. From: <sip:889@192.168.88.251>;tag=grberh41iu
  645. To: <sip:888@192.168.88.251>;tag=as35a3c936
  646. Call-ID: 61r71uq6tdlnd58mga7l
  647. CSeq: 5841 INVITE
  648. Server: Asterisk PBX 13.2.0
  649. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  650. Supported: replaces, timer
  651. Contact: <sip:888@192.168.88.251:5060;transport=WS>
  652. Content-Length: 0
  653.  
  654.  
  655. <------------>
  656. ast*CLI> sip set debug off
  657. SIP Debugging Disabled
  658. ast*CLI>
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