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  1. ==============================================================================
  2. ================== Avec "sip set debug peer my_phone" ========================
  3. ==============================================================================
  4.  
  5. root@SRV-VOIP:~# asterisk -r
  6. Asterisk 13.17.2, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  7. Created by Mark Spencer <markster@digium.com>
  8. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  9. This is free software, with components licensed under the GNU General Public
  10. License version 2 and other licenses; you are welcome to redistribute it under
  11. certain conditions. Type 'core show license' for details.
  12. =========================================================================
  13. Connected to Asterisk 13.17.2 currently running on SRV-VOIP (pid = 1837)
  14.  
  15. SRV-VOIP*CLI> sip set debug peer my_phone
  16. SIP Debugging Enabled for IP: 192.168.1.113
  17.  
  18. <--- SIP read from UDP:192.168.1.113:50414 --->
  19. INVITE sip:0673661284@192.168.1.58 SIP/2.0
  20. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---2c3b007dcf76f015;rport
  21. Max-Forwards: 70
  22. Contact: <sip:my_phone@192.168.1.113:50414;rinstance=4130ab4168da7dfc>
  23. To: <sip:0673661284@192.168.1.58>
  24. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=2989246a
  25. Call-ID: 88254YzhkZTA0YTFlMzVjMDcyYzQxM2EyMWUwYTU0Yzk1MGE
  26. CSeq: 1 INVITE
  27. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
  28. Content-Type: application/sdp
  29. Supported: replaces
  30. User-Agent: X-Lite release 5.0.3 stamp 88254
  31. Content-Length: 326
  32.  
  33. v=0
  34. o=- 13153514261005942 1 IN IP4 10.0.0.3
  35. s=X-Lite release 5.0.3 stamp 88254
  36. c=IN IP4 10.0.0.3
  37. t=0 0
  38. m=audio 59106 RTP/AVP 9 8 120 0 84 101
  39. a=rtpmap:120 opus/48000/2
  40. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
  41. a=rtpmap:84 speex/16000
  42. a=rtpmap:101 telephone-event/8000
  43. a=fmtp:101 0-15
  44. a=sendrecv
  45. <------------->
  46. --- (13 headers 12 lines) ---
  47. Sending to 192.168.1.113:50414 (NAT)
  48. Sending to 192.168.1.113:50414 (NAT)
  49. Using INVITE request as basis request - 88254YzhkZTA0YTFlMzVjMDcyYzQxM2EyMWUwYTU0Yzk1MGE
  50. Found peer 'my_phone' for 'my_phone' from 192.168.1.113:50414
  51.  
  52. <--- Reliably Transmitting (NAT) to 192.168.1.113:50414 --->
  53. SIP/2.0 401 Unauthorized
  54. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---2c3b007dcf76f015;received=192.168.1.113;rport=50414
  55. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=2989246a
  56. To: <sip:0673661284@192.168.1.58>;tag=as48e40aa0
  57. Call-ID: 88254YzhkZTA0YTFlMzVjMDcyYzQxM2EyMWUwYTU0Yzk1MGE
  58. CSeq: 1 INVITE
  59. Server: Asterisk PBX 13.17.2
  60. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  61. Supported: replaces, timer
  62. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5128c269"
  63. Content-Length: 0
  64.  
  65.  
  66. <------------>
  67. Scheduling destruction of SIP dialog '88254YzhkZTA0YTFlMzVjMDcyYzQxM2EyMWUwYTU0Yzk1MGE' in 32000 ms (Method: INVITE)
  68.  
  69. <--- SIP read from UDP:192.168.1.113:50414 --->
  70. ACK sip:0673661284@192.168.1.58 SIP/2.0
  71. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---2c3b007dcf76f015;rport
  72. Max-Forwards: 70
  73. To: <sip:0673661284@192.168.1.58>;tag=as48e40aa0
  74. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=2989246a
  75. Call-ID: 88254YzhkZTA0YTFlMzVjMDcyYzQxM2EyMWUwYTU0Yzk1MGE
  76. CSeq: 1 ACK
  77. Content-Length: 0
  78.  
  79. <------------->
  80. --- (8 headers 0 lines) ---
  81.  
  82. <--- SIP read from UDP:192.168.1.113:50414 --->
  83. INVITE sip:0673661284@192.168.1.58 SIP/2.0
  84. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---e309b621f1214946;rport
  85. Max-Forwards: 70
  86. Contact: <sip:my_phone@192.168.1.113:50414;rinstance=4130ab4168da7dfc>
  87. To: <sip:0673661284@192.168.1.58>
  88. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=2989246a
  89. Call-ID: 88254YzhkZTA0YTFlMzVjMDcyYzQxM2EyMWUwYTU0Yzk1MGE
  90. CSeq: 2 INVITE
  91. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
  92. Content-Type: application/sdp
  93. Supported: replaces
  94. User-Agent: X-Lite release 5.0.3 stamp 88254
  95. Authorization: Digest username="my_phone",realm="asterisk",nonce="5128c269",uri="sip:0673661284@192.168.1.58",response="dab457f749e452a73d007a75ae5bd7b2",algorithm=MD5
  96. Content-Length: 326
  97.  
  98. v=0
  99. o=- 13153514261005942 1 IN IP4 10.0.0.3
  100. s=X-Lite release 5.0.3 stamp 88254
  101. c=IN IP4 10.0.0.3
  102. t=0 0
  103. m=audio 59106 RTP/AVP 9 8 120 0 84 101
  104. a=rtpmap:120 opus/48000/2
  105. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
  106. a=rtpmap:84 speex/16000
  107. a=rtpmap:101 telephone-event/8000
  108. a=fmtp:101 0-15
  109. a=sendrecv
  110. <------------->
  111. --- (14 headers 12 lines) ---
  112. Sending to 192.168.1.113:50414 (NAT)
  113. Using INVITE request as basis request - 88254YzhkZTA0YTFlMzVjMDcyYzQxM2EyMWUwYTU0Yzk1MGE
  114. Found peer 'my_phone' for 'my_phone' from 192.168.1.113:50414
  115. Found RTP audio format 9
  116. Found RTP audio format 8
  117. Found RTP audio format 120
  118. Found RTP audio format 0
  119. Found RTP audio format 84
  120. Found RTP audio format 101
  121. Found audio description format opus for ID 120
  122. Found audio description format speex for ID 84
  123. Found audio description format telephone-event for ID 101
  124. Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|speex16|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  125. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  126. Peer audio RTP is at port 10.0.0.3:59106
  127. Looking for 0673661284 in home (domain 192.168.1.58)
  128. sip_route_dump: route/path hop: <sip:my_phone@192.168.1.113:50414;rinstance=4130ab4168da7dfc>
  129.  
  130. <--- Transmitting (NAT) to 192.168.1.113:50414 --->
  131. SIP/2.0 100 Trying
  132. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---e309b621f1214946;received=192.168.1.113;rport=50414
  133. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=2989246a
  134. To: <sip:0673661284@192.168.1.58>
  135. Call-ID: 88254YzhkZTA0YTFlMzVjMDcyYzQxM2EyMWUwYTU0Yzk1MGE
  136. CSeq: 2 INVITE
  137. Server: Asterisk PBX 13.17.2
  138. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  139. Supported: replaces, timer
  140. Contact: <sip:0673661284@192.168.1.58:5060>
  141. Content-Length: 0
  142.  
  143.  
  144. <------------>
  145. [Oct 26 17:57:29] WARNING[1898][C-0000001d]: chan_sip.c:24003 handle_response_invite: Received response: "Forbidden" from '"Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as1d4a4cb8'
  146.  
  147. <--- Reliably Transmitting (NAT) to 192.168.1.113:50414 --->
  148. SIP/2.0 503 Service Unavailable
  149. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---e309b621f1214946;received=192.168.1.113;rport=50414
  150. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=2989246a
  151. To: <sip:0673661284@192.168.1.58>;tag=as0c26b56d
  152. Call-ID: 88254YzhkZTA0YTFlMzVjMDcyYzQxM2EyMWUwYTU0Yzk1MGE
  153. CSeq: 2 INVITE
  154. Server: Asterisk PBX 13.17.2
  155. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  156. Supported: replaces, timer
  157. X-Asterisk-HangupCause: Call Rejected
  158. X-Asterisk-HangupCauseCode: 21
  159. Content-Length: 0
  160.  
  161.  
  162. <------------>
  163.  
  164. <--- SIP read from UDP:192.168.1.113:50414 --->
  165. ACK sip:0673661284@192.168.1.58 SIP/2.0
  166. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---e309b621f1214946;rport
  167. Max-Forwards: 70
  168. To: <sip:0673661284@192.168.1.58>;tag=as0c26b56d
  169. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=2989246a
  170. Call-ID: 88254YzhkZTA0YTFlMzVjMDcyYzQxM2EyMWUwYTU0Yzk1MGE
  171. CSeq: 2 ACK
  172. Content-Length: 0
  173.  
  174. <------------->
  175. --- (8 headers 0 lines) ---
  176. Really destroying SIP dialog '88254YzhkZTA0YTFlMzVjMDcyYzQxM2EyMWUwYTU0Yzk1MGE' Method: ACK
  177.  
  178.  
  179.  
  180.  
  181.  
  182.  
  183.  
  184. ==============================================================================
  185. ================== Avec activation "sip set debug on" ========================
  186. ==============================================================================
  187.  
  188.  
  189. root@SRV-VOIP:~# asterisk -r
  190. Asterisk 13.17.2, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  191. Created by Mark Spencer <markster@digium.com>
  192. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  193. This is free software, with components licensed under the GNU General Public
  194. License version 2 and other licenses; you are welcome to redistribute it under
  195. certain conditions. Type 'core show license' for details.
  196. =========================================================================
  197. Connected to Asterisk 13.17.2 currently running on SRV-VOIP (pid = 1837)
  198. SRV-VOIP*CLI> sip set debug on
  199. SIP Debugging re-enabled
  200.  
  201. <--- SIP read from UDP:192.168.1.113:50414 --->
  202. INVITE sip:0673661284@192.168.1.58 SIP/2.0
  203. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---673a5b6068f56c53;rport
  204. Max-Forwards: 70
  205. Contact: <sip:my_phone@192.168.1.113:50414;rinstance=4130ab4168da7dfc>
  206. To: <sip:0673661284@192.168.1.58>
  207. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=13599b72
  208. Call-ID: 88254MThmMTcxOWUwMzZlZjg5ZTcxNTI4MzQ5OGEzOWNmODg
  209. CSeq: 1 INVITE
  210. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
  211. Content-Type: application/sdp
  212. Supported: replaces
  213. User-Agent: X-Lite release 5.0.3 stamp 88254
  214. Content-Length: 326
  215.  
  216. v=0
  217. o=- 13153514530307787 1 IN IP4 10.0.0.3
  218. s=X-Lite release 5.0.3 stamp 88254
  219. c=IN IP4 10.0.0.3
  220. t=0 0
  221. m=audio 51190 RTP/AVP 9 8 120 0 84 101
  222. a=rtpmap:120 opus/48000/2
  223. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
  224. a=rtpmap:84 speex/16000
  225. a=rtpmap:101 telephone-event/8000
  226. a=fmtp:101 0-15
  227. a=sendrecv
  228. <------------->
  229. --- (13 headers 12 lines) ---
  230. Sending to 192.168.1.113:50414 (NAT)
  231. Sending to 192.168.1.113:50414 (NAT)
  232. Using INVITE request as basis request - 88254MThmMTcxOWUwMzZlZjg5ZTcxNTI4MzQ5OGEzOWNmODg
  233. Found peer 'my_phone' for 'my_phone' from 192.168.1.113:50414
  234.  
  235. <--- Reliably Transmitting (NAT) to 192.168.1.113:50414 --->
  236. SIP/2.0 401 Unauthorized
  237. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---673a5b6068f56c53;received=192.168.1.113;rport=50414
  238. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=13599b72
  239. To: <sip:0673661284@192.168.1.58>;tag=as5584d2bb
  240. Call-ID: 88254MThmMTcxOWUwMzZlZjg5ZTcxNTI4MzQ5OGEzOWNmODg
  241. CSeq: 1 INVITE
  242. Server: Asterisk PBX 13.17.2
  243. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  244. Supported: replaces, timer
  245. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="50aaed41"
  246. Content-Length: 0
  247.  
  248.  
  249. <------------>
  250. Scheduling destruction of SIP dialog '88254MThmMTcxOWUwMzZlZjg5ZTcxNTI4MzQ5OGEzOWNmODg' in 32000 ms (Method: INVITE)
  251.  
  252. <--- SIP read from UDP:192.168.1.113:50414 --->
  253. ACK sip:0673661284@192.168.1.58 SIP/2.0
  254. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---673a5b6068f56c53;rport
  255. Max-Forwards: 70
  256. To: <sip:0673661284@192.168.1.58>;tag=as5584d2bb
  257. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=13599b72
  258. Call-ID: 88254MThmMTcxOWUwMzZlZjg5ZTcxNTI4MzQ5OGEzOWNmODg
  259. CSeq: 1 ACK
  260. Content-Length: 0
  261.  
  262. <------------->
  263. --- (8 headers 0 lines) ---
  264.  
  265. <--- SIP read from UDP:192.168.1.113:50414 --->
  266. INVITE sip:0673661284@192.168.1.58 SIP/2.0
  267. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---d99bc96cf5b42068;rport
  268. Max-Forwards: 70
  269. Contact: <sip:my_phone@192.168.1.113:50414;rinstance=4130ab4168da7dfc>
  270. To: <sip:0673661284@192.168.1.58>
  271. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=13599b72
  272. Call-ID: 88254MThmMTcxOWUwMzZlZjg5ZTcxNTI4MzQ5OGEzOWNmODg
  273. CSeq: 2 INVITE
  274. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
  275. Content-Type: application/sdp
  276. Supported: replaces
  277. User-Agent: X-Lite release 5.0.3 stamp 88254
  278. Authorization: Digest username="my_phone",realm="asterisk",nonce="50aaed41",uri="sip:0673661284@192.168.1.58",response="1d77d0245428b9ef39f58bc61b9dd701",algorithm=MD5
  279. Content-Length: 326
  280.  
  281. v=0
  282. o=- 13153514530307787 1 IN IP4 10.0.0.3
  283. s=X-Lite release 5.0.3 stamp 88254
  284. c=IN IP4 10.0.0.3
  285. t=0 0
  286. m=audio 51190 RTP/AVP 9 8 120 0 84 101
  287. a=rtpmap:120 opus/48000/2
  288. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
  289. a=rtpmap:84 speex/16000
  290. a=rtpmap:101 telephone-event/8000
  291. a=fmtp:101 0-15
  292. a=sendrecv
  293. <------------->
  294. --- (14 headers 12 lines) ---
  295. Sending to 192.168.1.113:50414 (NAT)
  296. Using INVITE request as basis request - 88254MThmMTcxOWUwMzZlZjg5ZTcxNTI4MzQ5OGEzOWNmODg
  297. Found peer 'my_phone' for 'my_phone' from 192.168.1.113:50414
  298. Found RTP audio format 9
  299. Found RTP audio format 8
  300. Found RTP audio format 120
  301. Found RTP audio format 0
  302. Found RTP audio format 84
  303. Found RTP audio format 101
  304. Found audio description format opus for ID 120
  305. Found audio description format speex for ID 84
  306. Found audio description format telephone-event for ID 101
  307. Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|speex16|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  308. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  309. Peer audio RTP is at port 10.0.0.3:51190
  310. Looking for 0673661284 in home (domain 192.168.1.58)
  311. sip_route_dump: route/path hop: <sip:my_phone@192.168.1.113:50414;rinstance=4130ab4168da7dfc>
  312.  
  313. <--- Transmitting (NAT) to 192.168.1.113:50414 --->
  314. SIP/2.0 100 Trying
  315. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---d99bc96cf5b42068;received=192.168.1.113;rport=50414
  316. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=13599b72
  317. To: <sip:0673661284@192.168.1.58>
  318. Call-ID: 88254MThmMTcxOWUwMzZlZjg5ZTcxNTI4MzQ5OGEzOWNmODg
  319. CSeq: 2 INVITE
  320. Server: Asterisk PBX 13.17.2
  321. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  322. Supported: replaces, timer
  323. Contact: <sip:0673661284@192.168.1.58:5060>
  324. Content-Length: 0
  325.  
  326.  
  327. <------------>
  328. Audio is at 14692
  329. Adding codec ulaw to SDP
  330. Adding codec alaw to SDP
  331. Adding codec gsm to SDP
  332. Adding non-codec 0x1 (telephone-event) to SDP
  333. Reliably Transmitting (NAT) to 194.169.214.30:5060:
  334. INVITE sip:0673661284@sip.ippi.com SIP/2.0
  335. Via: SIP/2.0/UDP 90.32.16.40:5060;branch=z9hG4bK27dd73ca;rport
  336. Max-Forwards: 70
  337. From: "Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as08e897dd
  338. To: <sip:0673661284@sip.ippi.com>
  339. Contact: <sip:my_phone@90.32.16.40:5060>
  340. Call-ID: 280067e64e7c277e705b380c04aa9362@sip.ippi.com
  341. CSeq: 102 INVITE
  342. User-Agent: Asterisk PBX 13.17.2
  343. Date: Thu, 26 Oct 2017 18:01:58 GMT
  344. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  345. Supported: replaces, timer
  346. Content-Type: application/sdp
  347. Content-Length: 285
  348.  
  349. v=0
  350. o=root 1212009518 1212009518 IN IP4 90.32.16.40
  351. s=Asterisk PBX 13.17.2
  352. c=IN IP4 90.32.16.40
  353. t=0 0
  354. m=audio 14692 RTP/AVP 0 8 3 101
  355. a=rtpmap:0 PCMU/8000
  356. a=rtpmap:8 PCMA/8000
  357. a=rtpmap:3 GSM/8000
  358. a=rtpmap:101 telephone-event/8000
  359. a=fmtp:101 0-16
  360. a=maxptime:150
  361. a=sendrecv
  362.  
  363. ---
  364.  
  365. <--- SIP read from UDP:194.169.214.30:5060 --->
  366. SIP/2.0 407 Proxy Authentication Required
  367. Via: SIP/2.0/UDP 90.32.16.40:5060;received=90.32.16.40;branch=z9hG4bK27dd73ca;rport=5060
  368. From: "Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as08e897dd
  369. To: <sip:0673661284@sip.ippi.com>;tag=a910c8153188470b2841623c513a131f.ee4c
  370. Call-ID: 280067e64e7c277e705b380c04aa9362@sip.ippi.com
  371. CSeq: 102 INVITE
  372. Proxy-Authenticate: Digest realm="ippi.fr", nonce="59f224423d0d81638658291fe1b53871e545e36f"
  373. Server: OpenSIPS (1.8.2-tls (i386/linux))
  374. Content-Length: 0
  375.  
  376. <------------->
  377. --- (9 headers 0 lines) ---
  378. Transmitting (NAT) to 194.169.214.30:5060:
  379. ACK sip:0673661284@sip.ippi.com SIP/2.0
  380. Via: SIP/2.0/UDP 90.32.16.40:5060;branch=z9hG4bK27dd73ca;rport
  381. Max-Forwards: 70
  382. From: "Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as08e897dd
  383. To: <sip:0673661284@sip.ippi.com>;tag=a910c8153188470b2841623c513a131f.ee4c
  384. Contact: <sip:my_phone@90.32.16.40:5060>
  385. Call-ID: 280067e64e7c277e705b380c04aa9362@sip.ippi.com
  386. CSeq: 102 ACK
  387. User-Agent: Asterisk PBX 13.17.2
  388. Content-Length: 0
  389.  
  390.  
  391. ---
  392. Audio is at 14692
  393. Adding codec ulaw to SDP
  394. Adding codec alaw to SDP
  395. Adding codec gsm to SDP
  396. Adding non-codec 0x1 (telephone-event) to SDP
  397. Reliably Transmitting (NAT) to 194.169.214.30:5060:
  398. INVITE sip:0673661284@sip.ippi.com SIP/2.0
  399. Via: SIP/2.0/UDP 90.32.16.40:5060;branch=z9hG4bK23d6f458;rport
  400. Max-Forwards: 70
  401. From: "Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as08e897dd
  402. To: <sip:0673661284@sip.ippi.com>
  403. Contact: <sip:my_phone@90.32.16.40:5060>
  404. Call-ID: 280067e64e7c277e705b380c04aa9362@sip.ippi.com
  405. CSeq: 103 INVITE
  406. User-Agent: Asterisk PBX 13.17.2
  407. Proxy-Authorization: Digest username="Ujonathan", realm="ippi.fr", algorithm=MD5, uri="sip:0673661284@sip.ippi.com", nonce="59f224423d0d81638658291fe1b53871e545e36f", response="7ae84f61f96b92670034844bf2eded0e"
  408. Date: Thu, 26 Oct 2017 18:01:58 GMT
  409. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  410. Supported: replaces, timer
  411. Content-Type: application/sdp
  412. Content-Length: 285
  413.  
  414. v=0
  415. o=root 1212009518 1212009519 IN IP4 90.32.16.40
  416. s=Asterisk PBX 13.17.2
  417. c=IN IP4 90.32.16.40
  418. t=0 0
  419. m=audio 14692 RTP/AVP 0 8 3 101
  420. a=rtpmap:0 PCMU/8000
  421. a=rtpmap:8 PCMA/8000
  422. a=rtpmap:3 GSM/8000
  423. a=rtpmap:101 telephone-event/8000
  424. a=fmtp:101 0-16
  425. a=maxptime:150
  426. a=sendrecv
  427.  
  428. ---
  429.  
  430. <--- SIP read from UDP:194.169.214.30:5060 --->
  431. SIP/2.0 403 Fake FROM - use From=id next time
  432. Via: SIP/2.0/UDP 90.32.16.40:5060;received=90.32.16.40;branch=z9hG4bK23d6f458;rport=5060
  433. From: "Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as08e897dd
  434. To: <sip:0673661284@sip.ippi.com>;tag=a910c8153188470b2841623c513a131f.9b5c
  435. Call-ID: 280067e64e7c277e705b380c04aa9362@sip.ippi.com
  436. CSeq: 103 INVITE
  437. Server: OpenSIPS (1.8.2-tls (i386/linux))
  438. Content-Length: 0
  439.  
  440. <------------->
  441. --- (8 headers 0 lines) ---
  442. Transmitting (NAT) to 194.169.214.30:5060:
  443. ACK sip:0673661284@sip.ippi.com SIP/2.0
  444. Via: SIP/2.0/UDP 90.32.16.40:5060;branch=z9hG4bK23d6f458;rport
  445. Max-Forwards: 70
  446. From: "Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as08e897dd
  447. To: <sip:0673661284@sip.ippi.com>;tag=a910c8153188470b2841623c513a131f.9b5c
  448. Contact: <sip:my_phone@90.32.16.40:5060>
  449. Call-ID: 280067e64e7c277e705b380c04aa9362@sip.ippi.com
  450. CSeq: 103 ACK
  451. User-Agent: Asterisk PBX 13.17.2
  452. Content-Length: 0
  453.  
  454.  
  455. ---
  456. [Oct 26 18:01:58] WARNING[1898][C-0000001e]: chan_sip.c:24003 handle_response_invite: Received response: "Forbidden" from '"Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as08e897dd'
  457. Scheduling destruction of SIP dialog '280067e64e7c277e705b380c04aa9362@sip.ippi.com' in 32000 ms (Method: INVITE)
  458.  
  459. <--- Reliably Transmitting (NAT) to 192.168.1.113:50414 --->
  460. SIP/2.0 503 Service Unavailable
  461. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---d99bc96cf5b42068;received=192.168.1.113;rport=50414
  462. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=13599b72
  463. To: <sip:0673661284@192.168.1.58>;tag=as664057ea
  464. Call-ID: 88254MThmMTcxOWUwMzZlZjg5ZTcxNTI4MzQ5OGEzOWNmODg
  465. CSeq: 2 INVITE
  466. Server: Asterisk PBX 13.17.2
  467. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  468. Supported: replaces, timer
  469. X-Asterisk-HangupCause: Call Rejected
  470. X-Asterisk-HangupCauseCode: 21
  471. Content-Length: 0
  472.  
  473.  
  474. <------------>
  475.  
  476. <--- SIP read from UDP:192.168.1.113:50414 --->
  477. ACK sip:0673661284@192.168.1.58 SIP/2.0
  478. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---d99bc96cf5b42068;rport
  479. Max-Forwards: 70
  480. To: <sip:0673661284@192.168.1.58>;tag=as664057ea
  481. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=13599b72
  482. Call-ID: 88254MThmMTcxOWUwMzZlZjg5ZTcxNTI4MzQ5OGEzOWNmODg
  483. CSeq: 2 ACK
  484. Content-Length: 0
  485.  
  486. <------------->
  487. --- (8 headers 0 lines) ---
  488. Really destroying SIP dialog '88254MThmMTcxOWUwMzZlZjg5ZTcxNTI4MzQ5OGEzOWNmODg' Method: ACK
  489.  
  490. <--- SIP read from UDP:192.168.1.113:50414 --->
  491.  
  492.  
  493. <------------->
  494. Really destroying SIP dialog '280067e64e7c277e705b380c04aa9362@sip.ippi.com' Method: INVITE
  495.  
  496. <--- SIP read from UDP:185.107.83.26:5062 --->
  497. OPTIONS sip:100@90.32.16.40 SIP/2.0
  498. Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK-3712243567;rport
  499. Content-Length: 0
  500. From: "sipvicious"<sip:100@1.1.1.1>;tag=35613230313032383133633401313232363733343732
  501. Accept: application/sdp
  502. User-Agent: friendly-scanner
  503. To: "sipvicious"<sip:100@1.1.1.1>
  504. Contact: sip:100@127.0.0.1:5062
  505. CSeq: 1 OPTIONS
  506. Call-ID: 1109126168203218783661547
  507. Max-Forwards: 70
  508.  
  509. <------------->
  510. --- (11 headers 0 lines) ---
  511. Sending to 185.107.83.26:5062 (NAT)
  512. Looking for 100 in default (domain 90.32.16.40)
  513.  
  514. <--- Transmitting (NAT) to 185.107.83.26:5062 --->
  515. SIP/2.0 404 Not Found
  516. Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK-3712243567;received=185.107.83.26;rport=5062
  517. From: "sipvicious"<sip:100@1.1.1.1>;tag=35613230313032383133633401313232363733343732
  518. To: "sipvicious"<sip:100@1.1.1.1>;tag=as0abc8cf5
  519. Call-ID: 1109126168203218783661547
  520. CSeq: 1 OPTIONS
  521. Server: Asterisk PBX 13.17.2
  522. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  523. Supported: replaces, timer
  524. Accept: application/sdp
  525. Content-Length: 0
  526.  
  527.  
  528. <------------>
  529. Scheduling destruction of SIP dialog '1109126168203218783661547' in 32000 ms (Method: OPTIONS)
  530.  
  531. <--- SIP read from UDP:192.168.1.113:50414 --->
  532.  
  533.  
  534. <------------->
  535.  
  536. <--- SIP read from UDP:192.168.1.113:50414 --->
  537. INVITE sip:0673661284@192.168.1.58 SIP/2.0
  538. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---e87c714d49e0c704;rport
  539. Max-Forwards: 70
  540. Contact: <sip:my_phone@192.168.1.113:50414;rinstance=4130ab4168da7dfc>
  541. To: <sip:0673661284@192.168.1.58>
  542. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=c6480816
  543. Call-ID: 88254ODVhMmI0MTI0N2M4N2IxYWQ0YWJkMWQ3YWI0MTBiZjU
  544. CSeq: 1 INVITE
  545. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
  546. Content-Type: application/sdp
  547. Supported: replaces
  548. User-Agent: X-Lite release 5.0.3 stamp 88254
  549. Content-Length: 326
  550.  
  551. v=0
  552. o=- 13153514593541057 1 IN IP4 10.0.0.3
  553. s=X-Lite release 5.0.3 stamp 88254
  554. c=IN IP4 10.0.0.3
  555. t=0 0
  556. m=audio 61106 RTP/AVP 9 8 120 0 84 101
  557. a=rtpmap:120 opus/48000/2
  558. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
  559. a=rtpmap:84 speex/16000
  560. a=rtpmap:101 telephone-event/8000
  561. a=fmtp:101 0-15
  562. a=sendrecv
  563. <------------->
  564. --- (13 headers 12 lines) ---
  565. Sending to 192.168.1.113:50414 (NAT)
  566. Sending to 192.168.1.113:50414 (NAT)
  567. Using INVITE request as basis request - 88254ODVhMmI0MTI0N2M4N2IxYWQ0YWJkMWQ3YWI0MTBiZjU
  568. Found peer 'my_phone' for 'my_phone' from 192.168.1.113:50414
  569.  
  570. <--- Reliably Transmitting (NAT) to 192.168.1.113:50414 --->
  571. SIP/2.0 401 Unauthorized
  572. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---e87c714d49e0c704;received=192.168.1.113;rport=50414
  573. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=c6480816
  574. To: <sip:0673661284@192.168.1.58>;tag=as7137874b
  575. Call-ID: 88254ODVhMmI0MTI0N2M4N2IxYWQ0YWJkMWQ3YWI0MTBiZjU
  576. CSeq: 1 INVITE
  577. Server: Asterisk PBX 13.17.2
  578. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  579. Supported: replaces, timer
  580. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2f304ebd"
  581. Content-Length: 0
  582.  
  583.  
  584. <------------>
  585. Scheduling destruction of SIP dialog '88254ODVhMmI0MTI0N2M4N2IxYWQ0YWJkMWQ3YWI0MTBiZjU' in 32000 ms (Method: INVITE)
  586.  
  587. <--- SIP read from UDP:192.168.1.113:50414 --->
  588. ACK sip:0673661284@192.168.1.58 SIP/2.0
  589. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---e87c714d49e0c704;rport
  590. Max-Forwards: 70
  591. To: <sip:0673661284@192.168.1.58>;tag=as7137874b
  592. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=c6480816
  593. Call-ID: 88254ODVhMmI0MTI0N2M4N2IxYWQ0YWJkMWQ3YWI0MTBiZjU
  594. CSeq: 1 ACK
  595. Content-Length: 0
  596.  
  597. <------------->
  598. --- (8 headers 0 lines) ---
  599.  
  600. <--- SIP read from UDP:192.168.1.113:50414 --->
  601. INVITE sip:0673661284@192.168.1.58 SIP/2.0
  602. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---35880f1faaaf0049;rport
  603. Max-Forwards: 70
  604. Contact: <sip:my_phone@192.168.1.113:50414;rinstance=4130ab4168da7dfc>
  605. To: <sip:0673661284@192.168.1.58>
  606. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=c6480816
  607. Call-ID: 88254ODVhMmI0MTI0N2M4N2IxYWQ0YWJkMWQ3YWI0MTBiZjU
  608. CSeq: 2 INVITE
  609. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
  610. Content-Type: application/sdp
  611. Supported: replaces
  612. User-Agent: X-Lite release 5.0.3 stamp 88254
  613. Authorization: Digest username="my_phone",realm="asterisk",nonce="2f304ebd",uri="sip:0673661284@192.168.1.58",response="e37277c37b192dac1c82c24c5da846c7",algorithm=MD5
  614. Content-Length: 326
  615.  
  616. v=0
  617. o=- 13153514593541057 1 IN IP4 10.0.0.3
  618. s=X-Lite release 5.0.3 stamp 88254
  619. c=IN IP4 10.0.0.3
  620. t=0 0
  621. m=audio 61106 RTP/AVP 9 8 120 0 84 101
  622. a=rtpmap:120 opus/48000/2
  623. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
  624. a=rtpmap:84 speex/16000
  625. a=rtpmap:101 telephone-event/8000
  626. a=fmtp:101 0-15
  627. a=sendrecv
  628. <------------->
  629. --- (14 headers 12 lines) ---
  630. Sending to 192.168.1.113:50414 (NAT)
  631. Using INVITE request as basis request - 88254ODVhMmI0MTI0N2M4N2IxYWQ0YWJkMWQ3YWI0MTBiZjU
  632. Found peer 'my_phone' for 'my_phone' from 192.168.1.113:50414
  633. Found RTP audio format 9
  634. Found RTP audio format 8
  635. Found RTP audio format 120
  636. Found RTP audio format 0
  637. Found RTP audio format 84
  638. Found RTP audio format 101
  639. Found audio description format opus for ID 120
  640. Found audio description format speex for ID 84
  641. Found audio description format telephone-event for ID 101
  642. Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|speex16|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  643. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  644. Peer audio RTP is at port 10.0.0.3:61106
  645. Looking for 0673661284 in home (domain 192.168.1.58)
  646. sip_route_dump: route/path hop: <sip:my_phone@192.168.1.113:50414;rinstance=4130ab4168da7dfc>
  647.  
  648. <--- Transmitting (NAT) to 192.168.1.113:50414 --->
  649. SIP/2.0 100 Trying
  650. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---35880f1faaaf0049;received=192.168.1.113;rport=50414
  651. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=c6480816
  652. To: <sip:0673661284@192.168.1.58>
  653. Call-ID: 88254ODVhMmI0MTI0N2M4N2IxYWQ0YWJkMWQ3YWI0MTBiZjU
  654. CSeq: 2 INVITE
  655. Server: Asterisk PBX 13.17.2
  656. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  657. Supported: replaces, timer
  658. Contact: <sip:0673661284@192.168.1.58:5060>
  659. Content-Length: 0
  660.  
  661.  
  662. <------------>
  663. Audio is at 11412
  664. Adding codec ulaw to SDP
  665. Adding codec alaw to SDP
  666. Adding codec gsm to SDP
  667. Adding non-codec 0x1 (telephone-event) to SDP
  668. Reliably Transmitting (NAT) to 194.169.214.30:5060:
  669. INVITE sip:0673661284@sip.ippi.com SIP/2.0
  670. Via: SIP/2.0/UDP 90.32.16.40:5060;branch=z9hG4bK761115a4;rport
  671. Max-Forwards: 70
  672. From: "Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as6b0b47cf
  673. To: <sip:0673661284@sip.ippi.com>
  674. Contact: <sip:my_phone@90.32.16.40:5060>
  675. Call-ID: 6e2f64e71ab5b29a1cd0f8d15760f07d@sip.ippi.com
  676. CSeq: 102 INVITE
  677. User-Agent: Asterisk PBX 13.17.2
  678. Date: Thu, 26 Oct 2017 18:03:01 GMT
  679. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  680. Supported: replaces, timer
  681. Content-Type: application/sdp
  682. Content-Length: 283
  683.  
  684. v=0
  685. o=root 637274555 637274555 IN IP4 90.32.16.40
  686. s=Asterisk PBX 13.17.2
  687. c=IN IP4 90.32.16.40
  688. t=0 0
  689. m=audio 11412 RTP/AVP 0 8 3 101
  690. a=rtpmap:0 PCMU/8000
  691. a=rtpmap:8 PCMA/8000
  692. a=rtpmap:3 GSM/8000
  693. a=rtpmap:101 telephone-event/8000
  694. a=fmtp:101 0-16
  695. a=maxptime:150
  696. a=sendrecv
  697.  
  698. ---
  699.  
  700. <--- SIP read from UDP:194.169.214.30:5060 --->
  701. SIP/2.0 407 Proxy Authentication Required
  702. Via: SIP/2.0/UDP 90.32.16.40:5060;received=90.32.16.40;branch=z9hG4bK761115a4;rport=5060
  703. From: "Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as6b0b47cf
  704. To: <sip:0673661284@sip.ippi.com>;tag=a910c8153188470b2841623c513a131f.1506
  705. Call-ID: 6e2f64e71ab5b29a1cd0f8d15760f07d@sip.ippi.com
  706. CSeq: 102 INVITE
  707. Proxy-Authenticate: Digest realm="ippi.fr", nonce="59f224814fe24158b1e101da08a646244ec652b0"
  708. Server: OpenSIPS (1.8.2-tls (i386/linux))
  709. Content-Length: 0
  710.  
  711. <------------->
  712. --- (9 headers 0 lines) ---
  713. Transmitting (NAT) to 194.169.214.30:5060:
  714. ACK sip:0673661284@sip.ippi.com SIP/2.0
  715. Via: SIP/2.0/UDP 90.32.16.40:5060;branch=z9hG4bK761115a4;rport
  716. Max-Forwards: 70
  717. From: "Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as6b0b47cf
  718. To: <sip:0673661284@sip.ippi.com>;tag=a910c8153188470b2841623c513a131f.1506
  719. Contact: <sip:my_phone@90.32.16.40:5060>
  720. Call-ID: 6e2f64e71ab5b29a1cd0f8d15760f07d@sip.ippi.com
  721. CSeq: 102 ACK
  722. User-Agent: Asterisk PBX 13.17.2
  723. Content-Length: 0
  724.  
  725.  
  726. ---
  727. Audio is at 11412
  728. Adding codec ulaw to SDP
  729. Adding codec alaw to SDP
  730. Adding codec gsm to SDP
  731. Adding non-codec 0x1 (telephone-event) to SDP
  732. Reliably Transmitting (NAT) to 194.169.214.30:5060:
  733. INVITE sip:0673661284@sip.ippi.com SIP/2.0
  734. Via: SIP/2.0/UDP 90.32.16.40:5060;branch=z9hG4bK0048fcc9;rport
  735. Max-Forwards: 70
  736. From: "Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as6b0b47cf
  737. To: <sip:0673661284@sip.ippi.com>
  738. Contact: <sip:my_phone@90.32.16.40:5060>
  739. Call-ID: 6e2f64e71ab5b29a1cd0f8d15760f07d@sip.ippi.com
  740. CSeq: 103 INVITE
  741. User-Agent: Asterisk PBX 13.17.2
  742. Proxy-Authorization: Digest username="Ujonathan", realm="ippi.fr", algorithm=MD5, uri="sip:0673661284@sip.ippi.com", nonce="59f224814fe24158b1e101da08a646244ec652b0", response="a3de86ecf0d128754adbf232b0b7aca6"
  743. Date: Thu, 26 Oct 2017 18:03:01 GMT
  744. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  745. Supported: replaces, timer
  746. Content-Type: application/sdp
  747. Content-Length: 283
  748.  
  749. v=0
  750. o=root 637274555 637274556 IN IP4 90.32.16.40
  751. s=Asterisk PBX 13.17.2
  752. c=IN IP4 90.32.16.40
  753. t=0 0
  754. m=audio 11412 RTP/AVP 0 8 3 101
  755. a=rtpmap:0 PCMU/8000
  756. a=rtpmap:8 PCMA/8000
  757. a=rtpmap:3 GSM/8000
  758. a=rtpmap:101 telephone-event/8000
  759. a=fmtp:101 0-16
  760. a=maxptime:150
  761. a=sendrecv
  762.  
  763. ---
  764.  
  765. <--- SIP read from UDP:194.169.214.30:5060 --->
  766. SIP/2.0 403 Fake FROM - use From=id next time
  767. Via: SIP/2.0/UDP 90.32.16.40:5060;received=90.32.16.40;branch=z9hG4bK0048fcc9;rport=5060
  768. From: "Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as6b0b47cf
  769. To: <sip:0673661284@sip.ippi.com>;tag=a910c8153188470b2841623c513a131f.92ed
  770. Call-ID: 6e2f64e71ab5b29a1cd0f8d15760f07d@sip.ippi.com
  771. CSeq: 103 INVITE
  772. Server: OpenSIPS (1.8.2-tls (i386/linux))
  773. Content-Length: 0
  774.  
  775. <------------->
  776. --- (8 headers 0 lines) ---
  777. Transmitting (NAT) to 194.169.214.30:5060:
  778. ACK sip:0673661284@sip.ippi.com SIP/2.0
  779. Via: SIP/2.0/UDP 90.32.16.40:5060;branch=z9hG4bK0048fcc9;rport
  780. Max-Forwards: 70
  781. From: "Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as6b0b47cf
  782. To: <sip:0673661284@sip.ippi.com>;tag=a910c8153188470b2841623c513a131f.92ed
  783. Contact: <sip:my_phone@90.32.16.40:5060>
  784. Call-ID: 6e2f64e71ab5b29a1cd0f8d15760f07d@sip.ippi.com
  785. CSeq: 103 ACK
  786. User-Agent: Asterisk PBX 13.17.2
  787. Content-Length: 0
  788.  
  789.  
  790. ---
  791. [Oct 26 18:03:02] WARNING[1898][C-0000001f]: chan_sip.c:24003 handle_response_invite: Received response: "Forbidden" from '"Ujonathan" <sip:my_phone@sip.ippi.com>;tag=as6b0b47cf'
  792. Scheduling destruction of SIP dialog '6e2f64e71ab5b29a1cd0f8d15760f07d@sip.ippi.com' in 32000 ms (Method: INVITE)
  793.  
  794. <--- Reliably Transmitting (NAT) to 192.168.1.113:50414 --->
  795. SIP/2.0 503 Service Unavailable
  796. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---35880f1faaaf0049;received=192.168.1.113;rport=50414
  797. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=c6480816
  798. To: <sip:0673661284@192.168.1.58>;tag=as24225d28
  799. Call-ID: 88254ODVhMmI0MTI0N2M4N2IxYWQ0YWJkMWQ3YWI0MTBiZjU
  800. CSeq: 2 INVITE
  801. Server: Asterisk PBX 13.17.2
  802. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  803. Supported: replaces, timer
  804. X-Asterisk-HangupCause: Call Rejected
  805. X-Asterisk-HangupCauseCode: 21
  806. Content-Length: 0
  807.  
  808.  
  809. <------------>
  810.  
  811. <--- SIP read from UDP:192.168.1.113:50414 --->
  812. ACK sip:0673661284@192.168.1.58 SIP/2.0
  813. Via: SIP/2.0/UDP 10.0.0.3:50414;branch=z9hG4bK-524287-1---35880f1faaaf0049;rport
  814. Max-Forwards: 70
  815. To: <sip:0673661284@192.168.1.58>;tag=as24225d28
  816. From: "Ujonathan"<sip:my_phone@192.168.1.58>;tag=c6480816
  817. Call-ID: 88254ODVhMmI0MTI0N2M4N2IxYWQ0YWJkMWQ3YWI0MTBiZjU
  818. CSeq: 2 ACK
  819. Content-Length: 0
  820.  
  821. <------------->
  822. --- (8 headers 0 lines) ---
  823. Really destroying SIP dialog '88254ODVhMmI0MTI0N2M4N2IxYWQ0YWJkMWQ3YWI0MTBiZjU' Method: ACK
  824. [Oct 26 18:03:04] NOTICE[1898]: chan_sip.c:15722 sip_reregister: -- Re-registration for Ujonathan@sip.ippi.com
  825. REGISTER 12 headers, 0 lines
  826. Reliably Transmitting (no NAT) to 194.169.214.30:5060:
  827. REGISTER sip:sip.ippi.com SIP/2.0
  828. Via: SIP/2.0/UDP 90.32.16.40:5060;branch=z9hG4bK79d47d67
  829. Max-Forwards: 70
  830. From: <sip:Ujonathan@sip.ippi.com>;tag=as6d724c76
  831. To: <sip:Ujonathan@sip.ippi.com>
  832. Call-ID: 17f77554076c16c43338821c2fd65643@192.168.1.58
  833. CSeq: 189 REGISTER
  834. Supported: replaces, timer
  835. User-Agent: Asterisk PBX 13.17.2
  836. Authorization: Digest username="Ujonathan", realm="ippi.fr", algorithm=MD5, uri="sip:sip.ippi.com", nonce="59f223b23be8f5fc677fb830e4a2d960f3634bfe", response="535b794c24044cb772a2422470c89d8e"
  837. Expires: 120
  838. Contact: <sip:s@90.32.16.40:5060>
  839. Content-Length: 0
  840.  
  841.  
  842. ---
  843.  
  844. <--- SIP read from UDP:194.169.214.30:5060 --->
  845. SIP/2.0 200 OK
  846. Via: SIP/2.0/UDP 90.32.16.40:5060;received=90.32.16.40;rport=5060;branch=z9hG4bK79d47d67
  847. From: <sip:Ujonathan@sip.ippi.com>;tag=as6d724c76
  848. To: <sip:Ujonathan@sip.ippi.com>;tag=a910c8153188470b2841623c513a131f.6d53
  849. Call-ID: 17f77554076c16c43338821c2fd65643@192.168.1.58
  850. CSeq: 189 REGISTER
  851. Contact: <sip:s@90.32.16.40:5060>;expires=120
  852. Server: OpenSIPS (1.8.2-tls (i386/linux))
  853. Content-Length: 0
  854.  
  855. <------------->
  856. --- (9 headers 0 lines) ---
  857. [Oct 26 18:03:04] NOTICE[1898]: chan_sip.c:24538 handle_response_register: Outbound Registration: Expiry for sip.ippi.com is 120 sec (Scheduling reregistration in 105 s)
  858. Really destroying SIP dialog '17f77554076c16c43338821c2fd65643@192.168.1.58' Method: REGISTER
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