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  1. [2023-02-17 09:46:50] VERBOSE[645001] res_pjsip_logger.c: <--- Received SIP request (893 bytes) from UDP:192.168.0.14:59200 --->
  2. INVITE sip:4384082316@192.168.0.14;transport=UDP SIP/2.0
  3. Via: SIP/2.0/UDP 192.168.0.14:59200;branch=z9hG4bK-524287-1---a5d4fda2ba0d3ce7;rport
  4. Max-Forwards: 70
  5. Contact: <sip:801@192.168.0.14:59200;transport=UDP>
  6. To: <sip:4384082316@192.168.0.14>
  7. From: <sip:801@192.168.0.14;transport=UDP>;tag=daacb450
  8. Call-ID: mYKp18zHXrYBbNyNAtq6Pw..
  9. CSeq: 1 INVITE
  10. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  11. Content-Type: application/sdp
  12. User-Agent: Z 5.5.10 v2.10.17.3
  13. Allow-Events: presence, kpml, talk
  14. Content-Length: 329
  15.  
  16. v=0
  17. o=Z 872088606 1 IN IP4 192.168.0.14
  18. s=Z
  19. c=IN IP4 192.168.0.14
  20. t=0 0
  21. m=audio 59590 RTP/AVP 106 9 98 101 0 8 3
  22. a=rtpmap:106 opus/48000/2
  23. a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
  24. a=rtpmap:98 telephone-event/48000
  25. a=fmtp:98 0-16
  26. a=rtpmap:101 telephone-event/8000
  27. a=fmtp:101 0-16
  28. a=sendrecv
  29.  
  30. [2023-02-17 09:46:50] VERBOSE[760450] res_pjsip_logger.c: <--- Transmitting SIP response (517 bytes) to UDP:192.168.0.14:59200 --->
  31. SIP/2.0 401 Unauthorized
  32. Via: SIP/2.0/UDP 192.168.0.14:59200;rport=59200;received=192.168.0.14;branch=z9hG4bK-524287-1---a5d4fda2ba0d3ce7
  33. Call-ID: mYKp18zHXrYBbNyNAtq6Pw..
  34. From: <sip:801@192.168.0.14>;tag=daacb450
  35. To: <sip:4384082316@192.168.0.14>;tag=z9hG4bK-524287-1---a5d4fda2ba0d3ce7
  36. CSeq: 1 INVITE
  37. WWW-Authenticate: Digest realm="asterisk",nonce="1676645210/3591b0e35c8eb324dfe8604ababab1e4",opaque="24e3c5113aed0f11",algorithm=MD5,qop="auth"
  38. Server: Asterisk PBX GIT-18-c5c858287a
  39. Content-Length: 0
  40.  
  41.  
  42. [2023-02-17 09:46:50] VERBOSE[645001] res_pjsip_logger.c: <--- Received SIP request (360 bytes) from UDP:192.168.0.14:59200 --->
  43. ACK sip:4384082316@192.168.0.14;transport=UDP SIP/2.0
  44. Via: SIP/2.0/UDP 192.168.0.14:59200;branch=z9hG4bK-524287-1---a5d4fda2ba0d3ce7;rport
  45. Max-Forwards: 70
  46. To: <sip:4384082316@192.168.0.14>;tag=z9hG4bK-524287-1---a5d4fda2ba0d3ce7
  47. From: <sip:801@192.168.0.14;transport=UDP>;tag=daacb450
  48. Call-ID: mYKp18zHXrYBbNyNAtq6Pw..
  49. CSeq: 1 ACK
  50. Content-Length: 0
  51.  
  52.  
  53. [2023-02-17 09:46:50] VERBOSE[645001] res_pjsip_logger.c: <--- Received SIP request (1195 bytes) from UDP:192.168.0.14:59200 --->
  54. INVITE sip:4384082316@192.168.0.14;transport=UDP SIP/2.0
  55. Via: SIP/2.0/UDP 192.168.0.14:59200;branch=z9hG4bK-524287-1---2030d69687247245;rport
  56. Max-Forwards: 70
  57. Contact: <sip:801@192.168.0.14:59200;transport=UDP>
  58. To: <sip:4384082316@192.168.0.14>
  59. From: <sip:801@192.168.0.14;transport=UDP>;tag=daacb450
  60. Call-ID: mYKp18zHXrYBbNyNAtq6Pw..
  61. CSeq: 2 INVITE
  62. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  63. Content-Type: application/sdp
  64. User-Agent: Z 5.5.10 v2.10.17.3
  65. Authorization: Digest username="801",realm="asterisk",nonce="1676645210/3591b0e35c8eb324dfe8604ababab1e4",uri="sip:4384082316@192.168.0.14;transport=UDP",response="6389e287a47162f3fd075b4f0e0e4c4f",cnonce="023a57cf1870fe94a04eb419b29b4287",nc=00000001,qop=auth,algorithm=MD5,opaque="24e3c5113aed0f11"
  66. Allow-Events: presence, kpml, talk
  67. Content-Length: 329
  68.  
  69. v=0
  70. o=Z 872088606 1 IN IP4 192.168.0.14
  71. s=Z
  72. c=IN IP4 192.168.0.14
  73. t=0 0
  74. m=audio 59590 RTP/AVP 106 9 98 101 0 8 3
  75. a=rtpmap:106 opus/48000/2
  76. a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
  77. a=rtpmap:98 telephone-event/48000
  78. a=fmtp:98 0-16
  79. a=rtpmap:101 telephone-event/8000
  80. a=fmtp:101 0-16
  81. a=sendrecv
  82.  
  83. [2023-02-17 09:46:50] VERBOSE[738387] res_pjsip_logger.c: <--- Transmitting SIP response (325 bytes) to UDP:192.168.0.14:59200 --->
  84. SIP/2.0 100 Trying
  85. Via: SIP/2.0/UDP 192.168.0.14:59200;rport=59200;received=192.168.0.14;branch=z9hG4bK-524287-1---2030d69687247245
  86. Call-ID: mYKp18zHXrYBbNyNAtq6Pw..
  87. From: <sip:801@192.168.0.14>;tag=daacb450
  88. To: <sip:4384082316@192.168.0.14>
  89. CSeq: 2 INVITE
  90. Server: Asterisk PBX GIT-18-c5c858287a
  91. Content-Length: 0
  92.  
  93.  
  94. [2023-02-17 09:46:50] VERBOSE[789313][C-00000014] pbx_realtime.c: Executing [4384082316@internal:1] Dial("PJSIP/801-00000025", "PJSIP/4384082316@voipms")
  95. [2023-02-17 09:46:50] VERBOSE[789313][C-00000014] app_dial.c: Called PJSIP/4384082316@voipms
  96. [2023-02-17 09:46:50] VERBOSE[760450] res_pjsip_logger.c: <--- Transmitting SIP request (1029 bytes) to UDP:208.100.60.17:5060 --->
  97. INVITE sip:4384082316@atlanta.voip . ms SIP/2.0
  98. Via: SIP/2.0/UDP 192.168.0.14:5060;rport;branch=z9hG4bKPjac9f6835-f54b-431f-8292-e341464375aa
  99. From: <sip:801@192.168.0.14>;tag=9048f3a3-ebd0-4d72-9a08-7b772cfef9b8
  100. To: <sip:4384082316@atlanta.voip . ms>
  101. Contact: <sip:asterisk@192.168.0.14:5060>
  102. Call-ID: b54e38bc-74b8-4bab-a31d-410974729c0f
  103. CSeq: 24312 INVITE
  104. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  105. Supported: 100rel, timer, replaces, norefersub, histinfo
  106. Session-Expires: 1800
  107. Min-SE: 90
  108. Max-Forwards: 70
  109. User-Agent: Asterisk PBX GIT-18-c5c858287a
  110. Content-Type: application/sdp
  111. Content-Length: 346
  112.  
  113. v=0
  114. o=- 1821786222 1821786222 IN IP4 192.168.0.14
  115. s=Asterisk
  116. c=IN IP4 192.168.0.14
  117. t=0 0
  118. m=audio 14976 RTP/SAVP 0 8 101
  119. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TYJQ8T5EppW0f/Hz1MaCN1b17UqKsw3pPL53n4LM
  120. a=rtpmap:0 PCMU/8000
  121. a=rtpmap:8 PCMA/8000
  122. a=rtpmap:101 telephone-event/8000
  123. a=fmtp:101 0-16
  124. a=ptime:20
  125. a=maxptime:150
  126. a=sendrecv
  127.  
  128. [2023-02-17 09:46:50] VERBOSE[645001] res_pjsip_logger.c: <--- Received SIP response (564 bytes) from UDP:208.100.60.17:5060 --->
  129. SIP/2.0 100 Trying
  130. Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bKPjac9f6835-f54b-431f-8292-e341464375aa;received=192.168.0.14;rport=5060
  131. From: <sip:801@192.168.0.14:5060>;tag=9048f3a3-ebd0-4d72-9a08-7b772cfef9b8
  132. To: <sip:4384082316@atlanta.voip . ms>
  133. Call-ID: b54e38bc-74b8-4bab-a31d-410974729c0f
  134. CSeq: 24312 INVITE
  135. Server: voip . ms
  136. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  137. Supported: replaces, timer
  138. Session-Expires: 1800;refresher=uas
  139. Contact: <sip:4384082316@208.100.60.17:5060>
  140. Content-Length: 0
  141.  
  142.  
  143. [2023-02-17 09:46:50] VERBOSE[645001] res_pjsip_logger.c: <--- Received SIP response (934 bytes) from UDP:208.100.60.17:5060 --->
  144. SIP/2.0 200 OK
  145. Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bKPjac9f6835-f54b-431f-8292-e341464375aa;received=192.168.0.14;rport=5060
  146. From: <sip:801@192.168.0.14:5060>;tag=9048f3a3-ebd0-4d72-9a08-7b772cfef9b8
  147. To: <sip:4384082316@atlanta.voip . ms>;tag=as15747e29
  148. Call-ID: b54e38bc-74b8-4bab-a31d-410974729c0f
  149. CSeq: 24312 INVITE
  150. Server: voip . ms
  151. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  152. Supported: replaces, timer
  153. Session-Expires: 1800;refresher=uas
  154. Contact: <sip:4384082316@208.100.60.17:5060>
  155. Content-Type: application/sdp
  156. Require: timer
  157. Content-Length: 310
  158.  
  159. v=0
  160. o=root 1472725180 1472725180 IN IP4 208.100.60.17
  161. s=voip . ms
  162. c=IN IP4 208.100.60.17
  163. t=0 0
  164. m=audio 14518 RTP/SAVP 0 101
  165. a=rtpmap:0 PCMU/8000
  166. a=rtpmap:101 telephone-event/8000
  167. a=fmtp:101 0-16
  168. a=ptime:20
  169. a=sendrecv
  170. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NQrXbLa8ljEp3NfciKrOK/Ah/Oz54FMXfd8d8xUH
  171.  
  172. [2023-02-17 09:46:50] VERBOSE[760450] res_rtp_asterisk.c: 0x7fec04121360 -- Strict RTP learning after remote address set to: 208.100.60.17:14518
  173. [2023-02-17 09:46:50] VERBOSE[760450] res_pjsip_logger.c: <--- Transmitting SIP request (414 bytes) to UDP:208.100.60.17:5060 --->
  174. ACK sip:4384082316@208.100.60.17:5060 SIP/2.0
  175. Via: SIP/2.0/UDP 192.168.0.14:5060;rport;branch=z9hG4bKPj9041d1b8-e987-4ae1-bc56-e87e931b5830
  176. From: <sip:801@192.168.0.14>;tag=9048f3a3-ebd0-4d72-9a08-7b772cfef9b8
  177. To: <sip:4384082316@atlanta.voip . ms>;tag=as15747e29
  178. Call-ID: b54e38bc-74b8-4bab-a31d-410974729c0f
  179. CSeq: 24312 ACK
  180. Max-Forwards: 70
  181. User-Agent: Asterisk PBX GIT-18-c5c858287a
  182. Content-Length: 0
  183.  
  184.  
  185. [2023-02-17 09:46:50] VERBOSE[789313][C-00000014] app_dial.c: PJSIP/voipms-00000026 answered PJSIP/801-00000025
  186. [2023-02-17 09:46:50] VERBOSE[738387] res_rtp_asterisk.c: 0x7fec0411b7d0 -- Strict RTP learning after remote address set to: 192.168.0.14:59590
  187. [2023-02-17 09:46:50] VERBOSE[738387] res_pjsip_logger.c: <--- Transmitting SIP response (956 bytes) to UDP:192.168.0.14:59200 --->
  188. SIP/2.0 200 OK
  189. Via: SIP/2.0/UDP 192.168.0.14:59200;rport=59200;received=192.168.0.14;branch=z9hG4bK-524287-1---2030d69687247245
  190. Call-ID: mYKp18zHXrYBbNyNAtq6Pw..
  191. From: <sip:801@192.168.0.14>;tag=daacb450
  192. To: <sip:4384082316@192.168.0.14>;tag=301ee48c-6858-41a0-9974-e1f299a61eec
  193. CSeq: 2 INVITE
  194. Server: Asterisk PBX GIT-18-c5c858287a
  195. Contact: <sip:192.168.0.14:5060>
  196. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
  197. Supported: 100rel, timer, replaces, norefersub
  198. Content-Type: application/sdp
  199. Content-Length: 367
  200.  
  201. v=0
  202. o=- 872088606 3 IN IP4 192.168.0.14
  203. s=Asterisk
  204. c=IN IP4 192.168.0.14
  205. t=0 0
  206. m=audio 15332 RTP/AVP 0 8 9 3 106 101
  207. a=rtpmap:0 PCMU/8000
  208. a=rtpmap:8 PCMA/8000
  209. a=rtpmap:9 G722/8000
  210. a=rtpmap:3 GSM/8000
  211. a=rtpmap:106 opus/48000/2
  212. a=fmtp:106 sprop-maxcapturerate=16000
  213. a=rtpmap:101 telephone-event/8000
  214. a=fmtp:101 0-16
  215. a=ptime:20
  216. a=maxptime:60
  217. a=sendrecv
  218.  
  219. [2023-02-17 09:46:50] VERBOSE[789314][C-00000014] bridge_channel.c: Channel PJSIP/voipms-00000026 joined 'simple_bridge' basic-bridge <a7f5d6be-b690-406e-88bf-f38b3a823865>
  220. [2023-02-17 09:46:50] VERBOSE[789313][C-00000014] bridge_channel.c: Channel PJSIP/801-00000025 joined 'simple_bridge' basic-bridge <a7f5d6be-b690-406e-88bf-f38b3a823865>
  221. [2023-02-17 09:46:50] VERBOSE[789313][C-00000014] res_rtp_asterisk.c: 0x7fec0411b7d0 -- Strict RTP switching to RTP target address 192.168.0.14:59590 as source
  222. [2023-02-17 09:46:50] VERBOSE[645001] res_pjsip_logger.c: <--- Received SIP request (413 bytes) from UDP:192.168.0.14:59200 --->
  223. ACK sip:192.168.0.14:5060 SIP/2.0
  224. Via: SIP/2.0/UDP 192.168.0.14:59200;branch=z9hG4bK-524287-1---2be54b9b5264c473;rport
  225. Max-Forwards: 70
  226. Contact: <sip:801@192.168.0.14:59200;transport=UDP>
  227. To: <sip:4384082316@192.168.0.14>;tag=301ee48c-6858-41a0-9974-e1f299a61eec
  228. From: <sip:801@192.168.0.14>;tag=daacb450
  229. Call-ID: mYKp18zHXrYBbNyNAtq6Pw..
  230. CSeq: 2 ACK
  231. User-Agent: Z 5.5.10 v2.10.17.3
  232. Content-Length: 0
  233.  
  234.  
  235. [2023-02-17 09:46:51] VERBOSE[789314][C-00000014] res_rtp_asterisk.c: 0x7fec04121360 -- Strict RTP switching to RTP target address 208.100.60.17:14518 as source
  236. [2023-02-17 09:46:55] VERBOSE[789314][C-00000014] res_rtp_asterisk.c: 0x7fec04121360 -- Strict RTP learning complete - Locking on source address 208.100.60.17:14518
  237. [2023-02-17 09:46:55] VERBOSE[789313][C-00000014] res_rtp_asterisk.c: 0x7fec0411b7d0 -- Strict RTP learning complete - Locking on source address 192.168.0.14:59590
  238. [2023-02-17 09:46:57] VERBOSE[645001] res_pjsip_logger.c: <--- Received SIP request (425 bytes) from UDP:208.100.60.17:5060 --->
  239. BYE sip:asterisk@192.168.0.14:5060 SIP/2.0
  240. Via: SIP/2.0/UDP 208.100.60.17:5060;branch=z9hG4bK377bbfa4;rport
  241. Max-Forwards: 70
  242. From: <sip:4384082316@atlanta.voip . ms>;tag=as15747e29
  243. To: <sip:801@192.168.0.14:5060>;tag=9048f3a3-ebd0-4d72-9a08-7b772cfef9b8
  244. Call-ID: b54e38bc-74b8-4bab-a31d-410974729c0f
  245. CSeq: 102 BYE
  246. User-Agent: voip . ms
  247. X-Asterisk-HangupCause: Unknown
  248. X-Asterisk-HangupCauseCode: 0
  249. Content-Length: 0
  250.  
  251.  
  252. [2023-02-17 09:46:57] VERBOSE[760450] res_pjsip_logger.c: <--- Transmitting SIP response (358 bytes) to UDP:208.100.60.17:5060 --->
  253. SIP/2.0 200 OK
  254. Via: SIP/2.0/UDP 208.100.60.17:5060;rport=5060;received=208.100.60.17;branch=z9hG4bK377bbfa4
  255. Call-ID: b54e38bc-74b8-4bab-a31d-410974729c0f
  256. From: <sip:4384082316@atlanta.voip . ms>;tag=as15747e29
  257. To: <sip:801@192.168.0.14>;tag=9048f3a3-ebd0-4d72-9a08-7b772cfef9b8
  258. CSeq: 102 BYE
  259. Server: Asterisk PBX GIT-18-c5c858287a
  260. Content-Length: 0
  261.  
  262.  
  263. [2023-02-17 09:46:57] VERBOSE[789314][C-00000014] bridge_channel.c: Channel PJSIP/voipms-00000026 left 'simple_bridge' basic-bridge <a7f5d6be-b690-406e-88bf-f38b3a823865>
  264. [2023-02-17 09:46:57] VERBOSE[789313][C-00000014] bridge_channel.c: Channel PJSIP/801-00000025 left 'simple_bridge' basic-bridge <a7f5d6be-b690-406e-88bf-f38b3a823865>
  265. [2023-02-17 09:46:57] VERBOSE[789313][C-00000014] pbx.c: Spawn extension (internal, 4384082316, 1) exited non-zero on 'PJSIP/801-00000025'
  266. [2023-02-17 09:46:57] VERBOSE[760450] res_pjsip_logger.c: <--- Transmitting SIP request (427 bytes) to UDP:192.168.0.14:59200 --->
  267. BYE sip:801@192.168.0.14:59200;transport=UDP SIP/2.0
  268. Via: SIP/2.0/UDP 192.168.0.14:5060;rport;branch=z9hG4bKPj672111b7-08f2-4a1b-8028-ab1b62b010ff
  269. From: <sip:4384082316@192.168.0.14>;tag=301ee48c-6858-41a0-9974-e1f299a61eec
  270. To: <sip:801@192.168.0.14>;tag=daacb450
  271. Call-ID: mYKp18zHXrYBbNyNAtq6Pw..
  272. CSeq: 8784 BYE
  273. Reason: Q.850;cause=16
  274. Max-Forwards: 70
  275. User-Agent: Asterisk PBX GIT-18-c5c858287a
  276. Content-Length: 0
  277.  
  278.  
  279. [2023-02-17 09:46:57] VERBOSE[645001] res_pjsip_logger.c: <--- Received SIP response (393 bytes) from UDP:192.168.0.14:59200 --->
  280. SIP/2.0 200 OK
  281. Via: SIP/2.0/UDP 192.168.0.14:5060;rport=5060;branch=z9hG4bKPj672111b7-08f2-4a1b-8028-ab1b62b010ff
  282. Contact: <sip:801@192.168.0.14:59200;transport=UDP>
  283. To: <sip:801@192.168.0.14>;tag=daacb450
  284. From: <sip:4384082316@192.168.0.14>;tag=301ee48c-6858-41a0-9974-e1f299a61eec
  285. Call-ID: mYKp18zHXrYBbNyNAtq6Pw..
  286. CSeq: 8784 BYE
  287. User-Agent: Z 5.5.10 v2.10.17.3
  288. Content-Length: 0
  289.  
  290.  
  291.  
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