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  1. asavs0002l*CLI> channel originate PJSIP/047999712337@falemaisvoip application MusicOnHold
  2. -- Called 047999712337@falemaisvoip
  3.  
  4. <--- Transmitting SIP request (961 bytes) to UDP:179.124.44.234:5060 --->
  5. INVITE sip:047999712337@179.124.44.234 SIP/2.0
  6. Via: SIP/2.0/UDP 131.161.43.126:5060;rport;branch=z9hG4bKPjaea33ba7-73ff-4ddb-aa79-e007647b628b
  7. From: "Anonymous" <sip:2754375@179.124.44.234>;tag=6a4b1722-a5f9-41df-8359-eea30a78cdd7
  8. To: <sip:047999712337@179.124.44.234>
  9. Contact: <sip:2754375@131.161.43.126:5060>
  10. Call-ID: 8ed2caef-59fb-45ad-a6c3-ba9cd23cb89e
  11. CSeq: 22507 INVITE
  12. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  13. Supported: 100rel, timer, replaces, norefersub
  14. Session-Expires: 1800
  15. Min-SE: 90
  16. Max-Forwards: 70
  17. User-Agent: Asterisk PBX certified/13.13-cert7
  18. Content-Type: application/sdp
  19. Content-Length: 261
  20.  
  21. v=0
  22. o=- 79883847 79883847 IN IP4 192.168.44.6
  23. s=Asterisk
  24. c=IN IP4 131.161.43.126
  25. t=0 0
  26. m=audio 15896 RTP/AVP 18 8 101
  27. a=rtpmap:18 G729/8000
  28. a=rtpmap:8 PCMA/8000
  29. a=rtpmap:101 telephone-event/8000
  30. a=fmtp:101 0-16
  31. a=ptime:20
  32. a=maxptime:150
  33. a=sendrecv
  34.  
  35. <--- Received SIP response (563 bytes) from UDP:179.124.44.234:5060 --->
  36. SIP/2.0 407 Proxy Authentication Required
  37. Via: SIP/2.0/UDP 131.161.43.126:5060;received=131.161.43.126;rport=1363;branch=z9hG4bKPjaea33ba7-73ff-4ddb-aa79-e007647b628b
  38. From: "Anonymous" <sip:2754375@179.124.44.234>;tag=6a4b1722-a5f9-41df-8359-eea30a78cdd7
  39. To: <sip:047999712337@179.124.44.234>;tag=0eed131814d2d199891fd2a7a02f9106.29e7
  40. Call-ID: 8ed2caef-59fb-45ad-a6c3-ba9cd23cb89e
  41. CSeq: 22507 INVITE
  42. Proxy-Authenticate: Digest realm="179.124.44.234", nonce="5a1042620000b7d9263e1607fa32c8f7f03a6e26883196a8"
  43. Server: SIPTEK Softswitch [218/1822]
  44. Content-Length: 0
  45.  
  46.  
  47. <--- Transmitting SIP request (464 bytes) to UDP:179.124.44.234:5060 --->
  48. ACK sip:047999712337@179.124.44.234 SIP/2.0
  49. Via: SIP/2.0/UDP 131.161.43.126:5060;rport;branch=z9hG4bKPjaea33ba7-73ff-4ddb-aa79-e007647b628b
  50. From: "Anonymous" <sip:2754375@179.124.44.234>;tag=6a4b1722-a5f9-41df-8359-eea30a78cdd7
  51. To: <sip:047999712337@179.124.44.234>;tag=0eed131814d2d199891fd2a7a02f9106.29e7
  52. Call-ID: 8ed2caef-59fb-45ad-a6c3-ba9cd23cb89e
  53. CSeq: 22507 ACK
  54. Max-Forwards: 70
  55. User-Agent: Asterisk PBX certified/13.13-cert7
  56. Content-Length: 0
  57.  
  58.  
  59. <--- Transmitting SIP request (1175 bytes) to UDP:179.124.44.234:5060 --->
  60. INVITE sip:047999712337@179.124.44.234 SIP/2.0
  61. Via: SIP/2.0/UDP 131.161.43.126:5060;rport;branch=z9hG4bKPj93127498-562b-4c98-8d31-7916743ff8b6
  62. From: "Anonymous" <sip:2754375@179.124.44.234>;tag=6a4b1722-a5f9-41df-8359-eea30a78cdd7
  63. To: <sip:047999712337@179.124.44.234>
  64. Contact: <sip:2754375@131.161.43.126:5060>
  65. Call-ID: 8ed2caef-59fb-45ad-a6c3-ba9cd23cb89e
  66. CSeq: 22508 INVITE
  67. Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
  68. Supported: 100rel, timer, replaces, norefersub
  69. Session-Expires: 1800
  70. Min-SE: 90
  71. Max-Forwards: 70
  72. User-Agent: Asterisk PBX certified/13.13-cert7
  73. Proxy-Authorization: Digest username="2754375", realm="179.124.44.234", nonce="5a1042620000b7d9263e1607fa32c8f7f03a6e26883196a8", uri="sip:047999712337@179.124.44.234", response="ec4dc13a45b3f5c3a9b8
  74. 843c5832da6d"
  75. Content-Type: application/sdp
  76. Content-Length: 261
  77.  
  78. v=0
  79. o=- 79883847 79883847 IN IP4 192.168.44.6
  80. s=Asterisk
  81. c=IN IP4 131.161.43.126
  82. t=0 0
  83. m=audio 15896 RTP/AVP 18 8 101
  84. a=rtpmap:18 G729/8000
  85. a=rtpmap:8 PCMA/8000
  86. a=rtpmap:101 telephone-event/8000
  87. a=fmtp:101 0-16
  88. a=ptime:20
  89. a=maxptime:150 [172/1822]
  90. a=sendrecv
  91.  
  92. <--- Received SIP response (389 bytes) from UDP:179.124.44.234:5060 --->
  93. SIP/2.0 100 Trying
  94. Via: SIP/2.0/UDP 131.161.43.126:5060;received=131.161.43.126;rport=1363;branch=z9hG4bKPj93127498-562b-4c98-8d31-7916743ff8b6
  95. From: "Anonymous" <sip:2754375@179.124.44.234>;tag=6a4b1722-a5f9-41df-8359-eea30a78cdd7
  96. To: <sip:047999712337@179.124.44.234>
  97. Call-ID: 8ed2caef-59fb-45ad-a6c3-ba9cd23cb89e
  98. CSeq: 22508 INVITE
  99. Server: SIPTEK Softswitch
  100. Content-Length: 0
  101.  
  102.  
  103. <--- Received SIP response (395 bytes) from UDP:179.124.44.234:5060 --->
  104. SIP/2.0 100 Giving a try
  105. Via: SIP/2.0/UDP 131.161.43.126:5060;received=131.161.43.126;rport=1363;branch=z9hG4bKPj93127498-562b-4c98-8d31-7916743ff8b6
  106. From: "Anonymous" <sip:2754375@179.124.44.234>;tag=6a4b1722-a5f9-41df-8359-eea30a78cdd7
  107. To: <sip:047999712337@179.124.44.234>
  108. Call-ID: 8ed2caef-59fb-45ad-a6c3-ba9cd23cb89e
  109. CSeq: 22508 INVITE
  110. Server: SIPTEK Softswitch
  111. Content-Length: 0
  112.  
  113.  
  114. <--- Transmitting SIP request (436 bytes) to UDP:179.124.44.234:5060 --->
  115. OPTIONS sip:179.124.44.234 SIP/2.0
  116. Via: SIP/2.0/UDP 131.161.43.126:5060;rport;branch=z9hG4bKPj3fb31549-b55e-443c-86d9-bed41e5ae466
  117. From: <sip:2754375@179.124.44.234>;tag=cc392f23-acea-425b-a54a-526cc9448d65
  118. To: <sip:179.124.44.234>
  119. Contact: <sip:2754375@131.161.43.126:5060>
  120. Call-ID: 7e842d23-2b5d-4767-895a-7c30560dd266
  121. CSeq: 36692 OPTIONS
  122. Max-Forwards: 70
  123. User-Agent: Asterisk PBX certified/13.13-cert7
  124. Content-Length: 0
  125.  
  126.  
  127. <--- Received SIP response (403 bytes) from UDP:179.124.44.234:5060 --->
  128. SIP/2.0 200 OK
  129. Via: SIP/2.0/UDP 131.161.43.126:5060;received=131.161.43.126;rport=1363;branch=z9hG4bKPj3fb31549-b55e-443c-86d9-bed41e5ae466
  130. From: <sip:2754375@179.124.44.234>;tag=cc392f23-acea-425b-a54a-526cc9448d65
  131. To: <sip:179.124.44.234>;tag=0eed131814d2d199891fd2a7a02f9106.f182
  132. Call-ID: 7e842d23-2b5d-4767-895a-7c30560dd266
  133. CSeq: 36692 OPTIONS
  134. Server: SIPTEK Softswitch
  135. Content-Length: 0 [126/1822]
  136.  
  137.  
  138. <--- Received SIP response (958 bytes) from UDP:179.124.44.234:5060 --->
  139. SIP/2.0 183 Session Progress
  140. Via: SIP/2.0/UDP 131.161.43.126:5060;received=131.161.43.126;rport=1363;branch=z9hG4bKPj93127498-562b-4c98-8d31-7916743ff8b6
  141. From: "Anonymous" <sip:2754375@179.124.44.234>;tag=6a4b1722-a5f9-41df-8359-eea30a78cdd7
  142. To: <sip:047999712337@179.124.44.234>;tag=as0723d64d
  143. Call-ID: 8ed2caef-59fb-45ad-a6c3-ba9cd23cb89e
  144. CSeq: 22508 INVITE
  145. Server: Asterisk PBX 1.8.31.0
  146. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  147. Supported: replaces, timer
  148. Session-Expires: 1800;refresher=uas
  149. Contact: <sip:179.124.44.234;did=a63.bb7f8c82>
  150. Content-Type: application/sdp
  151. Require: timer
  152. Content-Length: 286
  153.  
  154. v=0
  155. o=root 755424222 755424222 IN IP4 179.124.44.235
  156. s=Asterisk PBX 1.8.31.0
  157. c=IN IP4 179.124.44.234
  158. t=0 0
  159. m=audio 46128 RTP/AVP 18 8 101
  160. a=rtpmap:18 G729/8000
  161. a=fmtp:18 annexb=no
  162. a=rtpmap:8 PCMA/8000
  163. a=rtpmap:101 telephone-event/8000
  164. a=fmtp:101 0-16
  165. a=ptime:20
  166. a=sendrecv
  167.  
  168. -- PJSIP/falemaisvoip-00000000 is making progress
  169. <--- Received SIP request (282 bytes) from UDP:179.124.44.234:5060 --->
  170. NOTIFY sip:131.161.43.126:1363 SIP/2.0
  171. Via: SIP/2.0/UDP 179.124.44.234:5060;branch=z9hG4bK7970022
  172. From: sip:keepalive@179.124.44.234;tag=5fb0ccb
  173. To: sip:131.161.43.126:1363
  174. Call-ID: 35a981bf-4e134dc6-3a8c9@179.124.44.234
  175. CSeq: 1 NOTIFY
  176. Event: keep-alive
  177. Content-Length: 0
  178.  
  179.  
  180. <--- Transmitting SIP response (494 bytes) to UDP:179.124.44.234:5060 --->
  181. SIP/2.0 401 Unauthorized [80/1822]
  182. Via: SIP/2.0/UDP 179.124.44.234:5060;rport=5060;received=179.124.44.234;branch=z9hG4bK7970022
  183. Call-ID: 35a981bf-4e134dc6-3a8c9@179.124.44.234
  184. From: <sip:keepalive@179.124.44.234>;tag=5fb0ccb
  185. To: <sip:131.161.43.126>;tag=z9hG4bK7970022
  186. CSeq: 1 NOTIFY
  187. WWW-Authenticate: Digest realm="asterisk",nonce="1511015004/89e3089c68570fdd77da5bfd75aae57a",opaque="5ca5d22c474df5a0",algorithm=md5,qop="auth"
  188. Server: Asterisk PBX certified/13.13-cert7
  189. Content-Length: 0
  190.  
  191.  
  192. <--- Received SIP response (614 bytes) from UDP:179.124.44.234:5060 --->
  193. SIP/2.0 180 Ringing
  194. Via: SIP/2.0/UDP 131.161.43.126:5060;received=131.161.43.126;rport=1363;branch=z9hG4bKPj93127498-562b-4c98-8d31-7916743ff8b6
  195. From: "Anonymous" <sip:2754375@179.124.44.234>;tag=6a4b1722-a5f9-41df-8359-eea30a78cdd7
  196. To: <sip:047999712337@179.124.44.234>;tag=as0723d64d
  197. Call-ID: 8ed2caef-59fb-45ad-a6c3-ba9cd23cb89e
  198. CSeq: 22508 INVITE
  199. Server: Asterisk PBX 1.8.31.0
  200. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  201. Supported: replaces, timer
  202. Session-Expires: 1800;refresher=uas
  203. Contact: <sip:179.124.44.234;did=a63.bb7f8c82>
  204. Content-Length: 0
  205.  
  206.  
  207. -- PJSIP/falemaisvoip-00000000 is ringing
  208. <--- Received SIP response (944 bytes) from UDP:179.124.44.234:5060 --->
  209. SIP/2.0 200 OK
  210. Via: SIP/2.0/UDP 131.161.43.126:5060;received=131.161.43.126;rport=1363;branch=z9hG4bKPj93127498-562b-4c98-8d31-7916743ff8b6
  211. From: "Anonymous" <sip:2754375@179.124.44.234>;tag=6a4b1722-a5f9-41df-8359-eea30a78cdd7
  212. To: <sip:047999712337@179.124.44.234>;tag=as0723d64d
  213. Call-ID: 8ed2caef-59fb-45ad-a6c3-ba9cd23cb89e
  214. CSeq: 22508 INVITE
  215. Server: Asterisk PBX 1.8.31.0
  216. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  217. Supported: replaces, timer
  218. Session-Expires: 1800;refresher=uas
  219. Contact: <sip:179.124.44.234;did=a63.bb7f8c82>
  220. Content-Type: application/sdp
  221. Require: timer
  222. Content-Length: 286
  223.  
  224. v=0
  225. o=root 755424222 755424222 IN IP4 179.124.44.235
  226. s=Asterisk PBX 1.8.31.0
  227. c=IN IP4 179.124.44.234 [34/1822]
  228. t=0 0
  229. m=audio 46128 RTP/AVP 18 8 101
  230. a=rtpmap:18 G729/8000
  231. a=fmtp:18 annexb=no
  232. a=rtpmap:8 PCMA/8000
  233. a=rtpmap:101 telephone-event/8000
  234. a=fmtp:101 0-16
  235. a=ptime:20
  236. a=sendrecv
  237.  
  238. <--- Transmitting SIP request (446 bytes) to UDP:179.124.44.234:5060 --->
  239. ACK sip:179.124.44.234:5060;did=a63.bb7f8c82 SIP/2.0
  240. Via: SIP/2.0/UDP 131.161.43.126:5060;rport;branch=z9hG4bKPj4305ddfd-7b7d-4e04-a6fc-facb63d9305b
  241. From: "Anonymous" <sip:2754375@179.124.44.234>;tag=6a4b1722-a5f9-41df-8359-eea30a78cdd7
  242. To: <sip:047999712337@179.124.44.234>;tag=as0723d64d
  243. Call-ID: 8ed2caef-59fb-45ad-a6c3-ba9cd23cb89e
  244. CSeq: 22508 ACK
  245. Max-Forwards: 70
  246. User-Agent: Asterisk PBX certified/13.13-cert7
  247. Content-Length: 0
  248.  
  249.  
  250. -- PJSIP/falemaisvoip-00000000 answered
  251. -- Started music on hold, class 'default', on channel 'PJSIP/falemaisvoip-00000000'
  252. <--- Received SIP request (465 bytes) from UDP:179.124.44.234:5060 --->
  253. BYE sip:2754375@131.161.43.126:1363 SIP/2.0
  254. Via: SIP/2.0/UDP 179.124.44.234:5060;branch=z9hG4bKb329.9eafada.0
  255. Max-Forwards: 69
  256. From: <sip:047999712337@179.124.44.234>;tag=as0723d64d
  257. To: "1132300606" <sip:2754375@179.124.44.234>;tag=6a4b1722-a5f9-41df-8359-eea30a78cdd7
  258. Call-ID: 8ed2caef-59fb-45ad-a6c3-ba9cd23cb89e
  259. CSeq: 102 BYE
  260. User-Agent: Asterisk PBX 1.8.31.0
  261. X-Asterisk-HangupCause: Normal Clearing
  262. X-Asterisk-HangupCauseCode: 16
  263. Content-Length: 0
  264.  
  265.  
  266. <--- Transmitting SIP response (390 bytes) to UDP:179.124.44.234:5060 --->
  267. SIP/2.0 200 OK
  268. Via: SIP/2.0/UDP 179.124.44.234:5060;rport=5060;received=179.124.44.234;branch=z9hG4bKb329.9eafada.0
  269. Call-ID: 8ed2caef-59fb-45ad-a6c3-ba9cd23cb89e
  270. From: <sip:047999712337@179.124.44.234>;tag=as0723d64d
  271. To: "1132300606" <sip:2754375@179.124.44.234>;tag=6a4b1722-a5f9-41df-8359-eea30a78cdd7
  272. CSeq: 102 BYE
  273. CSeq: 102 BYE
  274. Server: Asterisk PBX certified/13.13-cert7
  275. Content-Length: 0
  276.  
  277.  
  278. -- Stopped music on hold on PJSIP/falemaisvoip-00000000
  279. <--- Transmitting SIP request (436 bytes) to UDP:179.124.44.234:5060 --->
  280. OPTIONS sip:179.124.44.234 SIP/2.0
  281. Via: SIP/2.0/UDP 131.161.43.126:5060;rport;branch=z9hG4bKPj929cc82e-23a7-4fde-b259-44783157c427
  282. From: <sip:2754375@179.124.44.234>;tag=d423d9fc-c157-4097-820d-b1392e49a4fe
  283. To: <sip:179.124.44.234>
  284. Contact: <sip:2754375@131.161.43.126:5060>
  285. Call-ID: 817eaf1e-13fc-475b-a6ea-fbb3790eeb58
  286. CSeq: 19495 OPTIONS
  287. Max-Forwards: 70
  288. User-Agent: Asterisk PBX certified/13.13-cert7
  289. Content-Length: 0
  290.  
  291.  
  292. <--- Received SIP response (403 bytes) from UDP:179.124.44.234:5060 --->
  293. SIP/2.0 200 OK
  294. Via: SIP/2.0/UDP 131.161.43.126:5060;received=131.161.43.126;rport=1363;branch=z9hG4bKPj929cc82e-23a7-4fde-b259-44783157c427
  295. From: <sip:2754375@179.124.44.234>;tag=d423d9fc-c157-4097-820d-b1392e49a4fe
  296. To: <sip:179.124.44.234>;tag=0eed131814d2d199891fd2a7a02f9106.b907
  297. Call-ID: 817eaf1e-13fc-475b-a6ea-fbb3790eeb58
  298. CSeq: 19495 OPTIONS
  299. Server: SIPTEK Softswitch
  300. Content-Length: 0
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