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  1. ncc-roseville-asterisk*CLI> hangup request all
  2. ncc-roseville-asterisk*CLI>
  3. ncc-roseville-asterisk*CLI>
  4. ncc-roseville-asterisk*CLI>
  5. ncc-roseville-asterisk*CLI>
  6. ncc-roseville-asterisk*CLI>
  7. ncc-roseville-asterisk*CLI>
  8. ncc-roseville-asterisk*CLI>
  9. ncc-roseville-asterisk*CLI> sip set debug on
  10. SIP Debugging enabled
  11.  
  12. <--- SIP read from UDP:192.168.1.204:5088 --->
  13. INVITE sip:1000@192.168.1.201 SIP/2.0
  14. Via: SIP/2.0/UDP 192.168.1.204:5088;branch=z9hG4bK501403648;rport
  15. From: "Unit 99" <sip:99@192.168.1.201>;tag=1126257751
  16. To: <sip:1000@192.168.1.201>
  17. Call-ID: 809853710-5088-11@BJC.BGI.B.CAE
  18. CSeq: 100 INVITE
  19. Contact: "Unit 99" <sip:99@192.168.1.204:5088>
  20. Max-Forwards: 70
  21. User-Agent: Grandstream GXW4216 V2.3B 1.0.5.30
  22. Privacy: none
  23. P-Preferred-Identity: "Unit 99" <sip:99@192.168.1.201>
  24. Supported: replaces, path, timer, eventlist
  25. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  26. Content-Type: application/sdp
  27. Accept: application/sdp, application/dtmf-relay
  28. Content-Length: 395
  29.  
  30. v=0
  31. o=99 8028 8000 IN IP4 192.168.1.204
  32. s=SIP Call
  33. c=IN IP4 192.168.1.204
  34. t=0 0
  35. m=audio 50056 RTP/AVP 0 8 4 18 2 97 104 100
  36. a=sendrecv
  37. a=rtpmap:0 PCMU/8000
  38. a=ptime:20
  39. a=rtpmap:8 PCMA/8000
  40. a=rtpmap:4 G723/8000
  41. a=rtpmap:18 G729/8000
  42. a=fmtp:18 annexb=no
  43. a=rtpmap:2 G726-32/8000
  44. a=rtpmap:97 iLBC/8000
  45. a=fmtp:97 mode=20
  46. a=rtpmap:104 AAL2-G726-32/8000
  47. a=rtpmap:100 AAL2-G726-16/8000
  48. <------------->
  49. --- (16 headers 18 lines) ---
  50. Sending to 192.168.1.204:5088 (no NAT)
  51. Sending to 192.168.1.204:5088 (no NAT)
  52. Using INVITE request as basis request - 809853710-5088-11@BJC.BGI.B.CAE
  53. Found peer '99' for '99' from 192.168.1.204:5088
  54.  
  55. <--- Reliably Transmitting (no NAT) to 192.168.1.204:5088 --->
  56. SIP/2.0 401 Unauthorized
  57. Via: SIP/2.0/UDP 192.168.1.204:5088;branch=z9hG4bK501403648;received=192.168.1.204;rport=5088
  58. From: "Unit 99" <sip:99@192.168.1.201>;tag=1126257751
  59. To: <sip:1000@192.168.1.201>;tag=as68759cc8
  60. Call-ID: 809853710-5088-11@BJC.BGI.B.CAE
  61. CSeq: 100 INVITE
  62. Server: Asterisk PBX 15.6.1
  63. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  64. Supported: replaces, timer
  65. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="74943002"
  66. Content-Length: 0
  67.  
  68.  
  69. <------------>
  70. Scheduling destruction of SIP dialog '809853710-5088-11@BJC.BGI.B.CAE' in 32000 ms (Method: INVITE)
  71.  
  72. <--- SIP read from UDP:192.168.1.204:5088 --->
  73. ACK sip:1000@192.168.1.201 SIP/2.0
  74. Via: SIP/2.0/UDP 192.168.1.204:5088;branch=z9hG4bK501403648;rport
  75. From: "Unit 99" <sip:99@192.168.1.201>;tag=1126257751
  76. To: <sip:1000@192.168.1.201>;tag=as68759cc8
  77. Call-ID: 809853710-5088-11@BJC.BGI.B.CAE
  78. CSeq: 100 ACK
  79. Content-Length: 0
  80.  
  81. <------------->
  82. --- (7 headers 0 lines) ---
  83.  
  84. <--- SIP read from UDP:192.168.1.204:5088 --->
  85. INVITE sip:1000@192.168.1.201 SIP/2.0
  86. Via: SIP/2.0/UDP 192.168.1.204:5088;branch=z9hG4bK164017646;rport
  87. From: "Unit 99" <sip:99@192.168.1.201>;tag=1126257751
  88. To: <sip:1000@192.168.1.201>
  89. Call-ID: 809853710-5088-11@BJC.BGI.B.CAE
  90. CSeq: 101 INVITE
  91. Contact: "Unit 99" <sip:99@192.168.1.204:5088>
  92. Authorization: Digest username="99", realm="asterisk", nonce="74943002", uri="sip:1000@192.168.1.201", response="888b195cbfeb9efec5a2e9e3d13d102d", algorithm=MD5
  93. Max-Forwards: 70
  94. User-Agent: Grandstream GXW4216 V2.3B 1.0.5.30
  95. Privacy: none
  96. P-Preferred-Identity: "Unit 99" <sip:99@192.168.1.201>
  97. Supported: replaces, path, timer, eventlist
  98. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  99. Content-Type: application/sdp
  100. Accept: application/sdp, application/dtmf-relay
  101. Content-Length: 395
  102.  
  103. v=0
  104. o=99 8028 8000 IN IP4 192.168.1.204
  105. s=SIP Call
  106. c=IN IP4 192.168.1.204
  107. t=0 0
  108. m=audio 50056 RTP/AVP 0 8 4 18 2 97 104 100
  109. a=sendrecv
  110. a=rtpmap:0 PCMU/8000
  111. a=ptime:20
  112. a=rtpmap:8 PCMA/8000
  113. a=rtpmap:4 G723/8000
  114. a=rtpmap:18 G729/8000
  115. a=fmtp:18 annexb=no
  116. a=rtpmap:2 G726-32/8000
  117. a=rtpmap:97 iLBC/8000
  118. a=fmtp:97 mode=20
  119. a=rtpmap:104 AAL2-G726-32/8000
  120. a=rtpmap:100 AAL2-G726-16/8000
  121. <------------->
  122. --- (17 headers 18 lines) ---
  123. Sending to 192.168.1.204:5088 (no NAT)
  124. Using INVITE request as basis request - 809853710-5088-11@BJC.BGI.B.CAE
  125. Found peer '99' for '99' from 192.168.1.204:5088
  126. == Using SIP RTP CoS mark 5
  127. Found RTP audio format 0
  128. Found RTP audio format 8
  129. Found RTP audio format 4
  130. Found RTP audio format 18
  131. Found RTP audio format 2
  132. Found RTP audio format 97
  133. Found RTP audio format 104
  134. Found RTP audio format 100
  135. Found audio description format PCMU for ID 0
  136. Found audio description format PCMA for ID 8
  137. Found audio description format G723 for ID 4
  138. Found audio description format G729 for ID 18
  139. Found audio description format G726-32 for ID 2
  140. Found audio description format iLBC for ID 97
  141. Found audio description format AAL2-G726-32 for ID 104
  142. Found unknown media description format AAL2-G726-16 for ID 100
  143. Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|g726|g723|alaw|g729|ilbc|g726aal2)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  144. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
  145. > 0x7fb1580120e0 -- Strict RTP learning after remote address set to: 192.168.1.204:50056
  146. Peer audio RTP is at port 192.168.1.204:50056
  147. Looking for 1000 in adminphone (domain 192.168.1.201)
  148. sip_route_dump: route/path hop: <sip:99@192.168.1.204:5088>
  149.  
  150. <--- Transmitting (no NAT) to 192.168.1.204:5088 --->
  151. SIP/2.0 100 Trying
  152. Via: SIP/2.0/UDP 192.168.1.204:5088;branch=z9hG4bK164017646;received=192.168.1.204;rport=5088
  153. From: "Unit 99" <sip:99@192.168.1.201>;tag=1126257751
  154. To: <sip:1000@192.168.1.201>
  155. Call-ID: 809853710-5088-11@BJC.BGI.B.CAE
  156. CSeq: 101 INVITE
  157. Server: Asterisk PBX 15.6.1
  158. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  159. Supported: replaces, timer
  160. Session-Expires: 1800;refresher=uas
  161. Contact: <sip:1000@192.168.1.201:5060>
  162. Content-Length: 0
  163.  
  164.  
  165. <------------>
  166. -- Executing [1000@adminphone:1] Park("SIP/99-0000000f", "") in new stack
  167. Audio is at 10588
  168. Adding codec ulaw to SDP
  169. Adding codec alaw to SDP
  170.  
  171. <--- Reliably Transmitting (no NAT) to 192.168.1.204:5088 --->
  172. SIP/2.0 200 OK
  173. Via: SIP/2.0/UDP 192.168.1.204:5088;branch=z9hG4bK164017646;received=192.168.1.204;rport=5088
  174. From: "Unit 99" <sip:99@192.168.1.201>;tag=1126257751
  175. To: <sip:1000@192.168.1.201>;tag=as531b6e40
  176. Call-ID: 809853710-5088-11@BJC.BGI.B.CAE
  177. CSeq: 101 INVITE
  178. Server: Asterisk PBX 15.6.1
  179. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  180. Supported: replaces, timer
  181. Session-Expires: 1800;refresher=uas
  182. Contact: <sip:1000@192.168.1.201:5060>
  183. Content-Type: application/sdp
  184. Require: timer
  185. Content-Length: 209
  186.  
  187. v=0
  188. o=root 1976578294 1976578294 IN IP4 192.168.1.201
  189. s=Asterisk PBX 15.6.1
  190. c=IN IP4 192.168.1.201
  191. t=0 0
  192. m=audio 10588 RTP/AVP 0 8
  193. a=rtpmap:0 PCMU/8000
  194. a=rtpmap:8 PCMA/8000
  195. a=maxptime:150
  196. a=sendrecv
  197.  
  198. <------------>
  199.  
  200. <--- SIP read from UDP:192.168.1.204:5088 --->
  201. ACK sip:1000@192.168.1.201:5060 SIP/2.0
  202. Via: SIP/2.0/UDP 192.168.1.204:5088;branch=z9hG4bK1205682778;rport
  203. From: "Unit 99" <sip:99@192.168.1.201>;tag=1126257751
  204. To: <sip:1000@192.168.1.201>;tag=as531b6e40
  205. Call-ID: 809853710-5088-11@BJC.BGI.B.CAE
  206. CSeq: 101 ACK
  207. Contact: <sip:99@192.168.1.204:5088>
  208. Max-Forwards: 70
  209. Supported: replaces, path, timer, eventlist
  210. User-Agent: Grandstream GXW4216 V2.3B 1.0.5.30
  211. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  212. Content-Length: 0
  213.  
  214. <------------->
  215. --- (12 headers 0 lines) ---
  216. > 0x7fb1580120e0 -- Strict RTP switching to RTP target address 192.168.1.204:50056 as source
  217. -- Parking 'SIP/99-0000000f' in 'default' at space 701
  218. -- Channel SIP/99-0000000f joined 'holding_bridge' parking-bridge <af125cb5-0426-466d-a8e3-74c1438d1072>
  219. -- <SIP/99-0000000f> Playing 'digits/7.gsm' (language 'en')
  220. -- <SIP/99-0000000f> Playing 'digits/0.gsm' (language 'en')
  221. -- <SIP/99-0000000f> Playing 'digits/1.gsm' (language 'en')
  222. -- Started music on hold, class 'default', on channel 'SIP/99-0000000f'
  223. > 0x7fb1580120e0 -- Strict RTP learning complete - Locking on source address 192.168.1.204:50056
  224. Really destroying SIP dialog '527660459-5088-10@BJC.BGI.B.CAE' Method: BYE
  225.  
  226. <--- SIP read from UDP:192.168.1.110:5060 --->
  227.  
  228.  
  229. <------------->
  230.  
  231. <--- SIP read from UDP:192.168.1.204:5088 --->
  232. BYE sip:1000@192.168.1.201:5060 SIP/2.0
  233. Via: SIP/2.0/UDP 192.168.1.204:5088;branch=z9hG4bK1950633748;rport
  234. From: "Unit 99" <sip:99@192.168.1.201>;tag=1126257751
  235. To: <sip:1000@192.168.1.201>;tag=as531b6e40
  236. Call-ID: 809853710-5088-11@BJC.BGI.B.CAE
  237. CSeq: 102 BYE
  238. Contact: <sip:99@192.168.1.204:5088>
  239. Max-Forwards: 70
  240. Supported: replaces, path, timer, eventlist
  241. User-Agent: Grandstream GXW4216 V2.3B 1.0.5.30
  242. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
  243. Content-Length: 0
  244.  
  245. <------------->
  246. --- (12 headers 0 lines) ---
  247. Sending to 192.168.1.204:5088 (no NAT)
  248. Scheduling destruction of SIP dialog '809853710-5088-11@BJC.BGI.B.CAE' in 32000 ms (Method: BYE)
  249.  
  250. <--- Transmitting (no NAT) to 192.168.1.204:5088 --->
  251. SIP/2.0 200 OK
  252. Via: SIP/2.0/UDP 192.168.1.204:5088;branch=z9hG4bK1950633748;received=192.168.1.204;rport=5088
  253. From: "Unit 99" <sip:99@192.168.1.201>;tag=1126257751
  254. To: <sip:1000@192.168.1.201>;tag=as531b6e40
  255. Call-ID: 809853710-5088-11@BJC.BGI.B.CAE
  256. CSeq: 102 BYE
  257. Server: Asterisk PBX 15.6.1
  258. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  259. Supported: replaces, timer
  260. Content-Length: 0
  261.  
  262.  
  263. <------------>
  264. -- Stopped music on hold on SIP/99-0000000f
  265. -- Channel SIP/99-0000000f left 'holding_bridge' parking-bridge <af125cb5-0426-466d-a8e3-74c1438d1072>
  266. == Spawn extension (adminphone, 1000, 1) exited non-zero on 'SIP/99-0000000f'
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