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  1. SIP/2.0 487 Request Terminated
  2. Via: SIP/2.0/UDP 192.168.30.112:5060;branch=z9hG4bK2014e46e;received=192.168.30.112
  3. From: "100" <sip:100@192.168.30.4>;tag=000e38b6cf3b06df56efd82d-0c042a3b
  4. To: <sip:89139706969@192.168.30.4>;tag=as57971bd1
  5. Call-ID: 000e38b6-cf3b0077-1e1650de-65bd3bfb@192.168.30.112
  6. CSeq: 102 INVITE
  7. Server: Asterisk PBX 14.0.0
  8. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  9. Supported: replaces, timer
  10. Content-Length: 0
  11.  
  12.  
  13. ---
  14. Sending to 192.168.30.112:5060 (no NAT)
  15. Sending to 192.168.30.112:5060 (no NAT)
  16.  
  17. <--- Reliably Transmitting (no NAT) to 192.168.30.112:5060 --->
  18. SIP/2.0 401 Unauthorized
  19. Via: SIP/2.0/UDP 192.168.30.112:5060;branch=z9hG4bK0acc8022;received=192.168.30.112
  20. From: "100" <sip:100@192.168.30.4>;tag=000e38b6cf3b06e06820ceb9-2ab9ae12
  21. To: <sip:89139706969@192.168.30.4>;tag=as2b439a9b
  22. Call-ID: 000e38b6-cf3b0078-3f3e197c-6921f9d4@192.168.30.112
  23. CSeq: 101 INVITE
  24. Server: Asterisk PBX 14.0.0
  25. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  26. Supported: replaces, timer
  27. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0db4a3f1"
  28. Content-Length: 0
  29.  
  30.  
  31. <------------>
  32. Scheduling destruction of SIP dialog '000e38b6-cf3b0078-3f3e197c-6921f9d4@192.168.30.112' in 32000 ms (Method: INVITE)
  33. Sending to 192.168.30.112:5060 (no NAT)
  34. Found RTP audio format 0
  35. Found RTP audio format 8
  36. Found RTP audio format 18
  37. Found RTP audio format 101
  38. Found audio description format PCMU for ID 0
  39. Found audio description format PCMA for ID 8
  40. Found audio description format G729 for ID 18
  41. Found audio description format telephone-event for ID 101
  42. Capabilities: us - (alaw|ulaw|gsm), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
  43. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  44. Peer audio RTP is at port 192.168.30.112:24816
  45. Looking for 89139706969 in xyandrex (domain 192.168.30.4)
  46. sip_route_dump: route/path hop: <sip:100@192.168.30.112:5060;transport=udp>
  47.  
  48. <--- Transmitting (no NAT) to 192.168.30.112:5060 --->
  49. SIP/2.0 100 Trying
  50. Via: SIP/2.0/UDP 192.168.30.112:5060;branch=z9hG4bK0c880001;received=192.168.30.112
  51. From: "100" <sip:100@192.168.30.4>;tag=000e38b6cf3b06e06820ceb9-2ab9ae12
  52. To: <sip:89139706969@192.168.30.4>
  53. Call-ID: 000e38b6-cf3b0078-3f3e197c-6921f9d4@192.168.30.112
  54. CSeq: 102 INVITE
  55. Server: Asterisk PBX 14.0.0
  56. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  57. Supported: replaces, timer
  58. Contact: <sip:89139706969@192.168.30.4:5060>
  59. Content-Length: 0
  60.  
  61.  
  62. <------------>
  63. Retransmitting #5 (no NAT) to 192.168.30.112:5060:
  64. SIP/2.0 487 Request Terminated
  65. Via: SIP/2.0/UDP 192.168.30.112:5060;branch=z9hG4bK2014e46e;received=192.168.30.112
  66. From: "100" <sip:100@192.168.30.4>;tag=000e38b6cf3b06df56efd82d-0c042a3b
  67. To: <sip:89139706969@192.168.30.4>;tag=as57971bd1
  68. Call-ID: 000e38b6-cf3b0077-1e1650de-65bd3bfb@192.168.30.112
  69. CSeq: 102 INVITE
  70. Server: Asterisk PBX 14.0.0
  71. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  72. Supported: replaces, timer
  73. Content-Length: 0
  74.  
  75.  
  76. ---
  77. Audio is at 17674
  78. Adding codec alaw to SDP
  79. Adding codec ulaw to SDP
  80. Adding non-codec 0x1 (telephone-event) to SDP
  81.  
  82. <--- Transmitting (no NAT) to 192.168.30.112:5060 --->
  83. SIP/2.0 183 Session Progress
  84. Via: SIP/2.0/UDP 192.168.30.112:5060;branch=z9hG4bK0c880001;received=192.168.30.112
  85. From: "100" <sip:100@192.168.30.4>;tag=000e38b6cf3b06e06820ceb9-2ab9ae12
  86. To: <sip:89139706969@192.168.30.4>;tag=as16aa75c9
  87. Call-ID: 000e38b6-cf3b0078-3f3e197c-6921f9d4@192.168.30.112
  88. CSeq: 102 INVITE
  89. Server: Asterisk PBX 14.0.0
  90. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  91. Supported: replaces, timer
  92. Contact: <sip:89139706969@192.168.30.4:5060>
  93. Content-Type: application/sdp
  94. Content-Length: 273
  95.  
  96. v=0
  97. o=root 875420421 875420421 IN IP4 192.168.30.4
  98. s=Asterisk PBX 14.0.0
  99. c=IN IP4 192.168.30.4
  100. t=0 0
  101. m=audio 17674 RTP/AVP 8 0 101
  102. a=rtpmap:8 PCMA/8000
  103. a=rtpmap:0 PCMU/8000
  104. a=rtpmap:101 telephone-event/8000
  105. a=fmtp:101 0-16
  106. a=ptime:20
  107. a=maxptime:150
  108. a=sendrecv
  109.  
  110. <------------>
  111. Audio is at 17674
  112. Adding codec alaw to SDP
  113. Adding codec ulaw to SDP
  114. Adding non-codec 0x1 (telephone-event) to SDP
  115.  
  116. <--- Reliably Transmitting (no NAT) to 192.168.30.112:5060 --->
  117. SIP/2.0 200 OK
  118. Via: SIP/2.0/UDP 192.168.30.112:5060;branch=z9hG4bK0c880001;received=192.168.30.112
  119. From: "100" <sip:100@192.168.30.4>;tag=000e38b6cf3b06e06820ceb9-2ab9ae12
  120. To: <sip:89139706969@192.168.30.4>;tag=as16aa75c9
  121. Call-ID: 000e38b6-cf3b0078-3f3e197c-6921f9d4@192.168.30.112
  122. CSeq: 102 INVITE
  123. Server: Asterisk PBX 14.0.0
  124. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  125. Supported: replaces, timer
  126. Contact: <sip:89139706969@192.168.30.4:5060>
  127. Content-Type: application/sdp
  128. Content-Length: 273
  129.  
  130. v=0
  131. o=root 875420421 875420421 IN IP4 192.168.30.4
  132. s=Asterisk PBX 14.0.0
  133. c=IN IP4 192.168.30.4
  134. t=0 0
  135. m=audio 17674 RTP/AVP 8 0 101
  136. a=rtpmap:8 PCMA/8000
  137. a=rtpmap:0 PCMU/8000
  138. a=rtpmap:101 telephone-event/8000
  139. a=fmtp:101 0-16
  140. a=ptime:20
  141. a=maxptime:150
  142. a=sendrecv
  143.  
  144. <------------>
  145. Retransmitting #6 (no NAT) to 192.168.30.112:5060:
  146. SIP/2.0 487 Request Terminated
  147. Via: SIP/2.0/UDP 192.168.30.112:5060;branch=z9hG4bK2014e46e;received=192.168.30.112
  148. From: "100" <sip:100@192.168.30.4>;tag=000e38b6cf3b06df56efd82d-0c042a3b
  149. To: <sip:89139706969@192.168.30.4>;tag=as57971bd1
  150. Call-ID: 000e38b6-cf3b0077-1e1650de-65bd3bfb@192.168.30.112
  151. CSeq: 102 INVITE
  152. Server: Asterisk PBX 14.0.0
  153. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  154. Supported: replaces, timer
  155. Content-Length: 0
  156.  
  157.  
  158. ---
  159. Scheduling destruction of SIP dialog '000e38b6-cf3b0078-3f3e197c-6921f9d4@192.168.30.112' in 32000 ms (Method: ACK)
  160. set_destination: Parsing <sip:100@192.168.30.112:5060;transport=udp> for address/port to send to
  161. set_destination: set destination to 192.168.30.112:5060
  162. Reliably Transmitting (no NAT) to 192.168.30.112:5060:
  163. BYE sip:100@192.168.30.112:5060;transport=udp SIP/2.0
  164. Via: SIP/2.0/UDP 192.168.30.4:5060;branch=z9hG4bK07d0ce78
  165. Max-Forwards: 70
  166. From: <sip:89139706969@192.168.30.4>;tag=as16aa75c9
  167. To: "100" <sip:100@192.168.30.4>;tag=000e38b6cf3b06e06820ceb9-2ab9ae12
  168. Call-ID: 000e38b6-cf3b0078-3f3e197c-6921f9d4@192.168.30.112
  169. CSeq: 102 BYE
  170. User-Agent: Asterisk PBX 14.0.0
  171. Proxy-Authorization: Digest username="100", realm="asterisk", algorithm=MD5, uri="sip:192.168.30.4", nonce="0db4a3f1", response="ef34b73dddd93e0279eb1b84ec907119"
  172. X-Asterisk-HangupCause: Normal Clearing
  173. X-Asterisk-HangupCauseCode: 16
  174. Content-Length: 0
  175.  
  176.  
  177. ---
  178. Really destroying SIP dialog '000e38b6-cf3b0078-3f3e197c-6921f9d4@192.168.30.112' Method: ACK
  179. Retransmitting #7 (no NAT) to 192.168.30.112:5060:
  180. SIP/2.0 487 Request Terminated
  181. Via: SIP/2.0/UDP 192.168.30.112:5060;branch=z9hG4bK2014e46e;received=192.168.30.112
  182. From: "100" <sip:100@192.168.30.4>;tag=000e38b6cf3b06df56efd82d-0c042a3b
  183. To: <sip:89139706969@192.168.30.4>;tag=as57971bd1
  184. Call-ID: 000e38b6-cf3b0077-1e1650de-65bd3bfb@192.168.30.112
  185. CSeq: 102 INVITE
  186. Server: Asterisk PBX 14.0.0
  187. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  188. Supported: replaces, timer
  189. Content-Length: 0
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