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  1. Sending to 96.54.35.0 : 33713 (NAT)
  2.  
  3. <--- Transmitting (NAT) to 96.54.35.0:33713 --->
  4. SIP/2.0 100 Trying
  5. Via: SIP/2.0/UDP 96.54.35.0:33713;branch=z9hG4bK21073;received=96.54.35.0;rport=33713
  6. From: <sip:1000@sip.jjhosting.org>;tag=z9hG4bK55481774
  7. To: <sip:1000@sip.jjhosting.org>
  8. Call-ID: 353157173613@96.54.35.0
  9. CSeq: 2 REGISTER
  10. User-Agent: Asterisk PBX
  11. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  12. Supported: replaces
  13. Contact: <sip:1000@50.16.216.65>
  14. Content-Length: 0
  15.  
  16.  
  17. <------------>
  18. ip-10-196-191-202*CLI>
  19. <--- Transmitting (NAT) to 96.54.35.0:33713 --->
  20. SIP/2.0 200 OK
  21. Via: SIP/2.0/UDP 96.54.35.0:33713;branch=z9hG4bK21073;received=96.54.35.0;rport=33713
  22. From: <sip:1000@sip.jjhosting.org>;tag=z9hG4bK55481774
  23. To: <sip:1000@sip.jjhosting.org>;tag=as1341a83b
  24. Call-ID: 353157173613@96.54.35.0
  25. CSeq: 2 REGISTER
  26. User-Agent: Asterisk PBX
  27. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  28. Supported: replaces
  29. Expires: 3600
  30. Contact: <sip:1000@96.54.35.0:33713;transport=udp>;expires=3600
  31. Date: Tue, 07 Dec 2010 05:48:54 GMT
  32. Content-Length: 0
  33.  
  34.  
  35. <------------>
  36. Scheduling destruction of SIP dialog '353157173613@96.54.35.0' in 32000 ms (Method: REGISTER)
  37. ip-10-196-191-202*CLI>
  38. <--- SIP read from 96.54.35.0:33713 --->
  39. INVITE sip:12506528444@sip.jjhosting.org SIP/2.0
  40. Via: SIP/2.0/UDP 96.54.35.0:33713;rport;branch=z9hG4bK21934
  41. Max-Forwards: 70
  42. To: <sip:12506528444@sip.jjhosting.org>
  43. From: <sip:1000@sip.jjhosting.org>;tag=z9hG4bK32968633
  44. Call-ID: 003726791548@96.54.35.0
  45. CSeq: 1 INVITE
  46. Contact: <sip:1000@96.54.35.0:33713;transport=udp>
  47. Expires: 3600
  48. User-Agent: Sipdroid/1.5.7 beta/Galaxy
  49. Content-Length: 308
  50. Content-Type: application/sdp
  51.  
  52. v=0
  53. o=1000@sip.jjhosting.org 0 0 IN IP4 96.54.35.0
  54. s=Session SIP/SDP
  55. c=IN IP4 96.54.35.0
  56. t=0 0
  57. m=audio 21000 RTP/AVP 9 8 0 101
  58. a=rtpmap:9 G722/8000
  59. a=rtpmap:8 PCMA/8000
  60. a=rtpmap:0 PCMU/8000
  61. a=rtpmap:101 telephone-event/8000
  62. a=fmtp:101 0-15
  63. m=video 21070 RTP/AVP 103
  64. a=rtpmap:103 h263-1998/90000
  65.  
  66. <------------->
  67. --- (12 headers 13 lines) ---
  68. Sending to 96.54.35.0 : 33713 (NAT)
  69. Using INVITE request as basis request - 003726791548@96.54.35.0
  70.  
  71. <--- Reliably Transmitting (NAT) to 96.54.35.0:33713 --->
  72. SIP/2.0 407 Proxy Authentication Required
  73. Via: SIP/2.0/UDP 96.54.35.0:33713;branch=z9hG4bK21934;received=96.54.35.0;rport=33713
  74. From: <sip:1000@sip.jjhosting.org>;tag=z9hG4bK32968633
  75. To: <sip:12506528444@sip.jjhosting.org>;tag=as2fc7dd12
  76. Call-ID: 003726791548@96.54.35.0
  77. CSeq: 1 INVITE
  78. User-Agent: Asterisk PBX
  79. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  80. Supported: replaces
  81. Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="653930c1"
  82. Content-Length: 0
  83.  
  84.  
  85. <------------>
  86. Scheduling destruction of SIP dialog '003726791548@96.54.35.0' in 32000 ms (Method: INVITE)
  87. Found user '1000'
  88. ip-10-196-191-202*CLI>
  89. <--- SIP read from 96.54.35.0:33713 --->
  90. ACK sip:12506528444@sip.jjhosting.org SIP/2.0
  91. Via: SIP/2.0/UDP 96.54.35.0:33713;rport;branch=z9hG4bK21934
  92. Max-Forwards: 70
  93. To: <sip:12506528444@sip.jjhosting.org>;tag=as2fc7dd12
  94. From: <sip:1000@sip.jjhosting.org>;tag=z9hG4bK32968633
  95. Call-ID: 003726791548@96.54.35.0
  96. CSeq: 1 ACK
  97. User-Agent: Sipdroid/1.5.7 beta/Galaxy
  98. Content-Length: 0
  99.  
  100.  
  101. <------------->
  102. --- (9 headers 0 lines) ---
  103.  
  104. <--- SIP read from 96.54.35.0:33713 --->
  105. INVITE sip:12506528444@sip.jjhosting.org SIP/2.0
  106. Via: SIP/2.0/UDP 96.54.35.0:33713;rport;branch=z9hG4bK45638
  107. Max-Forwards: 70
  108. To: <sip:12506528444@sip.jjhosting.org>
  109. From: <sip:1000@sip.jjhosting.org>;tag=z9hG4bK32968633
  110. Call-ID: 003726791548@96.54.35.0
  111. CSeq: 2 INVITE
  112. Contact: <sip:1000@96.54.35.0:33713;transport=udp>
  113. Expires: 3600
  114. User-Agent: Sipdroid/1.5.7 beta/Galaxy
  115. Proxy-Authorization: Digest username="1000", realm="asterisk", nonce="653930c1", uri="sip:12506528444@sip.jjhosting.org", algorithm=MD5, response="1305b6657e65dfcf3ec0b0a9ccabd93d"
  116. Content-Length: 308
  117. Content-Type: application/sdp
  118.  
  119. v=0
  120. o=1000@sip.jjhosting.org 0 0 IN IP4 96.54.35.0
  121. s=Session SIP/SDP
  122. c=IN IP4 96.54.35.0
  123. t=0 0
  124. m=audio 21000 RTP/AVP 9 8 0 101
  125. a=rtpmap:9 G722/8000
  126. a=rtpmap:8 PCMA/8000
  127. a=rtpmap:0 PCMU/8000
  128. a=rtpmap:101 telephone-event/8000
  129. a=fmtp:101 0-15
  130. m=video 21070 RTP/AVP 103
  131. a=rtpmap:103 h263-1998/90000
  132.  
  133. <------------->
  134. --- (13 headers 13 lines) ---
  135. Sending to 96.54.35.0 : 33713 (NAT)
  136. Using INVITE request as basis request - 003726791548@96.54.35.0
  137. Found user '1000'
  138. Found RTP audio format 9
  139. Found RTP audio format 8
  140. Found RTP audio format 0
  141. Found RTP audio format 101
  142. Found RTP video format 103
  143. Peer audio RTP is at port 96.54.35.0:21000
  144. Found audio description format G722 for ID 9
  145. Found audio description format PCMA for ID 8
  146. Found audio description format PCMU for ID 0
  147. Found audio description format telephone-event for ID 101
  148. Found unknown media description format h263-1998 for ID 103
  149. Capabilities: us - 0x1004 (ulaw|g722), peer - audio=0x100c (ulaw|alaw|g722)/video=0x0 (nothing), combined - 0x1004 (ulaw|g722)
  150. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  151. Peer audio RTP is at port 96.54.35.0:21000
  152. Looking for 12506528444 in phones (domain sip.jjhosting.org)
  153. list_route: hop: <sip:1000@96.54.35.0:33713;transport=udp>
  154.  
  155. <--- Transmitting (NAT) to 96.54.35.0:33713 --->
  156. SIP/2.0 100 Trying
  157. Via: SIP/2.0/UDP 96.54.35.0:33713;branch=z9hG4bK45638;received=96.54.35.0;rport=33713
  158. From: <sip:1000@sip.jjhosting.org>;tag=z9hG4bK32968633
  159. To: <sip:12506528444@sip.jjhosting.org>
  160. Call-ID: 003726791548@96.54.35.0
  161. CSeq: 2 INVITE
  162. User-Agent: Asterisk PBX
  163. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  164. Supported: replaces
  165. Contact: <sip:12506528444@50.16.216.65>
  166. Content-Length: 0
  167.  
  168.  
  169. <------------>
  170. -- Executing [12506528444@phones:1] Set("SIP/1000-09c942c8", "CALLERID(number)=1234567890") in new stack
  171. -- Executing [12506528444@phones:2] Dial("SIP/1000-09c942c8", "SIP/rapidvox/12506528444") in new stack
  172. [Dec 7 05:48:55] WARNING[10551]: chan_sip.c:3024 sip_call: No audio format found to offer. Cancelling call to 12506528444
  173. -- Couldn't call rapidvox/12506528444
  174. == Everyone is busy/congested at this time (0:0/0/0)
  175. == Auto fallthrough, channel 'SIP/1000-09c942c8' status is 'CHANUNAVAIL'
  176.  
  177. <--- Transmitting (NAT) to 96.54.35.0:33713 --->
  178. SIP/2.0 503 Service Unavailable
  179. Via: SIP/2.0/UDP 96.54.35.0:33713;branch=z9hG4bK45638;received=96.54.35.0;rport=33713
  180. From: <sip:1000@sip.jjhosting.org>;tag=z9hG4bK32968633
  181. To: <sip:12506528444@sip.jjhosting.org>;tag=as1b960d89
  182. Call-ID: 003726791548@96.54.35.0
  183. CSeq: 2 INVITE
  184. User-Agent: Asterisk PBX
  185. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
  186. Supported: replaces
  187. Contact: <sip:12506528444@50.16.216.65>
  188. Content-Length: 0
  189.  
  190.  
  191. <------------>
  192. Really destroying SIP dialog '003726791548@96.54.35.0' Method: INVITE
  193. Reliably Transmitting (NAT) to 96.54.35.0:33713:
  194. NOTIFY sip:1000@96.54.35.0:33713;transport=udp SIP/2.0
  195. Via: SIP/2.0/UDP 50.16.216.65:5060;branch=z9hG4bK56eeaa37;rport
  196. From: "asterisk" <sip:asterisk@sip.jjhosting.org>;tag=as7b027373
  197. To: <sip:sip:1000@96.54.35.0:33713;transport=udp>;tag=z9hG4bK67219099
  198. Contact: <sip:asterisk@50.16.216.65>
  199. Call-ID: 915372541456@96.54.35.0
  200. CSeq: 109 NOTIFY
  201. User-Agent: Asterisk PBX
  202. Max-Forwards: 70
  203. Event: message-summary
  204. Content-Type: application/simple-message-summary
  205. Subscription-State: active
  206. Content-Length: 97
  207.  
  208. Messages-Waiting: no
  209. Message-Account: sip:asterisk@sip.jjhosting.org
  210. Voice-Message: 0/0 (0/0)
  211.  
  212. ---
  213. ip-10-196-191-202*CLI>
  214. <--- SIP read from 96.54.35.0:33713 --->
  215. SIP/2.0 200 OK
  216. Via: SIP/2.0/UDP 50.16.216.65:5060;branch=z9hG4bK56eeaa37;rport=5060
  217. To: <sip:sip:1000@96.54.35.0:33713;transport=udp>;tag=z9hG4bK67219099
  218. From: "asterisk" <sip:asterisk@sip.jjhosting.org>;tag=as7b027373
  219. Call-ID: 915372541456@96.54.35.0
  220. CSeq: 109 NOTIFY
  221. Server: Sipdroid/1.5.7 beta/Galaxy
  222. Content-Length: 0
  223.  
  224.  
  225. <------------->
  226. --- (8 headers 0 lines) ---
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