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Feb 26th, 2018
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  1. SIP Debugging Enabled for IP: 91.121.129.159
  2. == Using SIP RTP CoS mark 5
  3. -- Executing [0032488443596@maison:1] Dial("SIP/bureau-00000047", "SIP/0032488443596@OVH") in new stack
  4. == Using SIP RTP CoS mark 5
  5. Audio is at 19588
  6. Adding codec 100003 (ulaw) to SDP
  7. Adding codec 100004 (alaw) to SDP
  8. Adding non-codec 0x1 (telephone-event) to SDP
  9. Reliably Transmitting (no NAT) to 91.121.129.159:5060:
  10. INVITE sip:0032488443596@sip3.ovh.be SIP/2.0
  11. Via: SIP/2.0/UDP 145.239.198.3:5060;branch=z9hG4bK610afeac
  12. Max-Forwards: 70
  13. From: "Christophe Van Waesberghe" <sip:003210871427@sip3.ovh.be>;tag=as7a3ffe36
  14. To: <sip:0032488443596@sip3.ovh.be>
  15. Contact: <sip:003210871427@145.239.198.3:5060>
  16. Call-ID: 1d6d0a0e4829ca4024bd45663bbb29ae@sip3.ovh.be
  17. CSeq: 102 INVITE
  18. User-Agent: Asterisk PBX certified/11.6-cert17
  19. Date: Mon, 26 Feb 2018 16:43:22 GMT
  20. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  21. Supported: replaces, timer
  22. Content-Type: application/sdp
  23. Content-Length: 301
  24.  
  25. v=0
  26. o=root 473008871 473008871 IN IP4 145.239.198.3
  27. s=Asterisk PBX certified/11.6-cert17
  28. c=IN IP4 145.239.198.3
  29. t=0 0
  30. m=audio 19588 RTP/AVP 0 8 101
  31. a=rtpmap:0 PCMU/8000
  32. a=rtpmap:8 PCMA/8000
  33. a=rtpmap:101 telephone-event/8000
  34. a=fmtp:101 0-16
  35. a=silenceSupp:off - - - -
  36. a=ptime:20
  37. a=sendrecv
  38.  
  39. ---
  40. -- Called SIP/0032488443596@OVH
  41.  
  42. <--- SIP read from UDP:91.121.129.159:5060 --->
  43. SIP/2.0 100 Trying
  44. Call-ID: 1d6d0a0e4829ca4024bd45663bbb29ae@sip3.ovh.be
  45. CSeq: 102 INVITE
  46. From: "Christophe Van Waesberghe" <sip:003210871427@sip3.ovh.be>;tag=as7a3ffe36
  47. To: <sip:0032488443596@sip3.ovh.be>
  48. Via: SIP/2.0/UDP 145.239.198.3:5060;received=145.239.198.3;rport=5060;branch=z9hG4bK610afeac
  49. Content-Length: 0
  50.  
  51. <------------->
  52. --- (7 headers 0 lines) ---
  53.  
  54. <--- SIP read from UDP:91.121.129.159:5060 --->
  55. SIP/2.0 403 not registered
  56. Call-ID: 1d6d0a0e4829ca4024bd45663bbb29ae@sip3.ovh.be
  57. CSeq: 102 INVITE
  58. From: "Christophe Van Waesberghe" <sip:003210871427@sip3.ovh.be>;tag=as7a3ffe36
  59. To: <sip:0032488443596@sip3.ovh.be>;tag=02-21803-b0cb3248-18a1f8400
  60. Via: SIP/2.0/UDP 145.239.198.3:5060;received=145.239.198.3;rport=5060;branch=z9hG4bK610afeac
  61. Content-Length: 0
  62.  
  63. <------------->
  64. --- (7 headers 0 lines) ---
  65. Transmitting (no NAT) to 91.121.129.159:5060:
  66. ACK sip:0032488443596@sip3.ovh.be SIP/2.0
  67. Via: SIP/2.0/UDP 145.239.198.3:5060;branch=z9hG4bK610afeac
  68. Max-Forwards: 70
  69. From: "Christophe Van Waesberghe" <sip:003210871427@sip3.ovh.be>;tag=as7a3ffe36
  70. To: <sip:0032488443596@sip3.ovh.be>;tag=02-21803-b0cb3248-18a1f8400
  71. Contact: <sip:003210871427@145.239.198.3:5060>
  72. Call-ID: 1d6d0a0e4829ca4024bd45663bbb29ae@sip3.ovh.be
  73. CSeq: 102 ACK
  74. User-Agent: Asterisk PBX certified/11.6-cert17
  75. Content-Length: 0
  76.  
  77.  
  78. ---
  79. [Feb 26 16:43:22] WARNING[32][C-000000a1]: chan_sip.c:22914 handle_response_invite: Received response: "Forbidden" from '"Christophe Van Waesberghe" <sip:003210871427@sip3.ovh.be>;tag=as7a3ffe36'
  80. Scheduling destruction of SIP dialog '1d6d0a0e4829ca4024bd45663bbb29ae@sip3.ovh.be' in 6400 ms (Method: INVITE)
  81. == Everyone is busy/congested at this time (1:0/0/1)
  82. -- Auto fallthrough, channel 'SIP/bureau-00000047' status is 'CHANUNAVAIL'
  83. Really destroying SIP dialog '1d6d0a0e4829ca4024bd45663bbb29ae@sip3.ovh.be' Method: INVITE
  84. Reliably Transmitting (no NAT) to 91.121.129.159:5060:
  85. OPTIONS sip:sip3.ovh.be SIP/2.0
  86. Via: SIP/2.0/UDP 145.239.198.3:5060;branch=z9hG4bK4bbde1da
  87. Max-Forwards: 70
  88. From: "asterisk" <sip:003210871427@145.239.198.3>;tag=as10922fad
  89. To: <sip:sip3.ovh.be>
  90. Contact: <sip:003210871427@145.239.198.3:5060>
  91. Call-ID: 6f9eb6e415aa835c5f746eb913cac701@145.239.198.3:5060
  92. CSeq: 102 OPTIONS
  93. User-Agent: Asterisk PBX certified/11.6-cert17
  94. Date: Mon, 26 Feb 2018 16:43:40 GMT
  95. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  96. Supported: replaces, timer
  97. Content-Length: 0
  98.  
  99.  
  100. ---
  101.  
  102. <--- SIP read from UDP:91.121.129.159:5060 --->
  103. SIP/2.0 501 Not Implemented
  104. Call-ID: 6f9eb6e415aa835c5f746eb913cac701@145.239.198.3:5060
  105. CSeq: 102 OPTIONS
  106. From: "asterisk" <sip:003210871427@145.239.198.3>;tag=as10922fad
  107. To: <sip:sip3.ovh.be>;tag=02-18048-b0cb95a3-00984c642
  108. Via: SIP/2.0/UDP 145.239.198.3:5060;received=145.239.198.3;rport=5060;branch=z9hG4bK4bbde1da
  109. Content-Length: 0
  110.  
  111. <------------->
  112. --- (7 headers 0 lines) ---
  113. Really destroying SIP dialog '6f9eb6e415aa835c5f746eb913cac701@145.239.198.3:5060' Method: OPTIONS
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