miken32

Asterisk hack attempt

Jun 20th, 2013
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  1. <--- SIP read from UDP:74.209.242.21:5060 --->
  2. INVITE tel:30441212792250 SIP/2.0
  3. Via: SIP/2.0/UDP 74.209.242.21:5060;branch=z9hG4bKjgre21V6FWQpe0;rport
  4. Max-Forwards: 70
  5. From: <sip:74.209.242.21>;tag=FBGwvNfkYPmy
  6. To: <tel:30441212792250>
  7. Contact: <sip:74.209.242.21>
  8. CSeq: 101 INVITE
  9. Call-Id: ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ
  10. Route: <sip:111.222.333.444;lr>
  11. User-Agent: Cisco-SIPGateway/IOS-12.x
  12. Supported: 100rel,timer,resource-priority,replaces
  13. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, UPDATE, REFER SUBSCRIBE, NOTIFY, INFO
  14. Allow-Events: telephone-event
  15. Content-Type: application/sdp
  16. Content-Length: 306
  17.  
  18. v=0
  19. o=unknown 10839 10840 IN IP4 189.172.50.191
  20. s=SIP Call
  21. c=IN IP4 189.172.50.191
  22. t=0 0
  23. m=audio 11000 RTP/AVP 18 4 0 8 3 101
  24. a=rtpmap:18 G729/8000
  25. a=rtpmap:4 G723/8000
  26. a=rtpmap:0 PCMU/8000
  27. a=rtpmap:8 PCMA/8000
  28. a=rtpmap:3 GSM/8000
  29. a=rtpmap:101 telephone-event/8000
  30. a=fmtp:101 0-16
  31. a=sendrecv
  32. <------------->
  33. --- (15 headers 14 lines) ---
  34. == Using UDPTL TOS bits 184
  35. == Using UDPTL CoS mark 5
  36. [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:4942 do_setnat: Setting NAT on UDPTL to On
  37. [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:7515 sip_alloc: Allocating new SIP dialog for ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ - INVITE (No RTP)
  38. Sending to 74.209.242.21:5060 (NAT)
  39. [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:21905 handle_request_invite: Initializing initreq for method INVITE - callid ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ
  40. Using INVITE request as basis request - ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ
  41. No matching peer for '74.209.242.21' from '74.209.242.21:5060'
  42. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:345 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0xa539ce0'
  43. [2013-06-21 16:43:49] DEBUG[26527]: res_rtp_asterisk.c:483 ast_rtp_new: Allocated port 15664 for RTP instance '0xa539ce0'
  44. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:354 ast_rtp_instance_new: RTP instance '0xa539ce0' is setup and ready to go
  45. [2013-06-21 16:43:49] DEBUG[26527]: res_rtp_asterisk.c:2394 ast_rtp_prop_set: Setup RTCP on RTP instance '0xa539ce0'
  46. == Using SIP RTP TOS bits 184
  47. == Using SIP RTP CoS mark 5
  48. [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:4934 do_setnat: Setting NAT on RTP to On
  49. [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:4942 do_setnat: Setting NAT on UDPTL to On
  50. Found RTP audio format 18
  51. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0xb7b92370
  52. Found RTP audio format 4
  53. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 4 based on m type on 0xb7b92370
  54. Found RTP audio format 0
  55. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0xb7b92370
  56. Found RTP audio format 8
  57. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0xb7b92370
  58. Found RTP audio format 3
  59. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 3 based on m type on 0xb7b92370
  60. Found RTP audio format 101
  61. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0xb7b92370
  62. Found audio description format G729 for ID 18
  63. Found audio description format G723 for ID 4
  64. Found audio description format PCMU for ID 0
  65. Found audio description format PCMA for ID 8
  66. Found audio description format GSM for ID 3
  67. Found audio description format telephone-event for ID 101
  68. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0xb7b92370
  69. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 3 on 0xb7b92370
  70. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 4 on 0xb7b92370
  71. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0xb7b92370
  72. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0xb7b92370
  73. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0xb7b92370
  74. Capabilities: us - 0x5104 (ulaw|g729|g722|siren14), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
  75. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  76. [2013-06-21 16:43:49] DEBUG[26527]: res_rtp_asterisk.c:2415 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa539ce0'
  77. Peer audio RTP is at port 189.172.50.191:11000
  78. [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:9036 process_sdp: Peer doesn't provide T.38 UDPTL
  79. [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:22053 handle_request_invite: Checking SIP call limits for device
  80. [2013-06-21 16:43:49] DEBUG[26527]: sip/reqresp_parser.c:83 parse_uri_full: No supported scheme found in 'tel:30441212792250' using the scheme[s] sip:,sips:
  81. [2013-06-21 16:43:49] WARNING[26527]: chan_sip.c:14820 get_destination: Not a SIP header ()?
  82.  
  83. <--- Reliably Transmitting (NAT) to 74.209.242.21:5060 --->
  84. SIP/2.0 416 Unsupported URI scheme
  85. Via: SIP/2.0/UDP 74.209.242.21:5060;branch=z9hG4bKjgre21V6FWQpe0;received=74.209.242.21;rport=5060
  86. From: <sip:74.209.242.21>;tag=FBGwvNfkYPmy
  87. To: <tel:30441212792250>;tag=as4e56dc9e
  88. Call-ID: ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ
  89. CSeq: 101 INVITE
  90. Server: FPBX-2.11.0(1.8.7.2)
  91. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  92. Supported: replaces, timer
  93. Content-Length: 0
  94.  
  95.  
  96. <------------>
  97. Scheduling destruction of SIP dialog 'ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ' in 32000 ms (Method: INVITE)
  98.  
  99. <--- SIP read from UDP:74.209.242.21:5060 --->
  100. ACK tel:30441212792250 SIP/2.0
  101. Via: SIP/2.0/UDP 74.209.242.21:5060;branch=z9hG4bKjgre21V6FWQpe0;received=74.209.242.21;rport
  102. From: <sip:74.209.242.21>;tag=FBGwvNfkYPmy
  103. To: <tel:30441212792250>;tag=as4e56dc9e
  104. Call-Id: ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ
  105. CSeq: 101 ACK
  106. User-Agent: gR-LxuidiMCFoH-
  107. Max-Forwards: 70
  108. Allow: INVITE, ACK, CANCEL, BYE
  109. Content-Length: 0
  110.  
  111. <------------->
  112. --- (10 headers 0 lines) ---
  113. [2013-06-21 16:43:49] DEBUG[26527]: chan_sip.c:4011 __sip_ack: Stopping retransmission on 'ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ' of Response 101: Match Found
  114. Really destroying SIP dialog 'ssjYGQXHedjjYwRgw8XUpvzRLwaJBttgpvzC6w9zhqNApvzRLQ' Method: ACK
  115. [2013-06-21 16:43:49] DEBUG[26527]: rtp_engine.c:293 instance_destructor: Destroyed RTP instance '0xa539ce0'
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