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  1. sip set debug peer new
  2. SIP Debugging Enabled for IP: 91.218.111.145
  3.  
  4. <--- SIP read from UDP:91.218.111.145:5060 --->
  5. INVITE sip:s@172.16.20.123:5060 SIP/2.0
  6. Via: SIP/2.0/UDP 91.218.111.145:5060;rport;branch=z9hG4bK-fc2dd4b4030d11ea8da6782bcb0acaea;sig=21a152b6
  7. Via: SIP/2.0/UDP 2LgN1H6wRYmw;rport;branch=z9hG4bK-fc2dcce4030d11ea8da6782bcb0acaea;sig=72f8fc2f;gr=i_G8oJbx57UhszahK7Pv9qxJ2cABrSTRggDY4bMvhaxcFVfb0B7gEMABWoW8qdU4dQvuqEPDUo4WW11XWAdQb_2wUoqUOjA2tFF-9qxgV7byfbKMu2vUTc1VnTnhzdqZSqwkjoe_Xb1YcHZaCvkAOKK8
  8. From: <sip:79055797907@91.218.111.145:5060;user=phone>;tag=KS4I9eFb2m-SE8XhW9pvkhPElDKgOQAq
  9. To: <sip:74999599580@172.16.20.123:5060;user=phone>
  10. Call-ID: 1FFDA2806AFF233C6921282ED8E22036
  11. CSeq: 1 INVITE
  12. Contact: <sip:79055797907@91.218.111.145:5060;user=phone>
  13. Content-Type: application/sdp
  14. Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, SUBSCRIBE, UPDATE
  15. Max-Forwards: 69
  16. User-Agent: TS-v4.7.3-06b
  17. Cisco-Guid: 213454938-51253738-2335668267-3406482010
  18. Content-Length: 421
  19.  
  20. v=0
  21. o=- 1573316918 1573316918 IN IP4 91.218.111.145
  22. s=-
  23. c=IN IP4 91.218.111.145
  24. t=0 0
  25. m=audio 16506 RTP/AVP 8 0 18 96
  26. a=rtpmap:8 PCMA/8000
  27. a=rtpmap:0 PCMU/8000
  28. a=rtpmap:18 G729/8000
  29. a=fmtp:18 annexb=no
  30. a=rtpmap:96 telephone-event/8000
  31. a=fmtp:96 0-15
  32. a=ptime:20
  33. a=sendrecv
  34. a=silenceSupp:off - - - -
  35. m=image 16508 udptl t38
  36. a=T38FaxRateManagement:transferredTCF
  37. a=T38FaxUdpEC:t38UDPRedundancy
  38. a=sendrecv
  39. <------------->
  40. --- (14 headers 19 lines) ---
  41. Sending to 91.218.111.145:5060 (NAT)
  42. Using INVITE request as basis request - 1FFDA2806AFF233C6921282ED8E22036
  43. Found peer 'new' for '79055797907' from 91.218.111.145:5060
  44.   == Using SIP RTP TOS bits 184
  45.   == Using SIP RTP CoS mark 5
  46. Found RTP audio format 8
  47. Found RTP audio format 0
  48. Found RTP audio format 18
  49. Found RTP audio format 96
  50. Found audio description format PCMA for ID 8
  51. Found audio description format PCMU for ID 0
  52. Found audio description format G729 for ID 18
  53. Found audio description format telephone-event for ID 96
  54.  
  55. <--- Reliably Transmitting (NAT) to 91.218.111.145:5060 --->
  56. SIP/2.0 488 Not acceptable here
  57. Via: SIP/2.0/UDP 91.218.111.145:5060;branch=z9hG4bK-fc2dd4b4030d11ea8da6782bcb0acaea;sig=21a152b6;received=91.218.111.145;rport=5060
  58. Via: SIP/2.0/UDP 2LgN1H6wRYmw;rport;branch=z9hG4bK-fc2dcce4030d11ea8da6782bcb0acaea;sig=72f8fc2f;gr=i_G8oJbx57UhszahK7Pv9qxJ2cABrSTRggDY4bMvhaxcFVfb0B7gEMABWoW8qdU4dQvuqEPDUo4WW11XWAdQb_2wUoqUOjA2tFF-9qxgV7byfbKMu2vUTc1VnTnhzdqZSqwkjoe_Xb1YcHZaCvkAOKK8
  59. From: <sip:79055797907@91.218.111.145:5060;user=phone>;tag=KS4I9eFb2m-SE8XhW9pvkhPElDKgOQAq
  60. To: <sip:74999599580@172.16.20.123:5060;user=phone>;tag=as5970ceed
  61. Call-ID: 1FFDA2806AFF233C6921282ED8E22036
  62. CSeq: 1 INVITE
  63. Server: FPBX-2.8.1(1.8.20.0)
  64. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  65. Supported: replaces, timer
  66. Content-Length: 0
  67.  
  68.  
  69. <------------>
  70. Scheduling destruction of SIP dialog '1FFDA2806AFF233C6921282ED8E22036' in 6400 ms (Method: INVITE)
  71.  
  72. <--- SIP read from UDP:91.218.111.145:5060 --->
  73. ACK sip:s@172.16.20.123:5060 SIP/2.0
  74. Via: SIP/2.0/UDP 91.218.111.145:5060;rport;branch=z9hG4bK-fc2dd4b4030d11ea8da6782bcb0acaea;sig=21a152b6
  75. From: <sip:79055797907@91.218.111.145:5060;user=phone>;tag=KS4I9eFb2m-SE8XhW9pvkhPElDKgOQAq
  76. To: <sip:74999599580@172.16.20.123:5060;user=phone>;tag=as5970ceed
  77. Call-ID: 1FFDA2806AFF233C6921282ED8E22036
  78. Max-Forwards: 70
  79. CSeq: 1 ACK
  80. Content-Length: 0
  81.  
  82. <------------->
  83. --- (8 headers 0 lines) ---
  84. Really destroying SIP dialog '1FFDA2806AFF233C6921282ED8E22036' Method: ACK
  85. sbc*CLI> sip set debug off
RAW Paste Data
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