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  1. MITOL*CLI> sip set debug peer sim1
  2. SIP Debugging Enabled for IP: 192.168.0.100
  3.  
  4. <--- SIP read from UDP:192.168.0.100:5064 --->
  5. INVITE sip:receiver@192.168.0.25 SIP/2.0
  6. Via: SIP/2.0/UDP 192.168.0.100:5064;rport;branch=z9hG4bK8444cd044f
  7. From: "+79250287897" <sip:sim1@192.168.0.100:5064>;tag=446a7210
  8. To: <sip:receiver@192.168.0.25>
  9. Call-ID: 255e70cd5b3fb5e533befc303f9ecdda@192.168.0.100
  10. Contact: <sip:+79250287897@192.168.0.100:5064>
  11. CSeq: 1 INVITE
  12. Max-Forwards: 70
  13. Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
  14. Supported: replaces
  15. Content-Type: application/sdp
  16. User-Agent: Mv-37x (904290)
  17. Content-Length: 323
  18.  
  19. v=0
  20. o=CMI-SIPUA 723 0 IN IP4 192.168.0.100
  21. s=SIP CALL
  22. c=IN IP4 192.168.0.100
  23. t=0 0
  24. m=audio 20004 RTP/AVP 0 8 4 18 23 22 2 21 101
  25. a=rtpmap:23 G726-16/8000
  26. a=rtpmap:22 G726-24/8000
  27. a=rtpmap:2 G726-32/8000
  28. a=rtpmap:21 G726-40/8000
  29. a=rtpmap:101 telephone-event/8000
  30. a=fmtp:101 0-15
  31. a=fmtp:18 annexb=no
  32. a=sendrecv
  33. <------------->
  34. --- (13 headers 14 lines) ---
  35. Sending to 192.168.0.100:5064 (NAT)
  36. Using INVITE request as basis request - 255e70cd5b3fb5e533befc303f9ecdda@192.168.0.100
  37. Found peer 'sim1' for 'sim1' from 192.168.0.100:5064
  38.   == Using SIP RTP CoS mark 5
  39. Found RTP audio format 0
  40. Found RTP audio format 8
  41. Found RTP audio format 4
  42. Found RTP audio format 18
  43. Found RTP audio format 23
  44. Found RTP audio format 22
  45. Found RTP audio format 2
  46. Found RTP audio format 21
  47. Found RTP audio format 101
  48. Found unknown media description format G726-16 for ID 23
  49. Found unknown media description format G726-24 for ID 22
  50. Found audio description format G726-32 for ID 2
  51. Found unknown media description format G726-40 for ID 21
  52. Found audio description format telephone-event for ID 101
  53. Capabilities: us - 0x80000008090e (gsm|ulaw|alaw|g726|g729|h263|testlaw), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x90c (ulaw|alaw|g726|g729)
  54. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  55. Peer audio RTP is at port 192.168.0.100:20004
  56. Looking for receiver in sip (domain 192.168.0.25)
  57.  
  58. <--- Reliably Transmitting (NAT) to 192.168.0.100:5064 --->
  59. SIP/2.0 404 Not Found
  60. Via: SIP/2.0/UDP 192.168.0.100:5064;branch=z9hG4bK8444cd044f;received=192.168.0.100;rport=5064
  61. From: "+79250287897" <sip:sim1@192.168.0.100:5064>;tag=446a7210
  62. To: <sip:receiver@192.168.0.25>;tag=as36414709
  63. Call-ID: 255e70cd5b3fb5e533befc303f9ecdda@192.168.0.100
  64. CSeq: 1 INVITE
  65. Server: Asterisk PBX 1.8.8.2~dfsg-1
  66. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  67. Supported: replaces, timer
  68. Content-Length: 0
  69.  
  70.  
  71. <------------>
  72. [Mar 24 19:25:10] NOTICE[27874]: chan_sip.c:22147 handle_request_invite: Call from 'sim1' (192.168.0.100:5064) to extension 'receiver' rejected because extension not found in context 'sip'.
  73. Scheduling destruction of SIP dialog '255e70cd5b3fb5e533befc303f9ecdda@192.168.0.100' in 32000 ms (Method: INVITE)
  74.  
  75. <--- SIP read from UDP:192.168.0.100:5064 --->
  76. ACK sip:receiver@192.168.0.25 SIP/2.0
  77. Via: SIP/2.0/UDP 192.168.0.100:5064;branch=z9hG4bK8444cd044f
  78. From: "+79250287897" <sip:sim1@192.168.0.100:5064>;tag=446a7210
  79. To: <sip:receiver@192.168.0.25>;tag=as36414709
  80. Call-ID: 255e70cd5b3fb5e533befc303f9ecdda@192.168.0.100
  81. Contact: <sip:+79250287897@192.168.0.100:5064>
  82. CSeq: 1 ACK
  83. Max-Forwards: 70
  84. Content-Length: 0
  85.  
  86. <------------->
  87. --- (9 headers 0 lines) ---
  88. Really destroying SIP dialog '255e70cd5b3fb5e533befc303f9ecdda@192.168.0.100' Method: ACK
RAW Paste Data
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