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intercom sip debug

BGM Sep 7th, 2019 7 Never
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  1. SIP Debugging Enabled for IP: 192.168.1.145
  2.  
  3. Scheduling destruction of SIP dialog '2bd201bc225d37372f0e4d8e191f7ee5@192.168.1.13:5060' in 6400 ms (Method: INVITE)
  4. Audio is at 16584
  5. Adding codec ulaw to SDP
  6. Adding codec alaw to SDP
  7. Adding codec gsm to SDP
  8. Adding codec g726 to SDP
  9. Adding codec g722 to SDP
  10. Adding codec speex32 to SDP
  11. Adding codec g729 to SDP
  12. Adding non-codec 0x1 (telephone-event) to SDP
  13. Reliably Transmitting (no NAT) to 192.168.1.145:5060:
  14. INVITE sip:2545@192.168.1.145;intercom=true SIP/2.0
  15. Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK04e00efa
  16. Max-Forwards: 70
  17. From: "MyUser" <sip:2536@192.168.1.13>;tag=as1fc9deee
  18. To: <sip:2545@192.168.1.145;intercom=true>
  19. Contact: <sip:2536@192.168.1.13:5060>
  20. Call-ID: 02eed05f14d2d67825c19e0471f015bb@192.168.1.13:5060
  21. CSeq: 102 INVITE
  22. User-Agent: FPBX-14.0.13.4(13.22.0)
  23. Date: Sat, 07 Sep 2019 17:22:11 GMT
  24. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  25. Supported: replaces, timer
  26. Alert-Info: <http://127.0.0.1>;info=Alert-Info: Auto Answer
  27. Call-Info: <uri>;answer-after=0
  28. P-Asserted-Identity: "MyUser" <sip:2536@192.168.1.13>
  29. Content-Type: application/sdp
  30. Content-Length: 430
  31.  
  32. v=0
  33. o=root 1682910058 1682910058 IN IP4 192.168.1.13
  34. s=Asterisk PBX 13.22.0
  35. c=IN IP4 192.168.1.13
  36. t=0 0
  37. m=audio 16584 RTP/AVP 0 8 3 111 9 119 18 101
  38. a=rtpmap:0 PCMU/8000
  39. a=rtpmap:8 PCMA/8000
  40. a=rtpmap:3 GSM/8000
  41. a=rtpmap:111 G726-32/8000
  42. a=rtpmap:9 G722/8000
  43. a=rtpmap:119 speex/32000
  44. a=rtpmap:18 G729/8000
  45. a=fmtp:18 annexb=no
  46. a=rtpmap:101 telephone-event/8000
  47. a=fmtp:101 0-16
  48. a=ptime:20
  49. a=maxptime:60
  50. a=sendrecv
  51.  
  52. ---
  53.  
  54. <--- SIP read from UDP:192.168.1.145:5060 --->
  55. SIP/2.0 100 Trying
  56. Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK04e00efa
  57. From: "MyUser" <sip:2536@192.168.1.13>;tag=as1fc9deee
  58. To: "2545" <sip:2545@192.168.1.145;intercom=true>;tag=5585A5E7-4D190D8C
  59. CSeq: 102 INVITE
  60. Call-ID: 02eed05f14d2d67825c19e0471f015bb@192.168.1.13:5060
  61. Contact: <sip:2545@192.168.1.145>
  62. User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.7.0198
  63. Accept-Language: en
  64. Content-Length: 0
  65.  
  66. <------------->
  67. --- (10 headers 0 lines) ---
  68.  
  69. <--- SIP read from UDP:192.168.1.145:5060 --->
  70. SIP/2.0 180 Ringing
  71. Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK04e00efa
  72. From: "MyUser" <sip:2536@192.168.1.13>;tag=as1fc9deee
  73. To: "2545" <sip:2545@192.168.1.145;intercom=true>;tag=5585A5E7-4D190D8C
  74. CSeq: 102 INVITE
  75. Call-ID: 02eed05f14d2d67825c19e0471f015bb@192.168.1.13:5060
  76. Contact: <sip:2545@192.168.1.145>
  77. User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.7.0198
  78. Call-Info: <uri>;answer-after=0
  79. Allow-Events: talk,hold,conference
  80. Accept-Language: en
  81. Content-Length: 0
  82.  
  83.  
  84. Reliably Transmitting (no NAT) to 192.168.1.145:5060:
  85. CANCEL sip:2545@192.168.1.145;intercom=true SIP/2.0
  86. Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK04e00efa
  87. Max-Forwards: 70
  88. From: "MyUser" <sip:2536@192.168.1.13>;tag=as1fc9deee
  89. To: <sip:2545@192.168.1.145;intercom=true>
  90. Call-ID: 02eed05f14d2d67825c19e0471f015bb@192.168.1.13:5060
  91. CSeq: 102 CANCEL
  92. User-Agent: FPBX-14.0.13.4(13.22.0)
  93. Content-Length: 0
  94.  
  95.  
  96. ---
  97. Scheduling destruction of SIP dialog '02eed05f14d2d67825c19e0471f015bb@192.168.1.13:5060' in 6400 ms (Method: INVITE)
  98. [2019-09-07 12:22:16] WARNING[15571][C-00000eb6]: channel.c:5080 ast_prod: Prodding channel 'SIP/2536-0000135a' failed
  99.  
  100. <--- SIP read from UDP:192.168.1.145:5060 --->
  101. SIP/2.0 200 OK
  102. Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK04e00efa
  103. From: "MyUser" <sip:2536@192.168.1.13>;tag=as1fc9deee
  104. To: "2545" <sip:2545@192.168.1.145;intercom=true>
  105. CSeq: 102 CANCEL
  106. Call-ID: 02eed05f14d2d67825c19e0471f015bb@192.168.1.13:5060
  107. Contact: <sip:2545@192.168.1.145>
  108. User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.7.0198
  109. Accept-Language: en
  110. Content-Length: 0
  111.  
  112. <------------->
  113. --- (10 headers 0 lines) ---
  114.  
  115. <--- SIP read from UDP:192.168.1.145:5060 --->
  116. SIP/2.0 487 Request Cancelled
  117. Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK04e00efa
  118. From: "MyUser" <sip:2536@192.168.1.13>;tag=as1fc9deee
  119. To: "2545" <sip:2545@192.168.1.145;intercom=true>;tag=5585A5E7-4D190D8C
  120. CSeq: 102 INVITE
  121. Call-ID: 02eed05f14d2d67825c19e0471f015bb@192.168.1.13:5060
  122. Contact: <sip:2545@192.168.1.145>
  123. User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.2.7.0198
  124. Accept-Language: en
  125. Content-Length: 0
  126.  
  127. <------------->
  128. --- (10 headers 0 lines) ---
  129. Transmitting (no NAT) to 192.168.1.145:5060:
  130. ACK sip:2545@192.168.1.145 SIP/2.0
  131. Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK04e00efa
  132. Max-Forwards: 70
  133. From: "MyUser" <sip:2536@192.168.1.13>;tag=as1fc9deee
  134. To: <sip:2545@192.168.1.145;intercom=true>;tag=5585A5E7-4D190D8C
  135. Contact: <sip:2536@192.168.1.13:5060>
  136. Call-ID: 02eed05f14d2d67825c19e0471f015bb@192.168.1.13:5060
  137. CSeq: 102 ACK
  138. User-Agent: FPBX-14.0.13.4(13.22.0)
  139. Content-Length: 0
RAW Paste Data
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