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  1. <--- SIP read from UDP:209.87.247.153:5060 --->
  2. INVITE sip:[email protected];user=phone SIP/2.0
  3. Record-Route: <sip:209.87.247.153;lr=on;ftag=F31ED35F-8C62ED5C;nat=yes>
  4. Via: SIP/2.0/UDP 209.87.247.153;branch=z9hG4bKffbf.564aa823.0
  5. Via: SIP/2.0/UDP 172.16.0.250;rport=1024;received=24.52.251.44;branch=z9hG4bK249651b6C6452F97
  6. From: "Paul Belanger" <sip:[email protected]>;tag=F31ED35F-8C62ED5C
  7. To: <sip:[email protected];user=phone>
  8. CSeq: 2 INVITE
  9. Contact: <sip:[email protected];alias=24.52.251.44~1024~1>
  10. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  11. User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.4.0085
  12. Accept-Language: en
  13. Supported: 100rel,replaces
  14. Allow-Events: talk,hold,conference
  15. Max-Forwards: 69
  16. Content-Type: application/sdp
  17. Content-Length: 317
  18.  
  19. v=0
  20. o=- 1167611450 1167611450 IN IP4 209.87.247.153
  21. s=Polycom IP Phone
  22. c=IN IP4 209.87.247.153
  23. t=0 0
  24. a=sendrecv
  25. m=audio 50028 RTP/AVP 9 0 8 18 127
  26. a=rtpmap:9 G722/8000
  27. a=rtpmap:0 PCMU/8000
  28. a=rtpmap:8 PCMA/8000
  29. a=rtpmap:18 G729/8000
  30. a=fmtp:18 annexb=no
  31. a=rtpmap:127 telephone-event/8000
  32. a=nortpproxy:yes
  33. <------------->
  34. --- (17 headers 14 lines) ---
  35. Sending to 209.87.247.153:5060 (no NAT)
  36. Using INVITE request as basis request - [email protected]
  37. No matching peer for '213' from '209.87.247.153:5060'
  38. [2013-09-14 14:54:39.004] NOTICE[17470]: chan_sip.c:22005 handle_request_invite: Sending fake auth rejection for device "Paul Belanger" <sip:[email protected]>;tag=F31ED35F-8C62ED5C
  39.  
  40. <--- Reliably Transmitting (no NAT) to 209.87.247.153:5060 --->
  41. SIP/2.0 401 Unauthorized
  42. Via: SIP/2.0/UDP 209.87.247.153;branch=z9hG4bKffbf.564aa823.0;received=209.87.247.153
  43. Via: SIP/2.0/UDP 172.16.0.250;rport=1024;received=24.52.251.44;branch=z9hG4bK249651b6C6452F97
  44. From: "Paul Belanger" <sip:[email protected]>;tag=F31ED35F-8C62ED5C
  45. To: <sip:[email protected];user=phone>;tag=as013b91a6
  46. CSeq: 2 INVITE
  47. Server: Asterisk PBX 1.8.7.1-1kickstand0~ppa190.1~precise
  48. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  49. Supported: replaces, timer
  50. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2917521f"
  51. Content-Length: 0
  52.  
  53.  
  54. <------------>
  55. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
  56. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
  57.  
  58. <--- SIP read from UDP:209.87.247.153:5060 --->
  59. ACK sip:[email protected];user=phone SIP/2.0
  60. Via: SIP/2.0/UDP 209.87.247.153;branch=z9hG4bKffbf.564aa823.0
  61. From: "Paul Belanger" <sip:[email protected]>;tag=F31ED35F-8C62ED5C
  62. To: <sip:[email protected];user=phone>;tag=as013b91a6
  63. CSeq: 2 ACK
  64. Max-Forwards: 69
  65. Content-Length: 0
  66.  
  67. <------------->
  68. --- (8 headers 0 lines) ---
  69.  
  70. <--- SIP read from UDP:209.87.247.153:5060 --->
  71. INVITE sip:[email protected];user=phone SIP/2.0
  72. Record-Route: <sip:209.87.247.153;lr=on;ftag=F31ED35F-8C62ED5C;nat=yes>
  73. Via: SIP/2.0/UDP 209.87.247.153;branch=z9hG4bK00cf.034569d5.0
  74. Via: SIP/2.0/UDP 172.16.0.250;rport=1024;received=24.52.251.44;branch=z9hG4bK932714a26B8947D3
  75. From: "Paul Belanger" <sip:[email protected]>;tag=F31ED35F-8C62ED5C
  76. To: <sip:[email protected];user=phone>
  77. CSeq: 3 INVITE
  78. Contact: <sip:[email protected];alias=24.52.251.44~1024~1>
  79. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  80. User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.3.4.0085
  81. Accept-Language: en
  82. Supported: 100rel,replaces
  83. Allow-Events: talk,hold,conference
  84. Authorization: Digest username="[email protected]", realm="asterisk", nonce="2917521f", uri="sip:[email protected];user=phone", response="c0e79e9192311ee2fd1916569c80be45", algorithm=MD5
  85. Max-Forwards: 69
  86. Content-Type: application/sdp
  87. Content-Length: 317
  88.  
  89. v=0
  90. o=- 1167611450 1167611450 IN IP4 209.87.247.153
  91. s=Polycom IP Phone
  92. c=IN IP4 209.87.247.153
  93. t=0 0
  94. a=sendrecv
  95. m=audio 59210 RTP/AVP 9 0 8 18 127
  96. a=rtpmap:9 G722/8000
  97. a=rtpmap:0 PCMU/8000
  98. a=rtpmap:8 PCMA/8000
  99. a=rtpmap:18 G729/8000
  100. a=fmtp:18 annexb=no
  101. a=rtpmap:127 telephone-event/8000
  102. a=nortpproxy:yes
  103. <------------->
  104. --- (18 headers 14 lines) ---
  105. Sending to 209.87.247.153:5060 (no NAT)
  106. Using INVITE request as basis request - [email protected]
  107. No matching peer for '213' from '209.87.247.153:5060'
  108. [2013-09-14 14:54:39.069] NOTICE[17470]: chan_sip.c:22005 handle_request_invite: Sending fake auth rejection for device "Paul Belanger" <sip:[email protected]>;tag=F31ED35F-8C62ED5C
  109.  
  110. <--- Transmitting (no NAT) to 209.87.247.153:5060 --->
  111. SIP/2.0 403 Forbidden (Bad auth)
  112. Via: SIP/2.0/UDP 209.87.247.153;branch=z9hG4bK00cf.034569d5.0;received=209.87.247.153
  113. Via: SIP/2.0/UDP 172.16.0.250;rport=1024;received=24.52.251.44;branch=z9hG4bK932714a26B8947D3
  114. From: "Paul Belanger" <sip:[email protected]>;tag=F31ED35F-8C62ED5C
  115. To: <sip:[email protected];user=phone>;tag=as013b91a6
  116. CSeq: 3 INVITE
  117. Server: Asterisk PBX 1.8.7.1-1kickstand0~ppa190.1~precise
  118. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  119. Supported: replaces, timer
  120. Content-Length: 0
  121.  
  122.  
  123. <------------>
  124. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
  125.  
  126. <--- SIP read from UDP:209.87.247.153:5060 --->
  127. ACK sip:[email protected];user=phone SIP/2.0
  128. Via: SIP/2.0/UDP 209.87.247.153;branch=z9hG4bK00cf.034569d5.0
  129. From: "Paul Belanger" <sip:[email protected]>;tag=F31ED35F-8C62ED5C
  130. To: <sip:[email protected];user=phone>;tag=as013b91a6
  131. CSeq: 3 ACK
  132. Max-Forwards: 69
  133. Content-Length: 0
  134.  
  135. <------------->
  136. --- (8 headers 0 lines) ---
  137. pbx-01-prod*CLI> sip show peers
  138. Name/username Host Dyn Forcerport ACL Port Status Description
  139. kamailio (Unspecified) 0 UNKNOWN
  140. kamailio-01-prod (Unspecified) 0 UNKNOWN
  141. voip.ms (Unspecified) 0 UNKNOWN
  142. 3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0 offline]
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