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  1. [root@localhost ~]#
  2. [root@localhost ~]# asterisk -RvvvvT
  3. [Apr 11 19:32:02] Asterisk 13.12.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  4. [Apr 11 19:32:02] Created by Mark Spencer <markster@digium.com>
  5. [Apr 11 19:32:02] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  6. [Apr 11 19:32:02] This is free software, with components licensed under the GNU General Public
  7. [Apr 11 19:32:02] License version 2 and other licenses; you are welcome to redistribute it under
  8. [Apr 11 19:32:02] certain conditions. Type 'core show license' for details.
  9. [Apr 11 19:32:02] =========================================================================
  10. [Apr 11 19:32:02] Connected to Asterisk 13.12.1 currently running on localhost (pid = 1931)
  11. [2017-04-11 19:32:04] WARNING[18823]: res_pjsip_pubsub.c:639 subscription_get_handler_from_rdata: No registered subscribe handler for event as-feature-event
  12. localhost*CLI> sip set debug on
  13. SIP Debugging enabled
  14. [2017-04-11 19:32:19] WARNING[18823]: res_pjsip_pubsub.c:639 subscription_get_handler_from_rdata: No registered subscribe handler for event as-feature-event
  15. [Apr 11 19:32:23]
  16. [Apr 11 19:32:23] <--- SIP read from UDP:(voip.ms IP):5060 --->
  17. [Apr 11 19:32:23] INVITE sip:(My DID #)@(My WAN IP):38587 SIP/2.0
  18. [Apr 11 19:32:23] Via: SIP/2.0/UDP (voip.ms IP):5060;branch=z9hG4bK7219e273;rport
  19. [Apr 11 19:32:23] Max-Forwards: 70
  20. [Apr 11 19:32:23] From: "(My CNAM)" <sip:(My cellphone #)@(voip.ms IP)>;tag=as5a6903a6
  21. [Apr 11 19:32:23] To: <sip:(My DID #)@(My WAN IP):38587>
  22. [Apr 11 19:32:23] Contact: <sip:(My cellphone #)@(voip.ms IP):5060>
  23. [Apr 11 19:32:23] Call-ID: 2445802e2ba3c12b24ac9cb425b77392@(voip.ms IP):5060
  24. [Apr 11 19:32:23] CSeq: 102 INVITE
  25. [Apr 11 19:32:23] User-Agent: voip.ms
  26. [Apr 11 19:32:23] Date: Wed, 12 Apr 2017 02:32:23 GMT
  27. [Apr 11 19:32:23] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  28. [Apr 11 19:32:23] Supported: replaces, timer
  29. [Apr 11 19:32:23] Remote-Party-ID: "(My CNAM)" <sip:(My cellphone #)@(voip.ms IP)>;party=calling;privacy=off;screen=no
  30. [Apr 11 19:32:23] Content-Type: application/sdp
  31. [Apr 11 19:32:23] Content-Length: 274
  32. [Apr 11 19:32:23]
  33. [Apr 11 19:32:23] v=0
  34. [Apr 11 19:32:23] o=root 1615298891 1615298891 IN IP4 (voip.ms IP)
  35. [Apr 11 19:32:23] s=voip.ms
  36. [Apr 11 19:32:23] c=IN IP4 (voip.ms IP)
  37. [Apr 11 19:32:23] t=0 0
  38. [Apr 11 19:32:23] m=audio 17628 RTP/AVP 0 18 101
  39. [Apr 11 19:32:23] a=rtpmap:0 PCMU/8000
  40. [Apr 11 19:32:23] a=rtpmap:18 G729/8000
  41. [Apr 11 19:32:23] a=fmtp:18 annexb=no
  42. [Apr 11 19:32:23] a=rtpmap:101 telephone-event/8000
  43. [Apr 11 19:32:23] a=fmtp:101 0-16
  44. [Apr 11 19:32:23] a=ptime:20
  45. [Apr 11 19:32:23] a=sendrecv
  46. [Apr 11 19:32:23] <------------->
  47. [Apr 11 19:32:23] --- (15 headers 13 lines) ---
  48. [Apr 11 19:32:23] Sending to (voip.ms IP):5060 (NAT)
  49. [Apr 11 19:32:23] Sending to (voip.ms IP):5060 (NAT)
  50. [Apr 11 19:32:23] Using INVITE request as basis request - 2445802e2ba3c12b24ac9cb425b77392@(voip.ms IP):5060
  51. [Apr 11 19:32:23] Found peer 'sanjose voip.ms' for '(My cellphone #)' from (voip.ms IP):5060
  52. [Apr 11 19:32:23] == Using SIP RTP TOS bits 184
  53. [Apr 11 19:32:23] == Using SIP RTP CoS mark 5
  54. [Apr 11 19:32:23] Found RTP audio format 0
  55. [Apr 11 19:32:23] Found RTP audio format 18
  56. [Apr 11 19:32:23] Found RTP audio format 101
  57. [Apr 11 19:32:23] Found audio description format PCMU for ID 0
  58. [Apr 11 19:32:23] Found audio description format G729 for ID 18
  59. [Apr 11 19:32:23] Found audio description format telephone-event for ID 101
  60. [Apr 11 19:32:23] Capabilities: us - (ulaw|g729), peer - audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|g729)
  61. [Apr 11 19:32:23] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  62. [Apr 11 19:32:23] Peer audio RTP is at port (voip.ms IP):17628
  63. [Apr 11 19:32:23] Looking for (My DID #) in from-trunk (domain (My WAN IP))
  64. [Apr 11 19:32:23] sip_route_dump: route/path hop: <sip:(My cellphone #)@(voip.ms IP):5060>
  65. [Apr 11 19:32:23]
  66. [Apr 11 19:32:23] <--- Transmitting (NAT) to (voip.ms IP):5060 --->
  67. [Apr 11 19:32:23] SIP/2.0 100 Trying
  68. [Apr 11 19:32:23] Via: SIP/2.0/UDP (voip.ms IP):5060;branch=z9hG4bK7219e273;received=(voip.ms IP);rport=5060
  69. [Apr 11 19:32:23] From: "(My CNAM)" <sip:(My cellphone #)@(voip.ms IP)>;tag=as5a6903a6
  70. [Apr 11 19:32:23] To: <sip:(My DID #)@(My WAN IP):38587>
  71. [Apr 11 19:32:23] Call-ID: 2445802e2ba3c12b24ac9cb425b77392@(voip.ms IP):5060
  72. [Apr 11 19:32:23] CSeq: 102 INVITE
  73. [Apr 11 19:32:23] Server: FPBX-13.0.190.7(13.12.1)
  74. [Apr 11 19:32:23] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  75. [Apr 11 19:32:23] Supported: replaces, timer
  76. [Apr 11 19:32:23] Session-Expires: 1800;refresher=uas
  77. [Apr 11 19:32:23] Contact: <sip:(My DID #)@(My WAN IP):5160>
  78. [Apr 11 19:32:23] Content-Length: 0
  79. [Apr 11 19:32:23]
  80. [Apr 11 19:32:23]
  81. [Apr 11 19:32:23] <------------>
  82. [Apr 11 19:32:23] -- Executing [(My DID #)@from-trunk:1] Set("SIP/sanjose voip.ms-00000022", "__FROM_DID=(My DID #)") in new stack
  83. [Apr 11 19:32:23] -- Executing [(My DID #)@from-trunk:2] NoOp("SIP/sanjose voip.ms-00000022", "Received an unknown call with DID set to (My DID #)") in new stack
  84. [Apr 11 19:32:23] -- Executing [(My DID #)@from-trunk:3] Goto("SIP/sanjose voip.ms-00000022", "s,a2") in new stack
  85. [Apr 11 19:32:23] -- Goto (from-trunk,s,2)
  86. [Apr 11 19:32:23] -- Executing [s@from-trunk:2] Answer("SIP/sanjose voip.ms-00000022", "") in new stack
  87. [Apr 11 19:32:23] Audio is at 14022
  88. [Apr 11 19:32:23] Adding codec ulaw to SDP
  89. [Apr 11 19:32:23] Adding codec g729 to SDP
  90. [Apr 11 19:32:23] Adding non-codec 0x1 (telephone-event) to SDP
  91. [Apr 11 19:32:23]
  92. [Apr 11 19:32:23] <--- Reliably Transmitting (NAT) to (voip.ms IP):5060 --->
  93. [Apr 11 19:32:23] SIP/2.0 200 OK
  94. [Apr 11 19:32:23] Via: SIP/2.0/UDP (voip.ms IP):5060;branch=z9hG4bK7219e273;received=(voip.ms IP);rport=5060
  95. [Apr 11 19:32:23] From: "(My CNAM)" <sip:(My cellphone #)@(voip.ms IP)>;tag=as5a6903a6
  96. [Apr 11 19:32:23] To: <sip:(My DID #)@(My WAN IP):38587>;tag=as033f0eb5
  97. [Apr 11 19:32:23] Call-ID: 2445802e2ba3c12b24ac9cb425b77392@(voip.ms IP):5060
  98. [Apr 11 19:32:23] CSeq: 102 INVITE
  99. [Apr 11 19:32:23] Server: FPBX-13.0.190.7(13.12.1)
  100. [Apr 11 19:32:23] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  101. [Apr 11 19:32:23] Supported: replaces, timer
  102. [Apr 11 19:32:23] Session-Expires: 1800;refresher=uas
  103. [Apr 11 19:32:23] Contact: <sip:(My DID #)@(My WAN IP):5160>
  104. [Apr 11 19:32:23] Content-Type: application/sdp
  105. [Apr 11 19:32:23] Require: timer
  106. [Apr 11 19:32:23] Content-Length: 299
  107. [Apr 11 19:32:23]
  108. [Apr 11 19:32:23] v=0
  109. [Apr 11 19:32:23] o=root 761167634 761167634 IN IP4 (My WAN IP)
  110. [Apr 11 19:32:23] s=Asterisk PBX 13.12.1
  111. [Apr 11 19:32:23] c=IN IP4 (My WAN IP)
  112. [Apr 11 19:32:23] t=0 0
  113. [Apr 11 19:32:23] m=audio 14022 RTP/AVP 0 18 101
  114. [Apr 11 19:32:23] a=rtpmap:0 PCMU/8000
  115. [Apr 11 19:32:23] a=rtpmap:18 G729/8000
  116. [Apr 11 19:32:23] a=fmtp:18 annexb=no
  117. [Apr 11 19:32:23] a=rtpmap:101 telephone-event/8000
  118. [Apr 11 19:32:23] a=fmtp:101 0-16
  119. [Apr 11 19:32:23] a=ptime:20
  120. [Apr 11 19:32:23] a=maxptime:150
  121. [Apr 11 19:32:23] a=sendrecv
  122. [Apr 11 19:32:23]
  123. [Apr 11 19:32:23] <------------>
  124. [Apr 11 19:32:23]
  125. [Apr 11 19:32:23] <--- SIP read from UDP:(voip.ms IP):5060 --->
  126. [Apr 11 19:32:23] ACK sip:(My DID #)@(My WAN IP):5160 SIP/2.0
  127. [Apr 11 19:32:23] Via: SIP/2.0/UDP (voip.ms IP):5060;branch=z9hG4bK2b0e19ad;rport
  128. [Apr 11 19:32:23] Max-Forwards: 70
  129. [Apr 11 19:32:23] From: "(My CNAM)" <sip:(My cellphone #)@(voip.ms IP)>;tag=as5a6903a6
  130. [Apr 11 19:32:23] To: <sip:(My DID #)@(My WAN IP):38587>;tag=as033f0eb5
  131. [Apr 11 19:32:23] Contact: <sip:(My cellphone #)@(voip.ms IP):5060>
  132. [Apr 11 19:32:23] Call-ID: 2445802e2ba3c12b24ac9cb425b77392@(voip.ms IP):5060
  133. [Apr 11 19:32:23] CSeq: 102 ACK
  134. [Apr 11 19:32:23] User-Agent: voip.ms
  135. [Apr 11 19:32:23] Content-Length: 0
  136. [Apr 11 19:32:23]
  137. [Apr 11 19:32:23] <------------->
  138. [Apr 11 19:32:23] --- (10 headers 0 lines) ---
  139. [Apr 11 19:32:23] -- Executing [s@from-trunk:3] Log("SIP/sanjose voip.ms-00000022", "WARNING,Friendly Scanner from (voip.ms IP)") in new stack
  140. [2017-04-11 19:32:23] WARNING[2350][C-0000002d]: Ext. s:3 @ from-trunk: Friendly Scanner from (voip.ms IP)
  141. [Apr 11 19:32:23] -- Executing [s@from-trunk:4] Wait("SIP/sanjose voip.ms-00000022", "2") in new stack
  142. [Apr 11 19:32:23] > 0x218cfd0 -- Probation passed - setting RTP source address to (voip.ms IP):17628
  143. [Apr 11 19:32:25] -- Executing [s@from-trunk:5] Playback("SIP/sanjose voip.ms-00000022", "ss-noservice") in new stack
  144. [Apr 11 19:32:25] -- <SIP/sanjose voip.ms-00000022> Playing 'ss-noservice.ulaw' (language 'en')
  145. [Apr 11 19:32:30] -- Executing [s@from-trunk:6] SayAlpha("SIP/sanjose voip.ms-00000022", "(My DID #)") in new stack
  146. [Apr 11 19:32:30] -- <SIP/sanjose voip.ms-00000022> Playing 'digits/9.ulaw' (language 'en')
  147. localhost*CLI> quit
  148. [Apr 11 19:32:31] Asterisk cleanly ending (0).
  149. [Apr 11 19:32:31] Executing last minute cleanups
  150. [root@localhost ~]#
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