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  1. ipiphony*CLI>
  2.  
  3. <--- SIP read from UDP:67.53.0.131:11961 --->
  4. REGISTER sip:64.105.229.19:5060 SIP/2.0
  5. Via: SIP/2.0/UDP 192.168.155.113:64232;branch=z9hG4bK-d8754z-3f6ae63e830b1a2b-1---d8754z-;rport
  6. Max-Forwards: 70
  7. Contact: <sip:101@67.53.0.131:11961;transport=UDP;rinstance=7c739af14cb9d189>
  8. To: <sip:101@64.105.229.19:5060>
  9. From: <sip:101@64.105.229.19:5060>;tag=be2ce406
  10. Call-ID: NmMzMDQ4NWNjM2FkOGEyYzMzMjQ2MjEwZmY4Y2VhZDA.
  11. CSeq: 929 REGISTER
  12. Expires: 120
  13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
  14. Supported: replaces
  15. User-Agent: 3CXPhone 6.0.19920.0
  16. Authorization: Digest username="101",realm="asterisk",nonce="6cc9c7e6",uri="sip:64.105.229.19:5060",response="318b6dd6247d6799d06ae1c034bdf1e9",algorithm=MD5
  17. Content-Length: 0
  18.  
  19.  
  20. <------------->
  21. --- (14 headers 0 lines) ---
  22. Sending to 67.53.0.131 : 11961 (NAT)
  23.  
  24. <--- Transmitting (NAT) to 67.53.0.131:11961 --->
  25. SIP/2.0 401 Unauthorized
  26. Via: SIP/2.0/UDP 192.168.155.113:64232;branch=z9hG4bK-d8754z-3f6ae63e830b1a2b-1---d8754z-;received=67.53.0.131;rport=11961
  27. From: <sip:101@64.105.229.19:5060>;tag=be2ce406
  28. To: <sip:101@64.105.229.19:5060>;tag=as21f7f11e
  29. Call-ID: NmMzMDQ4NWNjM2FkOGEyYzMzMjQ2MjEwZmY4Y2VhZDA.
  30. CSeq: 929 REGISTER
  31. Server: Asterisk PBX 1.6.2.18
  32. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  33. Supported: replaces, timer
  34. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ffabb99", stale=true
  35. Content-Length: 0
  36.  
  37.  
  38. <------------>
  39. Scheduling destruction of SIP dialog 'NmMzMDQ4NWNjM2FkOGEyYzMzMjQ2MjEwZmY4Y2VhZDA.' in 32000 ms (Method: REGISTER)
  40.  
  41. <--- SIP read from UDP:74.222.60.237:5060 --->
  42. INVITE sip:101@64.105.229.19 SIP/2.0
  43. Via: SIP/2.0/UDP 74.222.60.237:5060;branch=z9hG4bK3442d6df;rport
  44. Max-Forwards: 70
  45. From: "2242384656" <sip:2242384656@74.222.60.237>;tag=as58ce53bb
  46. To: <sip:101@64.105.229.19>
  47. Contact: <sip:2242384656@74.222.60.237>
  48. Call-ID: 2b63dfcf2dff7e707755a516582175a1@74.222.60.237
  49. CSeq: 102 INVITE
  50. User-Agent: Asterisk PBX 1.6.2.18
  51. Date: Wed, 05 Oct 2011 14:56:18 GMT
  52. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  53. Supported: replaces
  54. Content-Type: application/sdp
  55. Content-Length: 239
  56.  
  57. v=0
  58. o=root 1371694416 1371694416 IN IP4 74.222.60.237
  59. s=Asterisk PBX 1.6.2.18
  60. c=IN IP4 74.222.60.237
  61. t=0 0
  62. m=audio 10062 RTP/AVP 0 101
  63. a=rtpmap:0 PCMU/8000
  64. a=rtpmap:101 telephone-event/8000
  65. a=fmtp:101 0-16
  66. a=ptime:20
  67. a=sendrecv
  68.  
  69. <------------->
  70. --- (14 headers 11 lines) ---
  71.   == Using SIP RTP TOS bits 184
  72.   == Using SIP RTP CoS mark 5
  73. Sending to 74.222.60.237 : 5060 (NAT)
  74. Using INVITE request as basis request - 2b63dfcf2dff7e707755a516582175a1@74.222.60.237
  75. Found peer 'ds-all' for '2242384656' from 74.222.60.237:5060
  76. Found RTP audio format 0
  77. Found RTP audio format 101
  78. Found audio description format PCMU for ID 0
  79. Found audio description format telephone-event for ID 101
  80. Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  81. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  82. Peer audio RTP is at port 74.222.60.237:10062
  83. Looking for 101 in from-trunk (domain 64.105.229.19)
  84. list_route: hop: <sip:2242384656@74.222.60.237>
  85.  
  86. <--- Transmitting (NAT) to 74.222.60.237:5060 --->
  87. SIP/2.0 100 Trying
  88. Via: SIP/2.0/UDP 74.222.60.237:5060;branch=z9hG4bK3442d6df;received=74.222.60.237;rport=5060
  89. From: "2242384656" <sip:2242384656@74.222.60.237>;tag=as58ce53bb
  90. To: <sip:101@64.105.229.19>
  91. Call-ID: 2b63dfcf2dff7e707755a516582175a1@74.222.60.237
  92. CSeq: 102 INVITE
  93. Server: Asterisk PBX 1.6.2.18
  94. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  95. Supported: replaces, timer
  96. Contact: <sip:101@64.105.229.19>
  97. Content-Length: 0
  98.  
  99.  
  100. <------------>
  101.     -- Executing [101@from-trunk:1] Macro("SIP/ds-all-00000033", "exten-vm,101,101") in new stack
  102.     -- Executing [s@macro-exten-vm:1] Macro("SIP/ds-all-00000033", "user-callerid,") in new stack
  103.     -- Executing [s@macro-user-callerid:1] Set("SIP/ds-all-00000033", "AMPUSER=2242384656") in new stack
  104.     -- Executing [s@macro-user-callerid:2] GotoIf("SIP/ds-all-00000033", "0?report") in new stack
  105.     -- Executing [s@macro-user-callerid:3] ExecIf("SIP/ds-all-00000033", "1?Set(REALCALLERIDNUM=2242384656)") in new stack
  106.     -- Executing [s@macro-user-callerid:4] Set("SIP/ds-all-00000033", "AMPUSER=") in new stack
  107.     -- Executing [s@macro-user-callerid:5] Set("SIP/ds-all-00000033", "AMPUSERCIDNAME=") in new stack
  108.     -- Executing [s@macro-user-callerid:6] GotoIf("SIP/ds-all-00000033", "1?report") in new stack
  109.     -- Goto (macro-user-callerid,s,9)
  110.     -- Executing [s@macro-user-callerid:9] GotoIf("SIP/ds-all-00000033", "0?continue") in new stack
  111.     -- Executing [s@macro-user-callerid:10] Set("SIP/ds-all-00000033", "__TTL=64") in new stack
  112.     -- Executing [s@macro-user-callerid:11] GotoIf("SIP/ds-all-00000033", "1?continue") in new stack
  113.     -- Goto (macro-user-callerid,s,18)
  114.     -- Executing [s@macro-user-callerid:18] NoOp("SIP/ds-all-00000033", "Using CallerID "2242384656" <2242384656>") in new stack
  115.     -- Executing [s@macro-exten-vm:2] Set("SIP/ds-all-00000033", "RingGroupMethod=none") in new stack
  116.     -- Executing [s@macro-exten-vm:3] Set("SIP/ds-all-00000033", "VMBOX=101") in new stack
  117.     -- Executing [s@macro-exten-vm:4] Set("SIP/ds-all-00000033", "__EXTTOCALL=101") in new stack
  118.     -- Executing [s@macro-exten-vm:5] Set("SIP/ds-all-00000033", "CFUEXT=") in new stack
  119.     -- Executing [s@macro-exten-vm:6] Set("SIP/ds-all-00000033", "CFBEXT=") in new stack
  120.     -- Executing [s@macro-exten-vm:7] Set("SIP/ds-all-00000033", "RT=15") in new stack
  121.     -- Executing [s@macro-exten-vm:8] Macro("SIP/ds-all-00000033", "record-enable,101,IN") in new stack
  122.     -- Executing [s@macro-record-enable:1] GotoIf("SIP/ds-all-00000033", "1?check") in new stack
  123.     -- Goto (macro-record-enable,s,4)
  124.     -- Executing [s@macro-record-enable:4] ExecIf("SIP/ds-all-00000033", "0?MacroExit()") in new stack
  125.     -- Executing [s@macro-record-enable:5] GotoIf("SIP/ds-all-00000033", "0?Group:OUT") in new stack
  126.     -- Goto (macro-record-enable,s,15)
  127.     -- Executing [s@macro-record-enable:15] GotoIf("SIP/ds-all-00000033", "1?IN") in new stack
  128.     -- Goto (macro-record-enable,s,20)
  129.     -- Executing [s@macro-record-enable:20] ExecIf("SIP/ds-all-00000033", "1?MacroExit()") in new stack
  130.     -- Executing [s@macro-exten-vm:9] Macro("SIP/ds-all-00000033", "dial-one,15,trTwW,101") in new stack
  131.     -- Executing [s@macro-dial-one:1] Set("SIP/ds-all-00000033", "DEXTEN=101") in new stack
  132.     -- Executing [s@macro-dial-one:2] Set("SIP/ds-all-00000033", "DIALSTATUS_CW=") in new stack
  133.     -- Executing [s@macro-dial-one:3] GosubIf("SIP/ds-all-00000033", "0?screen,1") in new stack
  134.     -- Executing [s@macro-dial-one:4] GosubIf("SIP/ds-all-00000033", "0?cf,1") in new stack
  135.     -- Executing [s@macro-dial-one:5] GotoIf("SIP/ds-all-00000033", "1?skip1") in new stack
  136.     -- Goto (macro-dial-one,s,8)
  137.     -- Executing [s@macro-dial-one:8] GotoIf("SIP/ds-all-00000033", "0?nodial") in new stack
  138.     -- Executing [s@macro-dial-one:9] GotoIf("SIP/ds-all-00000033", "0?continue") in new stack
  139.     -- Executing [s@macro-dial-one:10] Set("SIP/ds-all-00000033", "EXTHASCW=ENABLED") in new stack
  140.     -- Executing [s@macro-dial-one:11] GotoIf("SIP/ds-all-00000033", "0?next1:cwinusebusy") in new stack
  141.     -- Goto (macro-dial-one,s,23)
  142.     -- Executing [s@macro-dial-one:23] GotoIf("SIP/ds-all-00000033", "1?next3:continue") in new stack
  143.     -- Goto (macro-dial-one,s,24)
  144.     -- Executing [s@macro-dial-one:24] ExecIf("SIP/ds-all-00000033", "0?Set(DIALSTATUS_CW=BUSY)") in new stack
  145.     -- Executing [s@macro-dial-one:25] GotoIf("SIP/ds-all-00000033", "0?nodial") in new stack
  146.     -- Executing [s@macro-dial-one:26] GosubIf("SIP/ds-all-00000033", "1?dstring,1:dlocal,1") in new stack
  147.     -- Executing [dstring@macro-dial-one:1] Set("SIP/ds-all-00000033", "DSTRING=") in new stack
  148.     -- Executing [dstring@macro-dial-one:2] Set("SIP/ds-all-00000033", "DEVICES=101") in new stack
  149.     -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/ds-all-00000033", "0?Return()") in new stack
  150.     -- Executing [dstring@macro-dial-one:4] Set("SIP/ds-all-00000033", "LOOPCNT=1") in new stack
  151.     -- Executing [dstring@macro-dial-one:5] Set("SIP/ds-all-00000033", "ITER=1") in new stack
  152.     -- Executing [dstring@macro-dial-one:6] Set("SIP/ds-all-00000033", "THISDIAL=SIP/101") in new stack
  153.     -- Executing [dstring@macro-dial-one:7] GosubIf("SIP/ds-all-00000033", "1?zap2dahdi,1") in new stack
  154.     -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/ds-all-00000033", "0?Return()") in new stack
  155.     -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/ds-all-00000033", "NEWDIAL=") in new stack
  156.     -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/ds-all-00000033", "LOOPCNT2=1") in new stack
  157.     -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/ds-all-00000033", "ITER2=1") in new stack
  158.     -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/ds-all-00000033", "THISPART2=SIP/101") in new stack
  159.     -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/ds-all-00000033", "0?Set(THISPART2=DAHDI/101)") in new stack
  160.     -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/ds-all-00000033", "NEWDIAL=SIP/101&") in new stack
  161.     -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/ds-all-00000033", "ITER2=2") in new stack
  162.     -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/ds-all-00000033", "0?begin2") in new stack
  163.     -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/ds-all-00000033", "THISDIAL=SIP/101") in new stack
  164.     -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/ds-all-00000033", "") in new stack
  165.     -- Executing [dstring@macro-dial-one:8] Set("SIP/ds-all-00000033", "DSTRING=SIP/101&") in new stack
  166.     -- Executing [dstring@macro-dial-one:9] Set("SIP/ds-all-00000033", "ITER=2") in new stack
  167.     -- Executing [dstring@macro-dial-one:10] GotoIf("SIP/ds-all-00000033", "0?begin") in new stack
  168.     -- Executing [dstring@macro-dial-one:11] Set("SIP/ds-all-00000033", "DSTRING=SIP/101") in new stack
  169.     -- Executing [dstring@macro-dial-one:12] Return("SIP/ds-all-00000033", "") in new stack
  170.     -- Executing [s@macro-dial-one:27] GotoIf("SIP/ds-all-00000033", "0?nodial") in new stack
  171.     -- Executing [s@macro-dial-one:28] GotoIf("SIP/ds-all-00000033", "1?skiptrace") in new stack
  172.     -- Goto (macro-dial-one,s,30)
  173.     -- Executing [s@macro-dial-one:30] Set("SIP/ds-all-00000033", "D_OPTIONS=trTwW") in new stack
  174.     -- Executing [s@macro-dial-one:31] ExecIf("SIP/ds-all-00000033", "0?SIPAddHeader(Alert-Info: )") in new stack
  175.     -- Executing [s@macro-dial-one:32] ExecIf("SIP/ds-all-00000033", "0?SIPAddHeader()") in new stack
  176.     -- Executing [s@macro-dial-one:33] ExecIf("SIP/ds-all-00000033", "0?SetMusicOnHold()") in new stack
  177.     -- Executing [s@macro-dial-one:34] GosubIf("SIP/ds-all-00000033", "0?qwait,1") in new stack
  178.     -- Executing [s@macro-dial-one:35] Set("SIP/ds-all-00000033", "__CWIGNORE=") in new stack
  179.     -- Executing [s@macro-dial-one:36] Set("SIP/ds-all-00000033", "__KEEPCID=TRUE") in new stack
  180.     -- Executing [s@macro-dial-one:37] Dial("SIP/ds-all-00000033", "SIP/101,15,trTwW") in new stack
  181.   == Using SIP RTP TOS bits 184
  182.   == Using SIP RTP CoS mark 5
  183. Really destroying SIP dialog '25ec06ae35f7ac970ae6627b00b68035@127.0.1.1' Method: INVITE
  184.   == Everyone is busy/congested at this time (1:0/0/1)
  185.     -- Executing [s@macro-dial-one:38] ExecIf("SIP/ds-all-00000033", "0?Set(DIALSTATUS=)") in new stack
  186.     -- Executing [s@macro-dial-one:39] GosubIf("SIP/ds-all-00000033", "0?s-CHANUNAVAIL,1") in new stack
  187.     -- Executing [s@macro-dial-one:40] MacroExit("SIP/ds-all-00000033", "") in new stack
  188.     -- Executing [s@macro-exten-vm:10] GotoIf("SIP/ds-all-00000033", "0?exit,return") in new stack
  189.     -- Executing [s@macro-exten-vm:11] Set("SIP/ds-all-00000033", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
  190.     -- Executing [s@macro-exten-vm:12] GosubIf("SIP/ds-all-00000033", "0?docfu,1") in new stack
  191.     -- Executing [s@macro-exten-vm:13] GosubIf("SIP/ds-all-00000033", "0?docfb,1") in new stack
  192.     -- Executing [s@macro-exten-vm:14] Set("SIP/ds-all-00000033", "DIALSTATUS=CHANUNAVAIL") in new stack
  193.     -- Executing [s@macro-exten-vm:15] NoOp("SIP/ds-all-00000033", "Voicemail is '101'") in new stack
  194.     -- Executing [s@macro-exten-vm:16] GotoIf("SIP/ds-all-00000033", "0?s-CHANUNAVAIL,1") in new stack
  195.     -- Executing [s@macro-exten-vm:17] NoOp("SIP/ds-all-00000033", "Sending to Voicemail box 101") in new stack
  196.     -- Executing [s@macro-exten-vm:18] Macro("SIP/ds-all-00000033", "vm,101,CHANUNAVAIL,") in new stack
  197.     -- Executing [s@macro-vm:1] Macro("SIP/ds-all-00000033", "user-callerid,SKIPTTL") in new stack
  198.     -- Executing [s@macro-user-callerid:1] Set("SIP/ds-all-00000033", "AMPUSER=2242384656") in new stack
  199.     -- Executing [s@macro-user-callerid:2] GotoIf("SIP/ds-all-00000033", "0?report") in new stack
  200.     -- Executing [s@macro-user-callerid:3] ExecIf("SIP/ds-all-00000033", "0?Set(REALCALLERIDNUM=2242384656)") in new stack
  201.     -- Executing [s@macro-user-callerid:4] Set("SIP/ds-all-00000033", "AMPUSER=") in new stack
  202.     -- Executing [s@macro-user-callerid:5] Set("SIP/ds-all-00000033", "AMPUSERCIDNAME=") in new stack
  203.     -- Executing [s@macro-user-callerid:6] GotoIf("SIP/ds-all-00000033", "1?report") in new stack
  204.     -- Goto (macro-user-callerid,s,9)
  205.     -- Executing [s@macro-user-callerid:9] GotoIf("SIP/ds-all-00000033", "1?continue") in new stack
  206.     -- Goto (macro-user-callerid,s,18)
  207.     -- Executing [s@macro-user-callerid:18] NoOp("SIP/ds-all-00000033", "Using CallerID "2242384656" <2242384656>") in new stack
  208.     -- Executing [s@macro-vm:2] Set("SIP/ds-all-00000033", "VMGAIN=""") in new stack
  209.     -- Executing [s@macro-vm:3] GotoIf("SIP/ds-all-00000033", "1?vmx,1") in new stack
  210.     -- Goto (macro-vm,vmx,1)
  211.     -- Executing [vmx@macro-vm:1] Set("SIP/ds-all-00000033", "MEXTEN=101") in new stack
  212.     -- Executing [vmx@macro-vm:2] Set("SIP/ds-all-00000033", "MMODE=CHANUNAVAIL") in new stack
  213.     -- Executing [vmx@macro-vm:3] Set("SIP/ds-all-00000033", "RETVM=") in new stack
  214.     -- Executing [vmx@macro-vm:4] Set("SIP/ds-all-00000033", "MODE=unavail") in new stack
  215.     -- Executing [vmx@macro-vm:5] GotoIf("SIP/ds-all-00000033", "1?chknomsg") in new stack
  216.     -- Goto (macro-vm,vmx,7)
  217.     -- Executing [vmx@macro-vm:7] GotoIf("SIP/ds-all-00000033", "0?s-CHANUNAVAIL,1") in new stack
  218.     -- Executing [vmx@macro-vm:8] GotoIf("SIP/ds-all-00000033", "1?notdirect") in new stack
  219.     -- Goto (macro-vm,vmx,10)
  220.     -- Executing [vmx@macro-vm:10] NoOp("SIP/ds-all-00000033", "Checking if ext 101 is enabled: ") in new stack
  221.     -- Executing [vmx@macro-vm:11] GotoIf("SIP/ds-all-00000033", "1?s-CHANUNAVAIL,1") in new stack
  222.     -- Goto (macro-vm,s-CHANUNAVAIL,1)
  223.     -- Executing [s-CHANUNAVAIL@macro-vm:1] Macro("SIP/ds-all-00000033", "get-vmcontext,101") in new stack
  224.     -- Executing [s@macro-get-vmcontext:1] Set("SIP/ds-all-00000033", "VMCONTEXT=default") in new stack
  225.     -- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/ds-all-00000033", "0?200:300") in new stack
  226.     -- Goto (macro-get-vmcontext,s,300)
  227.     -- Executing [s@macro-get-vmcontext:300] NoOp("SIP/ds-all-00000033", "") in new stack
  228.     -- Executing [s-CHANUNAVAIL@macro-vm:2] VoiceMail("SIP/ds-all-00000033", "101@default,u") in new stack
  229. Audio is at 64.105.229.19 port 10056
  230. Adding codec 0x4 (ulaw) to SDP
  231. Adding non-codec 0x1 (telephone-event) to SDP
  232.  
  233. <--- Reliably Transmitting (NAT) to 74.222.60.237:5060 --->
  234. SIP/2.0 200 OK
  235. Via: SIP/2.0/UDP 74.222.60.237:5060;branch=z9hG4bK3442d6df;received=74.222.60.237;rport=5060
  236. From: "2242384656" <sip:2242384656@74.222.60.237>;tag=as58ce53bb
  237. To: <sip:101@64.105.229.19>;tag=as1c634fcf
  238. Call-ID: 2b63dfcf2dff7e707755a516582175a1@74.222.60.237
  239. CSeq: 102 INVITE
  240. Server: Asterisk PBX 1.6.2.18
  241. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  242. Supported: replaces, timer
  243. Contact: <sip:101@64.105.229.19>
  244. Content-Type: application/sdp
  245. Content-Length: 239
  246.  
  247. v=0
  248. o=root 1471446729 1471446729 IN IP4 64.105.229.19
  249. s=Asterisk PBX 1.6.2.18
  250. c=IN IP4 64.105.229.19
  251. t=0 0
  252. m=audio 10056 RTP/AVP 0 101
  253. a=rtpmap:0 PCMU/8000
  254. a=rtpmap:101 telephone-event/8000
  255. a=fmtp:101 0-16
  256. a=ptime:20
  257. a=sendrecv
  258.  
  259. <------------>
  260.     -- <SIP/ds-all-00000033> Playing 'vm-theperson.ulaw' (language 'en')
  261. Retransmitting #1 (NAT) to 74.222.60.237:5060:
  262. SIP/2.0 200 OK
  263. Via: SIP/2.0/UDP 74.222.60.237:5060;branch=z9hG4bK3442d6df;received=74.222.60.237;rport=5060
  264. From: "2242384656" <sip:2242384656@74.222.60.237>;tag=as58ce53bb
  265. To: <sip:101@64.105.229.19>;tag=as1c634fcf
  266. Call-ID: 2b63dfcf2dff7e707755a516582175a1@74.222.60.237
  267. CSeq: 102 INVITE
  268. Server: Asterisk PBX 1.6.2.18
  269. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  270. Supported: replaces, timer
  271. Contact: <sip:101@64.105.229.19>
  272. ontent-Type: application/sdp
  273. Content-Length: 239
  274.  
  275. v=0
  276. o=root 1471446729 1471446729 IN IP4 64.105.229.19
  277. s=Asterisk PBX 1.6.2.18
  278. c=IN IP4 64.105.229.19
  279. t=0 0
  280. m=audio 10056 RTP/AVP 0 101
  281. a=rtpmap:0 PCMU/8000
  282. a=rtpmap:101 telephone-event/8000
  283. a=fmtp:101 0-16
  284. a=ptime:20
  285. a=sendrecv
  286.  
  287. ---
  288. Retransmitting #2 (NAT) to 74.222.60.237:5060:
  289. SIP/2.0 200 OK
  290. Via: SIP/2.0/UDP 74.222.60.237:5060;branch=z9hG4bK3442d6df;received=74.222.60.237;rport=5060
  291. From: "2242384656" <sip:2242384656@74.222.60.237>;tag=as58ce53bb
  292. To: <sip:101@64.105.229.19>;tag=as1c634fcf
  293. Call-ID: 2b63dfcf2dff7e707755a516582175a1@74.222.60.237
  294. CSeq: 102 INVITE
  295. Server: Asterisk PBX 1.6.2.18
  296. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  297. Supported: replaces, timer
  298. Contact: <sip:101@64.105.229.19>
  299. ontent-Type: application/sdp
  300. Content-Length: 239
  301.  
  302. v=0
  303. o=root 1471446729 1471446729 IN IP4 64.105.229.19
  304. s=Asterisk PBX 1.6.2.18
  305. c=IN IP4 64.105.229.19
  306. t=0 0
  307. m=audio 10056 RTP/AVP 0 101
  308. a=rtpmap:0 PCMU/8000
  309. a=rtpmap:101 telephone-event/8000
  310. a=fmtp:101 0-16
  311. a=ptime:20
  312. a=sendrecv
  313.  
  314. ---
  315.     -- <SIP/ds-all-00000033> Playing 'digits/1.ulaw' (language 'en')
  316.  
  317. <--- SIP read from UDP:67.53.0.131:11961 --->
  318. REGISTER sip:64.105.229.19:5060 SIP/2.0
  319. Via: SIP/2.0/UDP 192.168.155.113:64232;branch=z9hG4bK-d8754z-3f6ae63e830b1a2b-1---d8754z-;rport
  320. Max-Forwards: 70
  321. Contact: <sip:101@67.53.0.131:11961;transport=UDP;rinstance=7c739af14cb9d189>
  322. To: <sip:101@64.105.229.19:5060>
  323. From: <sip:101@64.105.229.19:5060>;tag=be2ce406
  324. Call-ID: NmMzMDQ4NWNjM2FkOGEyYzMzMjQ2MjEwZmY4Y2VhZDA.
  325. CSeq: 929 REGISTER
  326. Expires: 120
  327. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
  328. Supported: replaces
  329. User-Agent: 3CXPhone 6.0.19920.0
  330. Authorization: Digest username="101",realm="asterisk",nonce="6cc9c7e6",uri="sip:64.105.229.19:5060",response="318b6dd6247d6799d06ae1c034bdf1e9",algorithm=MD5
  331. Content-Length: 0
  332.  
  333.  
  334. <------------->
  335. --- (14 headers 0 lines) ---
  336. Sending to 67.53.0.131 : 11961 (NAT)
  337.  
  338. <--- Transmitting (NAT) to 67.53.0.131:11961 --->
  339. SIP/2.0 401 Unauthorized
  340. Via: SIP/2.0/UDP 192.168.155.113:64232;branch=z9hG4bK-d8754z-3f6ae63e830b1a2b-1---d8754z-;received=67.53.0.131;rport=11961
  341. From: <sip:101@64.105.229.19:5060>;tag=be2ce406
  342. To: <sip:101@64.105.229.19:5060>;tag=as21f7f11e
  343. Call-ID: NmMzMDQ4NWNjM2FkOGEyYzMzMjQ2MjEwZmY4Y2VhZDA.
  344. CSeq: 929 REGISTER
  345. Server: Asterisk PBX 1.6.2.18
  346. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  347. Supported: replaces, timer
  348. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ffabb99", stale=true
  349. Content-Length: 0
  350.  
  351.  
  352. <------------>
  353. Scheduling destruction of SIP dialog 'NmMzMDQ4NWNjM2FkOGEyYzMzMjQ2MjEwZmY4Y2VhZDA.' in 32000 ms (Method: REGISTER)
  354.     -- <SIP/ds-all-00000033> Playing 'digits/0.ulaw' (language 'en')
  355.     -- <SIP/ds-all-00000033> Playing 'digits/1.ulaw' (language 'en')
  356. Retransmitting #3 (NAT) to 74.222.60.237:5060:
  357. SIP/2.0 200 OK
  358. Via: SIP/2.0/UDP 74.222.60.237:5060;branch=z9hG4bK3442d6df;received=74.222.60.237;rport=5060
  359. From: "2242384656" <sip:2242384656@74.222.60.237>;tag=as58ce53bb
  360. To: <sip:101@64.105.229.19>;tag=as1c634fcf
  361. Call-ID: 2b63dfcf2dff7e707755a516582175a1@74.222.60.237
  362. CSeq: 102 INVITE
  363. Server: Asterisk PBX 1.6.2.18
  364. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  365. Supported: replaces, timer
  366. Contact: <sip:101@64.105.229.19>
  367. ontent-Type: application/sdp
  368. Content-Length: 239
  369.  
  370. v=0
  371. o=root 1471446729 1471446729 IN IP4 64.105.229.19
  372. s=Asterisk PBX 1.6.2.18
  373. c=IN IP4 64.105.229.19
  374. t=0 0
  375. m=audio 10056 RTP/AVP 0 101
  376. a=rtpmap:0 PCMU/8000
  377. a=rtpmap:101 telephone-event/8000
  378. a=fmtp:101 0-16
  379. a=ptime:20
  380. a=sendrecv
  381.  
  382. ---
  383.     -- <SIP/ds-all-00000033> Playing 'vm-isunavail.ulaw' (language 'en')
  384.     -- <SIP/ds-all-00000033> Playing 'vm-intro.ulaw' (language 'en')
  385. ipiphony*CLI>
  386.  
  387.  
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