sip call

From scheder, 9 Months ago, written in Plain Text, viewed 3 times. This paste will run down the curtain in 1 Second.
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  1. Connected to Asterisk 13.18.4 currently running on freepbx (pid = 29653)
  2. freepbx*CLI> sip set debug on
  3. SIP Debugging re-enabled
  4.  
  5. <--- SIP read from UDP:192.168.10.53:5065 --->
  6. INVITE sip:218604000023@192.168.10.248;transport=UDP SIP/2.0
  7. Via: SIP/2.0/UDP 192.168.10.53:5065;branch=z9hG4bK-524287-1---475279bd37c91357
  8. Max-Forwards: 70
  9. Contact: <sip:113@192.168.10.53:5065;transport=UDP>
  10. To: <sip:218604000023@192.168.10.248;transport=UDP>
  11. From: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  12. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  13. CSeq: 1 INVITE
  14. Content-Type: application/sdp
  15. User-Agent: Z 3.14.38765 rv2.8.3
  16. Allow-Events: presence, kpml, talk
  17. Content-Length: 212
  18.  
  19. v=0
  20. o=Z 0 0 IN IP4 192.168.10.53
  21. s=Z
  22. c=IN IP4 192.168.10.53
  23. t=0 0
  24. m=audio 8000 RTP/AVP 18 3 0 101
  25. a=rtpmap:18 G729/8000
  26. a=fmtp:18 annexb=no
  27. a=rtpmap:101 telephone-event/8000
  28. a=fmtp:101 0-16
  29. a=sendrecv
  30. <------------->
  31. --- (12 headers 11 lines) ---
  32. Sending to 192.168.10.53:5065 (NAT)
  33. Sending to 192.168.10.53:5065 (NAT)
  34. Using INVITE request as basis request - x3keCHm36L7mGvHC4cavfQ..
  35. Found peer '113' for '113' from 192.168.10.53:5065
  36.  
  37. <--- Reliably Transmitting (no NAT) to 192.168.10.53:5065 --->
  38. SIP/2.0 401 Unauthorized
  39. Via: SIP/2.0/UDP 192.168.10.53:5065;branch=z9hG4bK-524287-1---475279bd37c91357;received=192.168.1053
  40. From: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  41. To: <sip:218604000023@192.168.10.248;transport=UDP>;tag=as0fd456d0
  42. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  43. CSeq: 1 INVITE
  44. Server: FPBX-14.0.3.2(13.18.4)
  45. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  46. Supported: replaces, timer
  47. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17cd8bdb"
  48. Content-Length: 0
  49.  
  50.  
  51. <------------>
  52. Scheduling destruction of SIP dialog 'x3keCHm36L7mGvHC4cavfQ..' in 6400 ms (Method: INVITE)
  53.  
  54. <--- SIP read from UDP:192.168.10.53:5065 --->
  55. ACK sip:218604000023@192.168.10.248;transport=UDP SIP/2.0
  56. Via: SIP/2.0/UDP 192.168.10.53:5065;branch=z9hG4bK-524287-1---475279bd37c91357
  57. Max-Forwards: 70
  58. To: <sip:218604000023@192.168.10.248;transport=UDP>;tag=as0fd456d0
  59. From: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  60. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  61. CSeq: 1 ACK
  62. Content-Length: 0
  63.  
  64. <------------->
  65. --- (8 headers 0 lines) ---
  66.  
  67. <--- SIP read from UDP:192.168.10.53:5065 --->
  68. INVITE sip:218604000023@192.168.10.248;transport=UDP SIP/2.0
  69. Via: SIP/2.0/UDP 192.168.10.53:5065;branch=z9hG4bK-524287-1---d0741c0557995556
  70. Max-Forwards: 70
  71. Contact: <sip:113@192.168.10.53:5065;transport=UDP>
  72. To: <sip:218604000023@192.168.10.248;transport=UDP>
  73. From: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  74. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  75. CSeq: 2 INVITE
  76. Content-Type: application/sdp
  77. User-Agent: Z 3.14.38765 rv2.8.3
  78. Authorization: Digest username="113",realm="asterisk",nonce="17cd8bdb",uri="sip:218604000023@192.18.10.248;transport=UDP",response="d7217b42c860065973bf91467e33605f",algorithm=MD5
  79. Allow-Events: presence, kpml, talk
  80. Content-Length: 212
  81.  
  82. v=0
  83. o=Z 0 0 IN IP4 192.168.10.53
  84. s=Z
  85. c=IN IP4 192.168.10.53
  86. t=0 0
  87. m=audio 8000 RTP/AVP 18 3 0 101
  88. a=rtpmap:18 G729/8000
  89. a=fmtp:18 annexb=no
  90. a=rtpmap:101 telephone-event/8000
  91. a=fmtp:101 0-16
  92. a=sendrecv
  93. <------------->
  94. --- (13 headers 11 lines) ---
  95. Sending to 192.168.10.53:5065 (no NAT)
  96. Using INVITE request as basis request - x3keCHm36L7mGvHC4cavfQ..
  97. Found peer '113' for '113' from 192.168.10.53:5065
  98.   == Using SIP VIDEO TOS bits 136
  99.   == Using SIP VIDEO CoS mark 6
  100.   == Using SIP RTP TOS bits 184
  101.   == Using SIP RTP CoS mark 5
  102. Found RTP audio format 18
  103. Found RTP audio format 3
  104. Found RTP audio format 0
  105. Found RTP audio format 101
  106. Found audio description format G729 for ID 18
  107. Found audio description format telephone-event for ID 101
  108. Capabilities: us - (ulaw|alaw|gsm|g726|g722|h264|mpeg4|h263), peer - audio=(ulaw|gsm|g729)/video=(othing)/text=(nothing), combined - (ulaw|gsm)
  109. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combine - 0x1 (telephone-event|)
  110.        > 0x7f1c74c0fcb0 -- Strict RTP learning after remote address set to: 192.168.10.53:8000
  111. Peer audio RTP is at port 192.168.10.53:8000
  112. Peer doesn't provide video
  113. Looking for 218604000023 in from-internal (domain 192.168.10.248)
  114. sip_route_dump: route/path hop: <sip:113@192.168.10.53:5065;transport=UDP>
  115.  
  116. <--- Transmitting (no NAT) to 192.168.10.53:5065 --->
  117. SIP/2.0 100 Trying
  118. Via: SIP/2.0/UDP 192.168.10.53:5065;branch=z9hG4bK-524287-1---d0741c0557995556;received=192.168.1053
  119. From: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  120. To: <sip:218604000023@192.168.10.248;transport=UDP>
  121. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  122. CSeq: 2 INVITE
  123. Server: FPBX-14.0.3.2(13.18.4)
  124. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  125. Supported: replaces, timer
  126. Contact: <sip:218604000023@192.168.10.248:5060>
  127. Content-Length: 0
  128.  
  129.  
  130. <------------>
  131.     -- Executing [218604000023@from-internal:1] Macro("SIP/113-00000054", "user-callerid,LIMIT,EXTRNAL,") in new stack
  132.     -- Executing [s@macro-user-callerid:1] Set("SIP/113-00000054", "TOUCH_MONITOR=1527196458.465")in new stack
  133.     -- Executing [s@macro-user-callerid:2] Set("SIP/113-00000054", "AMPUSER=113") in new stack
  134.     -- Executing [s@macro-user-callerid:3] GotoIf("SIP/113-00000054", "0?report") in new stack
  135.     -- Executing [s@macro-user-callerid:4] ExecIf("SIP/113-00000054", "1?Set(REALCALLERIDNUM=113)" in new stack
  136.     -- Executing [s@macro-user-callerid:5] Set("SIP/113-00000054", "AMPUSER=113") in new stack
  137.     -- Executing [s@macro-user-callerid:6] GotoIf("SIP/113-00000054", "0?limit") in new stack
  138.     -- Executing [s@macro-user-callerid:7] Set("SIP/113-00000054", "AMPUSERCIDNAME=Roy") in new stck
  139.     -- Executing [s@macro-user-callerid:8] ExecIf("SIP/113-00000054", "0?Set(__CIDMASQUERADING=TRU)") in new stack
  140.     -- Executing [s@macro-user-callerid:9] GotoIf("SIP/113-00000054", "0?report") in new stack
  141.     -- Executing [s@macro-user-callerid:10] Set("SIP/113-00000054", "AMPUSERCID=113") in new stack
  142.     -- Executing [s@macro-user-callerid:11] Set("SIP/113-00000054", "__DIAL_OPTIONS=HhTtr") in newstack
  143.     -- Executing [s@macro-user-callerid:12] Set("SIP/113-00000054", "CALLERID(all)="Roy" <113>") i new stack
  144.     -- Executing [s@macro-user-callerid:13] GotoIf("SIP/113-00000054", "0?limit") in new stack
  145.     -- Executing [s@macro-user-callerid:14] ExecIf("SIP/113-00000054", "1?Set(GROUP(concurrency_liit)=113)") in new stack
  146.     -- Executing [s@macro-user-callerid:15] ExecIf("SIP/113-00000054", "0?Set(CHANNEL(language)=)" in new stack
  147.     -- Executing [s@macro-user-callerid:16] NoOp("SIP/113-00000054", "Macro Depth is 1") in new stck
  148.     -- Executing [s@macro-user-callerid:17] GotoIf("SIP/113-00000054", "1?report2:macroerror") in ew stack
  149.     -- Goto (macro-user-callerid,s,19)
  150.     -- Executing [s@macro-user-callerid:19] GotoIf("SIP/113-00000054", "1?continue") in new stack
  151.     -- Goto (macro-user-callerid,s,37)
  152.     -- Executing [s@macro-user-callerid:37] Set("SIP/113-00000054", "CALLERID(number)=113") in newstack
  153.     -- Executing [s@macro-user-callerid:38] Set("SIP/113-00000054", "CALLERID(name)=Roy") in new sack
  154.     -- Executing [s@macro-user-callerid:39] GotoIf("SIP/113-00000054", "0?cnum") in new stack
  155.     -- Executing [s@macro-user-callerid:40] Set("SIP/113-00000054", "CDR(cnam)=Roy") in new stack
  156.     -- Executing [s@macro-user-callerid:41] Set("SIP/113-00000054", "CDR(cnum)=113") in new stack
  157.     -- Executing [s@macro-user-callerid:42] Set("SIP/113-00000054", "CHANNEL(language)=en") in newstack
  158.     -- Executing [218604000023@from-internal:2] Gosub("SIP/113-00000054", "sub-record-check,s,1(ou,218604000023,dontcare)") in new stack
  159.     -- Executing [s@sub-record-check:1] GotoIf("SIP/113-00000054", "0?initialized") in new stack
  160.     -- Executing [s@sub-record-check:2] Set("SIP/113-00000054", "__REC_STATUS=INITIALIZED") in newstack
  161.     -- Executing [s@sub-record-check:3] Set("SIP/113-00000054", "NOW=1527196458") in new stack
  162.     -- Executing [s@sub-record-check:4] Set("SIP/113-00000054", "__DAY=24") in new stack
  163.     -- Executing [s@sub-record-check:5] Set("SIP/113-00000054", "__MONTH=05") in new stack
  164.     -- Executing [s@sub-record-check:6] Set("SIP/113-00000054", "__YEAR=2018") in new stack
  165.     -- Executing [s@sub-record-check:7] Set("SIP/113-00000054", "__TIMESTR=20180524-211418") in ne stack
  166.     -- Executing [s@sub-record-check:8] Set("SIP/113-00000054", "__FROMEXTEN=113") in new stack
  167.     -- Executing [s@sub-record-check:9] Set("SIP/113-00000054", "__MON_FMT=wav") in new stack
  168.     -- Executing [s@sub-record-check:10] NoOp("SIP/113-00000054", "Recordings initialized") in newstack
  169.     -- Executing [s@sub-record-check:11] ExecIf("SIP/113-00000054", "0?Set(ARG3=dontcare)") in newstack
  170.     -- Executing [s@sub-record-check:12] Set("SIP/113-00000054", "REC_POLICY_MODE_SAVE=") in new sack
  171.     -- Executing [s@sub-record-check:13] ExecIf("SIP/113-00000054", "0?Set(REC_STATUS=NO)") in newstack
  172.     -- Executing [s@sub-record-check:14] GotoIf("SIP/113-00000054", "3?checkaction") in new stack
  173.     -- Goto (sub-record-check,s,17)
  174.     -- Executing [s@sub-record-check:17] GotoIf("SIP/113-00000054", "1?sub-record-check,out,1") innew stack
  175.     -- Goto (sub-record-check,out,1)
  176.     -- Executing [out@sub-record-check:1] NoOp("SIP/113-00000054", "Outbound Recording Check from 13 to 218604000023") in new stack
  177.     -- Executing [out@sub-record-check:2] Set("SIP/113-00000054", "RECMODE=dontcare") in new stack
  178.     -- Executing [out@sub-record-check:3] ExecIf("SIP/113-00000054", "1?Goto(routewins)") in new sack
  179.     -- Goto (sub-record-check,out,7)
  180.     -- Executing [out@sub-record-check:7] Gosub("SIP/113-00000054", "recordcheck,1(dontcare,out,21604000023)") in new stack
  181.     -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/113-00000054", "Starting recording chek against dontcare") in new stack
  182.     -- Executing [recordcheck@sub-record-check:2] Goto("SIP/113-00000054", "dontcare") in new stac
  183.     -- Goto (sub-record-check,recordcheck,3)
  184.     -- Executing [recordcheck@sub-record-check:3] Return("SIP/113-00000054", "") in new stack
  185.     -- Executing [out@sub-record-check:8] Return("SIP/113-00000054", "") in new stack
  186.     -- Executing [218604000023@from-internal:3] ExecIf("SIP/113-00000054", "0 ?Set(CDR(accountcode=)") in new stack
  187.     -- Executing [218604000023@from-internal:4] Set("SIP/113-00000054", "MOHCLASS=default") in newstack
  188.     -- Executing [218604000023@from-internal:5] Set("SIP/113-00000054", "_NODEST=") in new stack
  189.     -- Executing [218604000023@from-internal:6] Macro("SIP/113-00000054", "dialout-trunk,1,860400023,,off") in new stack
  190.     -- Executing [s@macro-dialout-trunk:1] Set("SIP/113-00000054", "DIAL_TRUNK=1") in new stack
  191.     -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/113-00000054", "0?sub-pincheck,s,1()") in ew stack
  192.     -- Executing [s@macro-dialout-trunk:3] ExecIf("SIP/113-00000054", "0?Set(CALLERID(num)=113)") n new stack
  193.     -- Executing [s@macro-dialout-trunk:4] GotoIf("SIP/113-00000054", "0?disabletrunk,1") in new sack
  194.     -- Executing [s@macro-dialout-trunk:5] Set("SIP/113-00000054", "DIAL_NUMBER=8604000023") in ne stack
  195.     -- Executing [s@macro-dialout-trunk:6] Set("SIP/113-00000054", "DIAL_TRUNK_OPTIONS=HhTtr") in ew stack
  196.     -- Executing [s@macro-dialout-trunk:7] Set("SIP/113-00000054", "OUTBOUND_GROUP=OUT_1") in new tack
  197.     -- Executing [s@macro-dialout-trunk:8] Set("SIP/113-00000054", "DIAL_TRUNK_OPTIONS=T") in new tack
  198.     -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/113-00000054", "0?nomax") in new stack
  199.     -- Executing [s@macro-dialout-trunk:10] GotoIf("SIP/113-00000054", "0?chanfull") in new stack
  200.     -- Executing [s@macro-dialout-trunk:11] GotoIf("SIP/113-00000054", "0?skipoutcid") in new stac
  201.     -- Executing [s@macro-dialout-trunk:12] Macro("SIP/113-00000054", "outbound-callerid,1") in ne stack
  202.     -- Executing [s@macro-outbound-callerid:1] NoOp("SIP/113-00000054", "113") in new stack
  203.     -- Executing [s@macro-outbound-callerid:2] NoOp("SIP/113-00000054", "") in new stack
  204.     -- Executing [s@macro-outbound-callerid:3] NoOp("SIP/113-00000054", "off") in new stack
  205.     -- Executing [s@macro-outbound-callerid:4] ExecIf("SIP/113-00000054", "0?Set(CALLERPRES(name-pes)=)") in new stack
  206.     -- Executing [s@macro-outbound-callerid:5] ExecIf("SIP/113-00000054", "0?Set(CALLERPRES(num-prs)=)") in new stack
  207.     -- Executing [s@macro-outbound-callerid:6] ExecIf("SIP/113-00000054", "0?Set(REALCALLERIDNUM=13)") in new stack
  208.     -- Executing [s@macro-outbound-callerid:7] GotoIf("SIP/113-00000054", "1?normcid") in new stac
  209.     -- Goto (macro-outbound-callerid,s,11)
  210.     -- Executing [s@macro-outbound-callerid:11] Set("SIP/113-00000054", "USEROUTCID=3434771015") i new stack
  211.     -- Executing [s@macro-outbound-callerid:12] Set("SIP/113-00000054", "EMERGENCYCID=") in new stck
  212.     -- Executing [s@macro-outbound-callerid:13] Set("SIP/113-00000054", "TRUNKOUTCID=2052553464") n new stack
  213.     -- Executing [s@macro-outbound-callerid:14] GotoIf("SIP/113-00000054", "1?trunkcid") in new stck
  214.     -- Goto (macro-outbound-callerid,s,19)
  215.     -- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/113-00000054", "1?Set(CALLERID(all)=202553464)") in new stack
  216.     -- Executing [s@macro-outbound-callerid:20] ExecIf("SIP/113-00000054", "1?Set(CALLERID(all)=344771015)") in new stack
  217.     -- Executing [s@macro-outbound-callerid:21] ExecIf("SIP/113-00000054", "0?Set(CALLERID(all)=)" in new stack
  218.     -- Executing [s@macro-outbound-callerid:22] ExecIf("SIP/113-00000054", "0?Set(CALLERPRES(name-res)=prohib_passed_screen)") in new stack
  219.     -- Executing [s@macro-outbound-callerid:23] ExecIf("SIP/113-00000054", "0?Set(CALLERPRES(num-pes)=prohib_passed_screen)") in new stack
  220.     -- Executing [s@macro-outbound-callerid:24] Set("SIP/113-00000054", "CDR(outbound_cnum)=343477015") in new stack
  221.     -- Executing [s@macro-outbound-callerid:25] Set("SIP/113-00000054", "CDR(outbound_cnam)=") in ew stack
  222.     -- Executing [s@macro-dialout-trunk:13] GosubIf("SIP/113-00000054", "0?sub-flp-1,s,1()") in ne stack
  223.     -- Executing [s@macro-dialout-trunk:14] Set("SIP/113-00000054", "OUTNUM=8604000023") in new stck
  224.     -- Executing [s@macro-dialout-trunk:15] Set("SIP/113-00000054", "custom=SIP/Jose_Trunk") in ne stack
  225.     -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/113-00000054", "0?Set(DIAL_TRUNK_OPTIONS=Msetmusic^default)T)") in new stack
  226.     -- Executing [s@macro-dialout-trunk:17] ExecIf("SIP/113-00000054", "0?Set(DIAL_TRUNK_OPTIONS=T(confirm))") in new stack
  227.     -- Executing [s@macro-dialout-trunk:18] Macro("SIP/113-00000054", "dialout-trunk-predial-hook,) in new stack
  228.     -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/113-00000054", "") in new sack
  229.     -- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/113-00000054", "0?bypass,1") in new stack
  230.     -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/113-00000054", "1?Set(CONNECTEDLINE(num,i)8604000023)") in new stack
  231.     -- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/113-00000054", "1?Set(CONNECTEDLINE(name,i=CID:3434771015)") in new stack
  232.     -- Executing [s@macro-dialout-trunk:22] ExecIf("SIP/113-00000054", "0?Set(CONNECTEDLINE(name,i=CID:(Hidden)3434771015)") in new stack
  233.     -- Executing [s@macro-dialout-trunk:23] GotoIf("SIP/113-00000054", "0?customtrunk") in new stak
  234.     -- Executing [s@macro-dialout-trunk:24] Dial("SIP/113-00000054", "SIP/Jose_Trunk/8604000023,30,T") in new stack
  235. Reliably Transmitting (no NAT) to 192.168.10.53:5065:
  236. OPTIONS sip:113@192.168.10.53:5065;rinstance=ad99fccfc44ccc18;transport=UDP SIP/2.0
  237. Via: SIP/2.0/UDP 192.168.10.248:5060;branch=z9hG4bK75d32274
  238. Max-Forwards: 70
  239. From: "Unknown" <sip:Unknown@192.168.10.248>;tag=as453c5363
  240. To: <sip:113@192.168.10.53:5065;rinstance=ad99fccfc44ccc18;transport=UDP>
  241. Contact: <sip:Unknown@192.168.10.248:5060>
  242. Call-ID: 3594cc17468f16265b03bef271a111c9@192.168.10.248:5060
  243. CSeq: 102 OPTIONS
  244. User-Agent: FPBX-14.0.3.2(13.18.4)
  245. Date: Thu, 24 May 2018 21:14:18 GMT
  246. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  247. Supported: replaces, timer
  248. Content-Length: 0
  249.  
  250.  
  251. ---
  252.   == Using SIP RTP TOS bits 184
  253.   == Using SIP RTP CoS mark 5
  254. Audio is at 12606
  255. Adding codec ulaw to SDP
  256. Adding non-codec 0x1 (telephone-event) to SDP
  257. Reliably Transmitting (NAT) to 72.9.246.170:5060:
  258. INVITE sip:8604000023@atlanta2.voip.ms SIP/2.0
  259. Via: SIP/2.0/UDP 192.168.10.248:5060;branch=z9hG4bK7d4416df;rport
  260. Max-Forwards: 70
  261. From: <sip:206996@192.168.10.248>;tag=as1259ff78
  262. To: <sip:8604000023@atlanta2.voip.ms>
  263. Contact: <sip:206996@192.168.10.248:5060>
  264. Call-ID: 5f6d47381845abd81f8d7dd70d33a367@192.168.10.248:5060
  265. CSeq: 102 INVITE
  266. User-Agent: FPBX-14.0.3.2(13.18.4)
  267. Date: Thu, 24 May 2018 21:14:18 GMT
  268. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  269. Supported: replaces, timer
  270. Remote-Party-ID: "3434771015" <sip:3434771015@192.168.10.248>;party=calling;privacy=off;screen=no
  271. Content-Type: application/sdp
  272. Content-Length: 242
  273.  
  274. v=0
  275. o=root 750590803 750590803 IN IP4 192.168.10.248
  276. s=Asterisk PBX 13.18.4
  277. c=IN IP4 192.168.10.248
  278. t=0 0
  279. m=audio 12606 RTP/AVP 0 101
  280. a=rtpmap:0 PCMU/8000
  281. a=rtpmap:101 telephone-event/8000
  282. a=fmtp:101 0-16
  283. a=maxptime:150
  284. a=sendrecv
  285.  
  286. ---
  287.     -- Called SIP/Jose_Trunk/8604000023
  288.  
  289. <--- SIP read from UDP:192.168.10.53:5065 --->
  290. SIP/2.0 200 OK
  291. Via: SIP/2.0/UDP 192.168.10.248:5060;branch=z9hG4bK75d32274
  292. Contact: <sip:192.168.10.53:5065>
  293. To: <sip:113@192.168.10.53:5065;rinstance=ad99fccfc44ccc18;transport=UDP>;tag=13098066
  294. From: "Unknown" <sip:Unknown@192.168.10.248>;tag=as453c5363
  295. Call-ID: 3594cc17468f16265b03bef271a111c9@192.168.10.248:5060
  296. CSeq: 102 OPTIONS
  297. Accept: application/sdp, application/sdp
  298. Accept-Language: en
  299. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  300. Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
  301. User-Agent: Z 3.14.38765 rv2.8.3
  302. Allow-Events: presence, kpml, talk
  303. Content-Length: 0
  304.  
  305. <------------->
  306. --- (14 headers 0 lines) ---
  307. Really destroying SIP dialog '3594cc17468f16265b03bef271a111c9@192.168.10.248:5060' Method: OPTION
  308.  
  309. <--- SIP read from UDP:72.9.246.170:5060 --->
  310. SIP/2.0 401 Unauthorized
  311. Via: SIP/2.0/UDP 192.168.10.248:5060;branch=z9hG4bK7d4416df;received=187.155.135.64;rport=40264
  312. From: <sip:206996@192.168.10.248>;tag=as1259ff78
  313. To: <sip:8604000023@atlanta2.voip.ms>;tag=as3c1afee5
  314. Call-ID: 5f6d47381845abd81f8d7dd70d33a367@192.168.10.248:5060
  315. CSeq: 102 INVITE
  316. Server: voip.ms
  317. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  318. Supported: replaces, timer
  319. WWW-Authenticate: Digest algorithm=MD5, realm="atlanta2.voip.ms", nonce="1ef44647"
  320. Content-Length: 0
  321.  
  322. <------------->
  323. --- (11 headers 0 lines) ---
  324. Transmitting (NAT) to 72.9.246.170:5060:
  325. ACK sip:8604000023@atlanta2.voip.ms SIP/2.0
  326. Via: SIP/2.0/UDP 192.168.10.248:5060;branch=z9hG4bK7d4416df;rport
  327. Max-Forwards: 70
  328. From: <sip:206996@192.168.10.248>;tag=as1259ff78
  329. To: <sip:8604000023@atlanta2.voip.ms>;tag=as3c1afee5
  330. Contact: <sip:206996@192.168.10.248:5060>
  331. Call-ID: 5f6d47381845abd81f8d7dd70d33a367@192.168.10.248:5060
  332. CSeq: 102 ACK
  333. User-Agent: FPBX-14.0.3.2(13.18.4)
  334. Content-Length: 0
  335.  
  336.  
  337. ---
  338. Audio is at 12606
  339. Adding codec ulaw to SDP
  340. Adding non-codec 0x1 (telephone-event) to SDP
  341. Reliably Transmitting (NAT) to 72.9.246.170:5060:
  342. INVITE sip:8604000023@atlanta2.voip.ms SIP/2.0
  343. Via: SIP/2.0/UDP 192.168.10.248:5060;branch=z9hG4bK2cf2a33b;rport
  344. Max-Forwards: 70
  345. From: <sip:206996@192.168.10.248>;tag=as1259ff78
  346. To: <sip:8604000023@atlanta2.voip.ms>
  347. Contact: <sip:206996@192.168.10.248:5060>
  348. Call-ID: 5f6d47381845abd81f8d7dd70d33a367@192.168.10.248:5060
  349. CSeq: 103 INVITE
  350. User-Agent: FPBX-14.0.3.2(13.18.4)
  351. Authorization: Digest username="206996", realm="atlanta2.voip.ms", algorithm=MD5, uri="sip:860400023@atlanta2.voip.ms", nonce="1ef44647", response="2e635547c53f43dab12f9a52b0057c49"
  352. Date: Thu, 24 May 2018 21:14:18 GMT
  353. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  354. Supported: replaces, timer
  355. Remote-Party-ID: "3434771015" <sip:3434771015@192.168.10.248>;party=calling;privacy=off;screen=no
  356. Content-Type: application/sdp
  357. Content-Length: 242
  358.  
  359. v=0
  360. o=root 750590803 750590804 IN IP4 192.168.10.248
  361. s=Asterisk PBX 13.18.4
  362. c=IN IP4 192.168.10.248
  363. t=0 0
  364. m=audio 12606 RTP/AVP 0 101
  365. a=rtpmap:0 PCMU/8000
  366. a=rtpmap:101 telephone-event/8000
  367. a=fmtp:101 0-16
  368. a=maxptime:150
  369. a=sendrecv
  370.  
  371. ---
  372.  
  373. <--- SIP read from UDP:72.9.246.170:5060 --->
  374. SIP/2.0 100 Trying
  375. Via: SIP/2.0/UDP 192.168.10.248:5060;branch=z9hG4bK2cf2a33b;received=187.155.135.64;rport=40264
  376. From: <sip:206996@192.168.10.248>;tag=as1259ff78
  377. To: <sip:8604000023@atlanta2.voip.ms>
  378. Call-ID: 5f6d47381845abd81f8d7dd70d33a367@192.168.10.248:5060
  379. CSeq: 103 INVITE
  380. Server: voip.ms
  381. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  382. Supported: replaces, timer
  383. Session-Expires: 1800;refresher=uas
  384. Contact: <sip:8604000023@72.9.246.170:5060>
  385. Content-Length: 0
  386.  
  387. <------------->
  388. --- (12 headers 0 lines) ---
  389.  
  390. <--- SIP read from UDP:72.9.246.170:5060 --->
  391. SIP/2.0 180 Ringing
  392. Via: SIP/2.0/UDP 192.168.10.248:5060;branch=z9hG4bK2cf2a33b;received=187.155.135.64;rport=40264
  393. From: <sip:206996@192.168.10.248>;tag=as1259ff78
  394. To: <sip:8604000023@atlanta2.voip.ms>;tag=as191b7847
  395. Call-ID: 5f6d47381845abd81f8d7dd70d33a367@192.168.10.248:5060
  396. CSeq: 103 INVITE
  397. Server: voip.ms
  398. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  399. Supported: replaces, timer
  400. Session-Expires: 1800;refresher=uas
  401. Contact: <sip:8604000023@72.9.246.170:5060>
  402. Content-Length: 0
  403.  
  404. <------------->
  405. --- (12 headers 0 lines) ---
  406. sip_route_dump: route/path hop: <sip:8604000023@72.9.246.170:5060>
  407.     -- SIP/Jose_Trunk-00000055 is ringing
  408.  
  409. <--- Transmitting (no NAT) to 192.168.10.53:5065 --->
  410. SIP/2.0 180 Ringing
  411. Via: SIP/2.0/UDP 192.168.10.53:5065;branch=z9hG4bK-524287-1---d0741c0557995556;received=192.168.1053
  412. From: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  413. To: <sip:218604000023@192.168.10.248;transport=UDP>;tag=as3d890bea
  414. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  415. CSeq: 2 INVITE
  416. Server: FPBX-14.0.3.2(13.18.4)
  417. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  418. Supported: replaces, timer
  419. Contact: <sip:218604000023@192.168.10.248:5060>
  420. Content-Length: 0
  421.  
  422.  
  423. <------------>
  424.  
  425. <--- SIP read from UDP:72.9.246.170:5060 --->
  426. SIP/2.0 200 OK
  427. Via: SIP/2.0/UDP 192.168.10.248:5060;branch=z9hG4bK2cf2a33b;received=187.155.135.64;rport=40264
  428. From: <sip:206996@192.168.10.248>;tag=as1259ff78
  429. To: <sip:8604000023@atlanta2.voip.ms>;tag=as191b7847
  430. Call-ID: 5f6d47381845abd81f8d7dd70d33a367@192.168.10.248:5060
  431. CSeq: 103 INVITE
  432. Server: voip.ms
  433. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  434. Supported: replaces, timer
  435. Session-Expires: 1800;refresher=uas
  436. Contact: <sip:8604000023@72.9.246.170:5060>
  437. Content-Type: application/sdp
  438. Require: timer
  439. Content-Length: 223
  440.  
  441. v=0
  442. o=root 1926909035 1926909035 IN IP4 72.9.246.170
  443. s=voip.ms
  444. c=IN IP4 72.9.246.170
  445. t=0 0
  446. m=audio 18146 RTP/AVP 0 101
  447. a=rtpmap:0 PCMU/8000
  448. a=rtpmap:101 telephone-event/8000
  449. a=fmtp:101 0-16
  450. a=ptime:20
  451. a=sendrecv
  452. <------------->
  453. --- (14 headers 11 lines) ---
  454. Found RTP audio format 0
  455. Found RTP audio format 101
  456. Found audio description format PCMU for ID 0
  457. Found audio description format telephone-event for ID 101
  458. Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
  459. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combine - 0x1 (telephone-event|)
  460.        > 0x7f1c8406f7c0 -- Strict RTP learning after remote address set to: 72.9.246.170:18146
  461. Peer audio RTP is at port 72.9.246.170:18146
  462. sip_route_dump: route/path hop: <sip:8604000023@72.9.246.170:5060>
  463. Transmitting (NAT) to 72.9.246.170:5060:
  464. ACK sip:8604000023@72.9.246.170:5060 SIP/2.0
  465. Via: SIP/2.0/UDP 192.168.10.248:5060;branch=z9hG4bK6b01ed1f;rport
  466. Max-Forwards: 70
  467. From: <sip:206996@192.168.10.248>;tag=as1259ff78
  468. To: <sip:8604000023@atlanta2.voip.ms>;tag=as191b7847
  469. Contact: <sip:206996@192.168.10.248:5060>
  470. Call-ID: 5f6d47381845abd81f8d7dd70d33a367@192.168.10.248:5060
  471. CSeq: 103 ACK
  472. User-Agent: FPBX-14.0.3.2(13.18.4)
  473. Content-Length: 0
  474.  
  475.  
  476. ---
  477.     -- SIP/Jose_Trunk-00000055 answered SIP/113-00000054
  478. Audio is at 13382
  479. Adding codec ulaw to SDP
  480. Adding codec gsm to SDP
  481. Adding non-codec 0x1 (telephone-event) to SDP
  482.  
  483. <--- Reliably Transmitting (no NAT) to 192.168.10.53:5065 --->
  484. SIP/2.0 200 OK
  485. Via: SIP/2.0/UDP 192.168.10.53:5065;branch=z9hG4bK-524287-1---d0741c0557995556;received=192.168.1053
  486. From: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  487. To: <sip:218604000023@192.168.10.248;transport=UDP>;tag=as3d890bea
  488. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  489. CSeq: 2 INVITE
  490. Server: FPBX-14.0.3.2(13.18.4)
  491. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  492. Supported: replaces, timer
  493. Contact: <sip:218604000023@192.168.10.248:5060>
  494. P-Asserted-Identity: "CID:3434771015" <sip:8604000023@192.168.10.248>
  495. Content-Type: application/sdp
  496. Content-Length: 265
  497.  
  498. v=0
  499. o=root 397966207 397966207 IN IP4 192.168.10.248
  500. s=Asterisk PBX 13.18.4
  501. c=IN IP4 192.168.10.248
  502. t=0 0
  503. m=audio 13382 RTP/AVP 0 3 101
  504. a=rtpmap:0 PCMU/8000
  505. a=rtpmap:3 GSM/8000
  506. a=rtpmap:101 telephone-event/8000
  507. a=fmtp:101 0-16
  508. a=maxptime:150
  509. a=sendrecv
  510.  
  511. <------------>
  512.     -- Channel SIP/Jose_Trunk-00000055 joined 'simple_bridge' basic-bridge <b951bb83-6ca8-4b6d-bc9-ca233e2d4234>
  513.     -- Channel SIP/113-00000054 joined 'simple_bridge' basic-bridge <b951bb83-6ca8-4b6d-bc94-ca2332d4234>
  514.        > 0x7f1c74c0fcb0 -- Strict RTP switching to RTP target address 192.168.10.53:8000 as source
  515.  
  516. <--- SIP read from UDP:192.168.10.53:5065 --->
  517. ACK sip:218604000023@192.168.10.248:5060 SIP/2.0
  518. Via: SIP/2.0/UDP 192.168.10.53:5065;branch=z9hG4bK-524287-1---40a5543a1e282325
  519. Max-Forwards: 70
  520. Contact: <sip:113@192.168.10.53:5065;transport=UDP>
  521. To: <sip:218604000023@192.168.10.248;transport=UDP>;tag=as3d890bea
  522. From: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  523. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  524. CSeq: 2 ACK
  525. User-Agent: Z 3.14.38765 rv2.8.3
  526. Content-Length: 0
  527.  
  528. <------------->
  529. --- (10 headers 0 lines) ---
  530.  
  531. <--- SIP read from UDP:192.168.10.53:5065 --->
  532. INVITE sip:218604000023@192.168.10.248:5060 SIP/2.0
  533. Via: SIP/2.0/UDP 192.168.10.53:5065;branch=z9hG4bK-524287-1---de02db6ad3deea33
  534. Max-Forwards: 70
  535. Contact: <sip:113@192.168.10.53:5065;transport=UDP>
  536. To: <sip:218604000023@192.168.10.248;transport=UDP>;tag=as3d890bea
  537. From: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  538. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  539. CSeq: 3 INVITE
  540. Content-Type: application/sdp
  541. User-Agent: Z 3.14.38765 rv2.8.3
  542. Authorization: Digest username="113",realm="asterisk",nonce="17cd8bdb",uri="sip:218604000023@192.18.10.248:5060",response="ea9e810998abdbb4d01b8dd5b3afed6d",algorithm=MD5
  543. Allow-Events: presence, kpml, talk
  544. Content-Length: 212
  545.  
  546. v=0
  547. o=Z 0 2 IN IP4 192.168.10.53
  548. s=Z
  549. c=IN IP4 192.168.10.53
  550. t=0 0
  551. m=audio 8000 RTP/AVP 18 3 0 101
  552. a=rtpmap:18 G729/8000
  553. a=fmtp:18 annexb=no
  554. a=rtpmap:101 telephone-event/8000
  555. a=fmtp:101 0-16
  556. a=inactive
  557. <------------->
  558. --- (13 headers 11 lines) ---
  559. Sending to 192.168.10.53:5065 (no NAT)
  560. Found RTP audio format 18
  561. Found RTP audio format 3
  562. Found RTP audio format 0
  563. Found RTP audio format 101
  564. Found audio description format G729 for ID 18
  565. Found audio description format telephone-event for ID 101
  566. Capabilities: us - (ulaw|alaw|gsm|g726|g722|h264|mpeg4|h263), peer - audio=(ulaw|gsm|g729)/video=(othing)/text=(nothing), combined - (ulaw|gsm)
  567. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combine - 0x1 (telephone-event|)
  568. Peer audio RTP is at port 192.168.10.53:8000
  569.  
  570. <--- Transmitting (no NAT) to 192.168.10.53:5065 --->
  571. SIP/2.0 100 Trying
  572. Via: SIP/2.0/UDP 192.168.10.53:5065;branch=z9hG4bK-524287-1---de02db6ad3deea33;received=192.168.1053
  573. From: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  574. To: <sip:218604000023@192.168.10.248;transport=UDP>;tag=as3d890bea
  575. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  576. CSeq: 3 INVITE
  577. Server: FPBX-14.0.3.2(13.18.4)
  578. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  579. Supported: replaces, timer
  580. Contact: <sip:218604000023@192.168.10.248:5060>
  581. Content-Length: 0
  582.  
  583.  
  584. <------------>
  585. Audio is at 13382
  586. Adding codec ulaw to SDP
  587. Adding codec gsm to SDP
  588. Adding non-codec 0x1 (telephone-event) to SDP
  589.  
  590. <--- Reliably Transmitting (no NAT) to 192.168.10.53:5065 --->
  591. SIP/2.0 200 OK
  592. Via: SIP/2.0/UDP 192.168.10.53:5065;branch=z9hG4bK-524287-1---de02db6ad3deea33;received=192.168.1053
  593. From: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  594. To: <sip:218604000023@192.168.10.248;transport=UDP>;tag=as3d890bea
  595. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  596. CSeq: 3 INVITE
  597. Server: FPBX-14.0.3.2(13.18.4)
  598. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  599. Supported: replaces, timer
  600. Contact: <sip:218604000023@192.168.10.248:5060>
  601. Content-Type: application/sdp
  602. Content-Length: 265
  603.  
  604. v=0
  605. o=root 397966207 397966208 IN IP4 192.168.10.248
  606. s=Asterisk PBX 13.18.4
  607. c=IN IP4 192.168.10.248
  608. t=0 0
  609. m=audio 13382 RTP/AVP 0 3 101
  610. a=rtpmap:0 PCMU/8000
  611. a=rtpmap:3 GSM/8000
  612. a=rtpmap:101 telephone-event/8000
  613. a=fmtp:101 0-16
  614. a=maxptime:150
  615. a=inactive
  616.  
  617. <------------>
  618.     -- Started music on hold, class 'default', on channel 'SIP/Jose_Trunk-00000055'
  619.  
  620. <--- SIP read from UDP:192.168.10.53:5065 --->
  621. ACK sip:218604000023@192.168.10.248:5060 SIP/2.0
  622. Via: SIP/2.0/UDP 192.168.10.53:5065;branch=z9hG4bK-524287-1---75c697380c676378
  623. Max-Forwards: 70
  624. Contact: <sip:113@192.168.10.53:5065;transport=UDP>
  625. To: <sip:218604000023@192.168.10.248;transport=UDP>;tag=as3d890bea
  626. From: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  627. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  628. CSeq: 3 ACK
  629. User-Agent: Z 3.14.38765 rv2.8.3
  630. Content-Length: 0
  631.  
  632. <------------->
  633. --- (10 headers 0 lines) ---
  634.        > 0x7f1c8406f7c0 -- Strict RTP switching to RTP target address 72.9.246.170:18146 as source
  635.        > 0x7f1c8406f7c0 -- Strict RTP learning complete - Locking on source address 72.9.246.170:1146
  636.  
  637. <--- SIP read from UDP:72.9.246.170:5060 --->
  638. BYE sip:206996@192.168.10.248:5060 SIP/2.0
  639. Via: SIP/2.0/UDP 72.9.246.170:5060;branch=z9hG4bK1f4ab9e3;rport
  640. Max-Forwards: 70
  641. From: <sip:8604000023@atlanta2.voip.ms>;tag=as191b7847
  642. To: <sip:206996@192.168.10.248>;tag=as1259ff78
  643. Call-ID: 5f6d47381845abd81f8d7dd70d33a367@192.168.10.248:5060
  644. CSeq: 102 BYE
  645. User-Agent: voip.ms
  646. Proxy-Authorization: Digest username="206996", realm="atlanta2.voip.ms", algorithm=MD5, uri="sip:alanta2.voip.ms", nonce="1ef44647", response="827d9d0b0b036b13044c198f234107d9"
  647. X-Asterisk-HangupCause: Normal Clearing
  648. X-Asterisk-HangupCauseCode: 16
  649. Content-Length: 0
  650.  
  651. <------------->
  652. --- (12 headers 0 lines) ---
  653. Sending to 72.9.246.170:5060 (NAT)
  654. Scheduling destruction of SIP dialog '5f6d47381845abd81f8d7dd70d33a367@192.168.10.248:5060' in 640 ms (Method: BYE)
  655.  
  656. <--- Transmitting (NAT) to 72.9.246.170:5060 --->
  657. SIP/2.0 200 OK
  658. Via: SIP/2.0/UDP 72.9.246.170:5060;branch=z9hG4bK1f4ab9e3;received=72.9.246.170;rport=5060
  659. From: <sip:8604000023@atlanta2.voip.ms>;tag=as191b7847
  660. To: <sip:206996@192.168.10.248>;tag=as1259ff78
  661. Call-ID: 5f6d47381845abd81f8d7dd70d33a367@192.168.10.248:5060
  662. CSeq: 102 BYE
  663. Server: FPBX-14.0.3.2(13.18.4)
  664. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  665. Supported: replaces, timer
  666. Content-Length: 0
  667.  
  668.  
  669. <------------>
  670.     -- Stopped music on hold on SIP/Jose_Trunk-00000055
  671.     -- Channel SIP/Jose_Trunk-00000055 left 'simple_bridge' basic-bridge <b951bb83-6ca8-4b6d-bc94-a233e2d4234>
  672.     -- Channel SIP/113-00000054 left 'simple_bridge' basic-bridge <b951bb83-6ca8-4b6d-bc94-ca233e24234>
  673.   == Spawn extension (macro-dialout-trunk, s, 24) exited non-zero on 'SIP/113-00000054' in macro 'ialout-trunk'
  674.   == Spawn extension (from-internal, 218604000023, 6) exited non-zero on 'SIP/113-00000054'
  675.     -- Executing [h@from-internal:1] Macro("SIP/113-00000054", "hangupcall") in new stack
  676.     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/113-00000054", "1?theend") in new stack
  677.     -- Goto (macro-hangupcall,s,3)
  678.     -- Executing [s@macro-hangupcall:3] ExecIf("SIP/113-00000054", "0?Set(CDR(recordingfile)=)") i new stack
  679.     -- Executing [s@macro-hangupcall:4] NoOp("SIP/113-00000054", "SIP/Jose_Trunk-00000055 monior fle= ") in new stack
  680.     -- Executing [s@macro-hangupcall:5] AGI("SIP/113-00000054", "attendedtransfer-rec-restart.php,IP/Jose_Trunk-00000055,") in new stack
  681.     -- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
  682.     -- <SIP/113-00000054>AGI Script attendedtransfer-rec-restart.php completed, returning 0
  683.     -- Executing [s@macro-hangupcall:6] Hangup("SIP/113-00000054", "") in new stack
  684.   == Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/113-00000054' in macro 'hangpcall'
  685.   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/113-00000054'
  686. Scheduling destruction of SIP dialog 'x3keCHm36L7mGvHC4cavfQ..' in 6400 ms (Method: ACK)
  687. set_destination: Parsing <sip:113@192.168.10.53:5065;transport=UDP> for address/port to send to
  688. set_destination: set destination to 192.168.10.53:5065
  689. Reliably Transmitting (no NAT) to 192.168.10.53:5065:
  690. BYE sip:113@192.168.10.53:5065;transport=UDP SIP/2.0
  691. Via: SIP/2.0/UDP 192.168.10.248:5060;branch=z9hG4bK07913e8e
  692. Max-Forwards: 70
  693. From: <sip:218604000023@192.168.10.248;transport=UDP>;tag=as3d890bea
  694. To: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  695. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  696. CSeq: 102 BYE
  697. User-Agent: FPBX-14.0.3.2(13.18.4)
  698. Proxy-Authorization: Digest username="113", realm="asterisk", algorithm=MD5, uri="sip:192.168.10.28", nonce="17cd8bdb", response="4e76f2dbcae3240f29734a013089c8e4"
  699. X-Asterisk-HangupCause: Normal Clearing
  700. X-Asterisk-HangupCauseCode: 16
  701. Content-Length: 0
  702.  
  703.  
  704. ---
  705.  
  706. <--- SIP read from UDP:192.168.10.53:5065 --->
  707. SIP/2.0 200 OK
  708. Via: SIP/2.0/UDP 192.168.10.248:5060;branch=z9hG4bK07913e8e
  709. Contact: <sip:113@192.168.10.53:5065;transport=UDP>
  710. To: <sip:113@192.168.10.248;transport=UDP>;tag=47670316
  711. From: <sip:218604000023@192.168.10.248;transport=UDP>;tag=as3d890bea
  712. Call-ID: x3keCHm36L7mGvHC4cavfQ..
  713. CSeq: 102 BYE
  714. User-Agent: Z 3.14.38765 rv2.8.3
  715. Content-Length: 0
  716.  
  717. <------------->
  718. --- (9 headers 0 lines) ---
  719. SIP Response message for INCOMING dialog BYE arrived
  720. Really destroying SIP dialog 'x3keCHm36L7mGvHC4cavfQ..' Method: ACK
  721. freepbx*CLI> exit
  722. Asterisk cleanly ending (0).
  723. Executing last minute cleanups
  724. [root@freepbx asterisk]#
  725.  

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