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  1. [root@localhost ~]#
  2. [root@localhost ~]# asterisk -rvvvvvvvvv
  3. Asterisk 13.14.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  4. Created by Mark Spencer <markster@digium.com>
  5. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  6. This is free software, with components licensed under the GNU General Public
  7. License version 2 and other licenses; you are welcome to redistribute it under
  8. certain conditions. Type 'core show license' for details.
  9. =========================================================================
  10. Connected to Asterisk 13.14.0 currently running on localhost (pid = 4221)
  11.  
  12. <--- SIP read from UDP:70.27.246.146:5060 --->
  13. jaK
  14. <------------->
  15.  
  16. <--- SIP read from UDP:70.27.246.146:5060 --->
  17. jaK
  18. <------------->
  19. localhost*CLI> quit
  20. Asterisk cleanly ending (0).
  21. Executing last minute cleanups
  22. [root@localhost ~]# asterisk -rvvvvvvvvv
  23. Asterisk 13.14.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  24. Created by Mark Spencer <markster@digium.com>
  25. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  26. This is free software, with components licensed under the GNU General Public
  27. License version 2 and other licenses; you are welcome to redistribute it under
  28. certain conditions. Type 'core show license' for details.
  29. =========================================================================
  30. Connected to Asterisk 13.14.0 currently running on localhost (pid = 4221)
  31.  
  32. <--- SIP read from UDP:70.27.246.146:5060 --->
  33. INVITE sip:18666556565@162.210.197.227:5160 SIP/2.0
  34. Via: SIP/2.0/UDP 192.168.2.4:5060;rport;branch=z9hG4bK399085227
  35. From: <sip:1000@162.210.197.227:5160>;tag=1161571400
  36. To: <sip:18666556565@162.210.197.227:5160>
  37. Call-ID: 1625739749
  38. CSeq: 20 INVITE
  39. Contact: <sip:1000@70.27.246.146>
  40. Content-Type: application/sdp
  41. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  42. Max-Forwards: 70
  43. User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
  44. Subject: Phone call
  45. Content-Length: 477
  46.  
  47. v=0
  48. o=1000 3766 1978 IN IP4 192.168.2.4
  49. s=Talk
  50. c=IN IP4 192.168.2.4
  51. t=0 0
  52. m=audio 7078 RTP/AVP 111 110 0 8 101
  53. a=rtpmap:111 speex/16000
  54. a=fmtp:111 vbr=on
  55. a=rtpmap:110 speex/8000
  56. a=fmtp:110 vbr=on
  57. a=rtpmap:0 PCMU/8000
  58. a=rtpmap:8 PCMA/8000
  59. a=rtpmap:101 telephone-event/8000
  60. a=fmtp:101 0-11
  61. m=video 9078 RTP/AVP 103 102 99
  62. a=rtpmap:103 VP8/90000
  63. a=rtpmap:102 H264/90000
  64. a=fmtp:102 profile-level-id=428014
  65. a=rtpmap:99 MP4V-ES/90000
  66. a=fmtp:99 profile-level-id=3
  67. <------------->
  68. --- (13 headers 20 lines) ---
  69. Sending to 70.27.246.146:5060 (NAT)
  70. Sending to 70.27.246.146:5060 (NAT)
  71. Using INVITE request as basis request - 1625739749
  72. Found peer '1000' for '1000' from 70.27.246.146:5060
  73.  
  74. <--- Reliably Transmitting (NAT) to 70.27.246.146:5060 --->
  75. SIP/2.0 401 Unauthorized
  76. Via: SIP/2.0/UDP 192.168.2.4:5060;branch=z9hG4bK399085227;received=70.27.246.146;rport=5060
  77. From: <sip:1000@162.210.197.227:5160>;tag=1161571400
  78. To: <sip:18666556565@162.210.197.227:5160>;tag=as559577f4
  79. Call-ID: 1625739749
  80. CSeq: 20 INVITE
  81. Server: FPBX-13.0.191.5(13.14.0)
  82. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  83. Supported: replaces, timer
  84. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6e9c8a81"
  85. Content-Length: 0
  86.  
  87.  
  88. <------------>
  89. Scheduling destruction of SIP dialog '1625739749' in 6400 ms (Method: INVITE)
  90.  
  91. <--- SIP read from UDP:70.27.246.146:5060 --->
  92. ACK sip:18666556565@162.210.197.227:5160 SIP/2.0
  93. Via: SIP/2.0/UDP 192.168.2.4:5060;rport;branch=z9hG4bK399085227
  94. From: <sip:1000@162.210.197.227:5160>;tag=1161571400
  95. To: <sip:18666556565@162.210.197.227:5160>;tag=as559577f4
  96. Call-ID: 1625739749
  97. CSeq: 20 ACK
  98. Content-Length: 0
  99.  
  100. <------------->
  101. --- (7 headers 0 lines) ---
  102.  
  103. <--- SIP read from UDP:70.27.246.146:5060 --->
  104. INVITE sip:18666556565@162.210.197.227:5160 SIP/2.0
  105. Via: SIP/2.0/UDP 192.168.2.4:5060;rport;branch=z9hG4bK930839631
  106. From: <sip:1000@162.210.197.227:5160>;tag=1161571400
  107. To: <sip:18666556565@162.210.197.227:5160>
  108. Call-ID: 1625739749
  109. CSeq: 21 INVITE
  110. Contact: <sip:1000@70.27.246.146>
  111. Authorization: Digest username="1000", realm="asterisk", nonce="6e9c8a81", uri="sip:18666556565@162.210.197.227:5160", response="39ea159782eb7bf0017e2a39a95cab83", algorithm=MD5
  112. Content-Type: application/sdp
  113. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  114. Max-Forwards: 70
  115. User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
  116. Subject: Phone call
  117. Content-Length: 477
  118.  
  119. v=0
  120. o=1000 3766 1978 IN IP4 192.168.2.4
  121. s=Talk
  122. c=IN IP4 192.168.2.4
  123. t=0 0
  124. m=audio 7078 RTP/AVP 111 110 0 8 101
  125. a=rtpmap:111 speex/16000
  126. a=fmtp:111 vbr=on
  127. a=rtpmap:110 speex/8000
  128. a=fmtp:110 vbr=on
  129. a=rtpmap:0 PCMU/8000
  130. a=rtpmap:8 PCMA/8000
  131. a=rtpmap:101 telephone-event/8000
  132. a=fmtp:101 0-11
  133. m=video 9078 RTP/AVP 103 102 99
  134. a=rtpmap:103 VP8/90000
  135. a=rtpmap:102 H264/90000
  136. a=fmtp:102 profile-level-id=428014
  137. a=rtpmap:99 MP4V-ES/90000
  138. a=fmtp:99 profile-level-id=3
  139. <------------->
  140. --- (14 headers 20 lines) ---
  141. Sending to 70.27.246.146:5060 (NAT)
  142. Using INVITE request as basis request - 1625739749
  143. Found peer '1000' for '1000' from 70.27.246.146:5060
  144.   == Using SIP RTP TOS bits 184
  145.   == Using SIP RTP CoS mark 5
  146. Found RTP audio format 111
  147. Found RTP audio format 110
  148. Found RTP audio format 0
  149. Found RTP audio format 8
  150. Found RTP audio format 101
  151. Found audio description format speex for ID 111
  152. Found audio description format speex for ID 110
  153. Found audio description format PCMU for ID 0
  154. Found audio description format PCMA for ID 8
  155. Found audio description format telephone-event for ID 101
  156. Found RTP video format 103
  157. Found RTP video format 102
  158. Found RTP video format 99
  159. Found video description format VP8 for ID 103
  160. Found video description format H264 for ID 102
  161. Found video description format MP4V-ES for ID 99
  162. Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|alaw|speex|speex16)/video=(mpeg4|h264|vp8)/text=(nothing), combined - (ulaw|alaw)
  163. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  164. Peer audio RTP is at port 192.168.2.4:7078
  165. Looking for 18666556565 in from-internal (domain 162.210.197.227)
  166. sip_route_dump: route/path hop: <sip:1000@70.27.246.146>
  167.  
  168. <--- Transmitting (NAT) to 70.27.246.146:5060 --->
  169. SIP/2.0 100 Trying
  170. Via: SIP/2.0/UDP 192.168.2.4:5060;branch=z9hG4bK930839631;received=70.27.246.146;rport=5060
  171. From: <sip:1000@162.210.197.227:5160>;tag=1161571400
  172. To: <sip:18666556565@162.210.197.227:5160>
  173. Call-ID: 1625739749
  174. CSeq: 21 INVITE
  175. Server: FPBX-13.0.191.5(13.14.0)
  176. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  177. Supported: replaces, timer
  178. Contact: <sip:18666556565@162.210.197.227:5160>
  179. Content-Length: 0
  180.  
  181.  
  182. <------------>
  183.     -- Executing [18666556565@from-internal:1] Macro("SIP/1000-00000031", "user-callerid,LIMIT,EXTERNAL,") in new stack
  184.     -- Executing [s@macro-user-callerid:1] Set("SIP/1000-00000031", "TOUCH_MONITOR=1491162002.49") in new stack
  185.     -- Executing [s@macro-user-callerid:2] Set("SIP/1000-00000031", "AMPUSER=1000") in new stack
  186.     -- Executing [s@macro-user-callerid:3] GotoIf("SIP/1000-00000031", "0?report") in new stack
  187.     -- Executing [s@macro-user-callerid:4] ExecIf("SIP/1000-00000031", "1?Set(REALCALLERIDNUM=1000)") in new stack
  188.     -- Executing [s@macro-user-callerid:5] Set("SIP/1000-00000031", "AMPUSER=1000") in new stack
  189.     -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1000-00000031", "0?limit") in new stack
  190.     -- Executing [s@macro-user-callerid:7] Set("SIP/1000-00000031", "AMPUSERCIDNAME=Claudio") in new stack
  191.     -- Executing [s@macro-user-callerid:8] GotoIf("SIP/1000-00000031", "0?report") in new stack
  192.     -- Executing [s@macro-user-callerid:9] Set("SIP/1000-00000031", "AMPUSERCID=1000") in new stack
  193.     -- Executing [s@macro-user-callerid:10] Set("SIP/1000-00000031", "__DIAL_OPTIONS=Ttr") in new stack
  194.     -- Executing [s@macro-user-callerid:11] Set("SIP/1000-00000031", "CALLERID(all)="Claudio" <1000>") in new stack
  195.     -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1000-00000031", "0?limit") in new stack
  196.     -- Executing [s@macro-user-callerid:13] ExecIf("SIP/1000-00000031", "1?Set(GROUP(concurrency_limit)=1000)") in new stack
  197.     -- Executing [s@macro-user-callerid:14] GotoIf("SIP/1000-00000031", "1?continue") in new stack
  198.     -- Goto (macro-user-callerid,s,27)
  199.     -- Executing [s@macro-user-callerid:27] Set("SIP/1000-00000031", "CALLERID(number)=1000") in new stack
  200.     -- Executing [s@macro-user-callerid:28] Set("SIP/1000-00000031", "CALLERID(name)=Claudio") in new stack
  201.     -- Executing [s@macro-user-callerid:29] GotoIf("SIP/1000-00000031", "0?cnum") in new stack
  202.     -- Executing [s@macro-user-callerid:30] Set("SIP/1000-00000031", "CDR(cnam)=Claudio") in new stack
  203.     -- Executing [s@macro-user-callerid:31] Set("SIP/1000-00000031", "CDR(cnum)=1000") in new stack
  204.     -- Executing [s@macro-user-callerid:32] Set("SIP/1000-00000031", "CHANNEL(language)=en") in new stack
  205.     -- Executing [18666556565@from-internal:2] Gosub("SIP/1000-00000031", "sub-record-check,s,1(out,18666556565,dontcare)") in new stack
  206.     -- Executing [s@sub-record-check:1] GotoIf("SIP/1000-00000031", "0?initialized") in new stack
  207.     -- Executing [s@sub-record-check:2] Set("SIP/1000-00000031", "__REC_STATUS=INITIALIZED") in new stack
  208.     -- Executing [s@sub-record-check:3] Set("SIP/1000-00000031", "NOW=1491162002") in new stack
  209.     -- Executing [s@sub-record-check:4] Set("SIP/1000-00000031", "__DAY=02") in new stack
  210.     -- Executing [s@sub-record-check:5] Set("SIP/1000-00000031", "__MONTH=04") in new stack
  211.     -- Executing [s@sub-record-check:6] Set("SIP/1000-00000031", "__YEAR=2017") in new stack
  212.     -- Executing [s@sub-record-check:7] Set("SIP/1000-00000031", "__TIMESTR=20170402-154002") in new stack
  213.     -- Executing [s@sub-record-check:8] Set("SIP/1000-00000031", "__FROMEXTEN=1000") in new stack
  214.     -- Executing [s@sub-record-check:9] Set("SIP/1000-00000031", "__MON_FMT=wav") in new stack
  215.     -- Executing [s@sub-record-check:10] NoOp("SIP/1000-00000031", "Recordings initialized") in new stack
  216.     -- Executing [s@sub-record-check:11] ExecIf("SIP/1000-00000031", "0?Set(ARG3=dontcare)") in new stack
  217.     -- Executing [s@sub-record-check:12] Set("SIP/1000-00000031", "REC_POLICY_MODE_SAVE=") in new stack
  218.     -- Executing [s@sub-record-check:13] ExecIf("SIP/1000-00000031", "0?Set(REC_STATUS=NO)") in new stack
  219.     -- Executing [s@sub-record-check:14] GotoIf("SIP/1000-00000031", "3?checkaction") in new stack
  220.     -- Goto (sub-record-check,s,17)
  221.     -- Executing [s@sub-record-check:17] GotoIf("SIP/1000-00000031", "1?sub-record-check,out,1") in new stack
  222.     -- Goto (sub-record-check,out,1)
  223.     -- Executing [out@sub-record-check:1] NoOp("SIP/1000-00000031", "Outbound Recording Check from 1000 to 18666556565") in new stack
  224.     -- Executing [out@sub-record-check:2] Set("SIP/1000-00000031", "RECMODE=dontcare") in new stack
  225.     -- Executing [out@sub-record-check:3] ExecIf("SIP/1000-00000031", "1?Goto(routewins)") in new stack
  226.     -- Goto (sub-record-check,out,7)
  227.     -- Executing [out@sub-record-check:7] Gosub("SIP/1000-00000031", "recordcheck,1(dontcare,out,18666556565)") in new stack
  228.     -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/1000-00000031", "Starting recording check against dontcare") in new stack
  229.     -- Executing [recordcheck@sub-record-check:2] Goto("SIP/1000-00000031", "dontcare") in new stack
  230.     -- Goto (sub-record-check,recordcheck,3)
  231.     -- Executing [recordcheck@sub-record-check:3] Return("SIP/1000-00000031", "") in new stack
  232.     -- Executing [out@sub-record-check:8] Return("SIP/1000-00000031", "") in new stack
  233.     -- Executing [18666556565@from-internal:3] Set("SIP/1000-00000031", "MOHCLASS=default") in new stack
  234.     -- Executing [18666556565@from-internal:4] ExecIf("SIP/1000-00000031", "1?Set(TRUNKCIDOVERRIDE=root)") in new stack
  235.     -- Executing [18666556565@from-internal:5] Set("SIP/1000-00000031", "_NODEST=") in new stack
  236.     -- Executing [18666556565@from-internal:6] Macro("SIP/1000-00000031", "dialout-trunk,2,18666556565,,off") in new stack
  237.     -- Executing [s@macro-dialout-trunk:1] Set("SIP/1000-00000031", "DIAL_TRUNK=2") in new stack
  238.     -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1000-00000031", "0?sub-pincheck,s,1()") in new stack
  239.     -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1000-00000031", "0?disabletrunk,1") in new stack
  240.     -- Executing [s@macro-dialout-trunk:4] Set("SIP/1000-00000031", "DIAL_NUMBER=18666556565") in new stack
  241.     -- Executing [s@macro-dialout-trunk:5] Set("SIP/1000-00000031", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
  242.     -- Executing [s@macro-dialout-trunk:6] Set("SIP/1000-00000031", "OUTBOUND_GROUP=OUT_2") in new stack
  243.     -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1000-00000031", "1?nomax") in new stack
  244.     -- Goto (macro-dialout-trunk,s,9)
  245.     -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1000-00000031", "0?skipoutcid") in new stack
  246.     -- Executing [s@macro-dialout-trunk:10] Set("SIP/1000-00000031", "DIAL_TRUNK_OPTIONS=T") in new stack
  247.     -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1000-00000031", "outbound-callerid,2") in new stack
  248.     -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1000-00000031", "0?Set(CALLERPRES(name-pres)=)") in new stack
  249.     -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1000-00000031", "0?Set(CALLERPRES(num-pres)=)") in new stack
  250.     -- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/1000-00000031", "0?Set(REALCALLERIDNUM=1000)") in new stack
  251.     -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/1000-00000031", "1?normcid") in new stack
  252.     -- Goto (macro-outbound-callerid,s,7)
  253.     -- Executing [s@macro-outbound-callerid:7] Set("SIP/1000-00000031", "USEROUTCID=") in new stack
  254.     -- Executing [s@macro-outbound-callerid:8] Set("SIP/1000-00000031", "EMERGENCYCID=") in new stack
  255.     -- Executing [s@macro-outbound-callerid:9] Set("SIP/1000-00000031", "TRUNKOUTCID=") in new stack
  256.     -- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/1000-00000031", "1?trunkcid") in new stack
  257.     -- Goto (macro-outbound-callerid,s,15)
  258.     -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/1000-00000031", "0?Set(CALLERID(all)=)") in new stack
  259.     -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/1000-00000031", "0?Set(CALLERID(all)=)") in new stack
  260.     -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/1000-00000031", "1?Set(CALLERID(all)=root)") in new stack
  261.     -- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/1000-00000031", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
  262.     -- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/1000-00000031", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
  263.     -- Executing [s@macro-outbound-callerid:20] Set("SIP/1000-00000031", "CDR(outbound_cnum)=") in new stack
  264. [2017-04-02 15:40:02] WARNING[4242]: func_cdr.c:383 cdr_write_callback: CDR requires a value (CDR(variable)=value)
  265.     -- Executing [s@macro-outbound-callerid:21] Set("SIP/1000-00000031", "CDR(outbound_cnam)=root") in new stack
  266.     -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/1000-00000031", "0?sub-flp-2,s,1()") in new stack
  267.     -- Executing [s@macro-dialout-trunk:13] Set("SIP/1000-00000031", "OUTNUM=18666556565") in new stack
  268.     -- Executing [s@macro-dialout-trunk:14] Set("SIP/1000-00000031", "custom=SIP/outgoing") in new stack
  269.     -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1000-00000031", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
  270.     -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/1000-00000031", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
  271.     -- Executing [s@macro-dialout-trunk:17] Macro("SIP/1000-00000031", "dialout-trunk-predial-hook,") in new stack
  272.     -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1000-00000031", "") in new stack
  273.     -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1000-00000031", "0?bypass,1") in new stack
  274.     -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/1000-00000031", "1?Set(CONNECTEDLINE(num,i)=18666556565)") in new stack
  275.     -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/1000-00000031", "1?Set(CONNECTEDLINE(name,i)=CID:)") in new stack
  276.     -- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/1000-00000031", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden))") in new stack
  277.     -- Executing [s@macro-dialout-trunk:22] GotoIf("SIP/1000-00000031", "0?customtrunk") in new stack
  278.     -- Executing [s@macro-dialout-trunk:23] Dial("SIP/1000-00000031", "SIP/outgoing/18666556565,300,T") in new stack
  279.   == Using SIP RTP TOS bits 184
  280.   == Using SIP RTP CoS mark 5
  281. Audio is at 10788
  282. Adding codec ulaw to SDP
  283. Adding non-codec 0x1 (telephone-event) to SDP
  284. Reliably Transmitting (NAT) to 74.63.41.218:5060:
  285. INVITE sip:18666556565@newyork.voip.ms SIP/2.0
  286. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK23ffe58f;rport
  287. Max-Forwards: 70
  288. From: "root" <sip:214068@162.210.197.227:5160>;tag=as57108701
  289. To: <sip:18666556565@newyork.voip.ms>
  290. Contact: <sip:214068@162.210.197.227:5160>
  291. Call-ID: 77f3a14413ee71743b00f3230263a9ca@162.210.197.227:5160
  292. CSeq: 102 INVITE
  293. User-Agent: FPBX-13.0.191.5(13.14.0)
  294. Date: Sun, 02 Apr 2017 19:40:02 GMT
  295. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  296. Supported: replaces, timer
  297. Content-Type: application/sdp
  298. Content-Length: 258
  299.  
  300. v=0
  301. o=root 1210407780 1210407780 IN IP4 162.210.197.227
  302. s=Asterisk PBX 13.14.0
  303. c=IN IP4 162.210.197.227
  304. t=0 0
  305. m=audio 10788 RTP/AVP 0 101
  306. a=rtpmap:0 PCMU/8000
  307. a=rtpmap:101 telephone-event/8000
  308. a=fmtp:101 0-16
  309. a=ptime:20
  310. a=maxptime:150
  311. a=sendrecv
  312.  
  313. ---
  314.     -- Called SIP/outgoing/18666556565
  315. Retransmitting #1 (NAT) to 74.63.41.218:5060:
  316. INVITE sip:18666556565@newyork.voip.ms SIP/2.0
  317. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK23ffe58f;rport
  318. Max-Forwards: 70
  319. From: "root" <sip:214068@162.210.197.227:5160>;tag=as57108701
  320. To: <sip:18666556565@newyork.voip.ms>
  321. Contact: <sip:214068@162.210.197.227:5160>
  322. Call-ID: 77f3a14413ee71743b00f3230263a9ca@162.210.197.227:5160
  323. CSeq: 102 INVITE
  324. User-Agent: FPBX-13.0.191.5(13.14.0)
  325. Date: Sun, 02 Apr 2017 19:40:02 GMT
  326. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  327. Supported: replaces, timer
  328. Content-Type: application/sdp
  329. Content-Length: 258
  330.  
  331. v=0
  332. o=root 1210407780 1210407780 IN IP4 162.210.197.227
  333. s=Asterisk PBX 13.14.0
  334. c=IN IP4 162.210.197.227
  335. t=0 0
  336. m=audio 10788 RTP/AVP 0 101
  337. a=rtpmap:0 PCMU/8000
  338. a=rtpmap:101 telephone-event/8000
  339. a=fmtp:101 0-16
  340. a=ptime:20
  341. a=maxptime:150
  342. a=sendrecv
  343.  
  344. ---
  345. Retransmitting #2 (NAT) to 74.63.41.218:5060:
  346. INVITE sip:18666556565@newyork.voip.ms SIP/2.0
  347. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK23ffe58f;rport
  348. Max-Forwards: 70
  349. From: "root" <sip:214068@162.210.197.227:5160>;tag=as57108701
  350. To: <sip:18666556565@newyork.voip.ms>
  351. Contact: <sip:214068@162.210.197.227:5160>
  352. Call-ID: 77f3a14413ee71743b00f3230263a9ca@162.210.197.227:5160
  353. CSeq: 102 INVITE
  354. User-Agent: FPBX-13.0.191.5(13.14.0)
  355. Date: Sun, 02 Apr 2017 19:40:02 GMT
  356. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  357. Supported: replaces, timer
  358. Content-Type: application/sdp
  359. Content-Length: 258
  360.  
  361. v=0
  362. o=root 1210407780 1210407780 IN IP4 162.210.197.227
  363. s=Asterisk PBX 13.14.0
  364. c=IN IP4 162.210.197.227
  365. t=0 0
  366. m=audio 10788 RTP/AVP 0 101
  367. a=rtpmap:0 PCMU/8000
  368. a=rtpmap:101 telephone-event/8000
  369. a=fmtp:101 0-16
  370. a=ptime:20
  371. a=maxptime:150
  372. a=sendrecv
  373.  
  374. ---
  375. Retransmitting #3 (NAT) to 74.63.41.218:5060:
  376. INVITE sip:18666556565@newyork.voip.ms SIP/2.0
  377. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK23ffe58f;rport
  378. Max-Forwards: 70
  379. From: "root" <sip:214068@162.210.197.227:5160>;tag=as57108701
  380. To: <sip:18666556565@newyork.voip.ms>
  381. Contact: <sip:214068@162.210.197.227:5160>
  382. Call-ID: 77f3a14413ee71743b00f3230263a9ca@162.210.197.227:5160
  383. CSeq: 102 INVITE
  384. User-Agent: FPBX-13.0.191.5(13.14.0)
  385. Date: Sun, 02 Apr 2017 19:40:02 GMT
  386. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  387. Supported: replaces, timer
  388. Content-Type: application/sdp
  389. Content-Length: 258
  390.  
  391. v=0
  392. o=root 1210407780 1210407780 IN IP4 162.210.197.227
  393. s=Asterisk PBX 13.14.0
  394. c=IN IP4 162.210.197.227
  395. t=0 0
  396. m=audio 10788 RTP/AVP 0 101
  397. a=rtpmap:0 PCMU/8000
  398. a=rtpmap:101 telephone-event/8000
  399. a=fmtp:101 0-16
  400. a=ptime:20
  401. a=maxptime:150
  402. a=sendrecv
  403.  
  404. ---
  405. Retransmitting #4 (NAT) to 74.63.41.218:5060:
  406. INVITE sip:18666556565@newyork.voip.ms SIP/2.0
  407. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK23ffe58f;rport
  408. Max-Forwards: 70
  409. From: "root" <sip:214068@162.210.197.227:5160>;tag=as57108701
  410. To: <sip:18666556565@newyork.voip.ms>
  411. Contact: <sip:214068@162.210.197.227:5160>
  412. Call-ID: 77f3a14413ee71743b00f3230263a9ca@162.210.197.227:5160
  413. CSeq: 102 INVITE
  414. User-Agent: FPBX-13.0.191.5(13.14.0)
  415. Date: Sun, 02 Apr 2017 19:40:02 GMT
  416. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  417. Supported: replaces, timer
  418. Content-Type: application/sdp
  419. Content-Length: 258
  420.  
  421. v=0
  422. o=root 1210407780 1210407780 IN IP4 162.210.197.227
  423. s=Asterisk PBX 13.14.0
  424. c=IN IP4 162.210.197.227
  425. t=0 0
  426. m=audio 10788 RTP/AVP 0 101
  427. a=rtpmap:0 PCMU/8000
  428. a=rtpmap:101 telephone-event/8000
  429. a=fmtp:101 0-16
  430. a=ptime:20
  431. a=maxptime:150
  432. a=sendrecv
  433.  
  434. ---
  435. Retransmitting #5 (NAT) to 74.63.41.218:5060:
  436. INVITE sip:18666556565@newyork.voip.ms SIP/2.0
  437. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK23ffe58f;rport
  438. Max-Forwards: 70
  439. From: "root" <sip:214068@162.210.197.227:5160>;tag=as57108701
  440. To: <sip:18666556565@newyork.voip.ms>
  441. Contact: <sip:214068@162.210.197.227:5160>
  442. Call-ID: 77f3a14413ee71743b00f3230263a9ca@162.210.197.227:5160
  443. CSeq: 102 INVITE
  444. User-Agent: FPBX-13.0.191.5(13.14.0)
  445. Date: Sun, 02 Apr 2017 19:40:02 GMT
  446. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  447. Supported: replaces, timer
  448. Content-Type: application/sdp
  449. Content-Length: 258
  450.  
  451. v=0
  452. o=root 1210407780 1210407780 IN IP4 162.210.197.227
  453. s=Asterisk PBX 13.14.0
  454. c=IN IP4 162.210.197.227
  455. t=0 0
  456. m=audio 10788 RTP/AVP 0 101
  457. a=rtpmap:0 PCMU/8000
  458. a=rtpmap:101 telephone-event/8000
  459. a=fmtp:101 0-16
  460. a=ptime:20
  461. a=maxptime:150
  462. a=sendrecv
  463.  
  464. ---
  465.  
  466. <--- SIP read from UDP:74.63.41.218:5060 --->
  467. SIP/2.0 401 Unauthorized
  468. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK23ffe58f;received=162.210.197.227;rport=5160
  469. From: "root" <sip:214068@162.210.197.227:5160>;tag=as57108701
  470. To: <sip:18666556565@newyork.voip.ms>;tag=as0b654b9e
  471. Call-ID: 77f3a14413ee71743b00f3230263a9ca@162.210.197.227:5160
  472. CSeq: 102 INVITE
  473. Server: voip.ms
  474. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  475. Supported: replaces, timer
  476. WWW-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", nonce="3cdb4fca"
  477. Content-Length: 0
  478.  
  479. <------------->
  480. --- (11 headers 0 lines) ---
  481. Transmitting (NAT) to 74.63.41.218:5060:
  482. ACK sip:18666556565@newyork.voip.ms SIP/2.0
  483. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK23ffe58f;rport
  484. Max-Forwards: 70
  485. From: "root" <sip:214068@162.210.197.227:5160>;tag=as57108701
  486. To: <sip:18666556565@newyork.voip.ms>;tag=as0b654b9e
  487. Contact: <sip:214068@162.210.197.227:5160>
  488. Call-ID: 77f3a14413ee71743b00f3230263a9ca@162.210.197.227:5160
  489. CSeq: 102 ACK
  490. User-Agent: FPBX-13.0.191.5(13.14.0)
  491. Content-Length: 0
  492.  
  493.  
  494. ---
  495. Audio is at 10788
  496. Adding codec ulaw to SDP
  497. Adding non-codec 0x1 (telephone-event) to SDP
  498. Reliably Transmitting (NAT) to 74.63.41.218:5060:
  499. INVITE sip:18666556565@newyork.voip.ms SIP/2.0
  500. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK65b5d091;rport
  501. Max-Forwards: 70
  502. From: "root" <sip:214068@162.210.197.227:5160>;tag=as57108701
  503. To: <sip:18666556565@newyork.voip.ms>
  504. Contact: <sip:214068@162.210.197.227:5160>
  505. Call-ID: 77f3a14413ee71743b00f3230263a9ca@162.210.197.227:5160
  506. CSeq: 103 INVITE
  507. User-Agent: FPBX-13.0.191.5(13.14.0)
  508. Authorization: Digest username="214068", realm="newyork.voip.ms", algorithm=MD5, uri="sip:18666556565@newyork.voip.ms", nonce="3cdb4fca", response="2bc83978a99bf8ca4879c70b2adb5a08"
  509. Date: Sun, 02 Apr 2017 19:40:05 GMT
  510. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  511. Supported: replaces, timer
  512. Content-Type: application/sdp
  513. Content-Length: 258
  514.  
  515. v=0
  516. o=root 1210407780 1210407781 IN IP4 162.210.197.227
  517. s=Asterisk PBX 13.14.0
  518. c=IN IP4 162.210.197.227
  519. t=0 0
  520. m=audio 10788 RTP/AVP 0 101
  521. a=rtpmap:0 PCMU/8000
  522. a=rtpmap:101 telephone-event/8000
  523. a=fmtp:101 0-16
  524. a=ptime:20
  525. a=maxptime:150
  526. a=sendrecv
  527.  
  528. ---
  529.  
  530. <--- SIP read from UDP:74.63.41.218:5060 --->
  531. SIP/2.0 100 Trying
  532. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK65b5d091;received=162.210.197.227;rport=5160
  533. From: "root" <sip:214068@162.210.197.227:5160>;tag=as57108701
  534. To: <sip:18666556565@newyork.voip.ms>
  535. Call-ID: 77f3a14413ee71743b00f3230263a9ca@162.210.197.227:5160
  536. CSeq: 103 INVITE
  537. Server: voip.ms
  538. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  539. Supported: replaces, timer
  540. Session-Expires: 1800;refresher=uas
  541. Contact: <sip:18666556565@74.63.41.218:5060>
  542. Content-Length: 0
  543.  
  544. <------------->
  545. --- (12 headers 0 lines) ---
  546.  
  547. <--- SIP read from UDP:74.63.41.218:5060 --->
  548. SIP/2.0 503 Service Unavailable
  549. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK65b5d091;received=162.210.197.227;rport=5160
  550. From: "root" <sip:214068@162.210.197.227:5160>;tag=as57108701
  551. To: <sip:18666556565@newyork.voip.ms>;tag=as719621ea
  552. Call-ID: 77f3a14413ee71743b00f3230263a9ca@162.210.197.227:5160
  553. CSeq: 103 INVITE
  554. Server: voip.ms
  555. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  556. Supported: replaces, timer
  557. Session-Expires: 1800;refresher=uas
  558. Content-Length: 0
  559.  
  560. <------------->
  561. --- (11 headers 0 lines) ---
  562.     -- Got SIP response 503 "Service Unavailable" back from 74.63.41.218:5060
  563. Transmitting (NAT) to 74.63.41.218:5060:
  564. ACK sip:18666556565@newyork.voip.ms SIP/2.0
  565. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK65b5d091;rport
  566. Max-Forwards: 70
  567. From: "root" <sip:214068@162.210.197.227:5160>;tag=as57108701
  568. To: <sip:18666556565@newyork.voip.ms>;tag=as719621ea
  569. Contact: <sip:214068@162.210.197.227:5160>
  570. Call-ID: 77f3a14413ee71743b00f3230263a9ca@162.210.197.227:5160
  571. CSeq: 103 ACK
  572. User-Agent: FPBX-13.0.191.5(13.14.0)
  573. Content-Length: 0
  574.  
  575.  
  576. ---
  577.     -- SIP/outgoing-00000032 is circuit-busy
  578.   == Everyone is busy/congested at this time (1:0/1/0)
  579.     -- Executing [s@macro-dialout-trunk:24] NoOp("SIP/1000-00000031", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack
  580.     -- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/1000-00000031", "0?continue,1:s-CONGESTION,1") in new stack
  581.     -- Goto (macro-dialout-trunk,s-CONGESTION,1)
  582.     -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/1000-00000031", "RC=34") in new stack
  583.     -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/1000-00000031", "34,1") in new stack
  584.     -- Goto (macro-dialout-trunk,34,1)
  585.     -- Executing [34@macro-dialout-trunk:1] Goto("SIP/1000-00000031", "continue,1") in new stack
  586.     -- Goto (macro-dialout-trunk,continue,1)
  587.     -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/1000-00000031", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks") in new stack
  588.     -- Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/1000-00000031", "1?Set(CALLERID(number)=1000)") in new stack
  589.     -- Executing [18666556565@from-internal:7] Macro("SIP/1000-00000031", "outisbusy,") in new stack
  590. [2017-04-02 15:40:06] WARNING[10684][C-0000001b]: app_macro.c:310 _macro_exec: No such context 'macro-outisbusy' for macro 'outisbusy'. Was called by 18666556565@from-internal
  591.     -- Executing [18666556565@from-internal:8] Hangup("SIP/1000-00000031", "") in new stack
  592.   == Spawn extension (from-internal, 18666556565, 8) exited non-zero on 'SIP/1000-00000031'
  593.     -- Executing [h@from-internal:1] Macro("SIP/1000-00000031", "hangupcall") in new stack
  594.     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1000-00000031", "1?theend") in new stack
  595.     -- Goto (macro-hangupcall,s,3)
  596.     -- Executing [s@macro-hangupcall:3] ExecIf("SIP/1000-00000031", "0?Set(CDR(recordingfile)=)") in new stack
  597.     -- Executing [s@macro-hangupcall:4] Hangup("SIP/1000-00000031", "") in new stack
  598.   == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/1000-00000031' in macro 'hangupcall'
  599.   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1000-00000031'
  600. Scheduling destruction of SIP dialog '1625739749' in 6400 ms (Method: INVITE)
  601.  
  602. <--- Reliably Transmitting (NAT) to 70.27.246.146:5060 --->
  603. SIP/2.0 503 Service Unavailable
  604. Via: SIP/2.0/UDP 192.168.2.4:5060;branch=z9hG4bK930839631;received=70.27.246.146;rport=5060
  605. From: <sip:1000@162.210.197.227:5160>;tag=1161571400
  606. To: <sip:18666556565@162.210.197.227:5160>;tag=as77da72ff
  607. Call-ID: 1625739749
  608. CSeq: 21 INVITE
  609. Server: FPBX-13.0.191.5(13.14.0)
  610. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  611. Supported: replaces, timer
  612. Content-Length: 0
  613.  
  614.  
  615. <------------>
  616. Really destroying SIP dialog '77f3a14413ee71743b00f3230263a9ca@162.210.197.227:5160' Method: INVITE
  617.  
  618. <--- SIP read from UDP:70.27.246.146:5060 --->
  619. ACK sip:18666556565@162.210.197.227:5160 SIP/2.0
  620. Via: SIP/2.0/UDP 192.168.2.4:5060;rport;branch=z9hG4bK930839631
  621. From: <sip:1000@162.210.197.227:5160>;tag=1161571400
  622. To: <sip:18666556565@162.210.197.227:5160>;tag=as77da72ff
  623. Call-ID: 1625739749
  624. CSeq: 21 ACK
  625. Content-Length: 0
  626.  
  627. <------------->
  628. --- (7 headers 0 lines) ---
  629.  
  630. <--- SIP read from UDP:70.27.246.146:5060 --->
  631. jaK
  632. <------------->
  633. localhost*CLI> quit
  634. Asterisk cleanly ending (0).
  635. Executing last minute cleanups
  636. [root@localhost ~]# asterisk -rvvvvvvvvv
  637. Asterisk 13.14.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  638. Created by Mark Spencer <markster@digium.com>
  639. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  640. This is free software, with components licensed under the GNU General Public
  641. License version 2 and other licenses; you are welcome to redistribute it under
  642. certain conditions. Type 'core show license' for details.
  643. =========================================================================
  644. Connected to Asterisk 13.14.0 currently running on localhost (pid = 4221)
  645.  
  646. <--- SIP read from UDP:70.27.246.146:5060 --->
  647. INVITE sip:15192556770@162.210.197.227:5160 SIP/2.0
  648. Via: SIP/2.0/UDP 192.168.2.4:5060;rport;branch=z9hG4bK1090311976
  649. From: <sip:1000@162.210.197.227:5160>;tag=965951055
  650. To: <sip:15192556770@162.210.197.227:5160>
  651. Call-ID: 1655555233
  652. CSeq: 20 INVITE
  653. Contact: <sip:1000@70.27.246.146>
  654. Content-Type: application/sdp
  655. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  656. Max-Forwards: 70
  657. User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
  658. Subject: Phone call
  659. Content-Length: 477
  660.  
  661. v=0
  662. o=1000 1493 3246 IN IP4 192.168.2.4
  663. s=Talk
  664. c=IN IP4 192.168.2.4
  665. t=0 0
  666. m=audio 7078 RTP/AVP 111 110 0 8 101
  667. a=rtpmap:111 speex/16000
  668. a=fmtp:111 vbr=on
  669. a=rtpmap:110 speex/8000
  670. a=fmtp:110 vbr=on
  671. a=rtpmap:0 PCMU/8000
  672. a=rtpmap:8 PCMA/8000
  673. a=rtpmap:101 telephone-event/8000
  674. a=fmtp:101 0-11
  675. m=video 9078 RTP/AVP 103 102 99
  676. a=rtpmap:103 VP8/90000
  677. a=rtpmap:102 H264/90000
  678. a=fmtp:102 profile-level-id=428014
  679. a=rtpmap:99 MP4V-ES/90000
  680. a=fmtp:99 profile-level-id=3
  681. <------------->
  682. --- (13 headers 20 lines) ---
  683. Sending to 70.27.246.146:5060 (no NAT)
  684. Sending to 70.27.246.146:5060 (no NAT)
  685. Using INVITE request as basis request - 1655555233
  686. Found peer '1000' for '1000' from 70.27.246.146:5060
  687.  
  688. <--- Reliably Transmitting (NAT) to 70.27.246.146:5060 --->
  689. SIP/2.0 401 Unauthorized
  690. Via: SIP/2.0/UDP 192.168.2.4:5060;branch=z9hG4bK1090311976;received=70.27.246.146;rport=5060
  691. From: <sip:1000@162.210.197.227:5160>;tag=965951055
  692. To: <sip:15192556770@162.210.197.227:5160>;tag=as222aa5b6
  693. Call-ID: 1655555233
  694. CSeq: 20 INVITE
  695. Server: FPBX-13.0.191.5(13.14.0)
  696. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  697. Supported: replaces, timer
  698. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="078579aa"
  699. Content-Length: 0
  700.  
  701.  
  702. <------------>
  703. Scheduling destruction of SIP dialog '1655555233' in 6400 ms (Method: INVITE)
  704.  
  705. <--- SIP read from UDP:70.27.246.146:5060 --->
  706. jaK
  707. <------------->
  708.  
  709. <--- SIP read from UDP:70.27.246.146:5060 --->
  710. ACK sip:15192556770@162.210.197.227:5160 SIP/2.0
  711. Via: SIP/2.0/UDP 192.168.2.4:5060;rport;branch=z9hG4bK1090311976
  712. From: <sip:1000@162.210.197.227:5160>;tag=965951055
  713. To: <sip:15192556770@162.210.197.227:5160>;tag=as222aa5b6
  714. Call-ID: 1655555233
  715. CSeq: 20 ACK
  716. Content-Length: 0
  717.  
  718. <------------->
  719. --- (7 headers 0 lines) ---
  720.  
  721. <--- SIP read from UDP:70.27.246.146:5060 --->
  722. INVITE sip:15192556770@162.210.197.227:5160 SIP/2.0
  723. Via: SIP/2.0/UDP 192.168.2.4:5060;rport;branch=z9hG4bK1688908289
  724. From: <sip:1000@162.210.197.227:5160>;tag=965951055
  725. To: <sip:15192556770@162.210.197.227:5160>
  726. Call-ID: 1655555233
  727. CSeq: 21 INVITE
  728. Contact: <sip:1000@70.27.246.146>
  729. Authorization: Digest username="1000", realm="asterisk", nonce="078579aa", uri="sip:15192556770@162.210.197.227:5160", response="1b7b5b119d3e1badde357b5341204bea", algorithm=MD5
  730. Content-Type: application/sdp
  731. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  732. Max-Forwards: 70
  733. User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
  734. Subject: Phone call
  735. Content-Length: 477
  736.  
  737. v=0
  738. o=1000 1493 3246 IN IP4 192.168.2.4
  739. s=Talk
  740. c=IN IP4 192.168.2.4
  741. t=0 0
  742. m=audio 7078 RTP/AVP 111 110 0 8 101
  743. a=rtpmap:111 speex/16000
  744. a=fmtp:111 vbr=on
  745. a=rtpmap:110 speex/8000
  746. a=fmtp:110 vbr=on
  747. a=rtpmap:0 PCMU/8000
  748. a=rtpmap:8 PCMA/8000
  749. a=rtpmap:101 telephone-event/8000
  750. a=fmtp:101 0-11
  751. m=video 9078 RTP/AVP 103 102 99
  752. a=rtpmap:103 VP8/90000
  753. a=rtpmap:102 H264/90000
  754. a=fmtp:102 profile-level-id=428014
  755. a=rtpmap:99 MP4V-ES/90000
  756. a=fmtp:99 profile-level-id=3
  757. <------------->
  758. --- (14 headers 20 lines) ---
  759. Sending to 70.27.246.146:5060 (NAT)
  760. Using INVITE request as basis request - 1655555233
  761. Found peer '1000' for '1000' from 70.27.246.146:5060
  762.   == Using SIP RTP TOS bits 184
  763.   == Using SIP RTP CoS mark 5
  764. Found RTP audio format 111
  765. Found RTP audio format 110
  766. Found RTP audio format 0
  767. Found RTP audio format 8
  768. Found RTP audio format 101
  769. Found audio description format speex for ID 111
  770. Found audio description format speex for ID 110
  771. Found audio description format PCMU for ID 0
  772. Found audio description format PCMA for ID 8
  773. Found audio description format telephone-event for ID 101
  774. Found RTP video format 103
  775. Found RTP video format 102
  776. Found RTP video format 99
  777. Found video description format VP8 for ID 103
  778. Found video description format H264 for ID 102
  779. Found video description format MP4V-ES for ID 99
  780. Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|alaw|speex|speex16)/video=(mpeg4|h264|vp8)/text=(nothing), combined - (ulaw|alaw)
  781. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  782. Peer audio RTP is at port 192.168.2.4:7078
  783. Looking for 15192556770 in from-internal (domain 162.210.197.227)
  784. sip_route_dump: route/path hop: <sip:1000@70.27.246.146>
  785.  
  786. <--- Transmitting (NAT) to 70.27.246.146:5060 --->
  787. SIP/2.0 100 Trying
  788. Via: SIP/2.0/UDP 192.168.2.4:5060;branch=z9hG4bK1688908289;received=70.27.246.146;rport=5060
  789. From: <sip:1000@162.210.197.227:5160>;tag=965951055
  790. To: <sip:15192556770@162.210.197.227:5160>
  791. Call-ID: 1655555233
  792. CSeq: 21 INVITE
  793. Server: FPBX-13.0.191.5(13.14.0)
  794. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  795. Supported: replaces, timer
  796. Contact: <sip:15192556770@162.210.197.227:5160>
  797. Content-Length: 0
  798.  
  799.  
  800. <------------>
  801.     -- Executing [15192556770@from-internal:1] Macro("SIP/1000-0000003b", "user-callerid,LIMIT,EXTERNAL,") in new stack
  802.     -- Executing [s@macro-user-callerid:1] Set("SIP/1000-0000003b", "TOUCH_MONITOR=1491162343.59") in new stack
  803.     -- Executing [s@macro-user-callerid:2] Set("SIP/1000-0000003b", "AMPUSER=1000") in new stack
  804.     -- Executing [s@macro-user-callerid:3] GotoIf("SIP/1000-0000003b", "0?report") in new stack
  805.     -- Executing [s@macro-user-callerid:4] ExecIf("SIP/1000-0000003b", "1?Set(REALCALLERIDNUM=1000)") in new stack
  806.     -- Executing [s@macro-user-callerid:5] Set("SIP/1000-0000003b", "AMPUSER=1000") in new stack
  807.     -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1000-0000003b", "0?limit") in new stack
  808.     -- Executing [s@macro-user-callerid:7] Set("SIP/1000-0000003b", "AMPUSERCIDNAME=Claudio") in new stack
  809.     -- Executing [s@macro-user-callerid:8] GotoIf("SIP/1000-0000003b", "0?report") in new stack
  810.     -- Executing [s@macro-user-callerid:9] Set("SIP/1000-0000003b", "AMPUSERCID=1000") in new stack
  811.     -- Executing [s@macro-user-callerid:10] Set("SIP/1000-0000003b", "__DIAL_OPTIONS=Ttr") in new stack
  812.     -- Executing [s@macro-user-callerid:11] Set("SIP/1000-0000003b", "CALLERID(all)="Claudio" <1000>") in new stack
  813.     -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1000-0000003b", "0?limit") in new stack
  814.     -- Executing [s@macro-user-callerid:13] ExecIf("SIP/1000-0000003b", "1?Set(GROUP(concurrency_limit)=1000)") in new stack
  815.     -- Executing [s@macro-user-callerid:14] GotoIf("SIP/1000-0000003b", "1?continue") in new stack
  816.     -- Goto (macro-user-callerid,s,27)
  817.     -- Executing [s@macro-user-callerid:27] Set("SIP/1000-0000003b", "CALLERID(number)=1000") in new stack
  818.     -- Executing [s@macro-user-callerid:28] Set("SIP/1000-0000003b", "CALLERID(name)=Claudio") in new stack
  819.     -- Executing [s@macro-user-callerid:29] GotoIf("SIP/1000-0000003b", "0?cnum") in new stack
  820.     -- Executing [s@macro-user-callerid:30] Set("SIP/1000-0000003b", "CDR(cnam)=Claudio") in new stack
  821.     -- Executing [s@macro-user-callerid:31] Set("SIP/1000-0000003b", "CDR(cnum)=1000") in new stack
  822.     -- Executing [s@macro-user-callerid:32] Set("SIP/1000-0000003b", "CHANNEL(language)=en") in new stack
  823.     -- Executing [15192556770@from-internal:2] Gosub("SIP/1000-0000003b", "sub-record-check,s,1(out,15192556770,dontcare)") in new stack
  824.     -- Executing [s@sub-record-check:1] GotoIf("SIP/1000-0000003b", "0?initialized") in new stack
  825.     -- Executing [s@sub-record-check:2] Set("SIP/1000-0000003b", "__REC_STATUS=INITIALIZED") in new stack
  826.     -- Executing [s@sub-record-check:3] Set("SIP/1000-0000003b", "NOW=1491162343") in new stack
  827.     -- Executing [s@sub-record-check:4] Set("SIP/1000-0000003b", "__DAY=02") in new stack
  828.     -- Executing [s@sub-record-check:5] Set("SIP/1000-0000003b", "__MONTH=04") in new stack
  829.     -- Executing [s@sub-record-check:6] Set("SIP/1000-0000003b", "__YEAR=2017") in new stack
  830.     -- Executing [s@sub-record-check:7] Set("SIP/1000-0000003b", "__TIMESTR=20170402-154543") in new stack
  831.     -- Executing [s@sub-record-check:8] Set("SIP/1000-0000003b", "__FROMEXTEN=1000") in new stack
  832.     -- Executing [s@sub-record-check:9] Set("SIP/1000-0000003b", "__MON_FMT=wav") in new stack
  833.     -- Executing [s@sub-record-check:10] NoOp("SIP/1000-0000003b", "Recordings initialized") in new stack
  834.     -- Executing [s@sub-record-check:11] ExecIf("SIP/1000-0000003b", "0?Set(ARG3=dontcare)") in new stack
  835.     -- Executing [s@sub-record-check:12] Set("SIP/1000-0000003b", "REC_POLICY_MODE_SAVE=") in new stack
  836.     -- Executing [s@sub-record-check:13] ExecIf("SIP/1000-0000003b", "0?Set(REC_STATUS=NO)") in new stack
  837.     -- Executing [s@sub-record-check:14] GotoIf("SIP/1000-0000003b", "3?checkaction") in new stack
  838.     -- Goto (sub-record-check,s,17)
  839.     -- Executing [s@sub-record-check:17] GotoIf("SIP/1000-0000003b", "1?sub-record-check,out,1") in new stack
  840.     -- Goto (sub-record-check,out,1)
  841.     -- Executing [out@sub-record-check:1] NoOp("SIP/1000-0000003b", "Outbound Recording Check from 1000 to 15192556770") in new stack
  842.     -- Executing [out@sub-record-check:2] Set("SIP/1000-0000003b", "RECMODE=dontcare") in new stack
  843.     -- Executing [out@sub-record-check:3] ExecIf("SIP/1000-0000003b", "1?Goto(routewins)") in new stack
  844.     -- Goto (sub-record-check,out,7)
  845.     -- Executing [out@sub-record-check:7] Gosub("SIP/1000-0000003b", "recordcheck,1(dontcare,out,15192556770)") in new stack
  846.     -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/1000-0000003b", "Starting recording check against dontcare") in new stack
  847.     -- Executing [recordcheck@sub-record-check:2] Goto("SIP/1000-0000003b", "dontcare") in new stack
  848.     -- Goto (sub-record-check,recordcheck,3)
  849.     -- Executing [recordcheck@sub-record-check:3] Return("SIP/1000-0000003b", "") in new stack
  850.     -- Executing [out@sub-record-check:8] Return("SIP/1000-0000003b", "") in new stack
  851.     -- Executing [15192556770@from-internal:3] Set("SIP/1000-0000003b", "MOHCLASS=default") in new stack
  852.     -- Executing [15192556770@from-internal:4] ExecIf("SIP/1000-0000003b", "1?Set(TRUNKCIDOVERRIDE=root)") in new stack
  853.     -- Executing [15192556770@from-internal:5] Set("SIP/1000-0000003b", "_NODEST=") in new stack
  854.     -- Executing [15192556770@from-internal:6] Macro("SIP/1000-0000003b", "dialout-trunk,2,15192556770,,off") in new stack
  855.     -- Executing [s@macro-dialout-trunk:1] Set("SIP/1000-0000003b", "DIAL_TRUNK=2") in new stack
  856.     -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1000-0000003b", "0?sub-pincheck,s,1()") in new stack
  857.     -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1000-0000003b", "0?disabletrunk,1") in new stack
  858.     -- Executing [s@macro-dialout-trunk:4] Set("SIP/1000-0000003b", "DIAL_NUMBER=15192556770") in new stack
  859.     -- Executing [s@macro-dialout-trunk:5] Set("SIP/1000-0000003b", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
  860.     -- Executing [s@macro-dialout-trunk:6] Set("SIP/1000-0000003b", "OUTBOUND_GROUP=OUT_2") in new stack
  861.     -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1000-0000003b", "1?nomax") in new stack
  862.     -- Goto (macro-dialout-trunk,s,9)
  863.     -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1000-0000003b", "0?skipoutcid") in new stack
  864.     -- Executing [s@macro-dialout-trunk:10] Set("SIP/1000-0000003b", "DIAL_TRUNK_OPTIONS=T") in new stack
  865.     -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1000-0000003b", "outbound-callerid,2") in new stack
  866.     -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1000-0000003b", "0?Set(CALLERPRES(name-pres)=)") in new stack
  867.     -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1000-0000003b", "0?Set(CALLERPRES(num-pres)=)") in new stack
  868.     -- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/1000-0000003b", "0?Set(REALCALLERIDNUM=1000)") in new stack
  869.     -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/1000-0000003b", "1?normcid") in new stack
  870.     -- Goto (macro-outbound-callerid,s,7)
  871.     -- Executing [s@macro-outbound-callerid:7] Set("SIP/1000-0000003b", "USEROUTCID=") in new stack
  872.     -- Executing [s@macro-outbound-callerid:8] Set("SIP/1000-0000003b", "EMERGENCYCID=") in new stack
  873.     -- Executing [s@macro-outbound-callerid:9] Set("SIP/1000-0000003b", "TRUNKOUTCID=") in new stack
  874.     -- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/1000-0000003b", "1?trunkcid") in new stack
  875.     -- Goto (macro-outbound-callerid,s,15)
  876.     -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/1000-0000003b", "0?Set(CALLERID(all)=)") in new stack
  877.     -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/1000-0000003b", "0?Set(CALLERID(all)=)") in new stack
  878.     -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/1000-0000003b", "1?Set(CALLERID(all)=root)") in new stack
  879.     -- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/1000-0000003b", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
  880.     -- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/1000-0000003b", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
  881.     -- Executing [s@macro-outbound-callerid:20] Set("SIP/1000-0000003b", "CDR(outbound_cnum)=") in new stack
  882. [2017-04-02 15:45:43] WARNING[4242]: func_cdr.c:383 cdr_write_callback: CDR requires a value (CDR(variable)=value)
  883. )�Executing [s@macro-outbound-callerid:21] Set("SIP/1000-0000003b", "CDR(outbound_cnam)=root") in new stack
  884.     -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/1000-0000003b", "0?sub-flp-2,s,1()") in new stack
  885.     -- Executing [s@macro-dialout-trunk:13] Set("SIP/1000-0000003b", "OUTNUM=15192556770") in new stack
  886.     -- Executing [s@macro-dialout-trunk:14] Set("SIP/1000-0000003b", "custom=SIP/outgoing") in new stack
  887.     -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1000-0000003b", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
  888.     -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/1000-0000003b", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
  889.     -- Executing [s@macro-dialout-trunk:17] Macro("SIP/1000-0000003b", "dialout-trunk-predial-hook,") in new stack
  890.     -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1000-0000003b", "") in new stack
  891.     -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1000-0000003b", "0?bypass,1") in new stack
  892.     -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/1000-0000003b", "1?Set(CONNECTEDLINE(num,i)=15192556770)") in new stack
  893.     -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/1000-0000003b", "1?Set(CONNECTEDLINE(name,i)=CID:)") in new stack
  894.     -- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/1000-0000003b", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden))") in new stack
  895.     -- Executing [s@macro-dialout-trunk:22] GotoIf("SIP/1000-0000003b", "0?customtrunk") in new stack
  896.     -- Executing [s@macro-dialout-trunk:23] Dial("SIP/1000-0000003b", "SIP/outgoing/15192556770,300,T") in new stack
  897.   == Using SIP RTP TOS bits 184
  898.   == Using SIP RTP CoS mark 5
  899. Audio is at 13584
  900. Adding codec ulaw to SDP
  901. Adding non-codec 0x1 (telephone-event) to SDP
  902. Reliably Transmitting (no NAT) to 74.63.41.218:5060:
  903. INVITE sip:15192556770@newyork.voip.ms SIP/2.0
  904. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK4532d3c2
  905. Max-Forwards: 70
  906. From: "root" <sip:214068@162.210.197.227:5160>;tag=as10a65e11
  907. To: <sip:15192556770@newyork.voip.ms>
  908. Contact: <sip:214068@162.210.197.227:5160>
  909. Call-ID: 3cf8c7f051573907071dcede5263b94e@162.210.197.227:5160
  910. CSeq: 102 INVITE
  911. User-Agent: FPBX-13.0.191.5(13.14.0)
  912. Date: Sun, 02 Apr 2017 19:45:43 GMT
  913. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  914. Supported: replaces, timer
  915. Content-Type: application/sdp
  916. Content-Length: 256
  917.  
  918. v=0
  919. o=root 375939993 375939993 IN IP4 162.210.197.227
  920. s=Asterisk PBX 13.14.0
  921. c=IN IP4 162.210.197.227
  922. t=0 0
  923. m=audio 13584 RTP/AVP 0 101
  924. a=rtpmap:0 PCMU/8000
  925. a=rtpmap:101 telephone-event/8000
  926. a=fmtp:101 0-16
  927. a=ptime:20
  928. a=maxptime:150
  929. a=sendrecv
  930.  
  931. ---
  932.     -- Called SIP/outgoing/15192556770
  933.  
  934. <--- SIP read from UDP:74.63.41.218:5060 --->
  935. SIP/2.0 401 Unauthorized
  936. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK4532d3c2;received=162.210.197.227;rport=5160
  937. From: "root" <sip:214068@162.210.197.227:5160>;tag=as10a65e11
  938. To: <sip:15192556770@newyork.voip.ms>;tag=as020ce5e8
  939. Call-ID: 3cf8c7f051573907071dcede5263b94e@162.210.197.227:5160
  940. CSeq: 102 INVITE
  941. Server: voip.ms
  942. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  943. Supported: replaces, timer
  944. WWW-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", nonce="43231d41"
  945. Content-Length: 0
  946.  
  947. <------------->
  948. --- (11 headers 0 lines) ---
  949. Transmitting (no NAT) to 74.63.41.218:5060:
  950. ACK sip:15192556770@newyork.voip.ms SIP/2.0
  951. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK4532d3c2
  952. Max-Forwards: 70
  953. From: "root" <sip:214068@162.210.197.227:5160>;tag=as10a65e11
  954. To: <sip:15192556770@newyork.voip.ms>;tag=as020ce5e8
  955. Contact: <sip:214068@162.210.197.227:5160>
  956. Call-ID: 3cf8c7f051573907071dcede5263b94e@162.210.197.227:5160
  957. CSeq: 102 ACK
  958. User-Agent: FPBX-13.0.191.5(13.14.0)
  959. Content-Length: 0
  960.  
  961.  
  962. ---
  963. Audio is at 13584
  964. Adding codec ulaw to SDP
  965. Adding non-codec 0x1 (telephone-event) to SDP
  966. Reliably Transmitting (no NAT) to 74.63.41.218:5060:
  967. INVITE sip:15192556770@newyork.voip.ms SIP/2.0
  968. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK1141ecfc
  969. Max-Forwards: 70
  970. From: "root" <sip:214068@162.210.197.227:5160>;tag=as10a65e11
  971. To: <sip:15192556770@newyork.voip.ms>
  972. Contact: <sip:214068@162.210.197.227:5160>
  973. Call-ID: 3cf8c7f051573907071dcede5263b94e@162.210.197.227:5160
  974. CSeq: 103 INVITE
  975. User-Agent: FPBX-13.0.191.5(13.14.0)
  976. Authorization: Digest username="214068", realm="newyork.voip.ms", algorithm=MD5, uri="sip:15192556770@newyork.voip.ms", nonce="43231d41", response="94bac0bb4c6d12074a865c55900b4025"
  977. Date: Sun, 02 Apr 2017 19:45:43 GMT
  978. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  979. Supported: replaces, timer
  980. Content-Type: application/sdp
  981. Content-Length: 256
  982.  
  983. v=0
  984. o=root 375939993 375939994 IN IP4 162.210.197.227
  985. s=Asterisk PBX 13.14.0
  986. c=IN IP4 162.210.197.227
  987. t=0 0
  988. m=audio 13584 RTP/AVP 0 101
  989. a=rtpmap:0 PCMU/8000
  990. a=rtpmap:101 telephone-event/8000
  991. a=fmtp:101 0-16
  992. a=ptime:20
  993. a=maxptime:150
  994. a=sendrecv
  995.  
  996. ---
  997.  
  998. <--- SIP read from UDP:74.63.41.218:5060 --->
  999. SIP/2.0 100 Trying
  1000. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK1141ecfc;received=162.210.197.227;rport=5160
  1001. From: "root" <sip:214068@162.210.197.227:5160>;tag=as10a65e11
  1002. To: <sip:15192556770@newyork.voip.ms>
  1003. Call-ID: 3cf8c7f051573907071dcede5263b94e@162.210.197.227:5160
  1004. CSeq: 103 INVITE
  1005. Server: voip.ms
  1006. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1007. Supported: replaces, timer
  1008. Session-Expires: 1800;refresher=uas
  1009. Contact: <sip:15192556770@74.63.41.218:5060>
  1010. Content-Length: 0
  1011.  
  1012. <------------->
  1013. --- (12 headers 0 lines) ---
  1014.  
  1015. <--- SIP read from UDP:74.63.41.218:5060 --->
  1016. SIP/2.0 503 Service Unavailable
  1017. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK1141ecfc;received=162.210.197.227;rport=5160
  1018. From: "root" <sip:214068@162.210.197.227:5160>;tag=as10a65e11
  1019. To: <sip:15192556770@newyork.voip.ms>;tag=as2845d8df
  1020. Call-ID: 3cf8c7f051573907071dcede5263b94e@162.210.197.227:5160
  1021. CSeq: 103 INVITE
  1022. Server: voip.ms
  1023. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1024. Supported: replaces, timer
  1025. Session-Expires: 1800;refresher=uas
  1026. Content-Length: 0
  1027.  
  1028. <------------->
  1029. --- (11 headers 0 lines) ---
  1030.     -- Got SIP response 503 "Service Unavailable" back from 74.63.41.218:5060
  1031. Transmitting (no NAT) to 74.63.41.218:5060:
  1032. ACK sip:15192556770@newyork.voip.ms SIP/2.0
  1033. Via: SIP/2.0/UDP 162.210.197.227:5160;branch=z9hG4bK1141ecfc
  1034. Max-Forwards: 70
  1035. From: "root" <sip:214068@162.210.197.227:5160>;tag=as10a65e11
  1036. To: <sip:15192556770@newyork.voip.ms>;tag=as2845d8df
  1037. Contact: <sip:214068@162.210.197.227:5160>
  1038. Call-ID: 3cf8c7f051573907071dcede5263b94e@162.210.197.227:5160
  1039. CSeq: 103 ACK
  1040. User-Agent: FPBX-13.0.191.5(13.14.0)
  1041. Content-Length: 0
  1042.  
  1043.  
  1044. ---
  1045.     -- SIP/outgoing-0000003c is circuit-busy
  1046.   == Everyone is busy/congested at this time (1:0/1/0)
  1047.     -- Executing [s@macro-dialout-trunk:24] NoOp("SIP/1000-0000003b", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack
  1048.     -- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/1000-0000003b", "0?continue,1:s-CONGESTION,1") in new stack
  1049.     -- Goto (macro-dialout-trunk,s-CONGESTION,1)
  1050.     -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/1000-0000003b", "RC=34") in new stack
  1051.     -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/1000-0000003b", "34,1") in new stack
  1052.     -- Goto (macro-dialout-trunk,34,1)
  1053.     -- Executing [34@macro-dialout-trunk:1] Goto("SIP/1000-0000003b", "continue,1") in new stack
  1054.     -- Goto (macro-dialout-trunk,continue,1)
  1055.     -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/1000-0000003b", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks") in new stack
  1056.     -- Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/1000-0000003b", "1?Set(CALLERID(number)=1000)") in new stack
  1057.     -- Executing [15192556770@from-internal:7] Macro("SIP/1000-0000003b", "outisbusy,") in new stack
  1058. [2017-04-02 15:45:43] WARNING[10984][C-00000020]: app_macro.c:310 _macro_exec: No such context 'macro-outisbusy' for macro 'outisbusy'. Was called by 15192556770@from-internal
  1059.     -- Executing [15192556770@from-internal:8] Hangup("SIP/1000-0000003b", "") in new stack
  1060.   == Spawn extension (from-internal, 15192556770, 8) exited non-zero on 'SIP/1000-0000003b'
  1061.     -- Executing [h@from-internal:1] Macro("SIP/1000-0000003b", "hangupcall") in new stack
  1062.     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1000-0000003b", "1?theend") in new stack
  1063.     -- Goto (macro-hangupcall,s,3)
  1064.     -- Executing [s@macro-hangupcall:3] ExecIf("SIP/1000-0000003b", "0?Set(CDR(recordingfile)=)") in new stack
  1065.     -- Executing [s@macro-hangupcall:4] Hangup("SIP/1000-0000003b", "") in new stack
  1066.   == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/1000-0000003b' in macro 'hangupcall'
  1067.   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1000-0000003b'
  1068. Scheduling destruction of SIP dialog '1655555233' in 6400 ms (Method: INVITE)
  1069.  
  1070. <--- Reliably Transmitting (NAT) to 70.27.246.146:5060 --->
  1071. SIP/2.0 503 Service Unavailable
  1072. Via: SIP/2.0/UDP 192.168.2.4:5060;branch=z9hG4bK1688908289;received=70.27.246.146;rport=5060
  1073. From: <sip:1000@162.210.197.227:5160>;tag=965951055
  1074. To: <sip:15192556770@162.210.197.227:5160>;tag=as3adb65b7
  1075. Call-ID: 1655555233
  1076. CSeq: 21 INVITE
  1077. Server: FPBX-13.0.191.5(13.14.0)
  1078. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1079. Supported: replaces, timer
  1080. Content-Length: 0
  1081.  
  1082.  
  1083. <------------>
  1084. Really destroying SIP dialog '3cf8c7f051573907071dcede5263b94e@162.210.197.227:5160' Method: INVITE
  1085.  
  1086. <--- SIP read from UDP:70.27.246.146:5060 --->
  1087. ACK sip:15192556770@162.210.197.227:5160 SIP/2.0
  1088. Via: SIP/2.0/UDP 192.168.2.4:5060;rport;branch=z9hG4bK1688908289
  1089. From: <sip:1000@162.210.197.227:5160>;tag=965951055
  1090. To: <sip:15192556770@162.210.197.227:5160>;tag=as3adb65b7
  1091. Call-ID: 1655555233
  1092. CSeq: 21 ACK
  1093. Content-Length: 0
  1094.  
  1095. <------------->
  1096. --- (7 headers 0 lines) ---
  1097. localhost*CLI> quit
  1098. Asterisk cleanly ending (0).
  1099. Executing last minute cleanups
  1100. [root@localhost ~]#
  1101.  

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