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  1. [root@voip ~]# asterisk -rvvvvvvvvv
  2. Asterisk 11.16.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
  3. Created by Mark Spencer <markster@digium.com>
  4. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  5. This is free software, with components licensed under the GNU General Public
  6. License version 2 and other licenses; you are welcome to redistribute it under
  7. certain conditions. Type 'core show license' for details.
  8. =========================================================================
  9. Connected to Asterisk 11.16.0 currently running on voip (pid = 2151)
  10. voip*CLI> sip set debug on
  11. SIP Debugging re-enabled
  12.  
  13. <--- SIP read from UDP:176.47.75.56:1201 --->
  14.  
  15.  
  16. <------------->
  17.  
  18. <--- SIP read from UDP:192.168.112.22:5060 --->
  19. INVITE sip:101@192.168.112.50;user=phone SIP/2.0
  20. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK151315024312073383
  21. From: 103 <sip:103@192.168.112.50:5060>;tag=3092610502
  22. To: "101" <sip:101@192.168.112.50;user=phone>
  23. Call-ID: 5120292566874-14872666530126@192.168.112.22
  24. CSeq: 1 INVITE
  25. Contact: <sip:103@192.168.112.22:5060>
  26. Max-Forwards: 70
  27. Supported: replaces, join, path
  28. User-Agent: DLINK DPH-400SE FRU2.2.876.445
  29. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
  30. Content-Type: application/sdp
  31. Content-Length: 346
  32.  
  33. v=0
  34. o=103 3271720515 2414021518 IN IP4 192.168.112.22
  35. s=A conversation
  36. c=IN IP4 192.168.112.22
  37. t=0 0
  38. m=audio 10016 RTP/AVP 8 0 9 4 2 18 101
  39. a=rtpmap:8 PCMA/8000
  40. a=rtpmap:0 PCMU/8000
  41. a=rtpmap:9 G722/8000
  42. a=rtpmap:4 G723/8000
  43. a=rtpmap:2 G726-32/8000
  44. a=rtpmap:18 G729/8000
  45. a=rtpmap:101 telephone-event/8000
  46. a=fmtp:101 0-15
  47. a=sendrecv
  48. <------------->
  49. --- (13 headers 15 lines) ---
  50. Sending to 192.168.112.22:5060 (NAT)
  51. Sending to 192.168.112.22:5060 (NAT)
  52. Using INVITE request as basis request - 5120292566874-14872666530126@192.168.112.22
  53. Found peer '103' for '103' from 192.168.112.22:5060
  54.  
  55. <--- Reliably Transmitting (NAT) to 192.168.112.22:5060 --->
  56. SIP/2.0 401 Unauthorized
  57. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK151315024312073383;received=192.168.112.22;rport=5060
  58. From: 103 <sip:103@192.168.112.50:5060>;tag=3092610502
  59. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as5be4054f
  60. Call-ID: 5120292566874-14872666530126@192.168.112.22
  61. CSeq: 1 INVITE
  62. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  63. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  64. Supported: replaces, timer
  65. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="04472d90"
  66. Content-Length: 0
  67.  
  68.  
  69. <------------>
  70. Scheduling destruction of SIP dialog '5120292566874-14872666530126@192.168.112.22' in 6400 ms (Method: INVITE)
  71.  
  72. <--- SIP read from UDP:192.168.112.22:5060 --->
  73. ACK sip:101@192.168.112.50;user=phone SIP/2.0
  74. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK151315024312073383
  75. From: 103 <sip:103@192.168.112.50:5060>;tag=3092610502
  76. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as5be4054f
  77. Call-ID: 5120292566874-14872666530126@192.168.112.22
  78. CSeq: 1 ACK
  79. Contact: <sip:103@192.168.112.22:5060>
  80. Max-Forwards: 70
  81. Content-Length: 0
  82.  
  83. <------------->
  84. --- (9 headers 0 lines) ---
  85.  
  86. <--- SIP read from UDP:192.168.112.22:5060 --->
  87. INVITE sip:101@192.168.112.50;user=phone SIP/2.0
  88. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK23234170832238621948
  89. From: 103 <sip:103@192.168.112.50:5060>;tag=3092610502
  90. To: "101" <sip:101@192.168.112.50;user=phone>
  91. Call-ID: 5120292566874-14872666530126@192.168.112.22
  92. CSeq: 2 INVITE
  93. Contact: <sip:103@192.168.112.22:5060>
  94. Authorization: Digest username="103", realm="asterisk", nonce="04472d90", uri="sip:101@192.168.112.50;user=phone", response="c79361a61e3dad5b7c77a1a100ce5b25", algorithm=MD5
  95. Max-Forwards: 70
  96. Supported: replaces, join, path
  97. User-Agent: DLINK DPH-400SE FRU2.2.876.445
  98. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
  99. Content-Type: application/sdp
  100. Content-Length: 346
  101.  
  102. v=0
  103. o=103 3271720515 2414021518 IN IP4 192.168.112.22
  104. s=A conversation
  105. c=IN IP4 192.168.112.22
  106. t=0 0
  107. m=audio 10016 RTP/AVP 8 0 9 4 2 18 101
  108. a=rtpmap:8 PCMA/8000
  109. a=rtpmap:0 PCMU/8000
  110. a=rtpmap:9 G722/8000
  111. a=rtpmap:4 G723/8000
  112. a=rtpmap:2 G726-32/8000
  113. a=rtpmap:18 G729/8000
  114. a=rtpmap:101 telephone-event/8000
  115. a=fmtp:101 0-15
  116. a=sendrecv
  117. <------------->
  118. --- (14 headers 15 lines) ---
  119. Sending to 192.168.112.22:5060 (NAT)
  120. Using INVITE request as basis request - 5120292566874-14872666530126@192.168.112.22
  121. Found peer '103' for '103' from 192.168.112.22:5060
  122.   == Using SIP RTP TOS bits 184
  123.   == Using SIP RTP CoS mark 5
  124. Found RTP audio format 8
  125. Found RTP audio format 0
  126. Found RTP audio format 9
  127. Found RTP audio format 4
  128. Found RTP audio format 2
  129. Found RTP audio format 18
  130. Found RTP audio format 101
  131. Found audio description format PCMA for ID 8
  132. Found audio description format PCMU for ID 0
  133. Found audio description format G722 for ID 9
  134. Found audio description format G723 for ID 4
  135. Found audio description format G726-32 for ID 2
  136. Found audio description format G729 for ID 18
  137. Found audio description format telephone-event for ID 101
  138. Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(g723|ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726)
  139. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  140. Peer audio RTP is at port 192.168.112.22:10016
  141. Looking for 101 in from-internal (domain 192.168.112.50)
  142. list_route: hop: <sip:103@192.168.112.22:5060>
  143.  
  144. <--- Transmitting (NAT) to 192.168.112.22:5060 --->
  145. SIP/2.0 100 Trying
  146. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK23234170832238621948;received=192.168.112.22;rport=5060
  147. From: 103 <sip:103@192.168.112.50:5060>;tag=3092610502
  148. To: "101" <sip:101@192.168.112.50;user=phone>
  149. Call-ID: 5120292566874-14872666530126@192.168.112.22
  150. CSeq: 2 INVITE
  151. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  152. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  153. Supported: replaces, timer
  154. Contact: <sip:101@192.168.112.50:5060>
  155. Content-Length: 0
  156.  
  157.  
  158. <------------>
  159.     -- Executing [101@from-internal:1] GotoIf("SIP/103-00000010", "1?ext-local,101,1") in new stack
  160.     -- Goto (ext-local,101,1)
  161.     -- Executing [101@ext-local:1] Set("SIP/103-00000010", "__RINGTIMER=15") in new stack
  162.     -- Executing [101@ext-local:2] Macro("SIP/103-00000010", "exten-vm,novm,101,0,0,0") in new stack
  163.     -- Executing [s@macro-exten-vm:1] Macro("SIP/103-00000010", "user-callerid,") in new stack
  164.     -- Executing [s@macro-user-callerid:1] Set("SIP/103-00000010", "TOUCH_MONITOR=1489526970.16") in new stack
  165.     -- Executing [s@macro-user-callerid:2] Set("SIP/103-00000010", "AMPUSER=103") in new stack
  166.     -- Executing [s@macro-user-callerid:3] GotoIf("SIP/103-00000010", "0?report") in new stack
  167.     -- Executing [s@macro-user-callerid:4] ExecIf("SIP/103-00000010", "1?Set(REALCALLERIDNUM=103)") in new stack
  168.     -- Executing [s@macro-user-callerid:5] Set("SIP/103-00000010", "AMPUSER=103") in new stack
  169.     -- Executing [s@macro-user-callerid:6] GotoIf("SIP/103-00000010", "0?limit") in new stack
  170.     -- Executing [s@macro-user-callerid:7] Set("SIP/103-00000010", "AMPUSERCIDNAME=Zubair Laptop") in new stack
  171.     -- Executing [s@macro-user-callerid:8] GotoIf("SIP/103-00000010", "0?report") in new stack
  172.     -- Executing [s@macro-user-callerid:9] Set("SIP/103-00000010", "AMPUSERCID=103") in new stack
  173.     -- Executing [s@macro-user-callerid:10] Set("SIP/103-00000010", "__DIAL_OPTIONS=Ttr") in new stack
  174.     -- Executing [s@macro-user-callerid:11] Set("SIP/103-00000010", "CALLERID(all)="Zubair Laptop" <103>") in new stack
  175.     -- Executing [s@macro-user-callerid:12] GotoIf("SIP/103-00000010", "0?limit") in new stack
  176.     -- Executing [s@macro-user-callerid:13] ExecIf("SIP/103-00000010", "0?Set(GROUP(concurrency_limit)=103)") in new stack
  177.     -- Executing [s@macro-user-callerid:14] GosubIf("SIP/103-00000010", "7?sub-ccss,s,1(macro-exten-vm,101)") in new stack
  178.     -- Executing [s@sub-ccss:1] ExecIf("SIP/103-00000010", "0?Return()") in new stack
  179.     -- Executing [s@sub-ccss:2] Set("SIP/103-00000010", "CCSS_SETUP=TRUE") in new stack
  180.     -- Executing [s@sub-ccss:3] GosubIf("SIP/103-00000010", "0?monitor_config,1(macro-exten-vm,101):monitor_default,1(macro-exten-vm,101)") in new stack
  181.     -- Executing [monitor_default@sub-ccss:1] GotoIf("SIP/103-00000010", "1?is_exten") in new stack
  182.     -- Goto (sub-ccss,monitor_default,4)
  183.     -- Executing [monitor_default@sub-ccss:4] Set("SIP/103-00000010", "CALLCOMPLETION(cc_monitor_policy)=generic") in new stack
  184.     -- Executing [monitor_default@sub-ccss:5] Set("SIP/103-00000010", "CALLCOMPLETION(cc_max_monitors)=5") in new stack
  185.     -- Executing [monitor_default@sub-ccss:6] Return("SIP/103-00000010", "TRUE") in new stack
  186.     -- Executing [s@sub-ccss:4] GosubIf("SIP/103-00000010", "7?agent_config,1():agent_default,1()") in new stack
  187.     -- Executing [agent_config@sub-ccss:1] Set("SIP/103-00000010", "CALLCOMPLETION(cc_agent_policy)=generic") in new stack
  188.     -- Executing [agent_config@sub-ccss:2] Set("SIP/103-00000010", "CALLCOMPLETION(cc_offer_timer)=30") in new stack
  189.     -- Executing [agent_config@sub-ccss:3] Set("SIP/103-00000010", "CALLCOMPLETION(ccbs_available_timer)=") in new stack
  190. [2017-03-15 00:29:30] WARNING[6855][C-00000008]: ccss.c:948 ast_set_ccbs_available_timer: 0 is an invalid value for ccbs_available_timer. Retaining value as 4800
  191.     -- Executing [agent_config@sub-ccss:4] Set("SIP/103-00000010", "CALLCOMPLETION(ccnr_available_timer)=") in new stack
  192. [2017-03-15 00:29:30] WARNING[6855][C-00000008]: ccss.c:918 ast_set_ccnr_available_timer: 0 is an invalid value for ccnr_available_timer. Retaining value as 7200
  193.     -- Executing [agent_config@sub-ccss:5] Set("SIP/103-00000010", "CALLCOMPLETION(cc_callback_macro)=ccss-default") in new stack
  194. [2017-03-15 00:29:30] WARNING[6855][C-00000008]: ccss.c:1000 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated.  Please use cc_callback_sub instead.
  195.     -- Executing [agent_config@sub-ccss:6] ExecIf("SIP/103-00000010", "1?Set(CALLCOMPLETION(cc_recall_timer)=)") in new stack
  196. [2017-03-15 00:29:30] WARNING[6855][C-00000008]: ccss.c:933 ast_set_cc_recall_timer: 0 is an invalid value for ccnr_available_timer. Retaining value as 20
  197.     -- Executing [agent_config@sub-ccss:7] ExecIf("SIP/103-00000010", "1?Set(CALLCOMPLETION(cc_max_agents)=)") in new stack
  198.     -- Executing [agent_config@sub-ccss:8] ExecIf("SIP/103-00000010", "0?Set(CALLCOMPLETION(cc_agent_dialstring)=Local/103_101@from-ccss-)") in new stack
  199.     -- Executing [agent_config@sub-ccss:9] Set("SIP/103-00000010", "CALLCOMPLETION(cc_callback_macro)=ccss-default") in new stack
  200. [2017-03-15 00:29:30] WARNING[6855][C-00000008]: ccss.c:1000 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated.  Please use cc_callback_sub instead.
  201.     -- Executing [agent_config@sub-ccss:10] Return("SIP/103-00000010", "") in new stack
  202.     -- Executing [s@sub-ccss:5] Set("SIP/103-00000010", "DB(AMPUSER/103/ccss/last_number)=101") in new stack
  203.     -- Executing [s@sub-ccss:6] Return("SIP/103-00000010", "") in new stack
  204.     -- Executing [s@macro-user-callerid:15] ExecIf("SIP/103-00000010", "0?Set(CHANNEL(language)=)") in new stack
  205.     -- Executing [s@macro-user-callerid:16] GotoIf("SIP/103-00000010", "0?continue") in new stack
  206.     -- Executing [s@macro-user-callerid:17] ExecIf("SIP/103-00000010", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
  207.     -- Executing [s@macro-user-callerid:18] Set("SIP/103-00000010", "__TTL=64") in new stack
  208.     -- Executing [s@macro-user-callerid:19] GotoIf("SIP/103-00000010", "1?continue") in new stack
  209.     -- Goto (macro-user-callerid,s,30)
  210.     -- Executing [s@macro-user-callerid:30] Set("SIP/103-00000010", "CALLERID(number)=103") in new stack
  211.     -- Executing [s@macro-user-callerid:31] Set("SIP/103-00000010", "CALLERID(name)=Zubair Laptop") in new stack
  212.     -- Executing [s@macro-user-callerid:32] Set("SIP/103-00000010", "CDR(cnum)=103") in new stack
  213.     -- Executing [s@macro-user-callerid:33] Set("SIP/103-00000010", "CDR(cnam)=Zubair Laptop") in new stack
  214.     -- Executing [s@macro-user-callerid:34] Set("SIP/103-00000010", "CHANNEL(language)=en") in new stack
  215.     -- Executing [s@macro-exten-vm:2] Set("SIP/103-00000010", "RingGroupMethod=none") in new stack
  216.     -- Executing [s@macro-exten-vm:3] Set("SIP/103-00000010", "__EXTTOCALL=101") in new stack
  217.     -- Executing [s@macro-exten-vm:4] Set("SIP/103-00000010", "__PICKUPMARK=101") in new stack
  218.     -- Executing [s@macro-exten-vm:5] Set("SIP/103-00000010", "RT=") in new stack
  219.     -- Executing [s@macro-exten-vm:6] ExecIf("SIP/103-00000010", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
  220.     -- Executing [s@macro-exten-vm:7] ExecIf("SIP/103-00000010", "0?MacroExit()") in new stack
  221.     -- Executing [s@macro-exten-vm:8] Gosub("SIP/103-00000010", "sub-record-check,s,1(exten,101,dontcare)") in new stack
  222.     -- Executing [s@sub-record-check:1] GotoIf("SIP/103-00000010", "0?initialized") in new stack
  223.     -- Executing [s@sub-record-check:2] Set("SIP/103-00000010", "__REC_STATUS=INITIALIZED") in new stack
  224.     -- Executing [s@sub-record-check:3] Set("SIP/103-00000010", "NOW=1489526970") in new stack
  225.     -- Executing [s@sub-record-check:4] Set("SIP/103-00000010", "__DAY=15") in new stack
  226.     -- Executing [s@sub-record-check:5] Set("SIP/103-00000010", "__MONTH=03") in new stack
  227.     -- Executing [s@sub-record-check:6] Set("SIP/103-00000010", "__YEAR=2017") in new stack
  228.     -- Executing [s@sub-record-check:7] Set("SIP/103-00000010", "__TIMESTR=20170315-002930") in new stack
  229.     -- Executing [s@sub-record-check:8] Set("SIP/103-00000010", "__FROMEXTEN=103") in new stack
  230.     -- Executing [s@sub-record-check:9] Set("SIP/103-00000010", "__MON_FMT=wav") in new stack
  231.     -- Executing [s@sub-record-check:10] NoOp("SIP/103-00000010", "Recordings initialized") in new stack
  232.     -- Executing [s@sub-record-check:11] ExecIf("SIP/103-00000010", "0?Set(ARG3=dontcare)") in new stack
  233.     -- Executing [s@sub-record-check:12] Set("SIP/103-00000010", "REC_POLICY_MODE_SAVE=") in new stack
  234.     -- Executing [s@sub-record-check:13] ExecIf("SIP/103-00000010", "0?Set(REC_STATUS=NO)") in new stack
  235.     -- Executing [s@sub-record-check:14] GotoIf("SIP/103-00000010", "5?checkaction") in new stack
  236.     -- Goto (sub-record-check,s,17)
  237.     -- Executing [s@sub-record-check:17] GotoIf("SIP/103-00000010", "1?sub-record-check,exten,1") in new stack
  238.     -- Goto (sub-record-check,exten,1)
  239.     -- Executing [exten@sub-record-check:1] NoOp("SIP/103-00000010", "Exten Recording Check between 103 and 101") in new stack
  240.     -- Executing [exten@sub-record-check:2] Set("SIP/103-00000010", "CALLTYPE=internal") in new stack
  241.     -- Executing [exten@sub-record-check:3] ExecIf("SIP/103-00000010", "0?Set(CALLTYPE=)") in new stack
  242.     -- Executing [exten@sub-record-check:4] Set("SIP/103-00000010", "CALLEE=dontcare") in new stack
  243.     -- Executing [exten@sub-record-check:5] ExecIf("SIP/103-00000010", "0?Set(CALLEE=dontcare)") in new stack
  244.     -- Executing [exten@sub-record-check:6] GotoIf("SIP/103-00000010", "0?callee") in new stack
  245.     -- Executing [exten@sub-record-check:7] GotoIf("SIP/103-00000010", "1?caller") in new stack
  246.     -- Goto (sub-record-check,exten,13)
  247.     -- Executing [exten@sub-record-check:13] Set("SIP/103-00000010", "RECMODE=dontcare") in new stack
  248.     -- Executing [exten@sub-record-check:14] ExecIf("SIP/103-00000010", "0?Set(RECMODE=dontcare)") in new stack
  249.     -- Executing [exten@sub-record-check:15] ExecIf("SIP/103-00000010", "1?Set(RECMODE=dontcare)") in new stack
  250.     -- Executing [exten@sub-record-check:16] Gosub("SIP/103-00000010", "recordcheck,1(dontcare,internal,101)") in new stack
  251.     -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/103-00000010", "Starting recording check against dontcare") in new stack
  252.     -- Executing [recordcheck@sub-record-check:2] Goto("SIP/103-00000010", "dontcare") in new stack
  253.     -- Goto (sub-record-check,recordcheck,3)
  254.     -- Executing [recordcheck@sub-record-check:3] Return("SIP/103-00000010", "") in new stack
  255.     -- Executing [exten@sub-record-check:17] Return("SIP/103-00000010", "") in new stack
  256.     -- Executing [s@macro-exten-vm:9] GotoIf("SIP/103-00000010", "1?macrodial") in new stack
  257.     -- Goto (macro-exten-vm,s,15)
  258.     -- Executing [s@macro-exten-vm:15] GosubIf("SIP/103-00000010", "0?clrheader,1()") in new stack
  259.     -- Executing [s@macro-exten-vm:16] Macro("SIP/103-00000010", "dial-one,,Ttr,101") in new stack
  260.     -- Executing [s@macro-dial-one:1] Set("SIP/103-00000010", "DEXTEN=101") in new stack
  261.     -- Executing [s@macro-dial-one:2] Set("SIP/103-00000010", "DIALSTATUS_CW=") in new stack
  262.     -- Executing [s@macro-dial-one:3] GosubIf("SIP/103-00000010", "0?screen,1()") in new stack
  263.     -- Executing [s@macro-dial-one:4] GosubIf("SIP/103-00000010", "0?cf,1()") in new stack
  264.     -- Executing [s@macro-dial-one:5] GotoIf("SIP/103-00000010", "1?skip1") in new stack
  265.     -- Goto (macro-dial-one,s,8)
  266.     -- Executing [s@macro-dial-one:8] GotoIf("SIP/103-00000010", "0?nodial") in new stack
  267.     -- Executing [s@macro-dial-one:9] GotoIf("SIP/103-00000010", "0?continue") in new stack
  268.     -- Executing [s@macro-dial-one:10] Set("SIP/103-00000010", "EXTHASCW=ENABLED") in new stack
  269.     -- Executing [s@macro-dial-one:11] GotoIf("SIP/103-00000010", "0?next1:cwinusebusy") in new stack
  270.     -- Goto (macro-dial-one,s,23)
  271.     -- Executing [s@macro-dial-one:23] GotoIf("SIP/103-00000010", "1?next3:continue") in new stack
  272.     -- Goto (macro-dial-one,s,24)
  273.     -- Executing [s@macro-dial-one:24] ExecIf("SIP/103-00000010", "0?Set(DIALSTATUS_CW=BUSY)") in new stack
  274.     -- Executing [s@macro-dial-one:25] GotoIf("SIP/103-00000010", "0?nodial") in new stack
  275.     -- Executing [s@macro-dial-one:26] GosubIf("SIP/103-00000010", "1?dstring,1():dlocal,1()") in new stack
  276.     -- Executing [dstring@macro-dial-one:1] Set("SIP/103-00000010", "DSTRING=") in new stack
  277.     -- Executing [dstring@macro-dial-one:2] Set("SIP/103-00000010", "DEVICES=101") in new stack
  278.     -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/103-00000010", "0?Return()") in new stack
  279.     -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/103-00000010", "0?Set(DEVICES=01)") in new stack
  280.     -- Executing [dstring@macro-dial-one:5] Set("SIP/103-00000010", "LOOPCNT=1") in new stack
  281.     -- Executing [dstring@macro-dial-one:6] Set("SIP/103-00000010", "ITER=1") in new stack
  282.     -- Executing [dstring@macro-dial-one:7] Set("SIP/103-00000010", "THISDIAL=SIP/101") in new stack
  283.     -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/103-00000010", "1?zap2dahdi,1()") in new stack
  284.     -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/103-00000010", "0?Return()") in new stack
  285.     -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/103-00000010", "NEWDIAL=") in new stack
  286.     -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/103-00000010", "LOOPCNT2=1") in new stack
  287.     -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/103-00000010", "ITER2=1") in new stack
  288.     -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/103-00000010", "THISPART2=SIP/101") in new stack
  289.     -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/103-00000010", "0?Set(THISPART2=DAHDI/101)") in new stack
  290.     -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/103-00000010", "NEWDIAL=SIP/101&") in new stack
  291.     -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/103-00000010", "ITER2=2") in new stack
  292.     -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/103-00000010", "0?begin2") in new stack
  293.     -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/103-00000010", "THISDIAL=SIP/101") in new stack
  294.     -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/103-00000010", "") in new stack
  295.     -- Executing [dstring@macro-dial-one:9] GotoIf("SIP/103-00000010", "1?doset") in new stack
  296.     -- Goto (macro-dial-one,dstring,13)
  297.     -- Executing [dstring@macro-dial-one:13] Set("SIP/103-00000010", "DSTRING=SIP/101&") in new stack
  298.     -- Executing [dstring@macro-dial-one:14] Set("SIP/103-00000010", "ITER=2") in new stack
  299.     -- Executing [dstring@macro-dial-one:15] GotoIf("SIP/103-00000010", "0?begin") in new stack
  300.     -- Executing [dstring@macro-dial-one:16] ExecIf("SIP/103-00000010", "0?Return()") in new stack
  301.     -- Executing [dstring@macro-dial-one:17] Set("SIP/103-00000010", "DSTRING=SIP/101") in new stack
  302.     -- Executing [dstring@macro-dial-one:18] Return("SIP/103-00000010", "") in new stack
  303.     -- Executing [s@macro-dial-one:27] GotoIf("SIP/103-00000010", "0?nodial") in new stack
  304.     -- Executing [s@macro-dial-one:28] GotoIf("SIP/103-00000010", "0?skiptrace") in new stack
  305.     -- Executing [s@macro-dial-one:29] GosubIf("SIP/103-00000010", "1?ctset,1():ctclear,1()") in new stack
  306.     -- Executing [ctset@macro-dial-one:1] Set("SIP/103-00000010", "DB(CALLTRACE/101)=103") in new stack
  307.     -- Executing [ctset@macro-dial-one:2] Return("SIP/103-00000010", "") in new stack
  308.     -- Executing [s@macro-dial-one:30] Set("SIP/103-00000010", "D_OPTIONS=Ttr") in new stack
  309.     -- Executing [s@macro-dial-one:31] ExecIf("SIP/103-00000010", "0?SIPAddHeader(Alert-Info: )") in new stack
  310.     -- Executing [s@macro-dial-one:32] ExecIf("SIP/103-00000010", "0?SIPAddHeader()") in new stack
  311.     -- Executing [s@macro-dial-one:33] ExecIf("SIP/103-00000010", "0?Set(CHANNEL(musicclass)=)") in new stack
  312.     -- Executing [s@macro-dial-one:34] GosubIf("SIP/103-00000010", "0?qwait,1()") in new stack
  313.     -- Executing [s@macro-dial-one:35] Set("SIP/103-00000010", "__CWIGNORE=") in new stack
  314.     -- Executing [s@macro-dial-one:36] Set("SIP/103-00000010", "__KEEPCID=TRUE") in new stack
  315.     -- Executing [s@macro-dial-one:37] GotoIf("SIP/103-00000010", "0?usegoto,1") in new stack
  316.     -- Executing [s@macro-dial-one:38] GotoIf("SIP/103-00000010", "0?godial") in new stack
  317.     -- Executing [s@macro-dial-one:39] Gosub("SIP/103-00000010", "sub-presencestate-display,s,1(101)") in new stack
  318. [2017-03-15 00:29:30] WARNING[6855][C-00000008]: func_presencestate.c:132 presence_read: PRESENCE_STATE unknown
  319.     -- Executing [s@sub-presencestate-display:1] Goto("SIP/103-00000010", "state-,1") in new stack
  320.     -- Goto (sub-presencestate-display,state-,1)
  321.     -- Executing [state-@sub-presencestate-display:1] Set("SIP/103-00000010", "PRESENCESTATE_DISPLAY=") in new stack
  322.     -- Executing [state-@sub-presencestate-display:2] Return("SIP/103-00000010", "") in new stack
  323.     -- Executing [s@macro-dial-one:40] Set("SIP/103-00000010", "CONNECTEDLINE(name,i)=Home IP Phone") in new stack
  324.     -- Executing [s@macro-dial-one:41] Set("SIP/103-00000010", "CONNECTEDLINE(num)=101") in new stack
  325.     -- Executing [s@macro-dial-one:42] Set("SIP/103-00000010", "D_OPTIONS=TtrI") in new stack
  326.     -- Executing [s@macro-dial-one:43] Macro("SIP/103-00000010", "dialout-one-predial-hook,") in new stack
  327.     -- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("SIP/103-00000010", "") in new stack
  328.     -- Executing [s@macro-dial-one:44] ExecIf("SIP/103-00000010", "0?Set(D_OPTIONS=trII)") in new stack
  329.     -- Executing [s@macro-dial-one:45] Dial("SIP/103-00000010", "SIP/101,,TtrI") in new stack
  330.   == Using SIP RTP TOS bits 184
  331.   == Using SIP RTP CoS mark 5
  332. Audio is at 14132
  333. Adding codec 100003 (ulaw) to SDP
  334. Adding codec 100004 (alaw) to SDP
  335. Adding codec 100002 (gsm) to SDP
  336. Adding codec 100011 (g726) to SDP
  337. Adding non-codec 0x1 (telephone-event) to SDP
  338. Reliably Transmitting (NAT) to 176.47.75.56:1201:
  339. INVITE sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066 SIP/2.0
  340. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK41e63364;rport
  341. Max-Forwards: 70
  342. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as5e7c112c
  343. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  344. Contact: <sip:103@127.0.0.1:5060>
  345. Call-ID: 5e5462401e245db129e769e413c684c5@127.0.0.1:5060
  346. CSeq: 102 INVITE
  347. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  348. Date: Tue, 14 Mar 2017 21:29:30 GMT
  349. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  350. Supported: replaces, timer
  351. P-Asserted-Identity: "Zubair Laptop" <sip:103@127.0.0.1>
  352. Content-Type: application/sdp
  353. Content-Length: 306
  354.  
  355. v=0
  356. o=root 954660483 954660483 IN IP4 127.0.0.1
  357. s=Asterisk PBX 11.16.0
  358. c=IN IP4 127.0.0.1
  359. t=0 0
  360. m=audio 14132 RTP/AVP 0 8 3 111 101
  361. a=rtpmap:0 PCMU/8000
  362. a=rtpmap:8 PCMA/8000
  363. a=rtpmap:3 GSM/8000
  364. a=rtpmap:111 G726-32/8000
  365. a=rtpmap:101 telephone-event/8000
  366. a=fmtp:101 0-16
  367. a=ptime:20
  368. a=sendrecv
  369.  
  370. ---
  371.     -- Called SIP/101
  372.  
  373. <--- Transmitting (NAT) to 192.168.112.22:5060 --->
  374. SIP/2.0 180 Ringing
  375. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK23234170832238621948;received=192.168.112.22;rport=5060
  376. From: 103 <sip:103@192.168.112.50:5060>;tag=3092610502
  377. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as5a100b8b
  378. Call-ID: 5120292566874-14872666530126@192.168.112.22
  379. CSeq: 2 INVITE
  380. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  381. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  382. Supported: replaces, timer
  383. Contact: <sip:101@192.168.112.50:5060>
  384. P-Asserted-Identity: "Home IP Phone" <sip:101@192.168.112.50>
  385. Content-Length: 0
  386.  
  387.  
  388. <------------>
  389.     -- Connected line update to SIP/103-00000010 prevented.
  390. Really destroying SIP dialog 'No0rJm9Tm619Brig8s8gmg..' Method: REGISTER
  391. Retransmitting #1 (NAT) to 176.47.75.56:1201:
  392. INVITE sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066 SIP/2.0
  393. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK41e63364;rport
  394. Max-Forwards: 70
  395. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as5e7c112c
  396. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  397. Contact: <sip:103@127.0.0.1:5060>
  398. Call-ID: 5e5462401e245db129e769e413c684c5@127.0.0.1:5060
  399. CSeq: 102 INVITE
  400. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  401. Date: Tue, 14 Mar 2017 21:29:30 GMT
  402. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  403. Supported: replaces, timer
  404. P-Asserted-Identity: "Zubair Laptop" <sip:103@127.0.0.1>
  405. Content-Type: application/sdp
  406. Content-Length: 306
  407.  
  408. v=0
  409. o=root 954660483 954660483 IN IP4 127.0.0.1
  410. s=Asterisk PBX 11.16.0
  411. c=IN IP4 127.0.0.1
  412. t=0 0
  413. m=audio 14132 RTP/AVP 0 8 3 111 101
  414. a=rtpmap:0 PCMU/8000
  415. a=rtpmap:8 PCMA/8000
  416. a=rtpmap:3 GSM/8000
  417. a=rtpmap:111 G726-32/8000
  418. a=rtpmap:101 telephone-event/8000
  419. a=fmtp:101 0-16
  420. a=ptime:20
  421. a=sendrecv
  422.  
  423. ---
  424.  
  425. <--- SIP read from UDP:176.47.75.56:1201 --->
  426. SIP/2.0 100 Trying
  427. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK41e63364;rport=5060;received=94.48.30.110
  428. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  429. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as5e7c112c
  430. Call-ID: 5e5462401e245db129e769e413c684c5@127.0.0.1:5060
  431. CSeq: 102 INVITE
  432. Content-Length: 0
  433.  
  434. <------------->
  435. --- (7 headers 0 lines) ---
  436.  
  437. <--- SIP read from UDP:176.47.75.56:1201 --->
  438. SIP/2.0 100 Trying
  439. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK41e63364;rport=5060;received=94.48.30.110
  440. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  441. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as5e7c112c
  442. Call-ID: 5e5462401e245db129e769e413c684c5@127.0.0.1:5060
  443. CSeq: 102 INVITE
  444. Content-Length: 0
  445.  
  446. <------------->
  447. --- (7 headers 0 lines) ---
  448.  
  449. <--- SIP read from UDP:176.47.75.56:1201 --->
  450. SIP/2.0 180 Ringing
  451. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK41e63364;rport=5060;received=94.48.30.110
  452. Contact: <sip:101@176.47.75.56:1201;transport=UDP>
  453. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=d0f6a362
  454. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as5e7c112c
  455. Call-ID: 5e5462401e245db129e769e413c684c5@127.0.0.1:5060
  456. CSeq: 102 INVITE
  457. User-Agent: Zoiper rv2.8.30
  458. Allow-Events: presence, kpml, talk
  459. Content-Length: 0
  460.  
  461. <------------->
  462. --- (10 headers 0 lines) ---
  463. list_route: hop: <sip:101@176.47.75.56:1201;transport=UDP>
  464.     -- SIP/101-00000011 is ringing
  465.  
  466. <--- Transmitting (NAT) to 192.168.112.22:5060 --->
  467. SIP/2.0 180 Ringing
  468. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK23234170832238621948;received=192.168.112.22;rport=5060
  469. From: 103 <sip:103@192.168.112.50:5060>;tag=3092610502
  470. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as5a100b8b
  471. Call-ID: 5120292566874-14872666530126@192.168.112.22
  472. CSeq: 2 INVITE
  473. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  474. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  475. Supported: replaces, timer
  476. Contact: <sip:101@192.168.112.50:5060>
  477. Content-Length: 0
  478.  
  479.  
  480. <------------>
  481.  
  482. <--- SIP read from UDP:176.47.75.56:1201 --->
  483. SIP/2.0 200 OK
  484. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK41e63364;rport=5060;received=94.48.30.110
  485. Contact: <sip:101@176.47.75.56:1201;transport=UDP>
  486. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=d0f6a362
  487. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as5e7c112c
  488. Call-ID: 5e5462401e245db129e769e413c684c5@127.0.0.1:5060
  489. CSeq: 102 INVITE
  490. Content-Type: application/sdp
  491. User-Agent: Zoiper rv2.8.30
  492. Allow-Events: presence, kpml, talk
  493. Content-Length: 235
  494.  
  495. v=0
  496. o=Zoiper 0 1 IN IP4 127.0.0.1
  497. s=Zoiper
  498. c=IN IP4 127.0.0.1
  499. t=0 0
  500. m=audio 33670 RTP/AVP 0 3 8 101
  501. a=rtpmap:0 PCMU/8000
  502. a=rtpmap:3 GSM/8000
  503. a=rtpmap:8 PCMA/8000
  504. a=rtpmap:101 telephone-event/8000
  505. a=fmtp:101 0-16
  506. a=sendrecv
  507. <------------->
  508. --- (11 headers 12 lines) ---
  509. Found RTP audio format 0
  510. Found RTP audio format 3
  511. Found RTP audio format 8
  512. Found RTP audio format 101
  513. Found audio description format PCMU for ID 0
  514. Found audio description format GSM for ID 3
  515. Found audio description format PCMA for ID 8
  516. Found audio description format telephone-event for ID 101
  517. Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)
  518. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  519. Peer audio RTP is at port 127.0.0.1:33670
  520. list_route: hop: <sip:101@176.47.75.56:1201;transport=UDP>
  521. set_destination: Parsing <sip:101@176.47.75.56:1201;transport=UDP> for address/port to send to
  522. set_destination: set destination to 176.47.75.56:1201
  523. Transmitting (NAT) to 176.47.75.56:1201:
  524. ACK sip:101@176.47.75.56:1201;transport=UDP SIP/2.0
  525. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK2a260d0f;rport
  526. Max-Forwards: 70
  527. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as5e7c112c
  528. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=d0f6a362
  529. Contact: <sip:103@127.0.0.1:5060>
  530. Call-ID: 5e5462401e245db129e769e413c684c5@127.0.0.1:5060
  531. CSeq: 102 ACK
  532. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  533. Content-Length: 0
  534.  
  535.  
  536. ---
  537.     -- Connected line update to SIP/103-00000010 prevented.
  538.     -- SIP/101-00000011 answered SIP/103-00000010
  539. Audio is at 18366
  540. Adding codec 100003 (ulaw) to SDP
  541. Adding codec 100004 (alaw) to SDP
  542. Adding codec 100011 (g726) to SDP
  543. Adding non-codec 0x1 (telephone-event) to SDP
  544.  
  545. <--- Reliably Transmitting (NAT) to 192.168.112.22:5060 --->
  546. SIP/2.0 200 OK
  547. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK23234170832238621948;received=192.168.112.22;rport=5060
  548. From: 103 <sip:103@192.168.112.50:5060>;tag=3092610502
  549. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as5a100b8b
  550. Call-ID: 5120292566874-14872666530126@192.168.112.22
  551. CSeq: 2 INVITE
  552. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  553. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  554. Supported: replaces, timer
  555. Contact: <sip:101@192.168.112.50:5060>
  556. P-Asserted-Identity: "Home IP Phone" <sip:101@192.168.112.50>
  557. Content-Type: application/sdp
  558. Content-Length: 289
  559.  
  560. v=0
  561. o=root 155873890 155873890 IN IP4 192.168.112.50
  562. s=Asterisk PBX 11.16.0
  563. c=IN IP4 192.168.112.50
  564. t=0 0
  565. m=audio 18366 RTP/AVP 0 8 2 101
  566. a=rtpmap:0 PCMU/8000
  567. a=rtpmap:8 PCMA/8000
  568. a=rtpmap:2 G726-32/8000
  569. a=rtpmap:101 telephone-event/8000
  570. a=fmtp:101 0-16
  571. a=ptime:20
  572. a=sendrecv
  573.  
  574. <------------>
  575. Retransmitting #1 (NAT) to 192.168.112.22:5060:
  576. SIP/2.0 200 OK
  577. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK23234170832238621948;received=192.168.112.22;rport=5060
  578. From: 103 <sip:103@192.168.112.50:5060>;tag=3092610502
  579. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as5a100b8b
  580. Call-ID: 5120292566874-14872666530126@192.168.112.22
  581. CSeq: 2 INVITE
  582. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  583. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  584. Supported: replaces, timer
  585. Contact: <sip:101@192.168.112.50:5060>
  586. P-Asserted-Identity: "Home IP Phone" <sip:101@192.168.112.50>
  587. Content-Type: application/sdp
  588. Content-Length: 289
  589.  
  590. v=0
  591. o=root 155873890 155873890 IN IP4 192.168.112.50
  592. s=Asterisk PBX 11.16.0
  593. c=IN IP4 192.168.112.50
  594. t=0 0
  595. m=audio 18366 RTP/AVP 0 8 2 101
  596. a=rtpmap:0 PCMU/8000
  597. a=rtpmap:8 PCMA/8000
  598. a=rtpmap:2 G726-32/8000
  599. a=rtpmap:101 telephone-event/8000
  600. a=fmtp:101 0-16
  601. a=ptime:20
  602. a=sendrecv
  603.  
  604. ---
  605.  
  606. <--- SIP read from UDP:192.168.112.22:5060 --->
  607. ACK sip:101@192.168.112.50:5060 SIP/2.0
  608. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK25108158902853724250
  609. From: 103 <sip:103@192.168.112.50:5060>;tag=3092610502
  610. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as5a100b8b
  611. Call-ID: 5120292566874-14872666530126@192.168.112.22
  612. CSeq: 2 ACK
  613. Contact: <sip:103@192.168.112.22:5060>
  614. Max-Forwards: 70
  615. User-Agent: DLINK DPH-400SE FRU2.2.876.445
  616. Content-Length: 0
  617.  
  618. <------------->
  619. --- (10 headers 0 lines) ---
  620.  
  621. <--- SIP read from UDP:192.168.112.22:5060 --->
  622. ACK sip:101@192.168.112.50:5060 SIP/2.0
  623. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK25108158902853724250
  624. From: 103 <sip:103@192.168.112.50:5060>;tag=3092610502
  625. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as5a100b8b
  626. Call-ID: 5120292566874-14872666530126@192.168.112.22
  627. CSeq: 2 ACK
  628. Contact: <sip:103@192.168.112.22:5060>
  629. Max-Forwards: 70
  630. User-Agent: DLINK DPH-400SE FRU2.2.876.445
  631. Content-Length: 0
  632.  
  633. <------------->
  634. --- (10 headers 0 lines) ---
  635.        > 0x25b4900 -- Probation passed - setting RTP source address to 192.168.112.22:10016
  636.  
  637. <--- SIP read from UDP:192.168.112.22:5060 --->
  638. OPTIONS sip:192.168.112.50:5060 SIP/2.0
  639. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK16211158522281411637
  640. From: 103 <sip:103@192.168.112.50:5060>;tag=1057715157
  641. To: <sip:192.168.112.50:5060>
  642. Call-ID: 105812199012519-257271351410050@192.168.112.22
  643. CSeq: 1 OPTIONS
  644. Max-Forwards: 70
  645. User-Agent: DLINK DPH-400SE FRU2.2.876.445
  646. Accept: application/sdp
  647. Content-Length: 0
  648.  
  649. <------------->
  650. --- (10 headers 0 lines) ---
  651. Sending to 192.168.112.22:5060 (NAT)
  652. Looking for s in from-sip-external (domain 192.168.112.50)
  653.  
  654. <--- Transmitting (NAT) to 192.168.112.22:5060 --->
  655. SIP/2.0 200 OK
  656. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK16211158522281411637;received=192.168.112.22;rport=5060
  657. From: 103 <sip:103@192.168.112.50:5060>;tag=1057715157
  658. To: <sip:192.168.112.50:5060>;tag=as58df9b73
  659. Call-ID: 105812199012519-257271351410050@192.168.112.22
  660. CSeq: 1 OPTIONS
  661. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  662. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  663. Supported: replaces, timer
  664. Contact: <sip:192.168.112.50:5060>
  665. Accept: application/sdp
  666. Content-Length: 0
  667.  
  668.  
  669. <------------>
  670. Scheduling destruction of SIP dialog '105812199012519-257271351410050@192.168.112.22' in 32000 ms (Method: OPTIONS)
  671.  
  672. <--- SIP read from UDP:192.168.112.22:5060 --->
  673. BYE sip:101@192.168.112.50:5060 SIP/2.0
  674. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK28467256961619919519
  675. From: 103 <sip:103@192.168.112.50:5060>;tag=3092610502
  676. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as5a100b8b
  677. Call-ID: 5120292566874-14872666530126@192.168.112.22
  678. CSeq: 3 BYE
  679. Contact: <sip:103@192.168.112.22:5060>
  680. Max-Forwards: 70
  681. User-Agent: DLINK DPH-400SE FRU2.2.876.445
  682. Content-Length: 0
  683.  
  684. <------------->
  685. --- (10 headers 0 lines) ---
  686. Sending to 192.168.112.22:5060 (NAT)
  687. Scheduling destruction of SIP dialog '5120292566874-14872666530126@192.168.112.22' in 6400 ms (Method: BYE)
  688.  
  689. <--- Transmitting (NAT) to 192.168.112.22:5060 --->
  690. SIP/2.0 200 OK
  691. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK28467256961619919519;received=192.168.112.22;rport=5060
  692. From: 103 <sip:103@192.168.112.50:5060>;tag=3092610502
  693. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as5a100b8b
  694. Call-ID: 5120292566874-14872666530126@192.168.112.22
  695. CSeq: 3 BYE
  696. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  697. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  698. Supported: replaces, timer
  699. Content-Length: 0
  700.  
  701.  
  702. <------------>
  703.     -- Executing [h@macro-dial-one:1] Macro("SIP/103-00000010", "hangupcall,") in new stack
  704.     -- Executing [s@macro-hangupcall:1] ExecIf("SIP/103-00000010", "0?Set(CDR(recordingfile)=.wav)") in new stack
  705.     -- Executing [s@macro-hangupcall:2] GotoIf("SIP/103-00000010", "1?theend") in new stack
  706.     -- Goto (macro-hangupcall,s,4)
  707.     -- Executing [s@macro-hangupcall:4] Hangup("SIP/103-00000010", "") in new stack
  708.   == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/103-00000010' in macro 'hangupcall'
  709.   == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/103-00000010'
  710. Scheduling destruction of SIP dialog '5e5462401e245db129e769e413c684c5@127.0.0.1:5060' in 11648 ms (Method: INVITE)
  711. set_destination: Parsing <sip:101@176.47.75.56:1201;transport=UDP> for address/port to send to
  712. set_destination: set destination to 176.47.75.56:1201
  713. Reliably Transmitting (NAT) to 176.47.75.56:1201:
  714. BYE sip:101@176.47.75.56:1201;transport=UDP SIP/2.0
  715. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0cfba1ce;rport
  716. Max-Forwards: 70
  717. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as5e7c112c
  718. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=d0f6a362
  719. Call-ID: 5e5462401e245db129e769e413c684c5@127.0.0.1:5060
  720. CSeq: 103 BYE
  721. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  722. X-Asterisk-HangupCause: Normal Clearing
  723. X-Asterisk-HangupCauseCode: 16
  724. Content-Length: 0
  725.  
  726.  
  727. ---
  728.   == Spawn extension (macro-dial-one, s, 45) exited non-zero on 'SIP/103-00000010' in macro 'dial-one'
  729.   == Spawn extension (macro-exten-vm, s, 16) exited non-zero on 'SIP/103-00000010' in macro 'exten-vm'
  730.   == Spawn extension (ext-local, 101, 2) exited non-zero on 'SIP/103-00000010'
  731. Retransmitting #1 (NAT) to 176.47.75.56:1201:
  732. BYE sip:101@176.47.75.56:1201;transport=UDP SIP/2.0
  733. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0cfba1ce;rport
  734. Max-Forwards: 70
  735. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as5e7c112c
  736. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=d0f6a362
  737. Call-ID: 5e5462401e245db129e769e413c684c5@127.0.0.1:5060
  738. CSeq: 103 BYE
  739. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  740. X-Asterisk-HangupCause: Normal Clearing
  741. X-Asterisk-HangupCauseCode: 16
  742. Content-Length: 0
  743.  
  744.  
  745. ---
  746. Retransmitting #2 (NAT) to 176.47.75.56:1201:
  747. BYE sip:101@176.47.75.56:1201;transport=UDP SIP/2.0
  748. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0cfba1ce;rport
  749. Max-Forwards: 70
  750. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as5e7c112c
  751. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=d0f6a362
  752. Call-ID: 5e5462401e245db129e769e413c684c5@127.0.0.1:5060
  753. CSeq: 103 BYE
  754. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  755. X-Asterisk-HangupCause: Normal Clearing
  756. X-Asterisk-HangupCauseCode: 16
  757. Content-Length: 0
  758.  
  759.  
  760. ---
  761.  
  762. <--- SIP read from UDP:176.47.75.56:1201 --->
  763. SIP/2.0 200 OK
  764. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0cfba1ce;rport=5060;received=94.48.30.110
  765. Contact: <sip:101@176.47.75.56:1201;transport=UDP>
  766. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=d0f6a362
  767. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as5e7c112c
  768. Call-ID: 5e5462401e245db129e769e413c684c5@127.0.0.1:5060
  769. CSeq: 103 BYE
  770. User-Agent: Zoiper rv2.8.30
  771. Content-Length: 0
  772.  
  773. <------------->
  774. --- (9 headers 0 lines) ---
  775. Really destroying SIP dialog '5e5462401e245db129e769e413c684c5@127.0.0.1:5060' Method: INVITE
  776.  
  777. <--- SIP read from UDP:176.47.75.56:1201 --->
  778. SIP/2.0 200 OK
  779. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0cfba1ce;rport=5060;received=94.48.30.110
  780. Contact: <sip:101@176.47.75.56:1201;transport=UDP>
  781. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=d0f6a362
  782. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as5e7c112c
  783. Call-ID: 5e5462401e245db129e769e413c684c5@127.0.0.1:5060
  784. CSeq: 103 BYE
  785. User-Agent: Zoiper rv2.8.30
  786. Content-Length: 0
  787.  
  788. <------------->
  789. --- (9 headers 0 lines) ---
  790.  
  791. <--- SIP read from UDP:176.47.75.56:1201 --->
  792. SIP/2.0 200 OK
  793. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0cfba1ce;rport=5060;received=94.48.30.110
  794. Contact: <sip:101@176.47.75.56:1201;transport=UDP>
  795. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=d0f6a362
  796. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as5e7c112c
  797. Call-ID: 5e5462401e245db129e769e413c684c5@127.0.0.1:5060
  798. CSeq: 103 BYE
  799. User-Agent: Zoiper rv2.8.30
  800. Content-Length: 0
  801.  
  802. <------------->
  803. --- (9 headers 0 lines) ---
  804.  
  805. <--- SIP read from UDP:176.47.75.56:1201 --->
  806.  
  807.  
  808. <------------->
  809. Really destroying SIP dialog '5120292566874-14872666530126@192.168.112.22' Method: BYE
  810.  
  811. <--- SIP read from UDP:176.47.75.56:1201 --->
  812. REGISTER sip:voip.nhksa.com;transport=UDP SIP/2.0
  813. Via: SIP/2.0/UDP 176.47.75.56:1201;branch=z9hG4bK-524287-1---23573b797c917752;rport
  814. Max-Forwards: 70
  815. Contact: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  816. To: <sip:101@voip.nhksa.com;transport=UDP>
  817. From: <sip:101@voip.nhksa.com;transport=UDP>;tag=76d1d834
  818. Call-ID: No0rJm9Tm619Brig8s8gmg..
  819. CSeq: 183 REGISTER
  820. Expires: 60
  821. User-Agent: Zoiper rv2.8.30
  822. Authorization: Digest username="101",realm="asterisk",nonce="2caa8365",uri="sip:voip.nhksa.com;transport=UDP",response="382f7382c4e402cdd9f0641480f8a27d",algorithm=MD5
  823. Allow-Events: presence, kpml, talk
  824. Content-Length: 0
  825.  
  826. <------------->
  827. --- (13 headers 0 lines) ---
  828. Sending to 176.47.75.56:1201 (NAT)
  829. Sending to 176.47.75.56:1201 (NAT)
  830.  
  831. <--- Transmitting (NAT) to 176.47.75.56:1201 --->
  832. SIP/2.0 401 Unauthorized
  833. Via: SIP/2.0/UDP 176.47.75.56:1201;branch=z9hG4bK-524287-1---23573b797c917752;received=176.47.75.56;rport=1201
  834. From: <sip:101@voip.nhksa.com;transport=UDP>;tag=76d1d834
  835. To: <sip:101@voip.nhksa.com;transport=UDP>;tag=as05b9c3b0
  836. Call-ID: No0rJm9Tm619Brig8s8gmg..
  837. CSeq: 183 REGISTER
  838. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  839. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  840. Supported: replaces, timer
  841. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="18e7d0c8"
  842. Content-Length: 0
  843.  
  844.  
  845. <------------>
  846. Scheduling destruction of SIP dialog 'No0rJm9Tm619Brig8s8gmg..' in 32000 ms (Method: REGISTER)
  847.  
  848. <--- SIP read from UDP:176.47.75.56:1201 --->
  849. REGISTER sip:voip.nhksa.com;transport=UDP SIP/2.0
  850. Via: SIP/2.0/UDP 176.47.75.56:1201;branch=z9hG4bK-524287-1---e86876943175401d;rport
  851. Max-Forwards: 70
  852. Contact: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  853. To: <sip:101@voip.nhksa.com;transport=UDP>
  854. From: <sip:101@voip.nhksa.com;transport=UDP>;tag=76d1d834
  855. Call-ID: No0rJm9Tm619Brig8s8gmg..
  856. CSeq: 184 REGISTER
  857. Expires: 60
  858. User-Agent: Zoiper rv2.8.30
  859. Authorization: Digest username="101",realm="asterisk",nonce="18e7d0c8",uri="sip:voip.nhksa.com;transport=UDP",response="10810c5aa623e439a1de0c7a45f728dc",algorithm=MD5
  860. Allow-Events: presence, kpml, talk
  861. Content-Length: 0
  862.  
  863. <------------->
  864. --- (13 headers 0 lines) ---
  865. Sending to 176.47.75.56:1201 (NAT)
  866. Reliably Transmitting (NAT) to 176.47.75.56:1201:
  867. OPTIONS sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066 SIP/2.0
  868. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK413e520b;rport
  869. Max-Forwards: 70
  870. From: "Unknown" <sip:Unknown@127.0.0.1>;tag=as166abe26
  871. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  872. Contact: <sip:Unknown@127.0.0.1:5060>
  873. Call-ID: 2945804457ec377f1f0b315e63cd0e16@127.0.0.1:5060
  874. CSeq: 102 OPTIONS
  875. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  876. Date: Tue, 14 Mar 2017 21:30:06 GMT
  877. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  878. Supported: replaces, timer
  879. Content-Length: 0
  880.  
  881.  
  882. ---
  883.  
  884. <--- Transmitting (NAT) to 176.47.75.56:1201 --->
  885. SIP/2.0 200 OK
  886. Via: SIP/2.0/UDP 176.47.75.56:1201;branch=z9hG4bK-524287-1---e86876943175401d;received=176.47.75.56;rport=1201
  887. From: <sip:101@voip.nhksa.com;transport=UDP>;tag=76d1d834
  888. To: <sip:101@voip.nhksa.com;transport=UDP>;tag=as05b9c3b0
  889. Call-ID: No0rJm9Tm619Brig8s8gmg..
  890. CSeq: 184 REGISTER
  891. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  892. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  893. Supported: replaces, timer
  894. Expires: 60
  895. Contact: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;expires=60
  896. Date: Tue, 14 Mar 2017 21:30:06 GMT
  897. Content-Length: 0
  898.  
  899.  
  900. <------------>
  901. Scheduling destruction of SIP dialog 'No0rJm9Tm619Brig8s8gmg..' in 32000 ms (Method: REGISTER)
  902.  
  903. <--- SIP read from UDP:176.47.75.56:1201 --->
  904. SIP/2.0 200 OK
  905. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK413e520b;rport=5060;received=94.48.30.110
  906. Contact: <sip:192.168.33.20:49514>
  907. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=2d569e62
  908. From: "Unknown" <sip:Unknown@127.0.0.1>;tag=as166abe26
  909. Call-ID: 2945804457ec377f1f0b315e63cd0e16@127.0.0.1:5060
  910. CSeq: 102 OPTIONS
  911. Accept: application/sdp, application/sdp
  912. Accept-Language: en
  913. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  914. Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
  915. User-Agent: Zoiper rv2.8.30
  916. Allow-Events: presence, kpml, talk
  917. Content-Length: 0
  918.  
  919. <------------->
  920. --- (14 headers 0 lines) ---
  921. Really destroying SIP dialog '2945804457ec377f1f0b315e63cd0e16@127.0.0.1:5060' Method: OPTIONS
  922. Reliably Transmitting (NAT) to 192.168.112.22:5060:
  923. OPTIONS sip:103@192.168.112.22:5060 SIP/2.0
  924. Via: SIP/2.0/UDP 192.168.112.50:5060;branch=z9hG4bK21d9ac1c;rport
  925. Max-Forwards: 70
  926. From: "Unknown" <sip:Unknown@192.168.112.50>;tag=as274ea4ad
  927. To: <sip:103@192.168.112.22:5060>
  928. Contact: <sip:Unknown@192.168.112.50:5060>
  929. Call-ID: 6c4d1301590a57f82e056f64269b1f2a@192.168.112.50:5060
  930. CSeq: 102 OPTIONS
  931. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  932. Date: Tue, 14 Mar 2017 21:30:12 GMT
  933. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  934. Supported: replaces, timer
  935. Content-Length: 0
  936.  
  937.  
  938. ---
  939.  
  940. <--- SIP read from UDP:192.168.112.22:5060 --->
  941. SIP/2.0 200 OK
  942. Via: SIP/2.0/UDP 192.168.112.50:5060;branch=z9hG4bK21d9ac1c;rport=5060
  943. From: "Unknown" <sip:Unknown@192.168.112.50>;tag=as274ea4ad
  944. To: <sip:103@192.168.112.22:5060>;tag=2783229473
  945. Call-ID: 6c4d1301590a57f82e056f64269b1f2a@192.168.112.50:5060
  946. CSeq: 102 OPTIONS
  947. Contact: <sip:103@192.168.112.22:5060>
  948. Supported: 100rel, replaces, timer
  949. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
  950. Accept: application/sdp, message/sipfrag, application/dtmf-relay
  951. Content-Length: 0
  952.  
  953. <------------->
  954. --- (11 headers 0 lines) ---
  955. Really destroying SIP dialog '6c4d1301590a57f82e056f64269b1f2a@192.168.112.50:5060' Method: OPTIONS
  956. Really destroying SIP dialog '105812199012519-257271351410050@192.168.112.22' Method: OPTIONS
  957.  

Replies to no Audio rss

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Re: no Audio Sarthor text 10 Months ago.

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Here you can reply to the paste above