Re: no Audio

From Sarthor, 10 Months ago, written in Plain Text, viewed 99 times. This paste is a reply to no Audio from Sarthor - view diff
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  1. <--- SIP read from UDP:192.168.112.22:5060 --->
  2. OPTIONS sip:192.168.112.50:5060 SIP/2.0
  3. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK594325025504016890
  4. From: 103 <sip:103@192.168.112.50:5060>;tag=2037914439
  5. To: <sip:192.168.112.50:5060>
  6. Call-ID: 28532544921450-14331440214468@192.168.112.22
  7. CSeq: 1 OPTIONS
  8. Max-Forwards: 70
  9. User-Agent: DLINK DPH-400SE FRU2.2.876.445
  10. Accept: application/sdp
  11. Content-Length: 0
  12.  
  13. <------------->
  14. --- (10 headers 0 lines) ---
  15. Sending to 192.168.112.22:5060 (NAT)
  16. Looking for s in from-sip-external (domain 192.168.112.50)
  17.  
  18. <--- Transmitting (NAT) to 192.168.112.22:5060 --->
  19. SIP/2.0 200 OK
  20. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK594325025504016890;received=192.168.112.22;rport=5060
  21. From: 103 <sip:103@192.168.112.50:5060>;tag=2037914439
  22. To: <sip:192.168.112.50:5060>;tag=as7f791c41
  23. Call-ID: 28532544921450-14331440214468@192.168.112.22
  24. CSeq: 1 OPTIONS
  25. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  26. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  27. Supported: replaces, timer
  28. Contact: <sip:192.168.112.50:5060>
  29. Accept: application/sdp
  30. Content-Length: 0
  31.  
  32.  
  33. <------------>
  34. Scheduling destruction of SIP dialog '28532544921450-14331440214468@192.168.112.22' in 32000 ms (Method: OPTIONS)
  35. Really destroying SIP dialog '134682258631245-119062809021907@192.168.112.22' Method: BYE
  36.  
  37. <--- SIP read from UDP:192.168.112.22:5060 --->
  38. INVITE sip:101@192.168.112.50;user=phone SIP/2.0
  39. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK22103294011485630689
  40. From: 103 <sip:103@192.168.112.50:5060>;tag=1813713356
  41. To: "101" <sip:101@192.168.112.50;user=phone>
  42. Call-ID: 21764281622511-100381027830739@192.168.112.22
  43. CSeq: 1 INVITE
  44. Contact: <sip:103@192.168.112.22:5060>
  45. Max-Forwards: 70
  46. Supported: replaces, join, path
  47. User-Agent: DLINK DPH-400SE FRU2.2.876.445
  48. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
  49. Content-Type: application/sdp
  50. Content-Length: 344
  51.  
  52. v=0
  53. o=103 445323572 165608355 IN IP4 192.168.112.22
  54. s=A conversation
  55. c=IN IP4 192.168.112.22
  56. t=0 0
  57. m=audio 10020 RTP/AVP 8 0 9 4 2 18 101
  58. a=rtpmap:8 PCMA/8000
  59. a=rtpmap:0 PCMU/8000
  60. a=rtpmap:9 G722/8000
  61. a=rtpmap:4 G723/8000
  62. a=rtpmap:2 G726-32/8000
  63. a=rtpmap:18 G729/8000
  64. a=rtpmap:101 telephone-event/8000
  65. a=fmtp:101 0-15
  66. a=sendrecv
  67. <------------->
  68. --- (13 headers 15 lines) ---
  69. Sending to 192.168.112.22:5060 (NAT)
  70. Sending to 192.168.112.22:5060 (NAT)
  71. Using INVITE request as basis request - 21764281622511-100381027830739@192.168.112.22
  72. Found peer '103' for '103' from 192.168.112.22:5060
  73.  
  74. <--- Reliably Transmitting (NAT) to 192.168.112.22:5060 --->
  75. SIP/2.0 401 Unauthorized
  76. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK22103294011485630689;received=192.168.112.22;rport=5060
  77. From: 103 <sip:103@192.168.112.50:5060>;tag=1813713356
  78. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as0c789b6d
  79. Call-ID: 21764281622511-100381027830739@192.168.112.22
  80. CSeq: 1 INVITE
  81. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  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="66040906"
  85. Content-Length: 0
  86.  
  87.  
  88. <------------>
  89. Scheduling destruction of SIP dialog '21764281622511-100381027830739@192.168.112.22' in 6400 ms (Method: INVITE)
  90.  
  91. <--- SIP read from UDP:192.168.112.22:5060 --->
  92. ACK sip:101@192.168.112.50;user=phone SIP/2.0
  93. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK22103294011485630689
  94. From: 103 <sip:103@192.168.112.50:5060>;tag=1813713356
  95. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as0c789b6d
  96. Call-ID: 21764281622511-100381027830739@192.168.112.22
  97. CSeq: 1 ACK
  98. Contact: <sip:103@192.168.112.22:5060>
  99. Max-Forwards: 70
  100. Content-Length: 0
  101.  
  102. <------------->
  103. --- (9 headers 0 lines) ---
  104.  
  105. <--- SIP read from UDP:192.168.112.22:5060 --->
  106. INVITE sip:101@192.168.112.50;user=phone SIP/2.0
  107. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK1416131351383425855
  108. From: 103 <sip:103@192.168.112.50:5060>;tag=1813713356
  109. To: "101" <sip:101@192.168.112.50;user=phone>
  110. Call-ID: 21764281622511-100381027830739@192.168.112.22
  111. CSeq: 2 INVITE
  112. Contact: <sip:103@192.168.112.22:5060>
  113. Authorization: Digest username="103", realm="asterisk", nonce="66040906", uri="sip:101@192.168.112.50;user=phone", response="670c4347a2077846b6619a38583a9ec2", algorithm=MD5
  114. Max-Forwards: 70
  115. Supported: replaces, join, path
  116. User-Agent: DLINK DPH-400SE FRU2.2.876.445
  117. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
  118. Content-Type: application/sdp
  119. Content-Length: 344
  120.  
  121. v=0
  122. o=103 445323572 165608355 IN IP4 192.168.112.22
  123. s=A conversation
  124. c=IN IP4 192.168.112.22
  125. t=0 0
  126. m=audio 10020 RTP/AVP 8 0 9 4 2 18 101
  127. a=rtpmap:8 PCMA/8000
  128. a=rtpmap:0 PCMU/8000
  129. a=rtpmap:9 G722/8000
  130. a=rtpmap:4 G723/8000
  131. a=rtpmap:2 G726-32/8000
  132. a=rtpmap:18 G729/8000
  133. a=rtpmap:101 telephone-event/8000
  134. a=fmtp:101 0-15
  135. a=sendrecv
  136. <------------->
  137. --- (14 headers 15 lines) ---
  138. Sending to 192.168.112.22:5060 (NAT)
  139. Using INVITE request as basis request - 21764281622511-100381027830739@192.168.112.22
  140. Found peer '103' for '103' from 192.168.112.22:5060
  141.   == Using SIP RTP TOS bits 184
  142.   == Using SIP RTP CoS mark 5
  143. Found RTP audio format 8
  144. Found RTP audio format 0
  145. Found RTP audio format 9
  146. Found RTP audio format 4
  147. Found RTP audio format 2
  148. Found RTP audio format 18
  149. Found RTP audio format 101
  150. Found audio description format PCMA for ID 8
  151. Found audio description format PCMU for ID 0
  152. Found audio description format G722 for ID 9
  153. Found audio description format G723 for ID 4
  154. Found audio description format G726-32 for ID 2
  155. Found audio description format G729 for ID 18
  156. Found audio description format telephone-event for ID 101
  157. Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(g723|ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726)
  158. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  159. Peer audio RTP is at port 192.168.112.22:10020
  160. Looking for 101 in from-internal (domain 192.168.112.50)
  161. list_route: hop: <sip:103@192.168.112.22:5060>
  162.  
  163. <--- Transmitting (NAT) to 192.168.112.22:5060 --->
  164. SIP/2.0 100 Trying
  165. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK1416131351383425855;received=192.168.112.22;rport=5060
  166. From: 103 <sip:103@192.168.112.50:5060>;tag=1813713356
  167. To: "101" <sip:101@192.168.112.50;user=phone>
  168. Call-ID: 21764281622511-100381027830739@192.168.112.22
  169. CSeq: 2 INVITE
  170. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  171. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  172. Supported: replaces, timer
  173. Contact: <sip:101@192.168.112.50:5060>
  174. Content-Length: 0
  175.  
  176.  
  177. <------------>
  178.     -- Executing [101@from-internal:1] GotoIf("SIP/103-00000002", "1?ext-local,101,1") in new stack
  179.     -- Goto (ext-local,101,1)
  180.     -- Executing [101@ext-local:1] Set("SIP/103-00000002", "__RINGTIMER=15") in new stack
  181.     -- Executing [101@ext-local:2] Macro("SIP/103-00000002", "exten-vm,novm,101,0,0,0") in new stack
  182.     -- Executing [s@macro-exten-vm:1] Macro("SIP/103-00000002", "user-callerid,") in new stack
  183.     -- Executing [s@macro-user-callerid:1] Set("SIP/103-00000002", "TOUCH_MONITOR=1489527475.2") in new stack
  184.     -- Executing [s@macro-user-callerid:2] Set("SIP/103-00000002", "AMPUSER=103") in new stack
  185.     -- Executing [s@macro-user-callerid:3] GotoIf("SIP/103-00000002", "0?report") in new stack
  186.     -- Executing [s@macro-user-callerid:4] ExecIf("SIP/103-00000002", "1?Set(REALCALLERIDNUM=103)") in new stack
  187.     -- Executing [s@macro-user-callerid:5] Set("SIP/103-00000002", "AMPUSER=103") in new stack
  188.     -- Executing [s@macro-user-callerid:6] GotoIf("SIP/103-00000002", "0?limit") in new stack
  189.     -- Executing [s@macro-user-callerid:7] Set("SIP/103-00000002", "AMPUSERCIDNAME=Zubair Laptop") in new stack
  190.     -- Executing [s@macro-user-callerid:8] GotoIf("SIP/103-00000002", "0?report") in new stack
  191.     -- Executing [s@macro-user-callerid:9] Set("SIP/103-00000002", "AMPUSERCID=103") in new stack
  192.     -- Executing [s@macro-user-callerid:10] Set("SIP/103-00000002", "__DIAL_OPTIONS=Ttr") in new stack
  193.     -- Executing [s@macro-user-callerid:11] Set("SIP/103-00000002", "CALLERID(all)="Zubair Laptop" <103>") in new stack
  194.     -- Executing [s@macro-user-callerid:12] GotoIf("SIP/103-00000002", "0?limit") in new stack
  195.     -- Executing [s@macro-user-callerid:13] ExecIf("SIP/103-00000002", "0?Set(GROUP(concurrency_limit)=103)") in new stack
  196.     -- Executing [s@macro-user-callerid:14] GosubIf("SIP/103-00000002", "7?sub-ccss,s,1(macro-exten-vm,101)") in new stack
  197.     -- Executing [s@sub-ccss:1] ExecIf("SIP/103-00000002", "0?Return()") in new stack
  198.     -- Executing [s@sub-ccss:2] Set("SIP/103-00000002", "CCSS_SETUP=TRUE") in new stack
  199.     -- Executing [s@sub-ccss:3] GosubIf("SIP/103-00000002", "0?monitor_config,1(macro-exten-vm,101):monitor_default,1(macro-exten-vm,101)") in new stack
  200.     -- Executing [monitor_default@sub-ccss:1] GotoIf("SIP/103-00000002", "1?is_exten") in new stack
  201.     -- Goto (sub-ccss,monitor_default,4)
  202.     -- Executing [monitor_default@sub-ccss:4] Set("SIP/103-00000002", "CALLCOMPLETION(cc_monitor_policy)=generic") in new stack
  203.     -- Executing [monitor_default@sub-ccss:5] Set("SIP/103-00000002", "CALLCOMPLETION(cc_max_monitors)=5") in new stack
  204.     -- Executing [monitor_default@sub-ccss:6] Return("SIP/103-00000002", "TRUE") in new stack
  205.     -- Executing [s@sub-ccss:4] GosubIf("SIP/103-00000002", "7?agent_config,1():agent_default,1()") in new stack
  206.     -- Executing [agent_config@sub-ccss:1] Set("SIP/103-00000002", "CALLCOMPLETION(cc_agent_policy)=generic") in new stack
  207.     -- Executing [agent_config@sub-ccss:2] Set("SIP/103-00000002", "CALLCOMPLETION(cc_offer_timer)=30") in new stack
  208.     -- Executing [agent_config@sub-ccss:3] Set("SIP/103-00000002", "CALLCOMPLETION(ccbs_available_timer)=") in new stack
  209. [2017-03-15 00:37:55] WARNING[8717][C-00000001]: ccss.c:948 ast_set_ccbs_available_timer: 0 is an invalid value for ccbs_available_timer. Retaining value as 4800
  210.     -- Executing [agent_config@sub-ccss:4] Set("SIP/103-00000002", "CALLCOMPLETION(ccnr_available_timer)=") in new stack
  211. [2017-03-15 00:37:55] WARNING[8717][C-00000001]: ccss.c:918 ast_set_ccnr_available_timer: 0 is an invalid value for ccnr_available_timer. Retaining value as 7200
  212.     -- Executing [agent_config@sub-ccss:5] Set("SIP/103-00000002", "CALLCOMPLETION(cc_callback_macro)=ccss-default") in new stack
  213. [2017-03-15 00:37:55] WARNING[8717][C-00000001]: ccss.c:1000 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated.  Please use cc_callback_sub instead.
  214.     -- Executing [agent_config@sub-ccss:6] ExecIf("SIP/103-00000002", "1?Set(CALLCOMPLETION(cc_recall_timer)=)") in new stack
  215. [2017-03-15 00:37:55] WARNING[8717][C-00000001]: ccss.c:933 ast_set_cc_recall_timer: 0 is an invalid value for ccnr_available_timer. Retaining value as 20
  216.     -- Executing [agent_config@sub-ccss:7] ExecIf("SIP/103-00000002", "1?Set(CALLCOMPLETION(cc_max_agents)=)") in new stack
  217.     -- Executing [agent_config@sub-ccss:8] ExecIf("SIP/103-00000002", "0?Set(CALLCOMPLETION(cc_agent_dialstring)=Local/103_101@from-ccss-)") in new stack
  218.     -- Executing [agent_config@sub-ccss:9] Set("SIP/103-00000002", "CALLCOMPLETION(cc_callback_macro)=ccss-default") in new stack
  219. [2017-03-15 00:37:55] WARNING[8717][C-00000001]: ccss.c:1000 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated.  Please use cc_callback_sub instead.
  220.     -- Executing [agent_config@sub-ccss:10] Return("SIP/103-00000002", "") in new stack
  221.     -- Executing [s@sub-ccss:5] Set("SIP/103-00000002", "DB(AMPUSER/103/ccss/last_number)=101") in new stack
  222.     -- Executing [s@sub-ccss:6] Return("SIP/103-00000002", "") in new stack
  223.     -- Executing [s@macro-user-callerid:15] ExecIf("SIP/103-00000002", "0?Set(CHANNEL(language)=)") in new stack
  224.     -- Executing [s@macro-user-callerid:16] GotoIf("SIP/103-00000002", "0?continue") in new stack
  225.     -- Executing [s@macro-user-callerid:17] ExecIf("SIP/103-00000002", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
  226.     -- Executing [s@macro-user-callerid:18] Set("SIP/103-00000002", "__TTL=64") in new stack
  227.     -- Executing [s@macro-user-callerid:19] GotoIf("SIP/103-00000002", "1?continue") in new stack
  228.     -- Goto (macro-user-callerid,s,30)
  229.     -- Executing [s@macro-user-callerid:30] Set("SIP/103-00000002", "CALLERID(number)=103") in new stack
  230.     -- Executing [s@macro-user-callerid:31] Set("SIP/103-00000002", "CALLERID(name)=Zubair Laptop") in new stack
  231.     -- Executing [s@macro-user-callerid:32] Set("SIP/103-00000002", "CDR(cnum)=103") in new stack
  232.     -- Executing [s@macro-user-callerid:33] Set("SIP/103-00000002", "CDR(cnam)=Zubair Laptop") in new stack
  233.     -- Executing [s@macro-user-callerid:34] Set("SIP/103-00000002", "CHANNEL(language)=en") in new stack
  234.     -- Executing [s@macro-exten-vm:2] Set("SIP/103-00000002", "RingGroupMethod=none") in new stack
  235.     -- Executing [s@macro-exten-vm:3] Set("SIP/103-00000002", "__EXTTOCALL=101") in new stack
  236.     -- Executing [s@macro-exten-vm:4] Set("SIP/103-00000002", "__PICKUPMARK=101") in new stack
  237.     -- Executing [s@macro-exten-vm:5] Set("SIP/103-00000002", "RT=") in new stack
  238.     -- Executing [s@macro-exten-vm:6] ExecIf("SIP/103-00000002", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
  239.     -- Executing [s@macro-exten-vm:7] ExecIf("SIP/103-00000002", "0?MacroExit()") in new stack
  240.     -- Executing [s@macro-exten-vm:8] Gosub("SIP/103-00000002", "sub-record-check,s,1(exten,101,dontcare)") in new stack
  241.     -- Executing [s@sub-record-check:1] GotoIf("SIP/103-00000002", "0?initialized") in new stack
  242.     -- Executing [s@sub-record-check:2] Set("SIP/103-00000002", "__REC_STATUS=INITIALIZED") in new stack
  243.     -- Executing [s@sub-record-check:3] Set("SIP/103-00000002", "NOW=1489527475") in new stack
  244.     -- Executing [s@sub-record-check:4] Set("SIP/103-00000002", "__DAY=15") in new stack
  245.     -- Executing [s@sub-record-check:5] Set("SIP/103-00000002", "__MONTH=03") in new stack
  246.     -- Executing [s@sub-record-check:6] Set("SIP/103-00000002", "__YEAR=2017") in new stack
  247.     -- Executing [s@sub-record-check:7] Set("SIP/103-00000002", "__TIMESTR=20170315-003755") in new stack
  248.     -- Executing [s@sub-record-check:8] Set("SIP/103-00000002", "__FROMEXTEN=103") in new stack
  249.     -- Executing [s@sub-record-check:9] Set("SIP/103-00000002", "__MON_FMT=wav") in new stack
  250.     -- Executing [s@sub-record-check:10] NoOp("SIP/103-00000002", "Recordings initialized") in new stack
  251.     -- Executing [s@sub-record-check:11] ExecIf("SIP/103-00000002", "0?Set(ARG3=dontcare)") in new stack
  252.     -- Executing [s@sub-record-check:12] Set("SIP/103-00000002", "REC_POLICY_MODE_SAVE=") in new stack
  253.     -- Executing [s@sub-record-check:13] ExecIf("SIP/103-00000002", "0?Set(REC_STATUS=NO)") in new stack
  254.     -- Executing [s@sub-record-check:14] GotoIf("SIP/103-00000002", "5?checkaction") in new stack
  255.     -- Goto (sub-record-check,s,17)
  256.     -- Executing [s@sub-record-check:17] GotoIf("SIP/103-00000002", "1?sub-record-check,exten,1") in new stack
  257.     -- Goto (sub-record-check,exten,1)
  258.     -- Executing [exten@sub-record-check:1] NoOp("SIP/103-00000002", "Exten Recording Check between 103 and 101") in new stack
  259.     -- Executing [exten@sub-record-check:2] Set("SIP/103-00000002", "CALLTYPE=internal") in new stack
  260.     -- Executing [exten@sub-record-check:3] ExecIf("SIP/103-00000002", "0?Set(CALLTYPE=)") in new stack
  261.     -- Executing [exten@sub-record-check:4] Set("SIP/103-00000002", "CALLEE=dontcare") in new stack
  262.     -- Executing [exten@sub-record-check:5] ExecIf("SIP/103-00000002", "0?Set(CALLEE=dontcare)") in new stack
  263.     -- Executing [exten@sub-record-check:6] GotoIf("SIP/103-00000002", "0?callee") in new stack
  264.     -- Executing [exten@sub-record-check:7] GotoIf("SIP/103-00000002", "1?caller") in new stack
  265.     -- Goto (sub-record-check,exten,13)
  266.     -- Executing [exten@sub-record-check:13] Set("SIP/103-00000002", "RECMODE=dontcare") in new stack
  267.     -- Executing [exten@sub-record-check:14] ExecIf("SIP/103-00000002", "0?Set(RECMODE=dontcare)") in new stack
  268.     -- Executing [exten@sub-record-check:15] ExecIf("SIP/103-00000002", "1?Set(RECMODE=dontcare)") in new stack
  269.     -- Executing [exten@sub-record-check:16] Gosub("SIP/103-00000002", "recordcheck,1(dontcare,internal,101)") in new stack
  270.     -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/103-00000002", "Starting recording check against dontcare") in new stack
  271.     -- Executing [recordcheck@sub-record-check:2] Goto("SIP/103-00000002", "dontcare") in new stack
  272.     -- Goto (sub-record-check,recordcheck,3)
  273.     -- Executing [recordcheck@sub-record-check:3] Return("SIP/103-00000002", "") in new stack
  274.     -- Executing [exten@sub-record-check:17] Return("SIP/103-00000002", "") in new stack
  275.     -- Executing [s@macro-exten-vm:9] GotoIf("SIP/103-00000002", "1?macrodial") in new stack
  276.     -- Goto (macro-exten-vm,s,15)
  277.     -- Executing [s@macro-exten-vm:15] GosubIf("SIP/103-00000002", "0?clrheader,1()") in new stack
  278.     -- Executing [s@macro-exten-vm:16] Macro("SIP/103-00000002", "dial-one,,Ttr,101") in new stack
  279.     -- Executing [s@macro-dial-one:1] Set("SIP/103-00000002", "DEXTEN=101") in new stack
  280.     -- Executing [s@macro-dial-one:2] Set("SIP/103-00000002", "DIALSTATUS_CW=") in new stack
  281.     -- Executing [s@macro-dial-one:3] GosubIf("SIP/103-00000002", "0?screen,1()") in new stack
  282.     -- Executing [s@macro-dial-one:4] GosubIf("SIP/103-00000002", "0?cf,1()") in new stack
  283.     -- Executing [s@macro-dial-one:5] GotoIf("SIP/103-00000002", "1?skip1") in new stack
  284.     -- Goto (macro-dial-one,s,8)
  285.     -- Executing [s@macro-dial-one:8] GotoIf("SIP/103-00000002", "0?nodial") in new stack
  286.     -- Executing [s@macro-dial-one:9] GotoIf("SIP/103-00000002", "0?continue") in new stack
  287.     -- Executing [s@macro-dial-one:10] Set("SIP/103-00000002", "EXTHASCW=ENABLED") in new stack
  288.     -- Executing [s@macro-dial-one:11] GotoIf("SIP/103-00000002", "0?next1:cwinusebusy") in new stack
  289.     -- Goto (macro-dial-one,s,23)
  290.     -- Executing [s@macro-dial-one:23] GotoIf("SIP/103-00000002", "1?next3:continue") in new stack
  291.     -- Goto (macro-dial-one,s,24)
  292.     -- Executing [s@macro-dial-one:24] ExecIf("SIP/103-00000002", "0?Set(DIALSTATUS_CW=BUSY)") in new stack
  293.     -- Executing [s@macro-dial-one:25] GotoIf("SIP/103-00000002", "0?nodial") in new stack
  294.     -- Executing [s@macro-dial-one:26] GosubIf("SIP/103-00000002", "1?dstring,1():dlocal,1()") in new stack
  295.     -- Executing [dstring@macro-dial-one:1] Set("SIP/103-00000002", "DSTRING=") in new stack
  296.     -- Executing [dstring@macro-dial-one:2] Set("SIP/103-00000002", "DEVICES=101") in new stack
  297.     -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/103-00000002", "0?Return()") in new stack
  298.     -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/103-00000002", "0?Set(DEVICES=01)") in new stack
  299.     -- Executing [dstring@macro-dial-one:5] Set("SIP/103-00000002", "LOOPCNT=1") in new stack
  300.     -- Executing [dstring@macro-dial-one:6] Set("SIP/103-00000002", "ITER=1") in new stack
  301.     -- Executing [dstring@macro-dial-one:7] Set("SIP/103-00000002", "THISDIAL=SIP/101") in new stack
  302.     -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/103-00000002", "1?zap2dahdi,1()") in new stack
  303.     -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/103-00000002", "0?Return()") in new stack
  304.     -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/103-00000002", "NEWDIAL=") in new stack
  305.     -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/103-00000002", "LOOPCNT2=1") in new stack
  306.     -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/103-00000002", "ITER2=1") in new stack
  307.     -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/103-00000002", "THISPART2=SIP/101") in new stack
  308.     -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/103-00000002", "0?Set(THISPART2=DAHDI/101)") in new stack
  309.     -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/103-00000002", "NEWDIAL=SIP/101&") in new stack
  310.     -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/103-00000002", "ITER2=2") in new stack
  311.     -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/103-00000002", "0?begin2") in new stack
  312.     -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/103-00000002", "THISDIAL=SIP/101") in new stack
  313.     -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/103-00000002", "") in new stack
  314.     -- Executing [dstring@macro-dial-one:9] GotoIf("SIP/103-00000002", "1?doset") in new stack
  315.     -- Goto (macro-dial-one,dstring,13)
  316.     -- Executing [dstring@macro-dial-one:13] Set("SIP/103-00000002", "DSTRING=SIP/101&") in new stack
  317.     -- Executing [dstring@macro-dial-one:14] Set("SIP/103-00000002", "ITER=2") in new stack
  318.     -- Executing [dstring@macro-dial-one:15] GotoIf("SIP/103-00000002", "0?begin") in new stack
  319.     -- Executing [dstring@macro-dial-one:16] ExecIf("SIP/103-00000002", "0?Return()") in new stack
  320.     -- Executing [dstring@macro-dial-one:17] Set("SIP/103-00000002", "DSTRING=SIP/101") in new stack
  321.     -- Executing [dstring@macro-dial-one:18] Return("SIP/103-00000002", "") in new stack
  322.     -- Executing [s@macro-dial-one:27] GotoIf("SIP/103-00000002", "0?nodial") in new stack
  323.     -- Executing [s@macro-dial-one:28] GotoIf("SIP/103-00000002", "0?skiptrace") in new stack
  324.     -- Executing [s@macro-dial-one:29] GosubIf("SIP/103-00000002", "1?ctset,1():ctclear,1()") in new stack
  325.     -- Executing [ctset@macro-dial-one:1] Set("SIP/103-00000002", "DB(CALLTRACE/101)=103") in new stack
  326.     -- Executing [ctset@macro-dial-one:2] Return("SIP/103-00000002", "") in new stack
  327.     -- Executing [s@macro-dial-one:30] Set("SIP/103-00000002", "D_OPTIONS=Ttr") in new stack
  328.     -- Executing [s@macro-dial-one:31] ExecIf("SIP/103-00000002", "0?SIPAddHeader(Alert-Info: )") in new stack
  329.     -- Executing [s@macro-dial-one:32] ExecIf("SIP/103-00000002", "0?SIPAddHeader()") in new stack
  330.     -- Executing [s@macro-dial-one:33] ExecIf("SIP/103-00000002", "0?Set(CHANNEL(musicclass)=)") in new stack
  331.     -- Executing [s@macro-dial-one:34] GosubIf("SIP/103-00000002", "0?qwait,1()") in new stack
  332.     -- Executing [s@macro-dial-one:35] Set("SIP/103-00000002", "__CWIGNORE=") in new stack
  333.     -- Executing [s@macro-dial-one:36] Set("SIP/103-00000002", "__KEEPCID=TRUE") in new stack
  334.     -- Executing [s@macro-dial-one:37] GotoIf("SIP/103-00000002", "0?usegoto,1") in new stack
  335.     -- Executing [s@macro-dial-one:38] GotoIf("SIP/103-00000002", "0?godial") in new stack
  336.     -- Executing [s@macro-dial-one:39] Gosub("SIP/103-00000002", "sub-presencestate-display,s,1(101)") in new stack
  337. [2017-03-15 00:37:55] WARNING[8717][C-00000001]: func_presencestate.c:132 presence_read: PRESENCE_STATE unknown
  338.     -- Executing [s@sub-presencestate-display:1] Goto("SIP/103-00000002", "state-,1") in new stack
  339.     -- Goto (sub-presencestate-display,state-,1)
  340.     -- Executing [state-@sub-presencestate-display:1] Set("SIP/103-00000002", "PRESENCESTATE_DISPLAY=") in new stack
  341.     -- Executing [state-@sub-presencestate-display:2] Return("SIP/103-00000002", "") in new stack
  342.     -- Executing [s@macro-dial-one:40] Set("SIP/103-00000002", "CONNECTEDLINE(name,i)=Home IP Phone") in new stack
  343.     -- Executing [s@macro-dial-one:41] Set("SIP/103-00000002", "CONNECTEDLINE(num)=101") in new stack
  344.     -- Executing [s@macro-dial-one:42] Set("SIP/103-00000002", "D_OPTIONS=TtrI") in new stack
  345.     -- Executing [s@macro-dial-one:43] Macro("SIP/103-00000002", "dialout-one-predial-hook,") in new stack
  346.     -- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("SIP/103-00000002", "") in new stack
  347.     -- Executing [s@macro-dial-one:44] ExecIf("SIP/103-00000002", "0?Set(D_OPTIONS=trII)") in new stack
  348.     -- Executing [s@macro-dial-one:45] Dial("SIP/103-00000002", "SIP/101,,TtrI") in new stack
  349.   == Using SIP RTP TOS bits 184
  350.   == Using SIP RTP CoS mark 5
  351. Audio is at 17410
  352. Adding codec 100003 (ulaw) to SDP
  353. Adding codec 100004 (alaw) to SDP
  354. Adding codec 100002 (gsm) to SDP
  355. Adding codec 100011 (g726) to SDP
  356. Adding non-codec 0x1 (telephone-event) to SDP
  357. Reliably Transmitting (NAT) to 176.47.75.56:1201:
  358. INVITE sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066 SIP/2.0
  359. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK55f21544;rport
  360. Max-Forwards: 70
  361. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as4ef21b45
  362. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  363. Contact: <sip:103@127.0.0.1:5060>
  364. Call-ID: 6079165a0706206d3fadc0c951449748@127.0.0.1:5060
  365. CSeq: 102 INVITE
  366. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  367. Date: Tue, 14 Mar 2017 21:37:55 GMT
  368. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  369. Supported: replaces, timer
  370. P-Asserted-Identity: "Zubair Laptop" <sip:103@127.0.0.1>
  371. Content-Type: application/sdp
  372. Content-Length: 308
  373.  
  374. v=0
  375. o=root 1885627301 1885627301 IN IP4 127.0.0.1
  376. s=Asterisk PBX 11.16.0
  377. c=IN IP4 127.0.0.1
  378. t=0 0
  379. m=audio 17410 RTP/AVP 0 8 3 111 101
  380. a=rtpmap:0 PCMU/8000
  381. a=rtpmap:8 PCMA/8000
  382. a=rtpmap:3 GSM/8000
  383. a=rtpmap:111 G726-32/8000
  384. a=rtpmap:101 telephone-event/8000
  385. a=fmtp:101 0-16
  386. a=ptime:20
  387. a=sendrecv
  388.  
  389. ---
  390.     -- Called SIP/101
  391.  
  392. <--- Transmitting (NAT) to 192.168.112.22:5060 --->
  393. SIP/2.0 180 Ringing
  394. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK1416131351383425855;received=192.168.112.22;rport=5060
  395. From: 103 <sip:103@192.168.112.50:5060>;tag=1813713356
  396. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as0bff9024
  397. Call-ID: 21764281622511-100381027830739@192.168.112.22
  398. CSeq: 2 INVITE
  399. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  400. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  401. Supported: replaces, timer
  402. Contact: <sip:101@192.168.112.50:5060>
  403. P-Asserted-Identity: "Home IP Phone" <sip:101@192.168.112.50>
  404. Content-Length: 0
  405.  
  406.  
  407. <------------>
  408.     -- Connected line update to SIP/103-00000002 prevented.
  409. Retransmitting #1 (NAT) to 176.47.75.56:1201:
  410. INVITE sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066 SIP/2.0
  411. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK55f21544;rport
  412. Max-Forwards: 70
  413. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as4ef21b45
  414. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  415. Contact: <sip:103@127.0.0.1:5060>
  416. Call-ID: 6079165a0706206d3fadc0c951449748@127.0.0.1:5060
  417. CSeq: 102 INVITE
  418. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  419. Date: Tue, 14 Mar 2017 21:37:55 GMT
  420. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  421. Supported: replaces, timer
  422. P-Asserted-Identity: "Zubair Laptop" <sip:103@127.0.0.1>
  423. Content-Type: application/sdp
  424. Content-Length: 308
  425.  
  426. v=0
  427. o=root 1885627301 1885627301 IN IP4 127.0.0.1
  428. s=Asterisk PBX 11.16.0
  429. c=IN IP4 127.0.0.1
  430. t=0 0
  431. m=audio 17410 RTP/AVP 0 8 3 111 101
  432. a=rtpmap:0 PCMU/8000
  433. a=rtpmap:8 PCMA/8000
  434. a=rtpmap:3 GSM/8000
  435. a=rtpmap:111 G726-32/8000
  436. a=rtpmap:101 telephone-event/8000
  437. a=fmtp:101 0-16
  438. a=ptime:20
  439. a=sendrecv
  440.  
  441. ---
  442. Retransmitting #2 (NAT) to 176.47.75.56:1201:
  443. INVITE sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066 SIP/2.0
  444. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK55f21544;rport
  445. Max-Forwards: 70
  446. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as4ef21b45
  447. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  448. Contact: <sip:103@127.0.0.1:5060>
  449. Call-ID: 6079165a0706206d3fadc0c951449748@127.0.0.1:5060
  450. CSeq: 102 INVITE
  451. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  452. Date: Tue, 14 Mar 2017 21:37:55 GMT
  453. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  454. Supported: replaces, timer
  455. P-Asserted-Identity: "Zubair Laptop" <sip:103@127.0.0.1>
  456. Content-Type: application/sdp
  457. Content-Length: 308
  458.  
  459. v=0
  460. o=root 1885627301 1885627301 IN IP4 127.0.0.1
  461. s=Asterisk PBX 11.16.0
  462. c=IN IP4 127.0.0.1
  463. t=0 0
  464. m=audio 17410 RTP/AVP 0 8 3 111 101
  465. a=rtpmap:0 PCMU/8000
  466. a=rtpmap:8 PCMA/8000
  467. a=rtpmap:3 GSM/8000
  468. a=rtpmap:111 G726-32/8000
  469. a=rtpmap:101 telephone-event/8000
  470. a=fmtp:101 0-16
  471. a=ptime:20
  472. a=sendrecv
  473.  
  474. ---
  475.  
  476. <--- SIP read from UDP:176.47.75.56:1201 --->
  477. SIP/2.0 180 Ringing
  478. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK55f21544;rport=5060;received=94.48.30.110
  479. Contact: <sip:101@176.47.75.56:1201;transport=UDP>
  480. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=e774972b
  481. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as4ef21b45
  482. Call-ID: 6079165a0706206d3fadc0c951449748@127.0.0.1:5060
  483. CSeq: 102 INVITE
  484. User-Agent: Zoiper rv2.8.30
  485. Allow-Events: presence, kpml, talk
  486. Content-Length: 0
  487.  
  488. <------------->
  489. --- (10 headers 0 lines) ---
  490. list_route: hop: <sip:101@176.47.75.56:1201;transport=UDP>
  491.     -- SIP/101-00000003 is ringing
  492.  
  493. <--- Transmitting (NAT) to 192.168.112.22:5060 --->
  494. SIP/2.0 180 Ringing
  495. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK1416131351383425855;received=192.168.112.22;rport=5060
  496. From: 103 <sip:103@192.168.112.50:5060>;tag=1813713356
  497. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as0bff9024
  498. Call-ID: 21764281622511-100381027830739@192.168.112.22
  499. CSeq: 2 INVITE
  500. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  501. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  502. Supported: replaces, timer
  503. Contact: <sip:101@192.168.112.50:5060>
  504. Content-Length: 0
  505.  
  506.  
  507. <------------>
  508.  
  509. <--- SIP read from UDP:176.47.75.56:1201 --->
  510. SIP/2.0 100 Trying
  511. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK55f21544;rport=5060;received=94.48.30.110
  512. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  513. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as4ef21b45
  514. Call-ID: 6079165a0706206d3fadc0c951449748@127.0.0.1:5060
  515. CSeq: 102 INVITE
  516. Content-Length: 0
  517.  
  518. <------------->
  519. --- (7 headers 0 lines) ---
  520.  
  521. <--- SIP read from UDP:176.47.75.56:1201 --->
  522. SIP/2.0 200 OK
  523. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK55f21544;rport=5060;received=94.48.30.110
  524. Contact: <sip:101@176.47.75.56:1201;transport=UDP>
  525. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=e774972b
  526. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as4ef21b45
  527. Call-ID: 6079165a0706206d3fadc0c951449748@127.0.0.1:5060
  528. CSeq: 102 INVITE
  529. Content-Type: application/sdp
  530. User-Agent: Zoiper rv2.8.30
  531. Allow-Events: presence, kpml, talk
  532. Content-Length: 235
  533.  
  534. v=0
  535. o=Zoiper 0 1 IN IP4 127.0.0.1
  536. s=Zoiper
  537. c=IN IP4 127.0.0.1
  538. t=0 0
  539. m=audio 33670 RTP/AVP 0 3 8 101
  540. a=rtpmap:0 PCMU/8000
  541. a=rtpmap:3 GSM/8000
  542. a=rtpmap:8 PCMA/8000
  543. a=rtpmap:101 telephone-event/8000
  544. a=fmtp:101 0-16
  545. a=sendrecv
  546. <------------->
  547. --- (11 headers 12 lines) ---
  548. Found RTP audio format 0
  549. Found RTP audio format 3
  550. Found RTP audio format 8
  551. Found RTP audio format 101
  552. Found audio description format PCMU for ID 0
  553. Found audio description format GSM for ID 3
  554. Found audio description format PCMA for ID 8
  555. Found audio description format telephone-event for ID 101
  556. Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)
  557. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  558. Peer audio RTP is at port 127.0.0.1:33670
  559. list_route: hop: <sip:101@176.47.75.56:1201;transport=UDP>
  560. set_destination: Parsing <sip:101@176.47.75.56:1201;transport=UDP> for address/port to send to
  561. set_destination: set destination to 176.47.75.56:1201
  562. Transmitting (NAT) to 176.47.75.56:1201:
  563. ACK sip:101@176.47.75.56:1201;transport=UDP SIP/2.0
  564. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK455a6562;rport
  565. Max-Forwards: 70
  566. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as4ef21b45
  567. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=e774972b
  568. Contact: <sip:103@127.0.0.1:5060>
  569. Call-ID: 6079165a0706206d3fadc0c951449748@127.0.0.1:5060
  570. CSeq: 102 ACK
  571. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  572. Content-Length: 0
  573.  
  574.  
  575. ---
  576.     -- Connected line update to SIP/103-00000002 prevented.
  577.     -- SIP/101-00000003 answered SIP/103-00000002
  578. Audio is at 16476
  579. Adding codec 100003 (ulaw) to SDP
  580. Adding codec 100004 (alaw) to SDP
  581. Adding codec 100011 (g726) to SDP
  582. Adding non-codec 0x1 (telephone-event) to SDP
  583.  
  584. <--- Reliably Transmitting (NAT) to 192.168.112.22:5060 --->
  585. SIP/2.0 200 OK
  586. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK1416131351383425855;received=192.168.112.22;rport=5060
  587. From: 103 <sip:103@192.168.112.50:5060>;tag=1813713356
  588. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as0bff9024
  589. Call-ID: 21764281622511-100381027830739@192.168.112.22
  590. CSeq: 2 INVITE
  591. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  592. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  593. Supported: replaces, timer
  594. Contact: <sip:101@192.168.112.50:5060>
  595. P-Asserted-Identity: "Home IP Phone" <sip:101@192.168.112.50>
  596. Content-Type: application/sdp
  597. Content-Length: 289
  598.  
  599. v=0
  600. o=root 605756334 605756334 IN IP4 192.168.112.50
  601. s=Asterisk PBX 11.16.0
  602. c=IN IP4 192.168.112.50
  603. t=0 0
  604. m=audio 16476 RTP/AVP 0 8 2 101
  605. a=rtpmap:0 PCMU/8000
  606. a=rtpmap:8 PCMA/8000
  607. a=rtpmap:2 G726-32/8000
  608. a=rtpmap:101 telephone-event/8000
  609. a=fmtp:101 0-16
  610. a=ptime:20
  611. a=sendrecv
  612.  
  613. <------------>
  614. Retransmitting #1 (NAT) to 192.168.112.22:5060:
  615. SIP/2.0 200 OK
  616. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK1416131351383425855;received=192.168.112.22;rport=5060
  617. From: 103 <sip:103@192.168.112.50:5060>;tag=1813713356
  618. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as0bff9024
  619. Call-ID: 21764281622511-100381027830739@192.168.112.22
  620. CSeq: 2 INVITE
  621. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  622. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  623. Supported: replaces, timer
  624. Contact: <sip:101@192.168.112.50:5060>
  625. P-Asserted-Identity: "Home IP Phone" <sip:101@192.168.112.50>
  626. Content-Type: application/sdp
  627. Content-Length: 289
  628.  
  629. v=0
  630. o=root 605756334 605756334 IN IP4 192.168.112.50
  631. s=Asterisk PBX 11.16.0
  632. c=IN IP4 192.168.112.50
  633. t=0 0
  634. m=audio 16476 RTP/AVP 0 8 2 101
  635. a=rtpmap:0 PCMU/8000
  636. a=rtpmap:8 PCMA/8000
  637. a=rtpmap:2 G726-32/8000
  638. a=rtpmap:101 telephone-event/8000
  639. a=fmtp:101 0-16
  640. a=ptime:20
  641. a=sendrecv
  642.  
  643. ---
  644.  
  645. <--- SIP read from UDP:192.168.112.22:5060 --->
  646. ACK sip:101@192.168.112.50:5060 SIP/2.0
  647. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK30926203701789123443
  648. From: 103 <sip:103@192.168.112.50:5060>;tag=1813713356
  649. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as0bff9024
  650. Call-ID: 21764281622511-100381027830739@192.168.112.22
  651. CSeq: 2 ACK
  652. Contact: <sip:103@192.168.112.22:5060>
  653. Max-Forwards: 70
  654. User-Agent: DLINK DPH-400SE FRU2.2.876.445
  655. Content-Length: 0
  656.  
  657. <------------->
  658. --- (10 headers 0 lines) ---
  659.  
  660. <--- SIP read from UDP:192.168.112.22:5060 --->
  661. ACK sip:101@192.168.112.50:5060 SIP/2.0
  662. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK30926203701789123443
  663. From: 103 <sip:103@192.168.112.50:5060>;tag=1813713356
  664. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as0bff9024
  665. Call-ID: 21764281622511-100381027830739@192.168.112.22
  666. CSeq: 2 ACK
  667. Contact: <sip:103@192.168.112.22:5060>
  668. Max-Forwards: 70
  669. User-Agent: DLINK DPH-400SE FRU2.2.876.445
  670. Content-Length: 0
  671.  
  672. <------------->
  673. --- (10 headers 0 lines) ---
  674.  
  675. <--- SIP read from UDP:176.47.75.56:1201 --->
  676. REGISTER sip:voip.nhksa.com;transport=UDP SIP/2.0
  677. Via: SIP/2.0/UDP 176.47.75.56:1201;branch=z9hG4bK-524287-1---746aa8f4f0717d57;rport
  678. Max-Forwards: 70
  679. Contact: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  680. To: <sip:101@voip.nhksa.com;transport=UDP>
  681. From: <sip:101@voip.nhksa.com;transport=UDP>;tag=76d1d834
  682. Call-ID: No0rJm9Tm619Brig8s8gmg..
  683. CSeq: 197 REGISTER
  684. Expires: 60
  685. User-Agent: Zoiper rv2.8.30
  686. Authorization: Digest username="101",realm="asterisk",nonce="589f34f1",uri="sip:voip.nhksa.com;transport=UDP",response="04e8fcce70c5fe4c7288ad5ffe782124",algorithm=MD5
  687. Allow-Events: presence, kpml, talk
  688. Content-Length: 0
  689.  
  690. <------------->
  691. --- (13 headers 0 lines) ---
  692. Sending to 176.47.75.56:1201 (NAT)
  693. Sending to 176.47.75.56:1201 (NAT)
  694.  
  695. <--- Transmitting (NAT) to 176.47.75.56:1201 --->
  696. SIP/2.0 401 Unauthorized
  697. Via: SIP/2.0/UDP 176.47.75.56:1201;branch=z9hG4bK-524287-1---746aa8f4f0717d57;received=176.47.75.56;rport=1201
  698. From: <sip:101@voip.nhksa.com;transport=UDP>;tag=76d1d834
  699. To: <sip:101@voip.nhksa.com;transport=UDP>;tag=as06e39793
  700. Call-ID: No0rJm9Tm619Brig8s8gmg..
  701. CSeq: 197 REGISTER
  702. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  703. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  704. Supported: replaces, timer
  705. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2cb5c8dc"
  706. Content-Length: 0
  707.  
  708.  
  709. <------------>
  710. Scheduling destruction of SIP dialog 'No0rJm9Tm619Brig8s8gmg..' in 32000 ms (Method: REGISTER)
  711.        > 0x224d2d0 -- Probation passed - setting RTP source address to 192.168.112.22:10020
  712.  
  713. <--- SIP read from UDP:176.47.75.56:1201 --->
  714. REGISTER sip:voip.nhksa.com;transport=UDP SIP/2.0
  715. Via: SIP/2.0/UDP 176.47.75.56:1201;branch=z9hG4bK-524287-1---481d63bafdb78861;rport
  716. Max-Forwards: 70
  717. Contact: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  718. To: <sip:101@voip.nhksa.com;transport=UDP>
  719. From: <sip:101@voip.nhksa.com;transport=UDP>;tag=76d1d834
  720. Call-ID: No0rJm9Tm619Brig8s8gmg..
  721. CSeq: 198 REGISTER
  722. Expires: 60
  723. User-Agent: Zoiper rv2.8.30
  724. Authorization: Digest username="101",realm="asterisk",nonce="2cb5c8dc",uri="sip:voip.nhksa.com;transport=UDP",response="3846565fa8202b82ba5263976748411c",algorithm=MD5
  725. Allow-Events: presence, kpml, talk
  726. Content-Length: 0
  727.  
  728. <------------->
  729. --- (13 headers 0 lines) ---
  730. Sending to 176.47.75.56:1201 (NAT)
  731. Reliably Transmitting (NAT) to 176.47.75.56:1201:
  732. OPTIONS sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066 SIP/2.0
  733. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK28de99ce;rport
  734. Max-Forwards: 70
  735. From: "Unknown" <sip:Unknown@127.0.0.1>;tag=as0578c02a
  736. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>
  737. Contact: <sip:Unknown@127.0.0.1:5060>
  738. Call-ID: 1c96267266269f831f1a2ec4150beaae@127.0.0.1:5060
  739. CSeq: 102 OPTIONS
  740. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  741. Date: Tue, 14 Mar 2017 21:37:59 GMT
  742. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  743. Supported: replaces, timer
  744. Content-Length: 0
  745.  
  746.  
  747. ---
  748.  
  749. <--- Transmitting (NAT) to 176.47.75.56:1201 --->
  750. SIP/2.0 200 OK
  751. Via: SIP/2.0/UDP 176.47.75.56:1201;branch=z9hG4bK-524287-1---481d63bafdb78861;received=176.47.75.56;rport=1201
  752. From: <sip:101@voip.nhksa.com;transport=UDP>;tag=76d1d834
  753. To: <sip:101@voip.nhksa.com;transport=UDP>;tag=as06e39793
  754. Call-ID: No0rJm9Tm619Brig8s8gmg..
  755. CSeq: 198 REGISTER
  756. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  757. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  758. Supported: replaces, timer
  759. Expires: 60
  760. Contact: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;expires=60
  761. Date: Tue, 14 Mar 2017 21:37:59 GMT
  762. Content-Length: 0
  763.  
  764.  
  765. <------------>
  766. Scheduling destruction of SIP dialog 'No0rJm9Tm619Brig8s8gmg..' in 32000 ms (Method: REGISTER)
  767.  
  768. <--- SIP read from UDP:176.47.75.56:1201 --->
  769. SIP/2.0 200 OK
  770. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK28de99ce;rport=5060;received=94.48.30.110
  771. Contact: <sip:192.168.33.20:49514>
  772. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=6234a13a
  773. From: "Unknown" <sip:Unknown@127.0.0.1>;tag=as0578c02a
  774. Call-ID: 1c96267266269f831f1a2ec4150beaae@127.0.0.1:5060
  775. CSeq: 102 OPTIONS
  776. Accept: application/sdp, application/sdp
  777. Accept-Language: en
  778. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  779. Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
  780. User-Agent: Zoiper rv2.8.30
  781. Allow-Events: presence, kpml, talk
  782. Content-Length: 0
  783.  
  784. <------------->
  785. --- (14 headers 0 lines) ---
  786. Really destroying SIP dialog '1c96267266269f831f1a2ec4150beaae@127.0.0.1:5060' Method: OPTIONS
  787.  
  788. <--- SIP read from UDP:176.47.75.56:1201 --->
  789.  
  790.  
  791. <------------->
  792.  
  793. <--- SIP read from UDP:192.168.112.22:5060 --->
  794. BYE sip:101@192.168.112.50:5060 SIP/2.0
  795. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK1755630339743020620
  796. From: 103 <sip:103@192.168.112.50:5060>;tag=1813713356
  797. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as0bff9024
  798. Call-ID: 21764281622511-100381027830739@192.168.112.22
  799. CSeq: 3 BYE
  800. Contact: <sip:103@192.168.112.22:5060>
  801. Max-Forwards: 70
  802. User-Agent: DLINK DPH-400SE FRU2.2.876.445
  803. Content-Length: 0
  804.  
  805. <------------->
  806. --- (10 headers 0 lines) ---
  807. Sending to 192.168.112.22:5060 (NAT)
  808. Scheduling destruction of SIP dialog '21764281622511-100381027830739@192.168.112.22' in 6400 ms (Method: BYE)
  809.  
  810. <--- Transmitting (NAT) to 192.168.112.22:5060 --->
  811. SIP/2.0 200 OK
  812. Via: SIP/2.0/UDP 192.168.112.22:5060;branch=z9hG4bK1755630339743020620;received=192.168.112.22;rport=5060
  813. From: 103 <sip:103@192.168.112.50:5060>;tag=1813713356
  814. To: "101" <sip:101@192.168.112.50;user=phone>;tag=as0bff9024
  815. Call-ID: 21764281622511-100381027830739@192.168.112.22
  816. CSeq: 3 BYE
  817. Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  818. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  819. Supported: replaces, timer
  820. Content-Length: 0
  821.  
  822.  
  823. <------------>
  824.     -- Executing [h@macro-dial-one:1] Macro("SIP/103-00000002", "hangupcall,") in new stack
  825.     -- Executing [s@macro-hangupcall:1] ExecIf("SIP/103-00000002", "0?Set(CDR(recordingfile)=.wav)") in new stack
  826.     -- Executing [s@macro-hangupcall:2] GotoIf("SIP/103-00000002", "1?theend") in new stack
  827.     -- Goto (macro-hangupcall,s,4)
  828.     -- Executing [s@macro-hangupcall:4] Hangup("SIP/103-00000002", "") in new stack
  829.   == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/103-00000002' in macro 'hangupcall'
  830.   == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/103-00000002'
  831. Scheduling destruction of SIP dialog '6079165a0706206d3fadc0c951449748@127.0.0.1:5060' in 6400 ms (Method: INVITE)
  832. set_destination: Parsing <sip:101@176.47.75.56:1201;transport=UDP> for address/port to send to
  833. set_destination: set destination to 176.47.75.56:1201
  834. Reliably Transmitting (NAT) to 176.47.75.56:1201:
  835. BYE sip:101@176.47.75.56:1201;transport=UDP SIP/2.0
  836. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6a98da27;rport
  837. Max-Forwards: 70
  838. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as4ef21b45
  839. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=e774972b
  840. Call-ID: 6079165a0706206d3fadc0c951449748@127.0.0.1:5060
  841. CSeq: 103 BYE
  842. User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
  843. X-Asterisk-HangupCause: Normal Clearing
  844. X-Asterisk-HangupCauseCode: 16
  845. Content-Length: 0
  846.  
  847.  
  848. ---
  849.   == Spawn extension (macro-dial-one, s, 45) exited non-zero on 'SIP/103-00000002' in macro 'dial-one'
  850.   == Spawn extension (macro-exten-vm, s, 16) exited non-zero on 'SIP/103-00000002' in macro 'exten-vm'
  851.   == Spawn extension (ext-local, 101, 2) exited non-zero on 'SIP/103-00000002'
  852.  
  853. <--- SIP read from UDP:176.47.75.56:1201 --->
  854. SIP/2.0 200 OK
  855. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK6a98da27;rport=5060;received=94.48.30.110
  856. Contact: <sip:101@176.47.75.56:1201;transport=UDP>
  857. To: <sip:101@176.47.75.56:1201;transport=UDP;rinstance=2d10da2def2bb066>;tag=e774972b
  858. From: "Zubair Laptop" <sip:103@127.0.0.1>;tag=as4ef21b45
  859. Call-ID: 6079165a0706206d3fadc0c951449748@127.0.0.1:5060
  860. CSeq: 103 BYE
  861. User-Agent: Zoiper rv2.8.30
  862. Content-Length: 0
  863.  
  864. <------------->
  865. --- (9 headers 0 lines) ---
  866. Really destroying SIP dialog '6079165a0706206d3fadc0c951449748@127.0.0.1:5060' Method: INVITE
  867. voip*CLI>
  868.  

Replies to Re: no Audio rss

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Re: Re: no Audio Sarthor text 10 Months ago.

Reply to "Re: no Audio "

Here you can reply to the paste above