problem

From radi, 1 Year ago, written in Plain Text, viewed 39 times.
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  1. ACK sip:1100000000@185.164.137.43:5160 SIP/2.0
  2. Via: SIP/2.0/UDP 178.75.237.247:27299;branch=z9hG4bK-524287-1---2f821ec1e6431b36;rport
  3. Max-Forwards: 70
  4. Contact: <sip:1100000001@178.75.237.247:27299;transport=UDP>
  5. To: <sip:1100000000@185.164.137.43:5160;transport=UDP>;tag=as51f6b667
  6. From: <sip:1100000001@185.164.137.43:5160;transport=UDP>;tag=9934d22a
  7. Call-ID: F3lWQUwSnJFR4BePDHmoag..
  8. CSeq: 2 ACK
  9. User-Agent: Zoiper rv2.8.26-mod
  10. Content-Length: 0
  11.  
  12. <------------->
  13. --- (10 headers 0 lines) ---
  14.        > 0x7fe6bc0dae50 -- Probation passed - setting RTP source address to 178.75.237.247:27180
  15.        > 0x7fe6bc0dae50 -- Probation passed - setting RTP source address to 178.75.237.247:27180
  16.  
  17. <--- SIP read from UDP:178.75.237.247:27299 --->
  18.  
  19.  
  20. <------------->
  21. Really destroying SIP dialog 'FzpdtwQOFXfrXu23nzia4g..' Method: REGISTER
  22. freepbx*CLI> exit
  23. Asterisk cleanly ending (0).
  24. Executing last minute cleanups
  25. [root@freepbx ~]# ^C
  26. [root@freepbx ~]# asterisk -rvvvvvv
  27. Asterisk 13.16.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  28. Created by Mark Spencer <markster@digium.com>
  29. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  30. This is free software, with components licensed under the GNU General Public
  31. License version 2 and other licenses; you are welcome to redistribute it under
  32. certain conditions. Type 'core show license' for details.
  33. =========================================================================
  34. Connected to Asterisk 13.16.0 currently running on freepbx (pid = 8197)
  35.  
  36. <--- SIP read from UDP:178.75.237.247:27040 --->
  37.  
  38.  
  39. <------------->
  40. Really destroying SIP dialog 'F3lWQUwSnJFR4BePDHmoag..' Method: BYE
  41. freepbx*CLI> sip set debug on
  42. SIP Debugging re-enabled
  43. freepbx*CLI> clear
  44. No such command 'clear' (type 'core show help clear' for other possible commands)
  45. freepbx*CLI> exit
  46. Asterisk cleanly ending (0).
  47. Executing last minute cleanups
  48. [root@freepbx ~]# clear
  49. [root@freepbx ~]# asterisk -rvvvvvv
  50. Asterisk 13.16.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  51. Created by Mark Spencer <markster@digium.com>
  52. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  53. This is free software, with components licensed under the GNU General Public
  54. License version 2 and other licenses; you are welcome to redistribute it under
  55. certain conditions. Type 'core show license' for details.
  56. =========================================================================
  57. Connected to Asterisk 13.16.0 currently running on freepbx (pid = 8197)
  58. freepbx*CLI> sip set debug on
  59. SIP Debugging re-enabled
  60.  
  61. <--- SIP read from UDP:178.75.237.247:27299 --->
  62. INVITE sip:1100000000@185.164.137.43:5160;transport=UDP SIP/2.0
  63. Via: SIP/2.0/UDP 178.75.237.247:27299;branch=z9hG4bK-524287-1---3b32f6dec6b0c888;rport
  64. Max-Forwards: 70
  65. Contact: <sip:1100000001@178.75.237.247:27299;transport=UDP>
  66. To: <sip:1100000000@185.164.137.43:5160;transport=UDP>
  67. From: <sip:1100000001@185.164.137.43:5160;transport=UDP>;tag=e6042e6c
  68. Call-ID: 01sSAl-uvzrXtD_d6kAn5A..
  69. CSeq: 1 INVITE
  70. Content-Type: application/sdp
  71. User-Agent: Zoiper rv2.8.26-mod
  72. Allow-Events: presence, kpml, talk
  73. Content-Length: 245
  74.  
  75. v=0
  76. o=Zoiper 0 0 IN IP4 178.75.237.247
  77. s=Zoiper
  78. c=IN IP4 178.75.237.247
  79. t=0 0
  80. m=audio 27396 RTP/AVP 3 0 8 101
  81. a=rtpmap:3 GSM/8000
  82. a=rtpmap:0 PCMU/8000
  83. a=rtpmap:8 PCMA/8000
  84. a=rtpmap:101 telephone-event/8000
  85. a=fmtp:101 0-16
  86. a=sendrecv
  87. <------------->
  88. --- (12 headers 12 lines) ---
  89. Sending to 178.75.237.247:27299 (NAT)
  90. Sending to 178.75.237.247:27299 (NAT)
  91. Using INVITE request as basis request - 01sSAl-uvzrXtD_d6kAn5A..
  92. Found peer '1100000001' for '1100000001' from 178.75.237.247:27299
  93.  
  94. <--- Reliably Transmitting (NAT) to 178.75.237.247:27299 --->
  95. SIP/2.0 401 Unauthorized
  96. Via: SIP/2.0/UDP 178.75.237.247:27299;branch=z9hG4bK-524287-1---3b32f6dec6b0c888;received=178.75.237.247;rport=27299
  97. From: <sip:1100000001@185.164.137.43:5160;transport=UDP>;tag=e6042e6c
  98. To: <sip:1100000000@185.164.137.43:5160;transport=UDP>;tag=as4222d69b
  99. Call-ID: 01sSAl-uvzrXtD_d6kAn5A..
  100. CSeq: 1 INVITE
  101. Server: FPBX-14.0.1.1(13.16.0)
  102. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  103. Supported: replaces, timer
  104. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="201235d8"
  105. Content-Length: 0
  106.  
  107.  
  108. <------------>
  109. Scheduling destruction of SIP dialog '01sSAl-uvzrXtD_d6kAn5A..' in 6400 ms (Method: INVITE)
  110.  
  111. <--- SIP read from UDP:178.75.237.247:27299 --->
  112. ACK sip:1100000000@185.164.137.43:5160;transport=UDP SIP/2.0
  113. Via: SIP/2.0/UDP 178.75.237.247:27299;branch=z9hG4bK-524287-1---3b32f6dec6b0c888;rport
  114. Max-Forwards: 70
  115. To: <sip:1100000000@185.164.137.43:5160;transport=UDP>;tag=as4222d69b
  116. From: <sip:1100000001@185.164.137.43:5160;transport=UDP>;tag=e6042e6c
  117. Call-ID: 01sSAl-uvzrXtD_d6kAn5A..
  118. CSeq: 1 ACK
  119. Content-Length: 0
  120.  
  121. <------------->
  122. --- (8 headers 0 lines) ---
  123.  
  124. <--- SIP read from UDP:178.75.237.247:27299 --->
  125. INVITE sip:1100000000@185.164.137.43:5160;transport=UDP SIP/2.0
  126. Via: SIP/2.0/UDP 178.75.237.247:27299;branch=z9hG4bK-524287-1---a492be17d3a66849;rport
  127. Max-Forwards: 70
  128. Contact: <sip:1100000001@178.75.237.247:27299;transport=UDP>
  129. To: <sip:1100000000@185.164.137.43:5160;transport=UDP>
  130. From: <sip:1100000001@185.164.137.43:5160;transport=UDP>;tag=e6042e6c
  131. Call-ID: 01sSAl-uvzrXtD_d6kAn5A..
  132. CSeq: 2 INVITE
  133. Content-Type: application/sdp
  134. User-Agent: Zoiper rv2.8.26-mod
  135. Authorization: Digest username="1100000001",realm="asterisk",nonce="201235d8",uri="sip:1100000000@185.164.137.43:5160;transport=UDP",response="bdcd102016ba8a05dc815e9aef80cc7b",algorithm=MD5
  136. Allow-Events: presence, kpml, talk
  137. Content-Length: 245
  138.  
  139. v=0
  140. o=Zoiper 0 0 IN IP4 178.75.237.247
  141. s=Zoiper
  142. c=IN IP4 178.75.237.247
  143. t=0 0
  144. m=audio 27396 RTP/AVP 3 0 8 101
  145. a=rtpmap:3 GSM/8000
  146. a=rtpmap:0 PCMU/8000
  147. a=rtpmap:8 PCMA/8000
  148. a=rtpmap:101 telephone-event/8000
  149. a=fmtp:101 0-16
  150. a=sendrecv
  151. <------------->
  152. --- (13 headers 12 lines) ---
  153. Sending to 178.75.237.247:27299 (NAT)
  154. Using INVITE request as basis request - 01sSAl-uvzrXtD_d6kAn5A..
  155. Found peer '1100000001' for '1100000001' from 178.75.237.247:27299
  156.   == Using SIP RTP TOS bits 184
  157.   == Using SIP RTP CoS mark 5
  158. Found RTP audio format 3
  159. Found RTP audio format 0
  160. Found RTP audio format 8
  161. Found RTP audio format 101
  162. Found audio description format GSM for ID 3
  163. Found audio description format PCMU for ID 0
  164. Found audio description format PCMA for ID 8
  165. Found audio description format telephone-event for ID 101
  166. Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
  167. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  168. Peer audio RTP is at port 178.75.237.247:27396
  169. Looking for 1100000000 in from-internal (domain 185.164.137.43)
  170. sip_route_dump: route/path hop: <sip:1100000001@178.75.237.247:27299;transport=UDP>
  171.  
  172. <--- Transmitting (NAT) to 178.75.237.247:27299 --->
  173. SIP/2.0 100 Trying
  174. Via: SIP/2.0/UDP 178.75.237.247:27299;branch=z9hG4bK-524287-1---a492be17d3a66849;received=178.75.237.247;rport=27299
  175. From: <sip:1100000001@185.164.137.43:5160;transport=UDP>;tag=e6042e6c
  176. To: <sip:1100000000@185.164.137.43:5160;transport=UDP>
  177. Call-ID: 01sSAl-uvzrXtD_d6kAn5A..
  178. CSeq: 2 INVITE
  179. Server: FPBX-14.0.1.1(13.16.0)
  180. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  181. Supported: replaces, timer
  182. Contact: <sip:1100000000@185.164.137.43:5160>
  183. Content-Length: 0
  184.  
  185.  
  186. <------------>
  187.     -- Executing [1100000000@from-internal:1] GotoIf("SIP/1100000001-00000020", "1?ext-local,1100000000,1:followme-check,1100000000,1") in new stack
  188.     -- Goto (ext-local,1100000000,1)
  189.     -- Executing [1100000000@ext-local:1] Set("SIP/1100000001-00000020", "__RINGTIMER=15") in new stack
  190.     -- Executing [1100000000@ext-local:2] Macro("SIP/1100000001-00000020", "exten-vm,novm,1100000000,0,0,0") in new stack
  191.     -- Executing [s@macro-exten-vm:1] Macro("SIP/1100000001-00000020", "user-callerid,") in new stack
  192.     -- Executing [s@macro-user-callerid:1] Set("SIP/1100000001-00000020", "TOUCH_MONITOR=1498937415.33") in new stack
  193.     -- Executing [s@macro-user-callerid:2] Set("SIP/1100000001-00000020", "AMPUSER=1100000001") in new stack
  194.     -- Executing [s@macro-user-callerid:3] GotoIf("SIP/1100000001-00000020", "0?report") in new stack
  195.     -- Executing [s@macro-user-callerid:4] ExecIf("SIP/1100000001-00000020", "1?Set(__REALCALLERIDNUM=1100000001)") in new stack
  196.     -- Executing [s@macro-user-callerid:5] Set("SIP/1100000001-00000020", "AMPUSER=1100000001") in new stack
  197.     -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1100000001-00000020", "0?limit") in new stack
  198.     -- Executing [s@macro-user-callerid:7] Set("SIP/1100000001-00000020", "AMPUSERCIDNAME=Nicole Foster") in new stack
  199.     -- Executing [s@macro-user-callerid:8] GotoIf("SIP/1100000001-00000020", "0?report") in new stack
  200.     -- Executing [s@macro-user-callerid:9] Set("SIP/1100000001-00000020", "AMPUSERCID=1100000001") in new stack
  201.     -- Executing [s@macro-user-callerid:10] Set("SIP/1100000001-00000020", "__DIAL_OPTIONS=Ttr") in new stack
  202.     -- Executing [s@macro-user-callerid:11] Set("SIP/1100000001-00000020", "CALLERID(all)="Nicole Foster" <1100000001>") in new stack
  203.     -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1100000001-00000020", "0?limit") in new stack
  204.     -- Executing [s@macro-user-callerid:13] ExecIf("SIP/1100000001-00000020", "0?Set(GROUP(concurrency_limit)=1100000001)") in new stack
  205.     -- Executing [s@macro-user-callerid:14] ExecIf("SIP/1100000001-00000020", "0?Set(CHANNEL(language)=)") in new stack
  206.     -- Executing [s@macro-user-callerid:15] GotoIf("SIP/1100000001-00000020", "0?continue") in new stack
  207.     -- Executing [s@macro-user-callerid:16] ExecIf("SIP/1100000001-00000020", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
  208.     -- Executing [s@macro-user-callerid:17] Set("SIP/1100000001-00000020", "__TTL=6") in new stack
  209.     -- Executing [s@macro-user-callerid:18] GotoIf("SIP/1100000001-00000020", "1?continue") in new stack
  210.     -- Goto (macro-user-callerid,s,29)
  211.     -- Executing [s@macro-user-callerid:29] Set("SIP/1100000001-00000020", "CALLERID(number)=1100000001") in new stack
  212.     -- Executing [s@macro-user-callerid:30] Set("SIP/1100000001-00000020", "CALLERID(name)=Nicole Foster") in new stack
  213.     -- Executing [s@macro-user-callerid:31] GotoIf("SIP/1100000001-00000020", "0?cnum") in new stack
  214.     -- Executing [s@macro-user-callerid:32] Set("SIP/1100000001-00000020", "CDR(cnam)=Nicole Foster") in new stack
  215.     -- Executing [s@macro-user-callerid:33] Set("SIP/1100000001-00000020", "CDR(cnum)=1100000001") in new stack
  216.     -- Executing [s@macro-user-callerid:34] Set("SIP/1100000001-00000020", "CHANNEL(language)=en") in new stack
  217.     -- Executing [s@macro-exten-vm:2] Set("SIP/1100000001-00000020", "RingGroupMethod=none") in new stack
  218.     -- Executing [s@macro-exten-vm:3] Set("SIP/1100000001-00000020", "__EXTTOCALL=1100000000") in new stack
  219.     -- Executing [s@macro-exten-vm:4] Set("SIP/1100000001-00000020", "__PICKUPMARK=1100000000") in new stack
  220.     -- Executing [s@macro-exten-vm:5] Set("SIP/1100000001-00000020", "RT=") in new stack
  221. [2017-07-01 19:30:15] ERROR[14644][C-00000012]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
  222.     -- Executing [s@macro-exten-vm:6] ExecIf("SIP/1100000001-00000020", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
  223. [2017-07-01 19:30:15] ERROR[14644][C-00000012]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
  224. [2017-07-01 19:30:15] ERROR[14644][C-00000012]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
  225.     -- Executing [s@macro-exten-vm:7] ExecIf("SIP/1100000001-00000020", "0?MacroExit()") in new stack
  226. [2017-07-01 19:30:15] ERROR[14644][C-00000012]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
  227. [2017-07-01 19:30:15] ERROR[14644][C-00000012]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
  228.     -- Executing [s@macro-exten-vm:8] ExecIf("SIP/1100000001-00000020", "0?Gosub(ext-intercom,*801100000000,1())") in new stack
  229. [2017-07-01 19:30:15] ERROR[14644][C-00000012]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
  230. [2017-07-01 19:30:15] ERROR[14644][C-00000012]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
  231.     -- Executing [s@macro-exten-vm:9] ExecIf("SIP/1100000001-00000020", "0?MacroExit()") in new stack
  232. [2017-07-01 19:30:15] ERROR[14644][C-00000012]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
  233. [2017-07-01 19:30:15] ERROR[14644][C-00000012]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
  234. [2017-07-01 19:30:15] WARNING[14644][C-00000012]: pbx_functions.c:460 func_args: Can't find trailing parenthesis for function 'DB(DEVICE/1100000000/dial'?
  235.     -- Executing [s@macro-exten-vm:10] ExecIf("SIP/1100000001-00000020", "0?ChanSpy(SIP/1100000000,q)") in new stack
  236. [2017-07-01 19:30:15] ERROR[14644][C-00000012]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
  237. [2017-07-01 19:30:15] WARNING[14644][C-00000012]: pbx_functions.c:460 func_args: Can't find trailing parenthesis for function 'DB(DEVICE/1100000000/dial'?
  238. [2017-07-01 19:30:15] ERROR[14644][C-00000012]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
  239.     -- Executing [s@macro-exten-vm:11] ExecIf("SIP/1100000001-00000020", "0?MacroExit()") in new stack
  240. [2017-07-01 19:30:15] ERROR[14644][C-00000012]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
  241.     -- Executing [s@macro-exten-vm:12] Gosub("SIP/1100000001-00000020", "sub-record-check,s,1(exten,1100000000,dontcare)") in new stack
  242.     -- Executing [s@sub-record-check:1] GotoIf("SIP/1100000001-00000020", "0?initialized") in new stack
  243.     -- Executing [s@sub-record-check:2] Set("SIP/1100000001-00000020", "__REC_STATUS=INITIALIZED") in new stack
  244.     -- Executing [s@sub-record-check:3] Set("SIP/1100000001-00000020", "NOW=1498937415") in new stack
  245.     -- Executing [s@sub-record-check:4] Set("SIP/1100000001-00000020", "__DAY=01") in new stack
  246.     -- Executing [s@sub-record-check:5] Set("SIP/1100000001-00000020", "__MONTH=07") in new stack
  247.     -- Executing [s@sub-record-check:6] Set("SIP/1100000001-00000020", "__YEAR=2017") in new stack
  248.     -- Executing [s@sub-record-check:7] Set("SIP/1100000001-00000020", "__TIMESTR=20170701-193015") in new stack
  249.     -- Executing [s@sub-record-check:8] Set("SIP/1100000001-00000020", "__FROMEXTEN=1100000001") in new stack
  250.     -- Executing [s@sub-record-check:9] Set("SIP/1100000001-00000020", "__MON_FMT=wav") in new stack
  251.     -- Executing [s@sub-record-check:10] NoOp("SIP/1100000001-00000020", "Recordings initialized") in new stack
  252.     -- Executing [s@sub-record-check:11] ExecIf("SIP/1100000001-00000020", "0?Set(ARG3=dontcare)") in new stack
  253.     -- Executing [s@sub-record-check:12] Set("SIP/1100000001-00000020", "REC_POLICY_MODE_SAVE=") in new stack
  254.     -- Executing [s@sub-record-check:13] ExecIf("SIP/1100000001-00000020", "0?Set(REC_STATUS=NO)") in new stack
  255.     -- Executing [s@sub-record-check:14] GotoIf("SIP/1100000001-00000020", "5?checkaction") in new stack
  256.     -- Goto (sub-record-check,s,17)
  257.     -- Executing [s@sub-record-check:17] GotoIf("SIP/1100000001-00000020", "1?sub-record-check,exten,1") in new stack
  258.     -- Goto (sub-record-check,exten,1)
  259.     -- Executing [exten@sub-record-check:1] NoOp("SIP/1100000001-00000020", "Exten Recording Check between 1100000001 and 1100000000") in new stack
  260.     -- Executing [exten@sub-record-check:2] Set("SIP/1100000001-00000020", "CALLTYPE=internal") in new stack
  261.     -- Executing [exten@sub-record-check:3] ExecIf("SIP/1100000001-00000020", "0?Set(CALLTYPE=)") in new stack
  262.     -- Executing [exten@sub-record-check:4] Set("SIP/1100000001-00000020", "CALLEE=dontcare") in new stack
  263.     -- Executing [exten@sub-record-check:5] ExecIf("SIP/1100000001-00000020", "0?Set(CALLEE=dontcare)") in new stack
  264.     -- Executing [exten@sub-record-check:6] GotoIf("SIP/1100000001-00000020", "0?callee") in new stack
  265.     -- Executing [exten@sub-record-check:7] GotoIf("SIP/1100000001-00000020", "1?caller") in new stack
  266.     -- Goto (sub-record-check,exten,13)
  267.     -- Executing [exten@sub-record-check:13] Set("SIP/1100000001-00000020", "RECMODE=dontcare") in new stack
  268.     -- Executing [exten@sub-record-check:14] ExecIf("SIP/1100000001-00000020", "0?Set(RECMODE=dontcare)") in new stack
  269.     -- Executing [exten@sub-record-check:15] ExecIf("SIP/1100000001-00000020", "1?Set(RECMODE=dontcare)") in new stack
  270.     -- Executing [exten@sub-record-check:16] Gosub("SIP/1100000001-00000020", "recordcheck,1(dontcare,internal,1100000000)") in new stack
  271.     -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/1100000001-00000020", "Starting recording check against dontcare") in new stack
  272.     -- Executing [recordcheck@sub-record-check:2] Goto("SIP/1100000001-00000020", "dontcare") in new stack
  273.     -- Goto (sub-record-check,recordcheck,3)
  274.     -- Executing [recordcheck@sub-record-check:3] Return("SIP/1100000001-00000020", "") in new stack
  275.     -- Executing [exten@sub-record-check:17] Return("SIP/1100000001-00000020", "") in new stack
  276.     -- Executing [s@macro-exten-vm:13] GotoIf("SIP/1100000001-00000020", "1?macrodial") in new stack
  277.     -- Goto (macro-exten-vm,s,19)
  278.     -- Executing [s@macro-exten-vm:19] GosubIf("SIP/1100000001-00000020", "0?clrheader,1()") in new stack
  279.     -- Executing [s@macro-exten-vm:20] Macro("SIP/1100000001-00000020", "dial-one,,Ttr,1100000000") in new stack
  280.     -- Executing [s@macro-dial-one:1] Set("SIP/1100000001-00000020", "DEXTEN=1100000000") in new stack
  281.     -- Executing [s@macro-dial-one:2] Set("SIP/1100000001-00000020", "__CRM_SOURCE=1100000001") in new stack
  282.     -- Executing [s@macro-dial-one:3] ExecIf("SIP/1100000001-00000020", "0?Set(EXTTOCALL=1100000000)") in new stack
  283.     -- Executing [s@macro-dial-one:4] Set("SIP/1100000001-00000020", "DIALSTATUS_CW=") in new stack
  284.     -- Executing [s@macro-dial-one:5] GosubIf("SIP/1100000001-00000020", "0?screen,1()") in new stack
  285.     -- Executing [s@macro-dial-one:6] GosubIf("SIP/1100000001-00000020", "0?cf,1()") in new stack
  286.     -- Executing [s@macro-dial-one:7] GotoIf("SIP/1100000001-00000020", "1?skip1") in new stack
  287.     -- Goto (macro-dial-one,s,10)
  288.     -- Executing [s@macro-dial-one:10] GotoIf("SIP/1100000001-00000020", "0?nodial") in new stack
  289.     -- Executing [s@macro-dial-one:11] GotoIf("SIP/1100000001-00000020", "0?continue") in new stack
  290.     -- Executing [s@macro-dial-one:12] Set("SIP/1100000001-00000020", "EXTHASCW=ENABLED") in new stack
  291.     -- Executing [s@macro-dial-one:13] GotoIf("SIP/1100000001-00000020", "0?next1:cwinusebusy") in new stack
  292.     -- Goto (macro-dial-one,s,25)
  293.     -- Executing [s@macro-dial-one:25] GotoIf("SIP/1100000001-00000020", "0?next3:continue") in new stack
  294.     -- Goto (macro-dial-one,s,27)
  295.     -- Executing [s@macro-dial-one:27] GotoIf("SIP/1100000001-00000020", "0?nodial") in new stack
  296.     -- Executing [s@macro-dial-one:28] GosubIf("SIP/1100000001-00000020", "1?dstring,1():dlocal,1()") in new stack
  297.     -- Executing [dstring@macro-dial-one:1] Set("SIP/1100000001-00000020", "DSTRING=") in new stack
  298.     -- Executing [dstring@macro-dial-one:2] Set("SIP/1100000001-00000020", "DEVICES=1100000000") in new stack
  299.     -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/1100000001-00000020", "0?Return()") in new stack
  300.     -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/1100000001-00000020", "0?Set(DEVICES=100000000)") in new stack
  301.     -- Executing [dstring@macro-dial-one:5] Set("SIP/1100000001-00000020", "LOOPCNT=1") in new stack
  302.     -- Executing [dstring@macro-dial-one:6] Set("SIP/1100000001-00000020", "ITER=1") in new stack
  303.     -- Executing [dstring@macro-dial-one:7] Set("SIP/1100000001-00000020", "THISDIAL=SIP/1100000000") in new stack
  304.     -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/1100000001-00000020", "1?zap2dahdi,1()") in new stack
  305.     -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/1100000001-00000020", "0?Return()") in new stack
  306.     -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/1100000001-00000020", "NEWDIAL=") in new stack
  307.     -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/1100000001-00000020", "LOOPCNT2=1") in new stack
  308.     -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/1100000001-00000020", "ITER2=1") in new stack
  309.     -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/1100000001-00000020", "THISPART2=SIP/1100000000") in new stack
  310.     -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/1100000001-00000020", "0?Set(THISPART2=DAHDI/1100000000)") in new stack
  311.     -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/1100000001-00000020", "NEWDIAL=SIP/1100000000&") in new stack
  312.     -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/1100000001-00000020", "ITER2=2") in new stack
  313.     -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/1100000001-00000020", "0?begin2") in new stack
  314.     -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/1100000001-00000020", "THISDIAL=SIP/1100000000") in new stack
  315.     -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/1100000001-00000020", "") in new stack
  316.     -- Executing [dstring@macro-dial-one:9] GotoIf("SIP/1100000001-00000020", "1?docheck") in new stack
  317.     -- Goto (macro-dial-one,dstring,14)
  318.     -- Executing [dstring@macro-dial-one:14] GotoIf("SIP/1100000001-00000020", "0?skipset") in new stack
  319.     -- Executing [dstring@macro-dial-one:15] Set("SIP/1100000001-00000020", "DSTRING=SIP/1100000000&") in new stack
  320.     -- Executing [dstring@macro-dial-one:16] Set("SIP/1100000001-00000020", "ITER=2") in new stack
  321.     -- Executing [dstring@macro-dial-one:17] GotoIf("SIP/1100000001-00000020", "0?begin") in new stack
  322.     -- Executing [dstring@macro-dial-one:18] ExecIf("SIP/1100000001-00000020", "0?Return()") in new stack
  323.     -- Executing [dstring@macro-dial-one:19] Set("SIP/1100000001-00000020", "DSTRING=SIP/1100000000") in new stack
  324.     -- Executing [dstring@macro-dial-one:20] Return("SIP/1100000001-00000020", "") in new stack
  325.     -- Executing [s@macro-dial-one:29] GotoIf("SIP/1100000001-00000020", "0?nodial") in new stack
  326.     -- Executing [s@macro-dial-one:30] GotoIf("SIP/1100000001-00000020", "0?skiptrace") in new stack
  327.     -- Executing [s@macro-dial-one:31] GosubIf("SIP/1100000001-00000020", "1?ctset,1():ctclear,1()") in new stack
  328.     -- Executing [ctset@macro-dial-one:1] Set("SIP/1100000001-00000020", "DB(CALLTRACE/1100000000)=1100000001") in new stack
  329.     -- Executing [ctset@macro-dial-one:2] Return("SIP/1100000001-00000020", "") in new stack
  330.     -- Executing [s@macro-dial-one:32] Set("SIP/1100000001-00000020", "D_OPTIONS=Ttr") in new stack
  331.     -- Executing [s@macro-dial-one:33] NoOp("SIP/1100000001-00000020", "Blind Transfer: , Attended Transfer: , User: 1100000001, Alert Info: ") in new stack
  332.     -- Executing [s@macro-dial-one:34] ExecIf("SIP/1100000001-00000020", "1?Set(ALERT_INFO=)") in new stack
  333.     -- Executing [s@macro-dial-one:35] ExecIf("SIP/1100000001-00000020", "0?Set(ALERT_INFO=)") in new stack
  334.     -- Executing [s@macro-dial-one:36] ExecIf("SIP/1100000001-00000020", "0?Set(ALERT_INFO=)") in new stack
  335.     -- Executing [s@macro-dial-one:37] ExecIf("SIP/1100000001-00000020", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
  336.     -- Executing [s@macro-dial-one:38] ExecIf("SIP/1100000001-00000020", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
  337.     -- Executing [s@macro-dial-one:39] GosubIf("SIP/1100000001-00000020", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
  338.     -- Executing [s@macro-dial-one:40] ExecIf("SIP/1100000001-00000020", "0?Set(CHANNEL(musicclass)=)") in new stack
  339.     -- Executing [s@macro-dial-one:41] GosubIf("SIP/1100000001-00000020", "0?qwait,1()") in new stack
  340.     -- Executing [s@macro-dial-one:42] Set("SIP/1100000001-00000020", "__CWIGNORE=") in new stack
  341.     -- Executing [s@macro-dial-one:43] Set("SIP/1100000001-00000020", "__KEEPCID=TRUE") in new stack
  342.     -- Executing [s@macro-dial-one:44] GotoIf("SIP/1100000001-00000020", "0?usegoto,1") in new stack
  343.     -- Executing [s@macro-dial-one:45] GotoIf("SIP/1100000001-00000020", "0?godial") in new stack
  344.     -- Executing [s@macro-dial-one:46] Gosub("SIP/1100000001-00000020", "sub-presencestate-display,s,1(1100000000)") in new stack
  345.     -- Executing [s@sub-presencestate-display:1] Goto("SIP/1100000001-00000020", "state-not_set,1") in new stack
  346.     -- Goto (sub-presencestate-display,state-not_set,1)
  347.     -- Executing [state-not_set@sub-presencestate-display:1] Set("SIP/1100000001-00000020", "PRESENCESTATE_DISPLAY=") in new stack
  348.     -- Executing [state-not_set@sub-presencestate-display:2] Return("SIP/1100000001-00000020", "") in new stack
  349.     -- Executing [s@macro-dial-one:47] Set("SIP/1100000001-00000020", "CONNECTEDLINE(name,i)=Radoslav Petrov") in new stack
  350.     -- Executing [s@macro-dial-one:48] Set("SIP/1100000001-00000020", "CONNECTEDLINE(num)=1100000000") in new stack
  351.     -- Executing [s@macro-dial-one:49] Set("SIP/1100000001-00000020", "D_OPTIONS=TtrI") in new stack
  352.     -- Executing [s@macro-dial-one:50] Macro("SIP/1100000001-00000020", "dialout-one-predial-hook,") in new stack
  353.     -- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("SIP/1100000001-00000020", "") in new stack
  354.     -- Executing [s@macro-dial-one:51] ExecIf("SIP/1100000001-00000020", "0?Set(D_OPTIONS=trII)") in new stack
  355.     -- Executing [s@macro-dial-one:52] NoOp("SIP/1100000001-00000020", "") in new stack
  356.     -- Executing [s@macro-dial-one:53] Dial("SIP/1100000001-00000020", "SIP/1100000000,,TtrIb(func-apply-sipheaders^s^1)") in new stack
  357.   == Using SIP RTP TOS bits 184
  358.   == Using SIP RTP CoS mark 5
  359.     -- SIP/1100000000-00000021 Internal Gosub(func-apply-sipheaders,s,1) start
  360.     -- Executing [s@func-apply-sipheaders:1] ExecIf("SIP/1100000000-00000021", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
  361.     -- Executing [s@func-apply-sipheaders:2] NoOp("SIP/1100000000-00000021", "Applying SIP Headers to channel") in new stack
  362.     -- Executing [s@func-apply-sipheaders:3] Set("SIP/1100000000-00000021", "SIPHEADERKEYS=") in new stack
  363.     -- Executing [s@func-apply-sipheaders:4] While("SIP/1100000000-00000021", "0") in new stack
  364.     -- Jumping to priority 8
  365.     -- Executing [s@func-apply-sipheaders:9] Return("SIP/1100000000-00000021", "") in new stack
  366.   == Spawn extension (from-internal, 1100000000, 1) exited non-zero on 'SIP/1100000000-00000021'
  367.     -- SIP/1100000000-00000021 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
  368. Audio is at 18740
  369. Adding codec ulaw to SDP
  370. Adding codec alaw to SDP
  371. Adding codec gsm to SDP
  372. Adding codec g726 to SDP
  373. Adding codec g722 to SDP
  374. Adding non-codec 0x1 (telephone-event) to SDP
  375. Reliably Transmitting (NAT) to 178.75.237.247:27040:
  376. INVITE sip:1100000000@178.75.237.247:27040;transport=UDP;rinstance=bfa9274abd4d9c27 SIP/2.0
  377. Via: SIP/2.0/UDP 185.164.137.43:5160;branch=z9hG4bK1f0b225f;rport
  378. Max-Forwards: 70
  379. From: "Nicole Foster" <sip:1100000001@185.164.137.43:5160>;tag=as1f74d5cb
  380. To: <sip:1100000000@178.75.237.247:27040;transport=UDP;rinstance=bfa9274abd4d9c27>
  381. Contact: <sip:1100000001@185.164.137.43:5160>
  382. Call-ID: 498d76834af41d9c4c3fa98e681ce1ec@185.164.137.43:5160
  383. CSeq: 102 INVITE
  384. User-Agent: FPBX-14.0.1.1(13.16.0)
  385. Date: Sat, 01 Jul 2017 19:30:15 GMT
  386. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  387. Supported: replaces, timer
  388. P-Asserted-Identity: "Nicole Foster" <sip:1100000001@185.164.137.43>
  389. Content-Type: application/sdp
  390. Content-Length: 358
  391.  
  392. v=0
  393. o=root 1070748849 1070748849 IN IP4 185.164.137.43
  394. s=Asterisk PBX 13.16.0
  395. c=IN IP4 185.164.137.43
  396. t=0 0
  397. m=audio 18740 RTP/AVP 0 8 3 111 9 101
  398. a=rtpmap:0 PCMU/8000
  399. a=rtpmap:8 PCMA/8000
  400. a=rtpmap:3 GSM/8000
  401. a=rtpmap:111 G726-32/8000
  402. a=rtpmap:9 G722/8000
  403. a=rtpmap:101 telephone-event/8000
  404. a=fmtp:101 0-16
  405. a=ptime:20
  406. a=maxptime:150
  407. a=sendrecv
  408.  
  409. ---
  410.     -- Called SIP/1100000000
  411.  
  412. <--- Transmitting (NAT) to 178.75.237.247:27299 --->
  413. SIP/2.0 180 Ringing
  414. Via: SIP/2.0/UDP 178.75.237.247:27299;branch=z9hG4bK-524287-1---a492be17d3a66849;received=178.75.237.247;rport=27299
  415. From: <sip:1100000001@185.164.137.43:5160;transport=UDP>;tag=e6042e6c
  416. To: <sip:1100000000@185.164.137.43:5160;transport=UDP>;tag=as65adc904
  417. Call-ID: 01sSAl-uvzrXtD_d6kAn5A..
  418. CSeq: 2 INVITE
  419. Server: FPBX-14.0.1.1(13.16.0)
  420. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  421. Supported: replaces, timer
  422. Contact: <sip:1100000000@185.164.137.43:5160>
  423. P-Asserted-Identity: "Radoslav Petrov" <sip:1100000000@185.164.137.43>
  424. Content-Length: 0
  425.  
  426.  
  427. <------------>
  428.     -- Connected line update to SIP/1100000001-00000020 prevented.
  429. Retransmitting #1 (NAT) to 178.75.237.247:27040:
  430. INVITE sip:1100000000@178.75.237.247:27040;transport=UDP;rinstance=bfa9274abd4d9c27 SIP/2.0
  431. Via: SIP/2.0/UDP 185.164.137.43:5160;branch=z9hG4bK1f0b225f;rport
  432. Max-Forwards: 70
  433. From: "Nicole Foster" <sip:1100000001@185.164.137.43:5160>;tag=as1f74d5cb
  434. To: <sip:1100000000@178.75.237.247:27040;transport=UDP;rinstance=bfa9274abd4d9c27>
  435. Contact: <sip:1100000001@185.164.137.43:5160>
  436. Call-ID: 498d76834af41d9c4c3fa98e681ce1ec@185.164.137.43:5160
  437. CSeq: 102 INVITE
  438. User-Agent: FPBX-14.0.1.1(13.16.0)
  439. Date: Sat, 01 Jul 2017 19:30:15 GMT
  440. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  441. Supported: replaces, timer
  442. P-Asserted-Identity: "Nicole Foster" <sip:1100000001@185.164.137.43>
  443. Content-Type: application/sdp
  444. Content-Length: 358
  445.  
  446. v=0
  447. o=root 1070748849 1070748849 IN IP4 185.164.137.43
  448. s=Asterisk PBX 13.16.0
  449. c=IN IP4 185.164.137.43
  450. t=0 0
  451. m=audio 18740 RTP/AVP 0 8 3 111 9 101
  452. a=rtpmap:0 PCMU/8000
  453. a=rtpmap:8 PCMA/8000
  454. a=rtpmap:3 GSM/8000
  455. a=rtpmap:111 G726-32/8000
  456. a=rtpmap:9 G722/8000
  457. a=rtpmap:101 telephone-event/8000
  458. a=fmtp:101 0-16
  459. a=ptime:20
  460. a=maxptime:150
  461. a=sendrecv
  462.  
  463. ---
  464.  
  465. <--- SIP read from UDP:178.75.237.247:27040 --->
  466. SIP/2.0 100 Trying
  467. Via: SIP/2.0/UDP 185.164.137.43:5160;branch=z9hG4bK1f0b225f;rport=5160
  468. To: <sip:1100000000@178.75.237.247:27040;transport=UDP;rinstance=bfa9274abd4d9c27>
  469. From: "Nicole Foster" <sip:1100000001@185.164.137.43:5160>;tag=as1f74d5cb
  470. Call-ID: 498d76834af41d9c4c3fa98e681ce1ec@185.164.137.43:5160
  471. CSeq: 102 INVITE
  472. Content-Length: 0
  473.  
  474. <------------->
  475. --- (7 headers 0 lines) ---
  476.  
  477. <--- SIP read from UDP:178.75.237.247:27040 --->
  478. SIP/2.0 180 Ringing
  479. Via: SIP/2.0/UDP 185.164.137.43:5160;branch=z9hG4bK1f0b225f;rport=5160
  480. Contact: <sip:1100000000@178.75.237.247:27040;transport=UDP>
  481. To: <sip:1100000000@178.75.237.247:27040;transport=UDP;rinstance=bfa9274abd4d9c27>;tag=26162c60
  482. From: "Nicole Foster" <sip:1100000001@185.164.137.43:5160>;tag=as1f74d5cb
  483. Call-ID: 498d76834af41d9c4c3fa98e681ce1ec@185.164.137.43:5160
  484. CSeq: 102 INVITE
  485. User-Agent: Zoiper rv2.8.26-mod
  486. Allow-Events: presence, kpml, talk
  487. Content-Length: 0
  488.  
  489. <------------->
  490. --- (10 headers 0 lines) ---
  491. sip_route_dump: route/path hop: <sip:1100000000@178.75.237.247:27040;transport=UDP>
  492.     -- SIP/1100000000-00000021 is ringing
  493.  
  494. <--- Transmitting (NAT) to 178.75.237.247:27299 --->
  495. SIP/2.0 180 Ringing
  496. Via: SIP/2.0/UDP 178.75.237.247:27299;branch=z9hG4bK-524287-1---a492be17d3a66849;received=178.75.237.247;rport=27299
  497. From: <sip:1100000001@185.164.137.43:5160;transport=UDP>;tag=e6042e6c
  498. To: <sip:1100000000@185.164.137.43:5160;transport=UDP>;tag=as65adc904
  499. Call-ID: 01sSAl-uvzrXtD_d6kAn5A..
  500. CSeq: 2 INVITE
  501. Server: FPBX-14.0.1.1(13.16.0)
  502. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  503. Supported: replaces, timer
  504. Contact: <sip:1100000000@185.164.137.43:5160>
  505. Content-Length: 0
  506.  
  507.  
  508. <------------>
  509. Really destroying SIP dialog 'Dv7pD_BZD80dLJ1vFd7tRg..' Method: REGISTER
  510.        > 0x7fe68c074520 -- Probation passed - setting RTP source address to 178.75.237.247:27180
  511.  
  512. <--- SIP read from UDP:178.75.237.247:27040 --->
  513. SIP/2.0 200 OK
  514. Via: SIP/2.0/UDP 185.164.137.43:5160;branch=z9hG4bK1f0b225f;rport=5160
  515. Contact: <sip:1100000000@178.75.237.247:27040;transport=UDP>
  516. To: <sip:1100000000@178.75.237.247:27040;transport=UDP;rinstance=bfa9274abd4d9c27>;tag=26162c60
  517. From: "Nicole Foster" <sip:1100000001@185.164.137.43:5160>;tag=as1f74d5cb
  518. Call-ID: 498d76834af41d9c4c3fa98e681ce1ec@185.164.137.43:5160
  519. CSeq: 102 INVITE
  520. Content-Type: application/sdp
  521. User-Agent: Zoiper rv2.8.26-mod
  522. Allow-Events: presence, kpml, talk
  523. Content-Length: 243
  524.  
  525. v=0
  526. o=Zoiper 0 1 IN IP4 192.168.0.103
  527. s=Zoiper
  528. c=IN IP4 192.168.0.103
  529. t=0 0
  530. m=audio 39260 RTP/AVP 0 3 8 101
  531. a=rtpmap:0 PCMU/8000
  532. a=rtpmap:3 GSM/8000
  533. a=rtpmap:8 PCMA/8000
  534. a=rtpmap:101 telephone-event/8000
  535. a=fmtp:101 0-16
  536. a=sendrecv
  537. <------------->
  538. --- (11 headers 12 lines) ---
  539. Found RTP audio format 0
  540. Found RTP audio format 3
  541. Found RTP audio format 8
  542. Found RTP audio format 101
  543. Found audio description format PCMU for ID 0
  544. Found audio description format GSM for ID 3
  545. Found audio description format PCMA for ID 8
  546. Found audio description format telephone-event for ID 101
  547. Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
  548. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  549. Peer audio RTP is at port 192.168.0.103:39260
  550. sip_route_dump: route/path hop: <sip:1100000000@178.75.237.247:27040;transport=UDP>
  551. Transmitting (NAT) to 178.75.237.247:27040:
  552. ACK sip:1100000000@178.75.237.247:27040;transport=UDP SIP/2.0
  553. Via: SIP/2.0/UDP 185.164.137.43:5160;branch=z9hG4bK50634a01;rport
  554. Max-Forwards: 70
  555. From: "Nicole Foster" <sip:1100000001@185.164.137.43:5160>;tag=as1f74d5cb
  556. To: <sip:1100000000@178.75.237.247:27040;transport=UDP;rinstance=bfa9274abd4d9c27>;tag=26162c60
  557. Contact: <sip:1100000001@185.164.137.43:5160>
  558. Call-ID: 498d76834af41d9c4c3fa98e681ce1ec@185.164.137.43:5160
  559. CSeq: 102 ACK
  560. User-Agent: FPBX-14.0.1.1(13.16.0)
  561. Content-Length: 0
  562.  
  563.  
  564. ---
  565.     -- Connected line update to SIP/1100000001-00000020 prevented.
  566.     -- SIP/1100000000-00000021 answered SIP/1100000001-00000020
  567. Audio is at 10696
  568. Adding codec ulaw to SDP
  569. Adding codec alaw to SDP
  570. Adding codec gsm to SDP
  571. Adding non-codec 0x1 (telephone-event) to SDP
  572.  
  573. <--- Reliably Transmitting (NAT) to 178.75.237.247:27299 --->
  574. SIP/2.0 200 OK
  575. Via: SIP/2.0/UDP 178.75.237.247:27299;branch=z9hG4bK-524287-1---a492be17d3a66849;received=178.75.237.247;rport=27299
  576. From: <sip:1100000001@185.164.137.43:5160;transport=UDP>;tag=e6042e6c
  577. To: <sip:1100000000@185.164.137.43:5160;transport=UDP>;tag=as65adc904
  578. Call-ID: 01sSAl-uvzrXtD_d6kAn5A..
  579. CSeq: 2 INVITE
  580. Server: FPBX-14.0.1.1(13.16.0)
  581. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  582. Supported: replaces, timer
  583. Contact: <sip:1100000000@185.164.137.43:5160>
  584. P-Asserted-Identity: "Radoslav Petrov" <sip:1100000000@185.164.137.43>
  585. Content-Type: application/sdp
  586. Content-Length: 299
  587.  
  588. v=0
  589. o=root 89875237 89875237 IN IP4 185.164.137.43
  590. s=Asterisk PBX 13.16.0
  591. c=IN IP4 185.164.137.43
  592. t=0 0
  593. m=audio 10696 RTP/AVP 0 8 3 101
  594. a=rtpmap:0 PCMU/8000
  595. a=rtpmap:8 PCMA/8000
  596. a=rtpmap:3 GSM/8000
  597. a=rtpmap:101 telephone-event/8000
  598. a=fmtp:101 0-16
  599. a=ptime:20
  600. a=maxptime:150
  601. a=sendrecv
  602.  
  603. <------------>
  604.     -- Channel SIP/1100000000-00000021 joined 'simple_bridge' basic-bridge <6e51b321-0db6-4f69-84c6-d73e64a1596e>
  605.     -- Channel SIP/1100000001-00000020 joined 'simple_bridge' basic-bridge <6e51b321-0db6-4f69-84c6-d73e64a1596e>
  606. Retransmitting #1 (NAT) to 178.75.237.247:27299:
  607. SIP/2.0 200 OK
  608. Via: SIP/2.0/UDP 178.75.237.247:27299;branch=z9hG4bK-524287-1---a492be17d3a66849;received=178.75.237.247;rport=27299
  609. From: <sip:1100000001@185.164.137.43:5160;transport=UDP>;tag=e6042e6c
  610. To: <sip:1100000000@185.164.137.43:5160;transport=UDP>;tag=as65adc904
  611. Call-ID: 01sSAl-uvzrXtD_d6kAn5A..
  612. CSeq: 2 INVITE
  613. Server: FPBX-14.0.1.1(13.16.0)
  614. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  615. Supported: replaces, timer
  616. Contact: <sip:1100000000@185.164.137.43:5160>
  617. P-Asserted-Identity: "Radoslav Petrov" <sip:1100000000@185.164.137.43>
  618. Content-Type: application/sdp
  619. Content-Length: 299
  620.  
  621. v=0
  622. o=root 89875237 89875237 IN IP4 185.164.137.43
  623. s=Asterisk PBX 13.16.0
  624. c=IN IP4 185.164.137.43
  625. t=0 0
  626. m=audio 10696 RTP/AVP 0 8 3 101
  627. a=rtpmap:0 PCMU/8000
  628. a=rtpmap:8 PCMA/8000
  629. a=rtpmap:3 GSM/8000
  630. a=rtpmap:101 telephone-event/8000
  631. a=fmtp:101 0-16
  632. a=ptime:20
  633. a=maxptime:150
  634. a=sendrecv
  635.  
  636. ---
  637.        > 0x7fe6b0009170 -- Probation passed - setting RTP source address to 178.75.237.247:27396
  638.  
  639. <--- SIP read from UDP:178.75.237.247:27299 --->
  640. ACK sip:1100000000@185.164.137.43:5160 SIP/2.0
  641. Via: SIP/2.0/UDP 178.75.237.247:27299;branch=z9hG4bK-524287-1---efce9665bc353d55;rport
  642. Max-Forwards: 70
  643. Contact: <sip:1100000001@178.75.237.247:27299;transport=UDP>
  644. To: <sip:1100000000@185.164.137.43:5160;transport=UDP>;tag=as65adc904
  645. From: <sip:1100000001@185.164.137.43:5160;transport=UDP>;tag=e6042e6c
  646. Call-ID: 01sSAl-uvzrXtD_d6kAn5A..
  647. CSeq: 2 ACK
  648. User-Agent: Zoiper rv2.8.26-mod
  649. Content-Length: 0
  650.  
  651. <------------->
  652. --- (10 headers 0 lines) ---
  653.  
  654. <--- SIP read from UDP:178.75.237.247:27299 --->
  655. ACK sip:1100000000@185.164.137.43:5160 SIP/2.0
  656. Via: SIP/2.0/UDP 178.75.237.247:27299;branch=z9hG4bK-524287-1---efce9665bc353d55;rport
  657. Max-Forwards: 70
  658. Contact: <sip:1100000001@178.75.237.247:27299;transport=UDP>
  659. To: <sip:1100000000@185.164.137.43:5160;transport=UDP>;tag=as65adc904
  660. From: <sip:1100000001@185.164.137.43:5160;transport=UDP>;tag=e6042e6c
  661. Call-ID: 01sSAl-uvzrXtD_d6kAn5A..
  662. CSeq: 2 ACK
  663. User-Agent: Zoiper rv2.8.26-mod
  664. Content-Length: 0
  665.  
  666. <------------->
  667. --- (10 headers 0 lines) ---
  668.        > 0x7fe68c074520 -- Probation passed - setting RTP source address to 178.75.237.247:27180
  669.        > 0x7fe68c074520 -- Probation passed - setting RTP source address to 178.75.237.247:27180
  670. freepbx*CLI>
  671.  

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