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  1. [root@FREEPBX ~]# asterisk -vvvr
  2. Asterisk 13.10.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  3. Created by Mark Spencer <markster@digium.com>
  4. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  5. This is free software, with components licensed under the GNU General Public
  6. License version 2 and other licenses; you are welcome to redistribute it under
  7. certain conditions. Type 'core show license' for details.
  8. =========================================================================
  9. Connected to Asterisk 13.10.0 currently running on FREEPBX (pid = 1672)
  10. == Using SIP RTP CoS mark 5
  11. -- Executing [761@from-internal:1] GotoIf("SIP/704-00000110", "1?ext-local,761,1:followme-check,761,1") in new stack
  12. -- Goto (ext-local,761,1)
  13. -- Executing [761@ext-local:1] Set("SIP/704-00000110", "__RINGTIMER=15") in new stack
  14. -- Executing [761@ext-local:2] Macro("SIP/704-00000110", "exten-vm,novm,761,0,0,0") in new stack
  15. -- Executing [s@macro-exten-vm:1] Macro("SIP/704-00000110", "user-callerid,") in new stack
  16. -- Executing [s@macro-user-callerid:1] Set("SIP/704-00000110", "TOUCH_MONITOR=1471022944.338") in new stack
  17. -- Executing [s@macro-user-callerid:2] Set("SIP/704-00000110", "AMPUSER=704") in new stack
  18. -- Executing [s@macro-user-callerid:3] GotoIf("SIP/704-00000110", "0?report") in new stack
  19. -- Executing [s@macro-user-callerid:4] ExecIf("SIP/704-00000110", "1?Set(REALCALLERIDNUM=704)") in new stack
  20. -- Executing [s@macro-user-callerid:5] Set("SIP/704-00000110", "AMPUSER=704") in new stack
  21. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/704-00000110", "0?limit") in new stack
  22. -- Executing [s@macro-user-callerid:7] Set("SIP/704-00000110", "AMPUSERCIDNAME=Mike") in new stack
  23. -- Executing [s@macro-user-callerid:8] GotoIf("SIP/704-00000110", "0?report") in new stack
  24. -- Executing [s@macro-user-callerid:9] Set("SIP/704-00000110", "AMPUSERCID=704") in new stack
  25. -- Executing [s@macro-user-callerid:10] Set("SIP/704-00000110", "__DIAL_OPTIONS=Ttr") in new stack
  26. -- Executing [s@macro-user-callerid:11] Set("SIP/704-00000110", "CALLERID(all)="Mike" <704>") in new stack
  27. -- Executing [s@macro-user-callerid:12] GotoIf("SIP/704-00000110", "0?limit") in new stack
  28. -- Executing [s@macro-user-callerid:13] ExecIf("SIP/704-00000110", "0?Set(GROUP(concurrency_limit)=704)") in new stack
  29. -- Executing [s@macro-user-callerid:14] ExecIf("SIP/704-00000110", "0?Set(CHANNEL(language)=)") in new stack
  30. -- Executing [s@macro-user-callerid:15] GotoIf("SIP/704-00000110", "0?continue") in new stack
  31. -- Executing [s@macro-user-callerid:16] ExecIf("SIP/704-00000110", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
  32. -- Executing [s@macro-user-callerid:17] Set("SIP/704-00000110", "__TTL=64") in new stack
  33. -- Executing [s@macro-user-callerid:18] GotoIf("SIP/704-00000110", "1?continue") in new stack
  34. -- Goto (macro-user-callerid,s,29)
  35. -- Executing [s@macro-user-callerid:29] Set("SIP/704-00000110", "CALLERID(number)=704") in new stack
  36. -- Executing [s@macro-user-callerid:30] Set("SIP/704-00000110", "CALLERID(name)=Mike") in new stack
  37. -- Executing [s@macro-user-callerid:31] Set("SIP/704-00000110", "CDR(cnum)=704") in new stack
  38. -- Executing [s@macro-user-callerid:32] Set("SIP/704-00000110", "CDR(cnam)=Mike") in new stack
  39. -- Executing [s@macro-user-callerid:33] Set("SIP/704-00000110", "CHANNEL(language)=en") in new stack
  40. -- Executing [s@macro-exten-vm:2] Set("SIP/704-00000110", "RingGroupMethod=none") in new stack
  41. -- Executing [s@macro-exten-vm:3] Set("SIP/704-00000110", "__EXTTOCALL=761") in new stack
  42. -- Executing [s@macro-exten-vm:4] Set("SIP/704-00000110", "__PICKUPMARK=761") in new stack
  43. -- Executing [s@macro-exten-vm:5] Set("SIP/704-00000110", "RT=") in new stack
  44. -- Executing [s@macro-exten-vm:6] ExecIf("SIP/704-00000110", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
  45. -- Executing [s@macro-exten-vm:7] ExecIf("SIP/704-00000110", "0?MacroExit()") in new stack
  46. -- Executing [s@macro-exten-vm:8] Gosub("SIP/704-00000110", "sub-record-check,s,1(exten,761,dontcare)") in new stack
  47. -- Executing [s@sub-record-check:1] GotoIf("SIP/704-00000110", "0?initialized") in new stack
  48. -- Executing [s@sub-record-check:2] Set("SIP/704-00000110", "__REC_STATUS=INITIALIZED") in new stack
  49. -- Executing [s@sub-record-check:3] Set("SIP/704-00000110", "NOW=1471022944") in new stack
  50. -- Executing [s@sub-record-check:4] Set("SIP/704-00000110", "__DAY=12") in new stack
  51. -- Executing [s@sub-record-check:5] Set("SIP/704-00000110", "__MONTH=08") in new stack
  52. -- Executing [s@sub-record-check:6] Set("SIP/704-00000110", "__YEAR=2016") in new stack
  53. -- Executing [s@sub-record-check:7] Set("SIP/704-00000110", "__TIMESTR=20160812-132904") in new stack
  54. -- Executing [s@sub-record-check:8] Set("SIP/704-00000110", "__FROMEXTEN=704") in new stack
  55. -- Executing [s@sub-record-check:9] Set("SIP/704-00000110", "__MON_FMT=wav") in new stack
  56. -- Executing [s@sub-record-check:10] NoOp("SIP/704-00000110", "Recordings initialized") in new stack
  57. -- Executing [s@sub-record-check:11] ExecIf("SIP/704-00000110", "0?Set(ARG3=dontcare)") in new stack
  58. -- Executing [s@sub-record-check:12] Set("SIP/704-00000110", "REC_POLICY_MODE_SAVE=") in new stack
  59. -- Executing [s@sub-record-check:13] ExecIf("SIP/704-00000110", "0?Set(REC_STATUS=NO)") in new stack
  60. -- Executing [s@sub-record-check:14] GotoIf("SIP/704-00000110", "5?checkaction") in new stack
  61. -- Goto (sub-record-check,s,17)
  62. -- Executing [s@sub-record-check:17] GotoIf("SIP/704-00000110", "1?sub-record-check,exten,1") in new stack
  63. -- Goto (sub-record-check,exten,1)
  64. -- Executing [exten@sub-record-check:1] NoOp("SIP/704-00000110", "Exten Recording Check between 704 and 761") in new stack
  65. -- Executing [exten@sub-record-check:2] Set("SIP/704-00000110", "CALLTYPE=internal") in new stack
  66. -- Executing [exten@sub-record-check:3] ExecIf("SIP/704-00000110", "0?Set(CALLTYPE=)") in new stack
  67. -- Executing [exten@sub-record-check:4] Set("SIP/704-00000110", "CALLEE=dontcare") in new stack
  68. -- Executing [exten@sub-record-check:5] ExecIf("SIP/704-00000110", "0?Set(CALLEE=dontcare)") in new stack
  69. -- Executing [exten@sub-record-check:6] GotoIf("SIP/704-00000110", "0?callee") in new stack
  70. -- Executing [exten@sub-record-check:7] GotoIf("SIP/704-00000110", "1?caller") in new stack
  71. -- Goto (sub-record-check,exten,13)
  72. -- Executing [exten@sub-record-check:13] Set("SIP/704-00000110", "RECMODE=dontcare") in new stack
  73. -- Executing [exten@sub-record-check:14] ExecIf("SIP/704-00000110", "0?Set(RECMODE=dontcare)") in new stack
  74. -- Executing [exten@sub-record-check:15] ExecIf("SIP/704-00000110", "1?Set(RECMODE=dontcare)") in new stack
  75. -- Executing [exten@sub-record-check:16] Gosub("SIP/704-00000110", "recordcheck,1(dontcare,internal,761)") in new stack
  76. -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/704-00000110", "Starting recording check against dontcare") in new stack
  77. -- Executing [recordcheck@sub-record-check:2] Goto("SIP/704-00000110", "dontcare") in new stack
  78. -- Goto (sub-record-check,recordcheck,3)
  79. -- Executing [recordcheck@sub-record-check:3] Return("SIP/704-00000110", "") in new stack
  80. -- Executing [exten@sub-record-check:17] Return("SIP/704-00000110", "") in new stack
  81. -- Executing [s@macro-exten-vm:9] GotoIf("SIP/704-00000110", "1?macrodial") in new stack
  82. -- Goto (macro-exten-vm,s,15)
  83. -- Executing [s@macro-exten-vm:15] GosubIf("SIP/704-00000110", "0?clrheader,1()") in new stack
  84. -- Executing [s@macro-exten-vm:16] Macro("SIP/704-00000110", "dial-one,,Ttr,761") in new stack
  85. -- Executing [s@macro-dial-one:1] Set("SIP/704-00000110", "DEXTEN=761") in new stack
  86. -- Executing [s@macro-dial-one:2] Set("SIP/704-00000110", "DIALSTATUS_CW=") in new stack
  87. -- Executing [s@macro-dial-one:3] GosubIf("SIP/704-00000110", "0?screen,1()") in new stack
  88. -- Executing [s@macro-dial-one:4] GosubIf("SIP/704-00000110", "0?cf,1()") in new stack
  89. -- Executing [s@macro-dial-one:5] GotoIf("SIP/704-00000110", "1?skip1") in new stack
  90. -- Goto (macro-dial-one,s,8)
  91. -- Executing [s@macro-dial-one:8] GotoIf("SIP/704-00000110", "0?nodial") in new stack
  92. -- Executing [s@macro-dial-one:9] GotoIf("SIP/704-00000110", "0?continue") in new stack
  93. -- Executing [s@macro-dial-one:10] Set("SIP/704-00000110", "EXTHASCW=ENABLED") in new stack
  94. -- Executing [s@macro-dial-one:11] GotoIf("SIP/704-00000110", "0?next1:cwinusebusy") in new stack
  95. -- Goto (macro-dial-one,s,23)
  96. -- Executing [s@macro-dial-one:23] GotoIf("SIP/704-00000110", "0?next3:continue") in new stack
  97. -- Goto (macro-dial-one,s,25)
  98. -- Executing [s@macro-dial-one:25] GotoIf("SIP/704-00000110", "0?nodial") in new stack
  99. -- Executing [s@macro-dial-one:26] GosubIf("SIP/704-00000110", "1?dstring,1():dlocal,1()") in new stack
  100. -- Executing [dstring@macro-dial-one:1] Set("SIP/704-00000110", "DSTRING=") in new stack
  101. -- Executing [dstring@macro-dial-one:2] Set("SIP/704-00000110", "DEVICES=761") in new stack
  102. -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/704-00000110", "0?Return()") in new stack
  103. -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/704-00000110", "0?Set(DEVICES=61)") in new stack
  104. -- Executing [dstring@macro-dial-one:5] Set("SIP/704-00000110", "LOOPCNT=1") in new stack
  105. -- Executing [dstring@macro-dial-one:6] Set("SIP/704-00000110", "ITER=1") in new stack
  106. -- Executing [dstring@macro-dial-one:7] Set("SIP/704-00000110", "THISDIAL=SIP/12132261066@audio1.join.me") in new stack
  107. -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/704-00000110", "1?zap2dahdi,1()") in new stack
  108. -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/704-00000110", "0?Return()") in new stack
  109. -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/704-00000110", "NEWDIAL=") in new stack
  110. -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/704-00000110", "LOOPCNT2=1") in new stack
  111. -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/704-00000110", "ITER2=1") in new stack
  112. -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/704-00000110", "THISPART2=SIP/12132261066@audio1.join.me") in new stack
  113. -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/704-00000110", "0?Set(THISPART2=DAHDI/12132261066@audio1.join.me)") in new stack
  114. -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/704-00000110", "NEWDIAL=SIP/12132261066@audio1.join.me&") in new stack
  115. -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/704-00000110", "ITER2=2") in new stack
  116. -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/704-00000110", "0?begin2") in new stack
  117. -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/704-00000110", "THISDIAL=SIP/12132261066@audio1.join.me") in new stack
  118. -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/704-00000110", "") in new stack
  119. -- Executing [dstring@macro-dial-one:9] GotoIf("SIP/704-00000110", "1?docheck") in new stack
  120. -- Goto (macro-dial-one,dstring,12)
  121. -- Executing [dstring@macro-dial-one:12] GotoIf("SIP/704-00000110", "0?skipset") in new stack
  122. -- Executing [dstring@macro-dial-one:13] Set("SIP/704-00000110", "DSTRING=SIP/12132261066@audio1.join.me&") in new stack
  123. -- Executing [dstring@macro-dial-one:14] Set("SIP/704-00000110", "ITER=2") in new stack
  124. -- Executing [dstring@macro-dial-one:15] GotoIf("SIP/704-00000110", "0?begin") in new stack
  125. -- Executing [dstring@macro-dial-one:16] ExecIf("SIP/704-00000110", "0?Return()") in new stack
  126. -- Executing [dstring@macro-dial-one:17] Set("SIP/704-00000110", "DSTRING=SIP/12132261066@audio1.join.me") in new stack
  127. -- Executing [dstring@macro-dial-one:18] Return("SIP/704-00000110", "") in new stack
  128. -- Executing [s@macro-dial-one:27] GotoIf("SIP/704-00000110", "0?nodial") in new stack
  129. -- Executing [s@macro-dial-one:28] GotoIf("SIP/704-00000110", "0?skiptrace") in new stack
  130. -- Executing [s@macro-dial-one:29] GosubIf("SIP/704-00000110", "1?ctset,1():ctclear,1()") in new stack
  131. -- Executing [ctset@macro-dial-one:1] Set("SIP/704-00000110", "DB(CALLTRACE/761)=704") in new stack
  132. -- Executing [ctset@macro-dial-one:2] Return("SIP/704-00000110", "") in new stack
  133. -- Executing [s@macro-dial-one:30] Set("SIP/704-00000110", "D_OPTIONS=Ttr") in new stack
  134. -- Executing [s@macro-dial-one:31] NoOp("SIP/704-00000110", "Blind Transfer: , Attended Transfer: , User: 704, Alert Info: ") in new stack
  135. -- Executing [s@macro-dial-one:32] ExecIf("SIP/704-00000110", "1?Set(ALERT_INFO=)") in new stack
  136. -- Executing [s@macro-dial-one:33] ExecIf("SIP/704-00000110", "0?Set(ALERT_INFO=)") in new stack
  137. -- Executing [s@macro-dial-one:34] ExecIf("SIP/704-00000110", "0?Set(ALERT_INFO=)") in new stack
  138. -- Executing [s@macro-dial-one:35] GosubIf("SIP/704-00000110", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
  139. -- Executing [s@macro-dial-one:36] ExecIf("SIP/704-00000110", "0?Set(CHANNEL(musicclass)=)") in new stack
  140. -- Executing [s@macro-dial-one:37] GosubIf("SIP/704-00000110", "0?qwait,1()") in new stack
  141. -- Executing [s@macro-dial-one:38] Set("SIP/704-00000110", "__CWIGNORE=") in new stack
  142. -- Executing [s@macro-dial-one:39] Set("SIP/704-00000110", "__KEEPCID=TRUE") in new stack
  143. -- Executing [s@macro-dial-one:40] GotoIf("SIP/704-00000110", "0?usegoto,1") in new stack
  144. -- Executing [s@macro-dial-one:41] GotoIf("SIP/704-00000110", "0?godial") in new stack
  145. -- Executing [s@macro-dial-one:42] Gosub("SIP/704-00000110", "sub-presencestate-display,s,1(761)") in new stack
  146. -- Executing [s@sub-presencestate-display:1] Goto("SIP/704-00000110", "state-not_set,1") in new stack
  147. -- Goto (sub-presencestate-display,state-not_set,1)
  148. -- Executing [state-not_set@sub-presencestate-display:1] Set("SIP/704-00000110", "PRESENCESTATE_DISPLAY=") in new stack
  149. -- Executing [state-not_set@sub-presencestate-display:2] Return("SIP/704-00000110", "") in new stack
  150. -- Executing [s@macro-dial-one:43] Set("SIP/704-00000110", "CONNECTEDLINE(name,i)=761") in new stack
  151. -- Executing [s@macro-dial-one:44] Set("SIP/704-00000110", "CONNECTEDLINE(num)=761") in new stack
  152. -- Executing [s@macro-dial-one:45] Set("SIP/704-00000110", "D_OPTIONS=TtrI") in new stack
  153. -- Executing [s@macro-dial-one:46] Macro("SIP/704-00000110", "dialout-one-predial-hook,") in new stack
  154. -- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("SIP/704-00000110", "") in new stack
  155. -- Executing [s@macro-dial-one:47] ExecIf("SIP/704-00000110", "0?Set(D_OPTIONS=trII)") in new stack
  156. -- Executing [s@macro-dial-one:48] Dial("SIP/704-00000110", "SIP/12132261066@audio1.join.me,,TtrIb(func-apply-sipheaders^s^1)") in new stack
  157. == Using SIP RTP CoS mark 5
  158. -- SIP/audio1.join.me-00000111 Internal Gosub(func-apply-sipheaders,s,1) start
  159. -- Executing [s@func-apply-sipheaders:1] NoOp("SIP/audio1.join.me-00000111", "Applying SIP Headers to channel") in new stack
  160. -- Executing [s@func-apply-sipheaders:2] Set("SIP/audio1.join.me-00000111", "SIPHEADERKEYS=") in new stack
  161. -- Executing [s@func-apply-sipheaders:3] While("SIP/audio1.join.me-00000111", "0") in new stack
  162. -- Jumping to priority 7
  163. -- Executing [s@func-apply-sipheaders:8] Return("SIP/audio1.join.me-00000111", "") in new stack
  164. == Spawn extension (default, 761, 1) exited non-zero on 'SIP/audio1.join.me-00000111'
  165. -- SIP/audio1.join.me-00000111 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
  166. -- Called SIP/12132261066@audio1.join.me
  167. [2016-08-12 13:29:36] WARNING[1730]: chan_sip.c:4059 retrans_pkt: Retransmission timeout reached on transmission 66386da03a7ee5761ca2d75f61230ec9@184.58.6isplay/AST/SIP+Retransmissions
  168. Packet timed out after 31999ms with no response
  169. [2016-08-12 13:29:36] WARNING[1730]: chan_sip.c:4083 retrans_pkt: Hanging up call 66386da03a7ee5761ca2d75f61230ec9@184.58.69.128:5160 - no reply to our cr
  170. == Everyone is busy/congested at this time (1:0/0/1)
  171. -- Executing [s@macro-dial-one:49] ExecIf("SIP/704-00000110", "0?MacroExit()") in new stack
  172. -- Executing [s@macro-dial-one:50] ExecIf("SIP/704-00000110", "0?Set(DIALSTATUS=)") in new stack
  173. -- Executing [s@macro-dial-one:51] GosubIf("SIP/704-00000110", "0?s-CHANUNAVAIL,1()") in new stack
  174. -- Executing [s@macro-dial-one:52] MacroExit("SIP/704-00000110", "") in new stack
  175. -- Executing [s@macro-exten-vm:17] Set("SIP/704-00000110", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
  176. -- Executing [s@macro-exten-vm:18] GosubIf("SIP/704-00000110", "0?docfu,1()") in new stack
  177. -- Executing [s@macro-exten-vm:19] GosubIf("SIP/704-00000110", "0?docfb,1()") in new stack
  178. -- Executing [s@macro-exten-vm:20] Set("SIP/704-00000110", "DIALSTATUS=CHANUNAVAIL") in new stack
  179. -- Executing [s@macro-exten-vm:21] ExecIf("SIP/704-00000110", "0?MacroExit()") in new stack
  180. -- Executing [s@macro-exten-vm:22] GotoIf("SIP/704-00000110", "1?s-CHANUNAVAIL,1") in new stack
  181. -- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
  182. -- Executing [s-CHANUNAVAIL@macro-exten-vm:1] GotoIf("SIP/704-00000110", "0?exit,1") in new stack
  183. -- Executing [s-CHANUNAVAIL@macro-exten-vm:2] PlayTones("SIP/704-00000110", "congestion") in new stack
  184. -- Executing [s-CHANUNAVAIL@macro-exten-vm:3] Congestion("SIP/704-00000110", "10") in new stack
  185. == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/704-00000110' in macro 'exten-vm'
  186. == Spawn extension (ext-local, 761, 2) exited non-zero on 'SIP/704-00000110'
  187. -- Executing [h@ext-local:1] Macro("SIP/704-00000110", "hangupcall,") in new stack
  188. -- Executing [s@macro-hangupcall:1] GotoIf("SIP/704-00000110", "1?theend") in new stack
  189. -- Goto (macro-hangupcall,s,3)
  190. -- Executing [s@macro-hangupcall:3] ExecIf("SIP/704-00000110", "0?Set(CDR(recordingfile)=)") in new stack
  191. -- Executing [s@macro-hangupcall:4] Hangup("SIP/704-00000110", "") in new stack
  192. == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/704-00000110' in macro 'hangupcall'
  193. == Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/704-00000110'
  194. FREEPBX*CLI> clear
  195. No such command 'clear' (type 'core show help clear' for other possible commands)
  196. FREEPBX*CLI> quit
  197. Asterisk cleanly ending (0).
  198. Executing last minute cleanups
  199. [root@FREEPBX ~]# clear
  200. [root@FREEPBX ~]# asterisk -vvvr
  201. Asterisk 13.10.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  202. Created by Mark Spencer <markster@digium.com>
  203. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  204. This is free software, with components licensed under the GNU General Public
  205. License version 2 and other licenses; you are welcome to redistribute it under
  206. certain conditions. Type 'core show license' for details.
  207. =========================================================================
  208. Connected to Asterisk 13.10.0 currently running on FREEPBX (pid = 1672)
  209. FREEPBX*CLI> sip set debug on
  210. SIP Debugging enabled
  211.  
  212. <--- SIP read from UDP:10.1.1.226:5060 --->
  213. INVITE sip:761@10.1.1.235:5061;user=phone SIP/2.0
  214. Via: SIP/2.0/UDP 10.1.1.226;rport;branch=z9hG4bK5e61868357B3123A
  215. From: "Mike" <sip:704@10.1.1.235:5061>;tag=4AA671E9-CD5264B8
  216. To: <sip:761@10.1.1.235;user=phone>
  217. CSeq: 1 INVITE
  218. Call-ID: 88faf30050df056679c3a357d003110b
  219. Contact: <sip:704@10.1.1.226>
  220. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  221. User-Agent: Polycom/5.5.0.20556 PolycomVVX-VVX_601-UA/5.5.0.20556
  222. Accept-Language: en
  223. Supported: replaces,100rel
  224. Allow-Events: conference,talk,hold
  225. Max-Forwards: 70
  226. Content-Type: application/sdp
  227. Content-Length: 558
  228.  
  229. v=0
  230. o=- 1471023250 1471023250 IN IP4 10.1.1.226
  231. s=Polycom IP Phone
  232. c=IN IP4 10.1.1.226
  233. b=AS:512
  234. t=0 0
  235. a=sendrecv
  236. m=audio 2310 RTP/AVP 115 9 102 0 8 18 127
  237. a=rtpmap:115 G7221/32000
  238. a=fmtp:115 bitrate=48000
  239. a=rtpmap:9 G722/8000
  240. a=rtpmap:102 G7221/16000
  241. a=fmtp:102 bitrate=32000
  242. a=rtpmap:0 PCMU/8000
  243. a=rtpmap:8 PCMA/8000
  244. a=rtpmap:18 G729/8000
  245. a=fmtp:18 annexb=no
  246. a=rtpmap:127 telephone-event/8000
  247. m=video 2312 RTP/AVP 109 34
  248. a=rtpmap:109 H264/90000
  249. a=fmtp:109 profile-level-id=42800d
  250. a=rtpmap:34 H263/90000
  251. a=fmtp:34 CIF=1;QCIF=1;SQCIF=1
  252. <------------->
  253. --- (15 headers 23 lines) ---
  254. Sending to 10.1.1.226:5060 (NAT)
  255. Sending to 10.1.1.226:5060 (NAT)
  256. Using INVITE request as basis request - 88faf30050df056679c3a357d003110b
  257. Found peer '704' for '704' from 10.1.1.226:5060
  258.  
  259. <--- Reliably Transmitting (no NAT) to 10.1.1.226:5060 --->
  260. SIP/2.0 401 Unauthorized
  261. Via: SIP/2.0/UDP 10.1.1.226;branch=z9hG4bK5e61868357B3123A;received=10.1.1.226;rport=5060
  262. From: "Mike" <sip:704@10.1.1.235:5061>;tag=4AA671E9-CD5264B8
  263. To: <sip:761@10.1.1.235;user=phone>;tag=as4562bb6f
  264. Call-ID: 88faf30050df056679c3a357d003110b
  265. CSeq: 1 INVITE
  266. Server: Asterisk PBX 13.10.0
  267. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  268. Supported: replaces, timer
  269. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="707cca83"
  270. Content-Length: 0
  271.  
  272.  
  273. <------------>
  274. Scheduling destruction of SIP dialog '88faf30050df056679c3a357d003110b' in 6400 ms (Method: INVITE)
  275.  
  276. <--- SIP read from UDP:10.1.1.226:5060 --->
  277. ACK sip:761@10.1.1.235:5061;user=phone SIP/2.0
  278. Via: SIP/2.0/UDP 10.1.1.226;rport;branch=z9hG4bK5e61868357B3123A
  279. From: "Mike" <sip:704@10.1.1.235:5061>;tag=4AA671E9-CD5264B8
  280. To: <sip:761@10.1.1.235;user=phone>;tag=as4562bb6f
  281. CSeq: 1 ACK
  282. Call-ID: 88faf30050df056679c3a357d003110b
  283. Contact: <sip:704@10.1.1.226>
  284. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  285. User-Agent: Polycom/5.5.0.20556 PolycomVVX-VVX_601-UA/5.5.0.20556
  286. Accept-Language: en
  287. Max-Forwards: 70
  288. Content-Length: 0
  289.  
  290. <------------->
  291. --- (12 headers 0 lines) ---
  292.  
  293. <--- SIP read from UDP:10.1.1.226:5060 --->
  294. INVITE sip:761@10.1.1.235:5061;user=phone SIP/2.0
  295. Via: SIP/2.0/UDP 10.1.1.226;rport;branch=z9hG4bK8e3cea8dD5FC28EC
  296. From: "Mike" <sip:704@10.1.1.235:5061>;tag=4AA671E9-CD5264B8
  297. To: <sip:761@10.1.1.235;user=phone>
  298. CSeq: 2 INVITE
  299. Call-ID: 88faf30050df056679c3a357d003110b
  300. Contact: <sip:704@10.1.1.226>
  301. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  302. User-Agent: Polycom/5.5.0.20556 PolycomVVX-VVX_601-UA/5.5.0.20556
  303. Accept-Language: en
  304. Supported: replaces,100rel
  305. Allow-Events: conference,talk,hold
  306. Authorization: Digest username="704", realm="asterisk", nonce="707cca83", uri="sip:761@10.1.1.235:5061;user=phone", response="828318e04746645f754ef8e3bf55e95d", algorithm=MD5
  307. Max-Forwards: 70
  308. Content-Type: application/sdp
  309. Content-Length: 558
  310.  
  311. v=0
  312. o=- 1471023250 1471023250 IN IP4 10.1.1.226
  313. s=Polycom IP Phone
  314. c=IN IP4 10.1.1.226
  315. b=AS:512
  316. t=0 0
  317. a=sendrecv
  318. m=audio 2310 RTP/AVP 115 9 102 0 8 18 127
  319. a=rtpmap:115 G7221/32000
  320. a=fmtp:115 bitrate=48000
  321. a=rtpmap:9 G722/8000
  322. a=rtpmap:102 G7221/16000
  323. a=fmtp:102 bitrate=32000
  324. a=rtpmap:0 PCMU/8000
  325. a=rtpmap:8 PCMA/8000
  326. a=rtpmap:18 G729/8000
  327. a=fmtp:18 annexb=no
  328. a=rtpmap:127 telephone-event/8000
  329. m=video 2312 RTP/AVP 109 34
  330. a=rtpmap:109 H264/90000
  331. a=fmtp:109 profile-level-id=42800d
  332. a=rtpmap:34 H263/90000
  333. a=fmtp:34 CIF=1;QCIF=1;SQCIF=1
  334. <------------->
  335. --- (16 headers 23 lines) ---
  336. Sending to 10.1.1.226:5060 (no NAT)
  337. Using INVITE request as basis request - 88faf30050df056679c3a357d003110b
  338. Found peer '704' for '704' from 10.1.1.226:5060
  339. == Using SIP RTP CoS mark 5
  340. Found RTP audio format 115
  341. Found RTP audio format 9
  342. Found RTP audio format 102
  343. Found RTP audio format 0
  344. Found RTP audio format 8
  345. Found RTP audio format 18
  346. Found RTP audio format 127
  347. Found audio description format G7221 for ID 115
  348. Found audio description format G722 for ID 9
  349. Found audio description format G7221 for ID 102
  350. Found audio description format PCMU for ID 0
  351. Found audio description format PCMA for ID 8
  352. Found audio description format G729 for ID 18
  353. Found audio description format telephone-event for ID 127
  354. Found RTP video format 109
  355. Found RTP video format 34
  356. Found video description format H264 for ID 109
  357. Found video description format H263 for ID 34
  358. Capabilities: us - (g722|opus|ulaw|alaw|gsm|g726), peer - audio=(ulaw|alaw|g722|g729|siren7|siren14)/video=(h263|h264)/text=(nothing), combined - (g722|ulaw|alaw)
  359. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  360. Peer audio RTP is at port 10.1.1.226:2310
  361. Looking for 761 in from-internal (domain 10.1.1.235)
  362. sip_route_dump: route/path hop: <sip:704@10.1.1.226>
  363.  
  364. <--- Transmitting (no NAT) to 10.1.1.226:5060 --->
  365. SIP/2.0 100 Trying
  366. Via: SIP/2.0/UDP 10.1.1.226;branch=z9hG4bK8e3cea8dD5FC28EC;received=10.1.1.226;rport=5060
  367. From: "Mike" <sip:704@10.1.1.235:5061>;tag=4AA671E9-CD5264B8
  368. To: <sip:761@10.1.1.235;user=phone>
  369. Call-ID: 88faf30050df056679c3a357d003110b
  370. CSeq: 2 INVITE
  371. Server: Asterisk PBX 13.10.0
  372. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  373. Supported: replaces, timer
  374. Contact: <sip:761@10.1.1.80:5160>
  375. Content-Length: 0
  376.  
  377.  
  378. <------------>
  379. -- Executing [761@from-internal:1] GotoIf("SIP/704-00000112", "1?ext-local,761,1:followme-check,761,1") in new stack
  380. -- Goto (ext-local,761,1)
  381. -- Executing [761@ext-local:1] Set("SIP/704-00000112", "__RINGTIMER=15") in new stack
  382. -- Executing [761@ext-local:2] Macro("SIP/704-00000112", "exten-vm,novm,761,0,0,0") in new stack
  383. -- Executing [s@macro-exten-vm:1] Macro("SIP/704-00000112", "user-callerid,") in new stack
  384. -- Executing [s@macro-user-callerid:1] Set("SIP/704-00000112", "TOUCH_MONITOR=1471023251.340") in new stack
  385. -- Executing [s@macro-user-callerid:2] Set("SIP/704-00000112", "AMPUSER=704") in new stack
  386. -- Executing [s@macro-user-callerid:3] GotoIf("SIP/704-00000112", "0?report") in new stack
  387. -- Executing [s@macro-user-callerid:4] ExecIf("SIP/704-00000112", "1?Set(REALCALLERIDNUM=704)") in new stack
  388. -- Executing [s@macro-user-callerid:5] Set("SIP/704-00000112", "AMPUSER=704") in new stack
  389. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/704-00000112", "0?limit") in new stack
  390. -- Executing [s@macro-user-callerid:7] Set("SIP/704-00000112", "AMPUSERCIDNAME=Mike") in new stack
  391. -- Executing [s@macro-user-callerid:8] GotoIf("SIP/704-00000112", "0?report") in new stack
  392. -- Executing [s@macro-user-callerid:9] Set("SIP/704-00000112", "AMPUSERCID=704") in new stack
  393. -- Executing [s@macro-user-callerid:10] Set("SIP/704-00000112", "__DIAL_OPTIONS=Ttr") in new stack
  394. -- Executing [s@macro-user-callerid:11] Set("SIP/704-00000112", "CALLERID(all)="Mike" <704>") in new stack
  395. -- Executing [s@macro-user-callerid:12] GotoIf("SIP/704-00000112", "0?limit") in new stack
  396. -- Executing [s@macro-user-callerid:13] ExecIf("SIP/704-00000112", "0?Set(GROUP(concurrency_limit)=704)") in new stack
  397. -- Executing [s@macro-user-callerid:14] ExecIf("SIP/704-00000112", "0?Set(CHANNEL(language)=)") in new stack
  398. -- Executing [s@macro-user-callerid:15] GotoIf("SIP/704-00000112", "0?continue") in new stack
  399. -- Executing [s@macro-user-callerid:16] ExecIf("SIP/704-00000112", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
  400. -- Executing [s@macro-user-callerid:17] Set("SIP/704-00000112", "__TTL=64") in new stack
  401. -- Executing [s@macro-user-callerid:18] GotoIf("SIP/704-00000112", "1?continue") in new stack
  402. -- Goto (macro-user-callerid,s,29)
  403. -- Executing [s@macro-user-callerid:29] Set("SIP/704-00000112", "CALLERID(number)=704") in new stack
  404. -- Executing [s@macro-user-callerid:30] Set("SIP/704-00000112", "CALLERID(name)=Mike") in new stack
  405. -- Executing [s@macro-user-callerid:31] Set("SIP/704-00000112", "CDR(cnum)=704") in new stack
  406. -- Executing [s@macro-user-callerid:32] Set("SIP/704-00000112", "CDR(cnam)=Mike") in new stack
  407. -- Executing [s@macro-user-callerid:33] Set("SIP/704-00000112", "CHANNEL(language)=en") in new stack
  408. -- Executing [s@macro-exten-vm:2] Set("SIP/704-00000112", "RingGroupMethod=none") in new stack
  409. -- Executing [s@macro-exten-vm:3] Set("SIP/704-00000112", "__EXTTOCALL=761") in new stack
  410. -- Executing [s@macro-exten-vm:4] Set("SIP/704-00000112", "__PICKUPMARK=761") in new stack
  411. -- Executing [s@macro-exten-vm:5] Set("SIP/704-00000112", "RT=") in new stack
  412. -- Executing [s@macro-exten-vm:6] ExecIf("SIP/704-00000112", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
  413. -- Executing [s@macro-exten-vm:7] ExecIf("SIP/704-00000112", "0?MacroExit()") in new stack
  414. -- Executing [s@macro-exten-vm:8] Gosub("SIP/704-00000112", "sub-record-check,s,1(exten,761,dontcare)") in new stack
  415. -- Executing [s@sub-record-check:1] GotoIf("SIP/704-00000112", "0?initialized") in new stack
  416. -- Executing [s@sub-record-check:2] Set("SIP/704-00000112", "__REC_STATUS=INITIALIZED") in new stack
  417. -- Executing [s@sub-record-check:3] Set("SIP/704-00000112", "NOW=1471023251") in new stack
  418. -- Executing [s@sub-record-check:4] Set("SIP/704-00000112", "__DAY=12") in new stack
  419. -- Executing [s@sub-record-check:5] Set("SIP/704-00000112", "__MONTH=08") in new stack
  420. -- Executing [s@sub-record-check:6] Set("SIP/704-00000112", "__YEAR=2016") in new stack
  421. -- Executing [s@sub-record-check:7] Set("SIP/704-00000112", "__TIMESTR=20160812-133411") in new stack
  422. -- Executing [s@sub-record-check:8] Set("SIP/704-00000112", "__FROMEXTEN=704") in new stack
  423. -- Executing [s@sub-record-check:9] Set("SIP/704-00000112", "__MON_FMT=wav") in new stack
  424. -- Executing [s@sub-record-check:10] NoOp("SIP/704-00000112", "Recordings initialized") in new stack
  425. -- Executing [s@sub-record-check:11] ExecIf("SIP/704-00000112", "0?Set(ARG3=dontcare)") in new stack
  426. -- Executing [s@sub-record-check:12] Set("SIP/704-00000112", "REC_POLICY_MODE_SAVE=") in new stack
  427. -- Executing [s@sub-record-check:13] ExecIf("SIP/704-00000112", "0?Set(REC_STATUS=NO)") in new stack
  428. -- Executing [s@sub-record-check:14] GotoIf("SIP/704-00000112", "5?checkaction") in new stack
  429. -- Goto (sub-record-check,s,17)
  430. -- Executing [s@sub-record-check:17] GotoIf("SIP/704-00000112", "1?sub-record-check,exten,1") in new stack
  431. -- Goto (sub-record-check,exten,1)
  432. -- Executing [exten@sub-record-check:1] NoOp("SIP/704-00000112", "Exten Recording Check between 704 and 761") in new stack
  433. -- Executing [exten@sub-record-check:2] Set("SIP/704-00000112", "CALLTYPE=internal") in new stack
  434. -- Executing [exten@sub-record-check:3] ExecIf("SIP/704-00000112", "0?Set(CALLTYPE=)") in new stack
  435. -- Executing [exten@sub-record-check:4] Set("SIP/704-00000112", "CALLEE=dontcare") in new stack
  436. -- Executing [exten@sub-record-check:5] ExecIf("SIP/704-00000112", "0?Set(CALLEE=dontcare)") in new stack
  437. -- Executing [exten@sub-record-check:6] GotoIf("SIP/704-00000112", "0?callee") in new stack
  438. -- Executing [exten@sub-record-check:7] GotoIf("SIP/704-00000112", "1?caller") in new stack
  439. -- Goto (sub-record-check,exten,13)
  440. -- Executing [exten@sub-record-check:13] Set("SIP/704-00000112", "RECMODE=dontcare") in new stack
  441. -- Executing [exten@sub-record-check:14] ExecIf("SIP/704-00000112", "0?Set(RECMODE=dontcare)") in new stack
  442. -- Executing [exten@sub-record-check:15] ExecIf("SIP/704-00000112", "1?Set(RECMODE=dontcare)") in new stack
  443. -- Executing [exten@sub-record-check:16] Gosub("SIP/704-00000112", "recordcheck,1(dontcare,internal,761)") in new stack
  444. -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/704-00000112", "Starting recording check against dontcare") in new stack
  445. -- Executing [recordcheck@sub-record-check:2] Goto("SIP/704-00000112", "dontcare") in new stack
  446. -- Goto (sub-record-check,recordcheck,3)
  447. -- Executing [recordcheck@sub-record-check:3] Return("SIP/704-00000112", "") in new stack
  448. -- Executing [exten@sub-record-check:17] Return("SIP/704-00000112", "") in new stack
  449. -- Executing [s@macro-exten-vm:9] GotoIf("SIP/704-00000112", "1?macrodial") in new stack
  450. -- Goto (macro-exten-vm,s,15)
  451. -- Executing [s@macro-exten-vm:15] GosubIf("SIP/704-00000112", "0?clrheader,1()") in new stack
  452. -- Executing [s@macro-exten-vm:16] Macro("SIP/704-00000112", "dial-one,,Ttr,761") in new stack
  453. -- Executing [s@macro-dial-one:1] Set("SIP/704-00000112", "DEXTEN=761") in new stack
  454. -- Executing [s@macro-dial-one:2] Set("SIP/704-00000112", "DIALSTATUS_CW=") in new stack
  455. -- Executing [s@macro-dial-one:3] GosubIf("SIP/704-00000112", "0?screen,1()") in new stack
  456. -- Executing [s@macro-dial-one:4] GosubIf("SIP/704-00000112", "0?cf,1()") in new stack
  457. -- Executing [s@macro-dial-one:5] GotoIf("SIP/704-00000112", "1?skip1") in new stack
  458. -- Goto (macro-dial-one,s,8)
  459. -- Executing [s@macro-dial-one:8] GotoIf("SIP/704-00000112", "0?nodial") in new stack
  460. -- Executing [s@macro-dial-one:9] GotoIf("SIP/704-00000112", "0?continue") in new stack
  461. -- Executing [s@macro-dial-one:10] Set("SIP/704-00000112", "EXTHASCW=ENABLED") in new stack
  462. -- Executing [s@macro-dial-one:11] GotoIf("SIP/704-00000112", "0?next1:cwinusebusy") in new stack
  463. -- Goto (macro-dial-one,s,23)
  464. -- Executing [s@macro-dial-one:23] GotoIf("SIP/704-00000112", "0?next3:continue") in new stack
  465. -- Goto (macro-dial-one,s,25)
  466. -- Executing [s@macro-dial-one:25] GotoIf("SIP/704-00000112", "0?nodial") in new stack
  467. -- Executing [s@macro-dial-one:26] GosubIf("SIP/704-00000112", "1?dstring,1():dlocal,1()") in new stack
  468. -- Executing [dstring@macro-dial-one:1] Set("SIP/704-00000112", "DSTRING=") in new stack
  469. -- Executing [dstring@macro-dial-one:2] Set("SIP/704-00000112", "DEVICES=761") in new stack
  470. -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/704-00000112", "0?Return()") in new stack
  471. -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/704-00000112", "0?Set(DEVICES=61)") in new stack
  472. -- Executing [dstring@macro-dial-one:5] Set("SIP/704-00000112", "LOOPCNT=1") in new stack
  473. -- Executing [dstring@macro-dial-one:6] Set("SIP/704-00000112", "ITER=1") in new stack
  474. -- Executing [dstring@macro-dial-one:7] Set("SIP/704-00000112", "THISDIAL=SIP/12132261066@audio1.join.me") in new stack
  475. -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/704-00000112", "1?zap2dahdi,1()") in new stack
  476. -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/704-00000112", "0?Return()") in new stack
  477. -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/704-00000112", "NEWDIAL=") in new stack
  478. -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/704-00000112", "LOOPCNT2=1") in new stack
  479. -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/704-00000112", "ITER2=1") in new stack
  480. -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/704-00000112", "THISPART2=SIP/12132261066@audio1.join.me") in new stack
  481. -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/704-00000112", "0?Set(THISPART2=DAHDI/12132261066@audio1.join.me)") in new stack
  482. -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/704-00000112", "NEWDIAL=SIP/12132261066@audio1.join.me&") in new stack
  483. -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/704-00000112", "ITER2=2") in new stack
  484. -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/704-00000112", "0?begin2") in new stack
  485. -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/704-00000112", "THISDIAL=SIP/12132261066@audio1.join.me") in new stack
  486. -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/704-00000112", "") in new stack
  487. -- Executing [dstring@macro-dial-one:9] GotoIf("SIP/704-00000112", "1?docheck") in new stack
  488. -- Goto (macro-dial-one,dstring,12)
  489. -- Executing [dstring@macro-dial-one:12] GotoIf("SIP/704-00000112", "0?skipset") in new stack
  490. -- Executing [dstring@macro-dial-one:13] Set("SIP/704-00000112", "DSTRING=SIP/12132261066@audio1.join.me&") in new stack
  491. -- Executing [dstring@macro-dial-one:14] Set("SIP/704-00000112", "ITER=2") in new stack
  492. -- Executing [dstring@macro-dial-one:15] GotoIf("SIP/704-00000112", "0?begin") in new stack
  493. -- Executing [dstring@macro-dial-one:16] ExecIf("SIP/704-00000112", "0?Return()") in new stack
  494. -- Executing [dstring@macro-dial-one:17] Set("SIP/704-00000112", "DSTRING=SIP/12132261066@audio1.join.me") in new stack
  495. -- Executing [dstring@macro-dial-one:18] Return("SIP/704-00000112", "") in new stack
  496. -- Executing [s@macro-dial-one:27] GotoIf("SIP/704-00000112", "0?nodial") in new stack
  497. -- Executing [s@macro-dial-one:28] GotoIf("SIP/704-00000112", "0?skiptrace") in new stack
  498. -- Executing [s@macro-dial-one:29] GosubIf("SIP/704-00000112", "1?ctset,1():ctclear,1()") in new stack
  499. -- Executing [ctset@macro-dial-one:1] Set("SIP/704-00000112", "DB(CALLTRACE/761)=704") in new stack
  500. -- Executing [ctset@macro-dial-one:2] Return("SIP/704-00000112", "") in new stack
  501. -- Executing [s@macro-dial-one:30] Set("SIP/704-00000112", "D_OPTIONS=Ttr") in new stack
  502. -- Executing [s@macro-dial-one:31] NoOp("SIP/704-00000112", "Blind Transfer: , Attended Transfer: , User: 704, Alert Info: ") in new stack
  503. -- Executing [s@macro-dial-one:32] ExecIf("SIP/704-00000112", "1?Set(ALERT_INFO=)") in new stack
  504. -- Executing [s@macro-dial-one:33] ExecIf("SIP/704-00000112", "0?Set(ALERT_INFO=)") in new stack
  505. -- Executing [s@macro-dial-one:34] ExecIf("SIP/704-00000112", "0?Set(ALERT_INFO=)") in new stack
  506. -- Executing [s@macro-dial-one:35] GosubIf("SIP/704-00000112", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
  507. -- Executing [s@macro-dial-one:36] ExecIf("SIP/704-00000112", "0?Set(CHANNEL(musicclass)=)") in new stack
  508. -- Executing [s@macro-dial-one:37] GosubIf("SIP/704-00000112", "0?qwait,1()") in new stack
  509. -- Executing [s@macro-dial-one:38] Set("SIP/704-00000112", "__CWIGNORE=") in new stack
  510. -- Executing [s@macro-dial-one:39] Set("SIP/704-00000112", "__KEEPCID=TRUE") in new stack
  511. -- Executing [s@macro-dial-one:40] GotoIf("SIP/704-00000112", "0?usegoto,1") in new stack
  512. -- Executing [s@macro-dial-one:41] GotoIf("SIP/704-00000112", "0?godial") in new stack
  513. -- Executing [s@macro-dial-one:42] Gosub("SIP/704-00000112", "sub-presencestate-display,s,1(761)") in new stack
  514. -- Executing [s@sub-presencestate-display:1] Goto("SIP/704-00000112", "state-not_set,1") in new stack
  515. -- Goto (sub-presencestate-display,state-not_set,1)
  516. -- Executing [state-not_set@sub-presencestate-display:1] Set("SIP/704-00000112", "PRESENCESTATE_DISPLAY=") in new stack
  517. -- Executing [state-not_set@sub-presencestate-display:2] Return("SIP/704-00000112", "") in new stack
  518. -- Executing [s@macro-dial-one:43] Set("SIP/704-00000112", "CONNECTEDLINE(name,i)=761") in new stack
  519. -- Executing [s@macro-dial-one:44] Set("SIP/704-00000112", "CONNECTEDLINE(num)=761") in new stack
  520. -- Executing [s@macro-dial-one:45] Set("SIP/704-00000112", "D_OPTIONS=TtrI") in new stack
  521. -- Executing [s@macro-dial-one:46] Macro("SIP/704-00000112", "dialout-one-predial-hook,") in new stack
  522. -- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("SIP/704-00000112", "") in new stack
  523. -- Executing [s@macro-dial-one:47] ExecIf("SIP/704-00000112", "0?Set(D_OPTIONS=trII)") in new stack
  524. -- Executing [s@macro-dial-one:48] Dial("SIP/704-00000112", "SIP/12132261066@audio1.join.me,,TtrIb(func-apply-sipheaders^s^1)") in new stack
  525. == Using SIP RTP CoS mark 5
  526. -- SIP/audio1.join.me-00000113 Internal Gosub(func-apply-sipheaders,s,1) start
  527. -- Executing [s@func-apply-sipheaders:1] NoOp("SIP/audio1.join.me-00000113", "Applying SIP Headers to channel") in new stack
  528. -- Executing [s@func-apply-sipheaders:2] Set("SIP/audio1.join.me-00000113", "SIPHEADERKEYS=") in new stack
  529. -- Executing [s@func-apply-sipheaders:3] While("SIP/audio1.join.me-00000113", "0") in new stack
  530. -- Jumping to priority 7
  531. -- Executing [s@func-apply-sipheaders:8] Return("SIP/audio1.join.me-00000113", "") in new stack
  532. == Spawn extension (default, 761, 1) exited non-zero on 'SIP/audio1.join.me-00000113'
  533. -- SIP/audio1.join.me-00000113 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
  534. Audio is at 15776
  535. Adding codec g722 to SDP
  536. Adding codec opus to SDP
  537. Adding codec ulaw to SDP
  538. Adding codec alaw to SDP
  539. Adding codec gsm to SDP
  540. Adding codec g726 to SDP
  541. Adding non-codec 0x1 (telephone-event) to SDP
  542. Reliably Transmitting (NAT) to 209.197.28.8:5060:
  543. INVITE sip:12132261066@audio1.join.me SIP/2.0
  544. Via: SIP/2.0/UDP 184.58.69.128:5160;branch=z9hG4bK0cff78fa;rport
  545. Max-Forwards: 70
  546. From: "Mike" <sip:704@184.58.69.128:5160>;tag=as2afe56ab
  547. To: <sip:12132261066@audio1.join.me>
  548. Contact: <sip:704@184.58.69.128:5160>
  549. Call-ID: 0c5aaae819b3575a3631215d5b2e3181@184.58.69.128:5160
  550. CSeq: 102 INVITE
  551. User-Agent: Asterisk PBX 13.10.0
  552. Date: Fri, 12 Aug 2016 17:34:11 GMT
  553. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  554. Supported: replaces, timer
  555. Content-Type: application/sdp
  556. Content-Length: 413
  557.  
  558. v=0
  559. o=root 1986799720 1986799720 IN IP4 184.58.69.128
  560. s=Asterisk PBX 13.10.0
  561. c=IN IP4 184.58.69.128
  562. t=0 0
  563. m=audio 15776 RTP/AVP 9 107 0 8 3 111 101
  564. a=rtpmap:9 G722/8000
  565. a=rtpmap:107 opus/48000/2
  566. a=fmtp:107 useinbandfec=1
  567. a=rtpmap:0 PCMU/8000
  568. a=rtpmap:8 PCMA/8000
  569. a=rtpmap:3 GSM/8000
  570. a=rtpmap:111 G726-32/8000
  571. a=rtpmap:101 telephone-event/8000
  572. a=fmtp:101 0-16
  573. a=ptime:20
  574. a=maxptime:60
  575. a=sendrecv
  576.  
  577. ---
  578. -- Called SIP/12132261066@audio1.join.me
  579.  
  580. <--- Transmitting (no NAT) to 10.1.1.226:5060 --->
  581. SIP/2.0 180 Ringing
  582. Via: SIP/2.0/UDP 10.1.1.226;branch=z9hG4bK8e3cea8dD5FC28EC;received=10.1.1.226;rport=5060
  583. From: "Mike" <sip:704@10.1.1.235:5061>;tag=4AA671E9-CD5264B8
  584. To: <sip:761@10.1.1.235;user=phone>;tag=as2400ea53
  585. Call-ID: 88faf30050df056679c3a357d003110b
  586. CSeq: 2 INVITE
  587. Server: Asterisk PBX 13.10.0
  588. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  589. Supported: replaces, timer
  590. Contact: <sip:761@10.1.1.80:5160>
  591. P-Asserted-Identity: "761" <sip:761@10.1.1.235>
  592. Content-Length: 0
  593.  
  594.  
  595. <------------>
  596. [2016-08-12 13:34:11] NOTICE[1730]: chan_sip.c:15596 sip_reregister: -- Re-registration for 56613@sip.nyc.didlogic.net
  597. REGISTER 12 headers, 0 lines
  598. Reliably Transmitting (NAT) to 162.217.100.10:5060:
  599. REGISTER sip:sip.nyc.didlogic.net SIP/2.0
  600. Via: SIP/2.0/UDP 184.58.69.128:5160;branch=z9hG4bK0b495e9b;rport
  601. Max-Forwards: 70
  602. From: <sip:56613@sip.nyc.didlogic.net>;tag=as0f0be050
  603. To: <sip:56613@sip.nyc.didlogic.net>
  604. Call-ID: 20c0d05f3e2d0292438e79ad1b637a59@[::1]
  605. CSeq: 110 REGISTER
  606. Supported: replaces, timer
  607. User-Agent: Asterisk PBX 13.10.0
  608. Authorization: Digest username="56613", realm="sip.nyc.didlogic.net", algorithm=MD5, uri="sip:sip.nyc.didlogic.net", nonce="V64IhFeuB1gfonyLPjeClDNoofHbVy0/O5cyiYA=", response="1201f0100b3bfc8a0795e021b726d3a2", qop=auth, cnonce="1178b0b7", nc=00000004
  609. Expires: 120
  610. Contact: <sip:s@184.58.69.128:5160>
  611. Content-Length: 0
  612.  
  613.  
  614. ---
  615.  
  616. <--- SIP read from UDP:162.217.100.10:5060 --->
  617. SIP/2.0 401 Unauthorized
  618. Via: SIP/2.0/UDP 184.58.69.128:5160;branch=z9hG4bK0b495e9b;rport=5160;received=184.58.69.128
  619. From: <sip:56613@sip.nyc.didlogic.net>;tag=as0f0be050
  620. To: <sip:56613@sip.nyc.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.0dde
  621. Call-ID: 20c0d05f3e2d0292438e79ad1b637a59@[::1]
  622. CSeq: 110 REGISTER
  623. WWW-Authenticate: Digest realm="sip.nyc.didlogic.net", nonce="V64Jv1euCJM8T+ATafLR5tC0nZa+BcAPO525CYA=", qop="auth"
  624. Content-Length: 0
  625.  
  626. <------------->
  627. --- (8 headers 0 lines) ---
  628. Responding to challenge, registration to domain/host name sip.nyc.didlogic.net
  629. REGISTER 12 headers, 0 lines
  630. Reliably Transmitting (NAT) to 162.217.100.10:5060:
  631. REGISTER sip:sip.nyc.didlogic.net SIP/2.0
  632. Via: SIP/2.0/UDP 184.58.69.128:5160;branch=z9hG4bK3496e799;rport
  633. Max-Forwards: 70
  634. From: <sip:56613@sip.nyc.didlogic.net>;tag=as0f0be050
  635. To: <sip:56613@sip.nyc.didlogic.net>
  636. Call-ID: 20c0d05f3e2d0292438e79ad1b637a59@[::1]
  637. CSeq: 111 REGISTER
  638. Supported: replaces, timer
  639. User-Agent: Asterisk PBX 13.10.0
  640. Authorization: Digest username="56613", realm="sip.nyc.didlogic.net", algorithm=MD5, uri="sip:sip.nyc.didlogic.net", nonce="V64Jv1euCJM8T+ATafLR5tC0nZa+BcAPO525CYA=", response="b8c358879108f7b5f7d9fefa334cec24", qop=auth, cnonce="48119a9e", nc=00000001
  641. Expires: 120
  642. Contact: <sip:s@184.58.69.128:5160>
  643. Content-Length: 0
  644.  
  645.  
  646. ---
  647. Retransmitting #1 (NAT) to 209.197.28.8:5060:
  648. INVITE sip:12132261066@audio1.join.me SIP/2.0
  649. Via: SIP/2.0/UDP 184.58.69.128:5160;branch=z9hG4bK0cff78fa;rport
  650. Max-Forwards: 70
  651. From: "Mike" <sip:704@184.58.69.128:5160>;tag=as2afe56ab
  652. To: <sip:12132261066@audio1.join.me>
  653. Contact: <sip:704@184.58.69.128:5160>
  654. Call-ID: 0c5aaae819b3575a3631215d5b2e3181@184.58.69.128:5160
  655. CSeq: 102 INVITE
  656. User-Agent: Asterisk PBX 13.10.0
  657. Date: Fri, 12 Aug 2016 17:34:11 GMT
  658. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  659. Supported: replaces, timer
  660. Content-Type: application/sdp
  661. Content-Length: 413
  662.  
  663. v=0
  664. o=root 1986799720 1986799720 IN IP4 184.58.69.128
  665. s=Asterisk PBX 13.10.0
  666. c=IN IP4 184.58.69.128
  667. t=0 0
  668. m=audio 15776 RTP/AVP 9 107 0 8 3 111 101
  669. a=rtpmap:9 G722/8000
  670. a=rtpmap:107 opus/48000/2
  671. a=fmtp:107 useinbandfec=1
  672. a=rtpmap:0 PCMU/8000
  673. a=rtpmap:8 PCMA/8000
  674. a=rtpmap:3 GSM/8000
  675. a=rtpmap:111 G726-32/8000
  676. a=rtpmap:101 telephone-event/8000
  677. a=fmtp:101 0-16
  678. a=ptime:20
  679. a=maxptime:60
  680. a=sendrecv
  681.  
  682. ---
  683.  
  684. <--- SIP read from UDP:162.217.100.10:5060 --->
  685. SIP/2.0 200 OK
  686. Via: SIP/2.0/UDP 184.58.69.128:5160;branch=z9hG4bK3496e799;rport=5160;received=184.58.69.128
  687. From: <sip:56613@sip.nyc.didlogic.net>;tag=as0f0be050
  688. To: <sip:56613@sip.nyc.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.a3d1
  689. Call-ID: 20c0d05f3e2d0292438e79ad1b637a59@[::1]
  690. CSeq: 111 REGISTER
  691. Contact: <sip:s@184.58.69.128:5160>;expires=120
  692. Content-Length: 0
  693.  
  694. <------------->
  695. --- (8 headers 0 lines) ---
  696. [2016-08-12 13:34:11] NOTICE[1730]: chan_sip.c:24377 handle_response_register: Outbound Registration: Expiry for sip.nyc.didlogic.net is 120 sec (Scheduling reregistration in 105 s)
  697. Really destroying SIP dialog '20c0d05f3e2d0292438e79ad1b637a59@[::1]' Method: REGISTER
  698. Retransmitting #2 (NAT) to 209.197.28.8:5060:
  699. INVITE sip:12132261066@audio1.join.me SIP/2.0
  700. Via: SIP/2.0/UDP 184.58.69.128:5160;branch=z9hG4bK0cff78fa;rport
  701. Max-Forwards: 70
  702. From: "Mike" <sip:704@184.58.69.128:5160>;tag=as2afe56ab
  703. To: <sip:12132261066@audio1.join.me>
  704. Contact: <sip:704@184.58.69.128:5160>
  705. Call-ID: 0c5aaae819b3575a3631215d5b2e3181@184.58.69.128:5160
  706. CSeq: 102 INVITE
  707. User-Agent: Asterisk PBX 13.10.0
  708. Date: Fri, 12 Aug 2016 17:34:11 GMT
  709. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  710. Supported: replaces, timer
  711. Content-Type: application/sdp
  712. Content-Length: 413
  713.  
  714. v=0
  715. o=root 1986799720 1986799720 IN IP4 184.58.69.128
  716. s=Asterisk PBX 13.10.0
  717. c=IN IP4 184.58.69.128
  718. t=0 0
  719. m=audio 15776 RTP/AVP 9 107 0 8 3 111 101
  720. a=rtpmap:9 G722/8000
  721. a=rtpmap:107 opus/48000/2
  722. a=fmtp:107 useinbandfec=1
  723. a=rtpmap:0 PCMU/8000
  724. a=rtpmap:8 PCMA/8000
  725. a=rtpmap:3 GSM/8000
  726. a=rtpmap:111 G726-32/8000
  727. a=rtpmap:101 telephone-event/8000
  728. a=fmtp:101 0-16
  729. a=ptime:20
  730. a=maxptime:60
  731. a=sendrecv
  732.  
  733. ---
  734. Retransmitting #3 (NAT) to 209.197.28.8:5060:
  735. INVITE sip:12132261066@audio1.join.me SIP/2.0
  736. Via: SIP/2.0/UDP 184.58.69.128:5160;branch=z9hG4bK0cff78fa;rport
  737. Max-Forwards: 70
  738. From: "Mike" <sip:704@184.58.69.128:5160>;tag=as2afe56ab
  739. To: <sip:12132261066@audio1.join.me>
  740. Contact: <sip:704@184.58.69.128:5160>
  741. Call-ID: 0c5aaae819b3575a3631215d5b2e3181@184.58.69.128:5160
  742. CSeq: 102 INVITE
  743. User-Agent: Asterisk PBX 13.10.0
  744. Date: Fri, 12 Aug 2016 17:34:11 GMT
  745. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  746. Supported: replaces, timer
  747. Content-Type: application/sdp
  748. Content-Length: 413
  749.  
  750. v=0
  751. o=root 1986799720 1986799720 IN IP4 184.58.69.128
  752. s=Asterisk PBX 13.10.0
  753. c=IN IP4 184.58.69.128
  754. t=0 0
  755. m=audio 15776 RTP/AVP 9 107 0 8 3 111 101
  756. a=rtpmap:9 G722/8000
  757. a=rtpmap:107 opus/48000/2
  758. a=fmtp:107 useinbandfec=1
  759. a=rtpmap:0 PCMU/8000
  760. a=rtpmap:8 PCMA/8000
  761. a=rtpmap:3 GSM/8000
  762. a=rtpmap:111 G726-32/8000
  763. a=rtpmap:101 telephone-event/8000
  764. a=fmtp:101 0-16
  765. a=ptime:20
  766. a=maxptime:60
  767. a=sendrecv
  768.  
  769. ---
  770. Retransmitting #4 (NAT) to 209.197.28.8:5060:
  771. INVITE sip:12132261066@audio1.join.me SIP/2.0
  772. Via: SIP/2.0/UDP 184.58.69.128:5160;branch=z9hG4bK0cff78fa;rport
  773. Max-Forwards: 70
  774. From: "Mike" <sip:704@184.58.69.128:5160>;tag=as2afe56ab
  775. To: <sip:12132261066@audio1.join.me>
  776. Contact: <sip:704@184.58.69.128:5160>
  777. Call-ID: 0c5aaae819b3575a3631215d5b2e3181@184.58.69.128:5160
  778. CSeq: 102 INVITE
  779. User-Agent: Asterisk PBX 13.10.0
  780. Date: Fri, 12 Aug 2016 17:34:11 GMT
  781. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  782. Supported: replaces, timer
  783. Content-Type: application/sdp
  784. Content-Length: 413
  785.  
  786. v=0
  787. o=root 1986799720 1986799720 IN IP4 184.58.69.128
  788. s=Asterisk PBX 13.10.0
  789. c=IN IP4 184.58.69.128
  790. t=0 0
  791. m=audio 15776 RTP/AVP 9 107 0 8 3 111 101
  792. a=rtpmap:9 G722/8000
  793. a=rtpmap:107 opus/48000/2
  794. a=fmtp:107 useinbandfec=1
  795. a=rtpmap:0 PCMU/8000
  796. a=rtpmap:8 PCMA/8000
  797. a=rtpmap:3 GSM/8000
  798. a=rtpmap:111 G726-32/8000
  799. a=rtpmap:101 telephone-event/8000
  800. a=fmtp:101 0-16
  801. a=ptime:20
  802. a=maxptime:60
  803. a=sendrecv
  804.  
  805. ---
  806. Retransmitting #5 (NAT) to 209.197.28.8:5060:
  807. INVITE sip:12132261066@audio1.join.me SIP/2.0
  808. Via: SIP/2.0/UDP 184.58.69.128:5160;branch=z9hG4bK0cff78fa;rport
  809. Max-Forwards: 70
  810. From: "Mike" <sip:704@184.58.69.128:5160>;tag=as2afe56ab
  811. To: <sip:12132261066@audio1.join.me>
  812. Contact: <sip:704@184.58.69.128:5160>
  813. Call-ID: 0c5aaae819b3575a3631215d5b2e3181@184.58.69.128:5160
  814. CSeq: 102 INVITE
  815. User-Agent: Asterisk PBX 13.10.0
  816. Date: Fri, 12 Aug 2016 17:34:11 GMT
  817. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  818. Supported: replaces, timer
  819. Content-Type: application/sdp
  820. Content-Length: 413
  821.  
  822. v=0
  823. o=root 1986799720 1986799720 IN IP4 184.58.69.128
  824. s=Asterisk PBX 13.10.0
  825. c=IN IP4 184.58.69.128
  826. t=0 0
  827. m=audio 15776 RTP/AVP 9 107 0 8 3 111 101
  828. a=rtpmap:9 G722/8000
  829. a=rtpmap:107 opus/48000/2
  830. a=fmtp:107 useinbandfec=1
  831. a=rtpmap:0 PCMU/8000
  832. a=rtpmap:8 PCMA/8000
  833. a=rtpmap:3 GSM/8000
  834. a=rtpmap:111 G726-32/8000
  835. a=rtpmap:101 telephone-event/8000
  836. a=fmtp:101 0-16
  837. a=ptime:20
  838. a=maxptime:60
  839. a=sendrecv
  840.  
  841. ---
  842. Reliably Transmitting (no NAT) to 10.1.1.227:5060:
  843. OPTIONS sip:703@10.1.1.227 SIP/2.0
  844. Via: SIP/2.0/UDP 10.1.1.80:5160;branch=z9hG4bK64de65c0
  845. Max-Forwards: 70
  846. From: "asterisk" <sip:asterisk@10.1.1.80:5160>;tag=as3b41d30f
  847. To: <sip:703@10.1.1.227>
  848. Contact: <sip:asterisk@10.1.1.80:5160>
  849. Call-ID: 2414f63360f26fa96c2fdcb1247ef96d@10.1.1.80:5160
  850. CSeq: 102 OPTIONS
  851. User-Agent: Asterisk PBX 13.10.0
  852. Date: Fri, 12 Aug 2016 17:34:40 GMT
  853. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  854. Supported: replaces, timer
  855. Content-Length: 0
  856.  
  857.  
  858. ---
  859.  
  860. <--- SIP read from UDP:10.1.1.227:5060 --->
  861. SIP/2.0 200 OK
  862. Via: SIP/2.0/UDP 10.1.1.80:5160;branch=z9hG4bK64de65c0
  863. From: "asterisk" <sip:asterisk@10.1.1.80:5160>;tag=as3b41d30f
  864. To: "Louisa" <sip:703@10.1.1.227>;tag=93882DC3-A4153D3E
  865. CSeq: 102 OPTIONS
  866. Call-ID: 2414f63360f26fa96c2fdcb1247ef96d@10.1.1.80:5160
  867. Contact: <sip:703@10.1.1.227>
  868. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  869. Supported: replaces,100rel,100rel,timer,replaces,norefersub,sdp-anat
  870. User-Agent: Polycom/5.5.0.20556 PolycomVVX-VVX_410-UA/5.5.0.20556
  871. Accept-Language: en
  872. Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
  873. Accept-Encoding: identity
  874. Content-Length: 0
  875.  
  876. <------------->
  877. --- (14 headers 0 lines) ---
  878. Really destroying SIP dialog '2414f63360f26fa96c2fdcb1247ef96d@10.1.1.80:5160' Method: OPTIONS
  879. Reliably Transmitting (no NAT) to 10.1.1.226:5060:
  880. OPTIONS sip:704@10.1.1.226 SIP/2.0
  881. Via: SIP/2.0/UDP 10.1.1.80:5160;branch=z9hG4bK580bede6
  882. Max-Forwards: 70
  883. From: "asterisk" <sip:asterisk@10.1.1.80:5160>;tag=as5438507b
  884. To: <sip:704@10.1.1.226>
  885. Contact: <sip:asterisk@10.1.1.80:5160>
  886. Call-ID: 745b7e162e8015e57e9cf1eb5a2ca078@10.1.1.80:5160
  887. CSeq: 102 OPTIONS
  888. User-Agent: Asterisk PBX 13.10.0
  889. Date: Fri, 12 Aug 2016 17:34:40 GMT
  890. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  891. Supported: replaces, timer
  892. Content-Length: 0
  893.  
  894.  
  895. ---
  896.  
  897. <--- SIP read from UDP:10.1.1.226:5060 --->
  898. SIP/2.0 200 OK
  899. Via: SIP/2.0/UDP 10.1.1.80:5160;branch=z9hG4bK580bede6
  900. From: "asterisk" <sip:asterisk@10.1.1.80:5160>;tag=as5438507b
  901. To: "Mike" <sip:704@10.1.1.226>;tag=F8BC06F1-A4060FE0
  902. CSeq: 102 OPTIONS
  903. Call-ID: 745b7e162e8015e57e9cf1eb5a2ca078@10.1.1.80:5160
  904. Contact: <sip:704@10.1.1.226>
  905. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  906. Supported: replaces,100rel,100rel,timer,replaces,norefersub,sdp-anat
  907. User-Agent: Polycom/5.5.0.20556 PolycomVVX-VVX_601-UA/5.5.0.20556
  908. Accept-Language: en
  909. Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
  910. Accept-Encoding: identity
  911. Content-Length: 0
  912.  
  913. <------------->
  914. --- (14 headers 0 lines) ---
  915. Really destroying SIP dialog '745b7e162e8015e57e9cf1eb5a2ca078@10.1.1.80:5160' Method: OPTIONS
  916. Reliably Transmitting (NAT) to 162.217.100.10:5060:
  917. OPTIONS sip:sip.nyc.didlogic.net SIP/2.0
  918. Via: SIP/2.0/UDP 184.58.69.128:5160;branch=z9hG4bK5187e0c3;rport
  919. Max-Forwards: 70
  920. From: "asterisk" <sip:56613@184.58.69.128:5160>;tag=as368f862e
  921. To: <sip:sip.nyc.didlogic.net>
  922. Contact: <sip:56613@184.58.69.128:5160>
  923. Call-ID: 7593c83379265ca347e384306a99a2d4@184.58.69.128:5160
  924. CSeq: 102 OPTIONS
  925. User-Agent: Asterisk PBX 13.10.0
  926. Date: Fri, 12 Aug 2016 17:34:41 GMT
  927. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  928. Supported: replaces, timer
  929. Content-Length: 0
  930.  
  931.  
  932. ---
  933.  
  934. <--- SIP read from UDP:162.217.100.10:5060 --->
  935. SIP/2.0 200 OK
  936. Via: SIP/2.0/UDP 184.58.69.128:5160;branch=z9hG4bK5187e0c3;rport=5160;received=184.58.69.128
  937. From: "asterisk" <sip:56613@184.58.69.128:5160>;tag=as368f862e
  938. To: <sip:sip.nyc.didlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.f92f
  939. Call-ID: 7593c83379265ca347e384306a99a2d4@184.58.69.128:5160
  940. CSeq: 102 OPTIONS
  941. Content-Length: 0
  942.  
  943. <------------->
  944. --- (7 headers 0 lines) ---
  945. Really destroying SIP dialog '7593c83379265ca347e384306a99a2d4@184.58.69.128:5160' Method: OPTIONS
  946. Retransmitting #6 (NAT) to 209.197.28.8:5060:
  947. INVITE sip:12132261066@audio1.join.me SIP/2.0
  948. Via: SIP/2.0/UDP 184.58.69.128:5160;branch=z9hG4bK0cff78fa;rport
  949. Max-Forwards: 70
  950. From: "Mike" <sip:704@184.58.69.128:5160>;tag=as2afe56ab
  951. To: <sip:12132261066@audio1.join.me>
  952. Contact: <sip:704@184.58.69.128:5160>
  953. Call-ID: 0c5aaae819b3575a3631215d5b2e3181@184.58.69.128:5160
  954. CSeq: 102 INVITE
  955. User-Agent: Asterisk PBX 13.10.0
  956. Date: Fri, 12 Aug 2016 17:34:11 GMT
  957. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  958. Supported: replaces, timer
  959. Content-Type: application/sdp
  960. Content-Length: 413
  961.  
  962. v=0
  963. o=root 1986799720 1986799720 IN IP4 184.58.69.128
  964. s=Asterisk PBX 13.10.0
  965. c=IN IP4 184.58.69.128
  966. t=0 0
  967. m=audio 15776 RTP/AVP 9 107 0 8 3 111 101
  968. a=rtpmap:9 G722/8000
  969. a=rtpmap:107 opus/48000/2
  970. a=fmtp:107 useinbandfec=1
  971. a=rtpmap:0 PCMU/8000
  972. a=rtpmap:8 PCMA/8000
  973. a=rtpmap:3 GSM/8000
  974. a=rtpmap:111 G726-32/8000
  975. a=rtpmap:101 telephone-event/8000
  976. a=fmtp:101 0-16
  977. a=ptime:20
  978. a=maxptime:60
  979. a=sendrecv
  980.  
  981. ---
  982. [2016-08-12 13:34:43] WARNING[1730]: chan_sip.c:4059 retrans_pkt: Retransmission timeout reached on transmission 0c5aaae819b3575a3631215d5b2e3181@184.58.69.128:5160 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  983. Packet timed out after 31999ms with no response
  984. [2016-08-12 13:34:43] WARNING[1730]: chan_sip.c:4083 retrans_pkt: Hanging up call 0c5aaae819b3575a3631215d5b2e3181@184.58.69.128:5160 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  985. Scheduling destruction of SIP dialog '0c5aaae819b3575a3631215d5b2e3181@184.58.69.128:5160' in 32000 ms (Method: INVITE)
  986. == Everyone is busy/congested at this time (1:0/0/1)
  987. -- Executing [s@macro-dial-one:49] ExecIf("SIP/704-00000112", "0?MacroExit()") in new stack
  988. -- Executing [s@macro-dial-one:50] ExecIf("SIP/704-00000112", "0?Set(DIALSTATUS=)") in new stack
  989. -- Executing [s@macro-dial-one:51] GosubIf("SIP/704-00000112", "0?s-CHANUNAVAIL,1()") in new stack
  990. -- Executing [s@macro-dial-one:52] MacroExit("SIP/704-00000112", "") in new stack
  991. -- Executing [s@macro-exten-vm:17] Set("SIP/704-00000112", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
  992. -- Executing [s@macro-exten-vm:18] GosubIf("SIP/704-00000112", "0?docfu,1()") in new stack
  993. -- Executing [s@macro-exten-vm:19] GosubIf("SIP/704-00000112", "0?docfb,1()") in new stack
  994. -- Executing [s@macro-exten-vm:20] Set("SIP/704-00000112", "DIALSTATUS=CHANUNAVAIL") in new stack
  995. -- Executing [s@macro-exten-vm:21] ExecIf("SIP/704-00000112", "0?MacroExit()") in new stack
  996. -- Executing [s@macro-exten-vm:22] GotoIf("SIP/704-00000112", "1?s-CHANUNAVAIL,1") in new stack
  997. -- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
  998. -- Executing [s-CHANUNAVAIL@macro-exten-vm:1] GotoIf("SIP/704-00000112", "0?exit,1") in new stack
  999. -- Executing [s-CHANUNAVAIL@macro-exten-vm:2] PlayTones("SIP/704-00000112", "congestion") in new stack
  1000. -- Executing [s-CHANUNAVAIL@macro-exten-vm:3] Congestion("SIP/704-00000112", "10") in new stack
  1001.  
  1002. <--- Reliably Transmitting (no NAT) to 10.1.1.226:5060 --->
  1003. SIP/2.0 503 Service Unavailable
  1004. Via: SIP/2.0/UDP 10.1.1.226;branch=z9hG4bK8e3cea8dD5FC28EC;received=10.1.1.226;rport=5060
  1005. From: "Mike" <sip:704@10.1.1.235:5061>;tag=4AA671E9-CD5264B8
  1006. To: <sip:761@10.1.1.235;user=phone>;tag=as2400ea53
  1007. Call-ID: 88faf30050df056679c3a357d003110b
  1008. CSeq: 2 INVITE
  1009. Server: Asterisk PBX 13.10.0
  1010. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  1011. Supported: replaces, timer
  1012. X-Asterisk-HangupCause: No user responding
  1013. X-Asterisk-HangupCauseCode: 18
  1014. Content-Length: 0
  1015.  
  1016.  
  1017. <------------>
  1018. == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/704-00000112' in macro 'exten-vm'
  1019. == Spawn extension (ext-local, 761, 2) exited non-zero on 'SIP/704-00000112'
  1020. -- Executing [h@ext-local:1] Macro("SIP/704-00000112", "hangupcall,") in new stack
  1021. -- Executing [s@macro-hangupcall:1] GotoIf("SIP/704-00000112", "1?theend") in new stack
  1022. -- Goto (macro-hangupcall,s,3)
  1023. -- Executing [s@macro-hangupcall:3] ExecIf("SIP/704-00000112", "0?Set(CDR(recordingfile)=)") in new stack
  1024. -- Executing [s@macro-hangupcall:4] Hangup("SIP/704-00000112", "") in new stack
  1025. Really destroying SIP dialog '0c5aaae819b3575a3631215d5b2e3181@184.58.69.128:5160' Method: INVITE
  1026. == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/704-00000112' in macro 'hangupcall'
  1027. == Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/704-00000112'
  1028.  
  1029. <--- SIP read from UDP:10.1.1.226:5060 --->
  1030. ACK sip:761@10.1.1.235:5061;user=phone SIP/2.0
  1031. Via: SIP/2.0/UDP 10.1.1.226;rport;branch=z9hG4bK8e3cea8dD5FC28EC
  1032. From: "Mike" <sip:704@10.1.1.235:5061>;tag=4AA671E9-CD5264B8
  1033. To: <sip:761@10.1.1.235;user=phone>;tag=as2400ea53
  1034. CSeq: 2 ACK
  1035. Call-ID: 88faf30050df056679c3a357d003110b
  1036. Contact: <sip:704@10.1.1.226>
  1037. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
  1038. User-Agent: Polycom/5.5.0.20556 PolycomVVX-VVX_601-UA/5.5.0.20556
  1039. Accept-Language: en
  1040. Max-Forwards: 70
  1041. Content-Length: 0
  1042.  
  1043. <------------->
  1044. --- (12 headers 0 lines) ---
  1045. Really destroying SIP dialog '88faf30050df056679c3a357d003110b' Method: ACK
  1046. FREEPBX*CLI>
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