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May 24th, 2012
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  1. -- Executing [8017478048@from-trunk:5] ExecIf("SIP/vitel-inbound-00000031", "0 ?Set(CALLERID(name)=8014100203)") in new stack
  2. -- Executing [8017478048@from-trunk:6] Set("SIP/vitel-inbound-00000031", "CHANNEL(musicclass)=moh") in new stack
  3. -- Executing [8017478048@from-trunk:7] Set("SIP/vitel-inbound-00000031", "__MOHCLASS=moh") in new stack
  4. -- Executing [8017478048@from-trunk:8] Ringing("SIP/vitel-inbound-00000031", "") in new stack
  5.  
  6. <--- Transmitting (NAT) to 66.241.96.221:5060 --->
  7. SIP/2.0 180 Ringing
  8. Via: SIP/2.0/UDP 66.241.96.221:5060;branch=z9hG4bK242209e1;received=66.241.96.221;rport=5060
  9. From: "8014100203" <sip:8014100203@66.241.96.221>;tag=as6056c176
  10. To: <sip:8017478048@24.2.66.2:5060>;tag=as34b2bc52
  11. Call-ID: 7c1454a27f4333035455d68e0f484ae0@66.241.96.221
  12. CSeq: 102 INVITE
  13. Server: FPBX-2.10.0(1.8.7.1)
  14. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  15. Supported: replaces, timer
  16. Contact: <sip:8017478048@24.2.66.2:5060>
  17. Content-Length: 0
  18.  
  19.  
  20. <------------>
  21. -- Executing [8017478048@from-trunk:9] Wait("SIP/vitel-inbound-00000031", "2") in new stack
  22. -- Executing [8017478048@from-trunk:10] Set("SIP/vitel-inbound-00000031", "__CALLINGPRES_SV=allowed_not_screened") in new stack
  23. -- Executing [8017478048@from-trunk:11] Set("SIP/vitel-inbound-00000031", "CALLERPRES()=allowed_not_screened") in new stack
  24. -- Executing [8017478048@from-trunk:12] Goto("SIP/vitel-inbound-00000031", "ivr-2,s,1") in new stack
  25. -- Goto (ivr-2,s,1)
  26. -- Executing [s@ivr-2:1] Set("SIP/vitel-inbound-00000031", "TIMEOUT_LOOPCOUNT=0") in new stack
  27. -- Executing [s@ivr-2:2] Set("SIP/vitel-inbound-00000031", "INVALID_LOOPCOUNT=0") in new stack
  28. -- Executing [s@ivr-2:3] Set("SIP/vitel-inbound-00000031", "_IVR_CONTEXT_ivr-2=") in new stack
  29. -- Executing [s@ivr-2:4] Set("SIP/vitel-inbound-00000031", "_IVR_CONTEXT=ivr-2") in new stack
  30. -- Executing [s@ivr-2:5] Set("SIP/vitel-inbound-00000031", "__IVR_RETVM=RETURN") in new stack
  31. -- Executing [s@ivr-2:6] GotoIf("SIP/vitel-inbound-00000031", "0?skip") in new stack
  32. -- Executing [s@ivr-2:7] Answer("SIP/vitel-inbound-00000031", "") in new stack
  33. Audio is at 5060
  34. Adding codec 0x4 (ulaw) to SDP
  35. Adding codec 0x8 (alaw) to SDP
  36. Adding codec 0x2 (gsm) to SDP
  37. Adding non-codec 0x1 (telephone-event) to SDP
  38.  
  39. <--- Reliably Transmitting (NAT) to 66.241.96.221:5060 --->
  40. SIP/2.0 200 OK
  41. Via: SIP/2.0/UDP 66.241.96.221:5060;branch=z9hG4bK242209e1;received=66.241.96.221;rport=5060
  42. From: "8014100203" <sip:8014100203@66.241.96.221>;tag=as6056c176
  43. To: <sip:8017478048@24.2.66.2:5060>;tag=as34b2bc52
  44. Call-ID: 7c1454a27f4333035455d68e0f484ae0@66.241.96.221
  45. CSeq: 102 INVITE
  46. Server: FPBX-2.10.0(1.8.7.1)
  47. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  48. Supported: replaces, timer
  49. Contact: <sip:8017478048@24.2.66.2:5060>
  50. Content-Type: application/sdp
  51. Content-Length: 277
  52.  
  53. v=0
  54. o=root 1917445873 1917445873 IN IP4 24.2.66.2
  55. s=Asterisk PBX 1.8.7.1
  56. c=IN IP4 24.2.66.2
  57. t=0 0
  58. m=audio 16808 RTP/AVP 0 8 3 101
  59. a=rtpmap:0 PCMU/8000
  60. a=rtpmap:8 PCMA/8000
  61. a=rtpmap:3 GSM/8000
  62. a=rtpmap:101 telephone-event/8000
  63. a=fmtp:101 0-16
  64. a=ptime:20
  65. a=sendrecv
  66.  
  67. <------------>
  68.  
  69. <--- SIP read from UDP:66.241.96.221:5060 --->
  70. ACK sip:8017478048@24.2.66.2:5060 SIP/2.0
  71. Via: SIP/2.0/UDP 66.241.96.221:5060;branch=z9hG4bK444485e3;rport
  72. From: "8014100203" <sip:8014100203@66.241.96.221>;tag=as6056c176
  73. To: <sip:8017478048@24.2.66.2:5060>;tag=as34b2bc52
  74. Contact: <sip:8014100203@66.241.96.221>
  75. Call-ID: 7c1454a27f4333035455d68e0f484ae0@66.241.96.221
  76. CSeq: 102 ACK
  77. User-Agent: packetrino
  78. Max-Forwards: 70
  79. Content-Length: 0
  80.  
  81. <------------->
  82. --- (10 headers 0 lines) ---
  83. -- Executing [s@ivr-2:8] Wait("SIP/vitel-inbound-00000031", "1") in new stack
  84. -- Executing [s@ivr-2:9] Set("SIP/vitel-inbound-00000031", "IVR_MSG=custom/ccwtechivr") in new stack
  85. -- Executing [s@ivr-2:10] Set("SIP/vitel-inbound-00000031", "TIMEOUT(digit)=3") in new stack
  86. -- Digit timeout set to 3.000
  87. -- Executing [s@ivr-2:11] ExecIf("SIP/vitel-inbound-00000031", "1?Background(custom/ccwtechivr)") in new stack
  88. -- <SIP/vitel-inbound-00000031> Playing 'custom/ccwtechivr.slin' (language 'en')
  89. == CDR updated on SIP/vitel-inbound-00000031
  90. -- Executing [1@ivr-2:1] Goto("SIP/vitel-inbound-00000031", "timeconditions,1,1") in new stack
  91. -- Goto (timeconditions,1,1)
  92. -- Executing [1@timeconditions:1] GotoIfTime("SIP/vitel-inbound-00000031", "08:00-17:00,mon-thu,*,*?truestate") in new stack
  93. -- Executing [1@timeconditions:2] GotoIf("SIP/vitel-inbound-00000031", "0?truegoto") in new stack
  94. -- Executing [1@timeconditions:3] ExecIf("SIP/vitel-inbound-00000031", "0?Set(DB(TC/1)=)") in new stack
  95. -- Executing [1@timeconditions:4] GotoIf("SIP/vitel-inbound-00000031", "1?ext-group,602,1") in new stack
  96. -- Goto (ext-group,602,1)
  97. -- Executing [602@ext-group:1] Macro("SIP/vitel-inbound-00000031", "user-callerid,") in new stack
  98. -- Executing [s@macro-user-callerid:1] Set("SIP/vitel-inbound-00000031", "AMPUSER=8014100203") in new stack
  99. -- Executing [s@macro-user-callerid:2] GotoIf("SIP/vitel-inbound-00000031", "0?report") in new stack
  100. -- Executing [s@macro-user-callerid:3] ExecIf("SIP/vitel-inbound-00000031", "1?Set(REALCALLERIDNUM=8014100203)") in new stack
  101. -- Executing [s@macro-user-callerid:4] Set("SIP/vitel-inbound-00000031", "AMPUSER=") in new stack
  102. -- Executing [s@macro-user-callerid:5] Set("SIP/vitel-inbound-00000031", "AMPUSERCIDNAME=") in new stack
  103. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/vitel-inbound-00000031", "1?report") in new stack
  104. -- Goto (macro-user-callerid,s,13)
  105. -- Executing [s@macro-user-callerid:13] GotoIf("SIP/vitel-inbound-00000031", "0?continue") in new stack
  106. -- Executing [s@macro-user-callerid:14] Set("SIP/vitel-inbound-00000031", "__TTL=64") in new stack
  107. -- Executing [s@macro-user-callerid:15] GotoIf("SIP/vitel-inbound-00000031", "1?continue") in new stack
  108. -- Goto (macro-user-callerid,s,26)
  109. -- Executing [s@macro-user-callerid:26] Set("SIP/vitel-inbound-00000031", "CALLERID(number)=8014100203") in new stack
  110. -- Executing [s@macro-user-callerid:27] Set("SIP/vitel-inbound-00000031", "CALLERID(name)=SALT LAKE, UT") in new stack
  111. -- Executing [s@macro-user-callerid:28] Set("SIP/vitel-inbound-00000031", "CHANNEL(language)=en") in new stack
  112. -- Executing [602@ext-group:2] Macro("SIP/vitel-inbound-00000031", "blkvm-setifempty,") in new stack
  113. -- Executing [s@macro-blkvm-setifempty:1] GotoIf("SIP/vitel-inbound-00000031", "1?init") in new stack
  114. -- Goto (macro-blkvm-setifempty,s,4)
  115. -- Executing [s@macro-blkvm-setifempty:4] Set("SIP/vitel-inbound-00000031", "__BLKVM_CHANNEL=SIP/vitel-inbound-00000031") in new stack
  116. -- Executing [s@macro-blkvm-setifempty:5] Set("SIP/vitel-inbound-00000031", "SHARED(BLKVM,SIP/vitel-inbound-00000031)=TRUE") in new stack
  117. -- Executing [s@macro-blkvm-setifempty:6] Set("SIP/vitel-inbound-00000031", "GOSUB_RETVAL=TRUE") in new stack
  118. -- Executing [s@macro-blkvm-setifempty:7] MacroExit("SIP/vitel-inbound-00000031", "") in new stack
  119. -- Executing [602@ext-group:3] GotoIf("SIP/vitel-inbound-00000031", "1?skipov") in new stack
  120. -- Goto (ext-group,602,6)
  121. -- Executing [602@ext-group:6] Set("SIP/vitel-inbound-00000031", "RRNODEST=") in new stack
  122. -- Executing [602@ext-group:7] Set("SIP/vitel-inbound-00000031", "__NODEST=602") in new stack
  123. -- Executing [602@ext-group:8] GosubIf("SIP/vitel-inbound-00000031", "0?sub-rgsetcid,s,1()") in new stack
  124. -- Executing [602@ext-group:9] Gosub("SIP/vitel-inbound-00000031", "sub-record-check,s,1(rg,602,dontcare)") in new stack
  125. -- Executing [s@sub-record-check:1] GotoIf("SIP/vitel-inbound-00000031", "1?check") in new stack
  126. -- Goto (sub-record-check,s,6)
  127. -- Executing [s@sub-record-check:6] Set("SIP/vitel-inbound-00000031", "__MON_FMT=wav") in new stack
  128. -- Executing [s@sub-record-check:7] GotoIf("SIP/vitel-inbound-00000031", "1?next") in new stack
  129. -- Goto (sub-record-check,s,10)
  130. -- Executing [s@sub-record-check:10] ExecIf("SIP/vitel-inbound-00000031", "0?Return()") in new stack
  131. -- Executing [s@sub-record-check:11] GotoIf("SIP/vitel-inbound-00000031", "0?rg,1") in new stack
  132. -- Executing [s@sub-record-check:12] Set("SIP/vitel-inbound-00000031", "__REC_STATUS=INITIALIZED") in new stack
  133. -- Executing [s@sub-record-check:13] ExecIf("SIP/vitel-inbound-00000031", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
  134. -- Executing [s@sub-record-check:14] Set("SIP/vitel-inbound-00000031", "NOW=1337856316") in new stack
  135. -- Executing [s@sub-record-check:15] Set("SIP/vitel-inbound-00000031", "__DAY=24") in new stack
  136. -- Executing [s@sub-record-check:16] Set("SIP/vitel-inbound-00000031", "__MONTH=05") in new stack
  137. -- Executing [s@sub-record-check:17] Set("SIP/vitel-inbound-00000031", "__YEAR=2012") in new stack
  138. -- Executing [s@sub-record-check:18] Set("SIP/vitel-inbound-00000031", "__TIMESTR=20120524-044516") in new stack
  139. -- Executing [s@sub-record-check:19] Set("SIP/vitel-inbound-00000031", "__FROMEXTEN=8014100203") in new stack
  140. -- Executing [s@sub-record-check:20] Set("SIP/vitel-inbound-00000031", "__CALLFILENAME=rg-602-8014100203-20120524-044516-1337856306.69") in new stack
  141. -- Executing [s@sub-record-check:21] Goto("SIP/vitel-inbound-00000031", "rg,1") in new stack
  142. -- Goto (sub-record-check,rg,1)
  143. -- Executing [rg@sub-record-check:1] GosubIf("SIP/vitel-inbound-00000031", "0?record,1(rg,dontcare,8014100203)") in new stack
  144. -- Executing [rg@sub-record-check:2] Return("SIP/vitel-inbound-00000031", "") in new stack
  145. -- Executing [602@ext-group:10] Set("SIP/vitel-inbound-00000031", "RingGroupMethod=ringall") in new stack
  146. -- Executing [602@ext-group:11] GotoIf("SIP/vitel-inbound-00000031", "0?DIALGRP") in new stack
  147. -- Executing [602@ext-group:12] Answer("SIP/vitel-inbound-00000031", "") in new stack
  148. -- Executing [602@ext-group:13] Wait("SIP/vitel-inbound-00000031", "1") in new stack
  149. -- Executing [602@ext-group:14] Playback("SIP/vitel-inbound-00000031", "custom/plzwait") in new stack
  150. -- <SIP/vitel-inbound-00000031> Playing 'custom/plzwait.slin' (language 'en')
  151. -- Executing [602@ext-group:15] Macro("SIP/vitel-inbound-00000031", "dial,20,m(moh)t,101-8012096684#") in new stack
  152. -- Executing [s@macro-dial:1] GotoIf("SIP/vitel-inbound-00000031", "0?dial") in new stack
  153. -- Executing [s@macro-dial:2] SetMusicOnHold("SIP/vitel-inbound-00000031", "moh") in new stack
  154. -- Executing [s@macro-dial:3] AGI("SIP/vitel-inbound-00000031", "dialparties.agi") in new stack
  155. -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  156. dialparties.agi: Starting New Dialparties.agi
  157. dialparties.agi: Caller ID name is 'SALT LAKE, UT' number is '8014100203'
  158. > dialparties.agi: USE_CONFIRMATION: 'FALSE'
  159. > dialparties.agi: RINGGROUP_INDEX: ''
  160. dialparties.agi: Methodology of ring is 'ringall'
  161. -- dialparties.agi: Added extension 101 to extension map
  162. -- dialparties.agi: Added extension 8012096684# to extension map
  163. -- dialparties.agi: Extension 101 cf is disabled
  164. -- dialparties.agi: Extension 8012096684# cf is disabled
  165. -- dialparties.agi: Extension 101 do not disturb is disabled
  166. > dialparties.agi: extnum 101 has: cw: 1; hascfb: 0 [] hascfu: 0 []
  167. -- dialparties.agi: dbset CALLTRACE/101 to 8014100203
  168. > dialparties.agi: extnum 8012096684# has: cw: 0; hascfb: 0 [] hascfu: 0 []
  169. > dialparties.agi: Built External dialstring component for 8012096684: Local/8012096684@from-internal/n
  170. -- dialparties.agi: Filtered ARG3: 101-8012096684
  171. > dialparties.agi: NODEST: 602 adding M(auto-blkvm) to dialopts: m(moh)tM(auto-blkvm)
  172. > dialparties.agi: NODEST: 602 blkvm enabled macro already in dialopts: m(moh)tM(auto-blkvm)
  173. -- <SIP/vitel-inbound-00000031>AGI Script dialparties.agi completed, returning 0
  174. -- Executing [s@macro-dial:7] Dial("SIP/vitel-inbound-00000031", "SIP/101&Local/8012096684@from-internal/n,20,m(moh)tM(auto-blkvm)") in new stack
  175. == Using SIP RTP TOS bits 184
  176. == Using SIP RTP CoS mark 5
  177. Audio is at 5060
  178. Adding codec 0x4 (ulaw) to SDP
  179. Adding codec 0x8 (alaw) to SDP
  180. Adding codec 0x2 (gsm) to SDP
  181. Adding non-codec 0x1 (telephone-event) to SDP
  182. Reliably Transmitting (no NAT) to 10.0.0.21:5060:
  183. INVITE sip:101@10.0.0.21:5060 SIP/2.0
  184. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK36e91778
  185. Max-Forwards: 70
  186. From: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
  187. To: <sip:101@10.0.0.21:5060>
  188. Contact: <sip:8014100203@24.2.66.2:5060>
  189. Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
  190. CSeq: 102 INVITE
  191. User-Agent: FPBX-2.10.0(1.8.7.1)
  192. Date: Thu, 24 May 2012 10:45:20 GMT
  193. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  194. Supported: replaces, timer
  195. Content-Type: application/sdp
  196. Content-Length: 277
  197.  
  198. v=0
  199. o=root 1090802124 1090802124 IN IP4 24.2.66.2
  200. s=Asterisk PBX 1.8.7.1
  201. c=IN IP4 24.2.66.2
  202. t=0 0
  203. m=audio 18128 RTP/AVP 0 8 3 101
  204. a=rtpmap:0 PCMU/8000
  205. a=rtpmap:8 PCMA/8000
  206. a=rtpmap:3 GSM/8000
  207. a=rtpmap:101 telephone-event/8000
  208. a=fmtp:101 0-16
  209. a=ptime:20
  210. a=sendrecv
  211.  
  212. ---
  213. -- Called SIP/101
  214. -- Called Local/8012096684@from-internal/n
  215. -- Started music on hold, class 'default', on SIP/vitel-inbound-00000031
  216. -- SIP/101-00000032 connected line has changed. Saving it until answer for SIP/vitel-inbound-00000031
  217. -- Executing [8012096684@from-internal:1] Macro("Local/8012096684@from-internal-6479;2", "user-callerid,LIMIT,") in new stack
  218. -- Executing [s@macro-user-callerid:1] Set("Local/8012096684@from-internal-6479;2", "AMPUSER=8014100203") in new stack
  219. -- Executing [s@macro-user-callerid:2] GotoIf("Local/8012096684@from-internal-6479;2", "0?report") in new stack
  220. -- Executing [s@macro-user-callerid:3] ExecIf("Local/8012096684@from-internal-6479;2", "1?Set(REALCALLERIDNUM=8014100203)") in new stack
  221. -- Executing [s@macro-user-callerid:4] Set("Local/8012096684@from-internal-6479;2", "AMPUSER=") in new stack
  222. -- Executing [s@macro-user-callerid:5] Set("Local/8012096684@from-internal-6479;2", "AMPUSERCIDNAME=") in new stack
  223. -- Executing [s@macro-user-callerid:6] GotoIf("Local/8012096684@from-internal-6479;2", "1?report") in new stack
  224. -- Goto (macro-user-callerid,s,13)
  225. -- Executing [s@macro-user-callerid:13] GotoIf("Local/8012096684@from-internal-6479;2", "1?continue") in new stack
  226. -- Goto (macro-user-callerid,s,26)
  227. -- Executing [s@macro-user-callerid:26] Set("Local/8012096684@from-internal-6479;2", "CALLERID(number)=8014100203") in new stack
  228. -- Executing [s@macro-user-callerid:27] Set("Local/8012096684@from-internal-6479;2", "CALLERID(name)=SALT LAKE, UT") in new stack
  229. -- Executing [s@macro-user-callerid:28] Set("Local/8012096684@from-internal-6479;2", "CHANNEL(language)=en") in new stack
  230. -- Executing [8012096684@from-internal:2] Set("Local/8012096684@from-internal-6479;2", "MOHCLASS=moh") in new stack
  231. -- Executing [8012096684@from-internal:3] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(TRUNKCIDOVERRIDE=<8014100203>)") in new stack
  232. -- Executing [8012096684@from-internal:4] Set("Local/8012096684@from-internal-6479;2", "_NODEST=") in new stack
  233. -- Executing [8012096684@from-internal:5] Gosub("Local/8012096684@from-internal-6479;2", "sub-record-check,s,1(out,8012096684,)") in new stack
  234. -- Executing [s@sub-record-check:1] GotoIf("Local/8012096684@from-internal-6479;2", "1?check") in new stack
  235. -- Goto (sub-record-check,s,6)
  236. -- Executing [s@sub-record-check:6] Set("Local/8012096684@from-internal-6479;2", "__MON_FMT=wav") in new stack
  237. -- Executing [s@sub-record-check:7] GotoIf("Local/8012096684@from-internal-6479;2", "1?next") in new stack
  238. -- Goto (sub-record-check,s,10)
  239. -- Executing [s@sub-record-check:10] ExecIf("Local/8012096684@from-internal-6479;2", "0?Return()") in new stack
  240. -- Executing [s@sub-record-check:11] GotoIf("Local/8012096684@from-internal-6479;2", "1?out,1") in new stack
  241. -- Goto (sub-record-check,out,1)
  242. -- Executing [out@sub-record-check:1] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(__REC_POLICY_MODE=)") in new stack
  243. -- Executing [out@sub-record-check:2] GosubIf("Local/8012096684@from-internal-6479;2", "0?record,1(exten,8012096684,8014100203)") in new stack
  244. -- Executing [out@sub-record-check:3] Return("Local/8012096684@from-internal-6479;2", "") in new stack
  245. -- Executing [8012096684@from-internal:6] Macro("Local/8012096684@from-internal-6479;2", "dialout-trunk,2,18012096684,") in new stack
  246. -- Executing [s@macro-dialout-trunk:1] Set("Local/8012096684@from-internal-6479;2", "DIAL_TRUNK=2") in new stack
  247. -- Executing [s@macro-dialout-trunk:2] GosubIf("Local/8012096684@from-internal-6479;2", "0?sub-pincheck,s,1()") in new stack
  248. -- Executing [s@macro-dialout-trunk:3] GotoIf("Local/8012096684@from-internal-6479;2", "0?disabletrunk,1") in new stack
  249. -- Executing [s@macro-dialout-trunk:4] Set("Local/8012096684@from-internal-6479;2", "DIAL_NUMBER=18012096684") in new stack
  250. -- Executing [s@macro-dialout-trunk:5] Set("Local/8012096684@from-internal-6479;2", "DIAL_TRUNK_OPTIONS=tr") in new stack
  251.  
  252. <--- SIP read from UDP:10.0.0.21:5060 --->
  253. SIP/2.0 100 Trying
  254. To: <sip:101@10.0.0.21:5060>
  255. From: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
  256. Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
  257. CSeq: 102 INVITE
  258. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK36e91778
  259. Server: Linksys/PAP2T-5.1.6(LS)
  260. Content-Length: 0
  261.  
  262. <------------->
  263. --- (8 headers 0 lines) ---
  264. -- Executing [s@macro-dialout-trunk:6] Set("Local/8012096684@from-internal-6479;2", "OUTBOUND_GROUP=OUT_2") in new stack
  265. -- Executing [s@macro-dialout-trunk:7] GotoIf("Local/8012096684@from-internal-6479;2", "0?nomax") in new stack
  266. -- Executing [s@macro-dialout-trunk:8] GotoIf("Local/8012096684@from-internal-6479;2", "0?chanfull") in new stack
  267. -- Executing [s@macro-dialout-trunk:9] GotoIf("Local/8012096684@from-internal-6479;2", "0?skipoutcid") in new stack
  268. -- Executing [s@macro-dialout-trunk:10] Set("Local/8012096684@from-internal-6479;2", "DIAL_TRUNK_OPTIONS=") in new stack
  269. -- Executing [s@macro-dialout-trunk:11] Macro("Local/8012096684@from-internal-6479;2", "outbound-callerid,2") in new stack
  270. -- Executing [s@macro-outbound-callerid:1] ExecIf("Local/8012096684@from-internal-6479;2", "1?Set(CALLERPRES()=allowed_not_screened)") in new stack
  271. -- Executing [s@macro-outbound-callerid:2] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(REALCALLERIDNUM=8014100203)") in new stack
  272. -- Executing [s@macro-outbound-callerid:3] GotoIf("Local/8012096684@from-internal-6479;2", "0?normcid") in new stack
  273. -- Executing [s@macro-outbound-callerid:4] Set("Local/8012096684@from-internal-6479;2", "USEROUTCID=8014100203") in new stack
  274. -- Executing [s@macro-outbound-callerid:5] GotoIf("Local/8012096684@from-internal-6479;2", "1?bypass") in new stack
  275. -- Goto (macro-outbound-callerid,s,7)
  276. -- Executing [s@macro-outbound-callerid:7] Set("Local/8012096684@from-internal-6479;2", "EMERGENCYCID=") in new stack
  277. -- Executing [s@macro-outbound-callerid:8] Set("Local/8012096684@from-internal-6479;2", "TRUNKOUTCID=<8017478048>") in new stack
  278. -- Executing [s@macro-outbound-callerid:9] GotoIf("Local/8012096684@from-internal-6479;2", "1?trunkcid") in new stack
  279. -- Goto (macro-outbound-callerid,s,12)
  280. -- Executing [s@macro-outbound-callerid:12] ExecIf("Local/8012096684@from-internal-6479;2", "1?Set(CALLERID(all)=<8017478048>)") in new stack
  281. -- Executing [s@macro-outbound-callerid:13] ExecIf("Local/8012096684@from-internal-6479;2", "1?Set(CALLERID(all)=8014100203)") in new stack
  282. -- Executing [s@macro-outbound-callerid:14] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(CALLERID(all)=)") in new stack
  283. -- Executing [s@macro-outbound-callerid:15] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
  284. -- Executing [s@macro-dialout-trunk:12] GosubIf("Local/8012096684@from-internal-6479;2", "0?sub-flp-2,s,1()") in new stack
  285. -- Executing [s@macro-dialout-trunk:13] Set("Local/8012096684@from-internal-6479;2", "OUTNUM=18012096684") in new stack
  286. -- Executing [s@macro-dialout-trunk:14] Set("Local/8012096684@from-internal-6479;2", "custom=SIP/vitel-outbound") in new stack
  287. -- Executing [s@macro-dialout-trunk:15] ExecIf("Local/8012096684@from-internal-6479;2", "1?Set(DIAL_TRUNK_OPTIONS=M(setmusic^moh))") in new stack
  288. -- Executing [s@macro-dialout-trunk:16] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^moh)M(confirm))") in new stack
  289. -- Executing [s@macro-dialout-trunk:17] Macro("Local/8012096684@from-internal-6479;2", "dialout-trunk-predial-hook,") in new stack
  290. -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("Local/8012096684@from-internal-6479;2", "") in new stack
  291. -- Executing [s@macro-dialout-trunk:18] GotoIf("Local/8012096684@from-internal-6479;2", "0?bypass,1") in new stack
  292. -- Executing [s@macro-dialout-trunk:19] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(CONNECTEDLINE(num,i)=18012096684)") in new stack
  293. -- Executing [s@macro-dialout-trunk:20] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(CONNECTEDLINE(name,i)=CID:8014100203)") in new stack
  294. -- Executing [s@macro-dialout-trunk:21] GotoIf("Local/8012096684@from-internal-6479;2", "0?customtrunk") in new stack
  295. -- Executing [s@macro-dialout-trunk:22] Dial("Local/8012096684@from-internal-6479;2", "SIP/vitel-outbound/18012096684,300,M(setmusic^moh)") in new stack
  296. == Using SIP RTP TOS bits 184
  297. == Using SIP RTP CoS mark 5
  298.  
  299. <--- SIP read from UDP:10.0.0.21:5060 --->
  300. SIP/2.0 180 Ringing
  301. To: <sip:101@10.0.0.21:5060>;tag=c95afd287417c8d0i0
  302. From: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
  303. Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
  304. CSeq: 102 INVITE
  305. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK36e91778
  306. Server: Linksys/PAP2T-5.1.6(LS)
  307. Content-Length: 0
  308.  
  309. <------------->
  310. --- (8 headers 0 lines) ---
  311. -- SIP/101-00000032 is ringing
  312. Audio is at 5060
  313. Adding codec 0x4 (ulaw) to SDP
  314. Adding codec 0x8 (alaw) to SDP
  315. Adding codec 0x2 (gsm) to SDP
  316. Adding non-codec 0x1 (telephone-event) to SDP
  317. Reliably Transmitting (NAT) to 64.2.142.17:5060:
  318. INVITE sip:18012096684@outbound.vitelity.net SIP/2.0
  319. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK0aaf31c1;rport
  320. Max-Forwards: 70
  321. From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
  322. To: <sip:18012096684@outbound.vitelity.net>
  323. Contact: <sip:ccwt_pbx@24.2.66.2:5060>
  324. Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
  325. CSeq: 102 INVITE
  326. User-Agent: FPBX-2.10.0(1.8.7.1)
  327. Date: Thu, 24 May 2012 10:45:20 GMT
  328. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  329. Supported: replaces, timer
  330. Remote-Party-ID: "8014100203" <sip:8014100203@24.2.66.2>;party=calling;privacy=off;screen=no
  331. Content-Type: application/sdp
  332. Content-Length: 275
  333.  
  334. v=0
  335. o=root 740845620 740845620 IN IP4 24.2.66.2
  336. s=Asterisk PBX 1.8.7.1
  337. c=IN IP4 24.2.66.2
  338. t=0 0
  339. m=audio 19396 RTP/AVP 0 8 3 101
  340. a=rtpmap:0 PCMU/8000
  341. a=rtpmap:8 PCMA/8000
  342. a=rtpmap:3 GSM/8000
  343. a=rtpmap:101 telephone-event/8000
  344. a=fmtp:101 0-16
  345. a=ptime:20
  346. a=sendrecv
  347.  
  348. ---
  349. -- Called SIP/vitel-outbound/18012096684
  350.  
  351. <--- SIP read from UDP:64.2.142.17:5060 --->
  352. SIP/2.0 407 Proxy Authentication Required
  353. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK0aaf31c1;received=24.2.66.2;rport=5060
  354. From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
  355. To: <sip:18012096684@outbound.vitelity.net>;tag=as0c51706e
  356. Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
  357. CSeq: 102 INVITE
  358. User-Agent: Asterisk PBX
  359. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  360. Supported: replaces
  361. Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4b0c3da9"
  362. Content-Length: 0
  363.  
  364. <------------->
  365. --- (11 headers 0 lines) ---
  366. Transmitting (NAT) to 64.2.142.17:5060:
  367. ACK sip:18012096684@outbound.vitelity.net SIP/2.0
  368. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK0aaf31c1;rport
  369. Max-Forwards: 70
  370. From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
  371. To: <sip:18012096684@outbound.vitelity.net>;tag=as0c51706e
  372. Contact: <sip:ccwt_pbx@24.2.66.2:5060>
  373. Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
  374. CSeq: 102 ACK
  375. User-Agent: FPBX-2.10.0(1.8.7.1)
  376. Content-Length: 0
  377.  
  378.  
  379. ---
  380. Audio is at 5060
  381. Adding codec 0x4 (ulaw) to SDP
  382. Adding codec 0x8 (alaw) to SDP
  383. Adding codec 0x2 (gsm) to SDP
  384. Adding non-codec 0x1 (telephone-event) to SDP
  385. Reliably Transmitting (NAT) to 64.2.142.17:5060:
  386. INVITE sip:18012096684@outbound.vitelity.net SIP/2.0
  387. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;rport
  388. Max-Forwards: 70
  389. From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
  390. To: <sip:18012096684@outbound.vitelity.net>
  391. Contact: <sip:ccwt_pbx@24.2.66.2:5060>
  392. Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
  393. CSeq: 103 INVITE
  394. User-Agent: FPBX-2.10.0(1.8.7.1)
  395. Proxy-Authorization: Digest username="ccwt_pbx", realm="asterisk", algorithm=MD5, uri="sip:18012096684@outbound.vitelity.net", nonce="4b0c3da9", response="2f75855df13197b2a86df0821110ed2a"
  396. Date: Thu, 24 May 2012 10:45:20 GMT
  397. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  398. Supported: replaces, timer
  399. Remote-Party-ID: "8014100203" <sip:8014100203@24.2.66.2>;party=calling;privacy=off;screen=no
  400. Content-Type: application/sdp
  401. Content-Length: 275
  402.  
  403. v=0
  404. o=root 740845620 740845621 IN IP4 24.2.66.2
  405. s=Asterisk PBX 1.8.7.1
  406. c=IN IP4 24.2.66.2
  407. t=0 0
  408. m=audio 19396 RTP/AVP 0 8 3 101
  409. a=rtpmap:0 PCMU/8000
  410. a=rtpmap:8 PCMA/8000
  411. a=rtpmap:3 GSM/8000
  412. a=rtpmap:101 telephone-event/8000
  413. a=fmtp:101 0-16
  414. a=ptime:20
  415. a=sendrecv
  416.  
  417. ---
  418.  
  419. <--- SIP read from UDP:64.2.142.17:5060 --->
  420. SIP/2.0 100 Trying
  421. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;received=24.2.66.2;rport=5060
  422. From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
  423. To: <sip:18012096684@outbound.vitelity.net>
  424. Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
  425. CSeq: 103 INVITE
  426. User-Agent: Asterisk PBX
  427. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  428. Supported: replaces
  429. Contact: <sip:18012096684@64.2.142.17>
  430. Content-Length: 0
  431.  
  432. <------------->
  433. --- (11 headers 0 lines) ---
  434.  
  435. <--- SIP read from UDP:64.2.142.17:5060 --->
  436. SIP/2.0 183 Session Progress
  437. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;received=24.2.66.2;rport=5060
  438. From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
  439. To: <sip:18012096684@outbound.vitelity.net>;tag=as43c26d7d
  440. Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
  441. CSeq: 103 INVITE
  442. User-Agent: Asterisk PBX
  443. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  444. Supported: replaces
  445. Contact: <sip:18012096684@64.2.142.17>
  446. ontent-Type: application/sdp
  447. Content-Length: 285
  448.  
  449. v=0
  450. o=root 15443 15443 IN IP4 64.2.142.17
  451. s=session
  452. c=IN IP4 64.2.142.17
  453. t=0 0
  454. m=audio 15014 RTP/AVP 0 8 3 101
  455. a=rtpmap:0 PCMU/8000
  456. a=rtpmap:8 PCMA/8000
  457. a=rtpmap:3 GSM/8000
  458. a=rtpmap:101 telephone-event/8000
  459. a=fmtp:101 0-16
  460. a=silenceSupp:off - - - -
  461. a=ptime:20
  462. a=sendrecv
  463. <------------->
  464. --- (12 headers 14 lines) ---
  465. Found RTP audio format 0
  466. Found RTP audio format 8
  467. Found RTP audio format 3
  468. Found RTP audio format 101
  469. Found audio description format PCMU for ID 0
  470. Found audio description format PCMA for ID 8
  471. Found audio description format GSM for ID 3
  472. Found audio description format telephone-event for ID 101
  473. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
  474. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  475. Peer audio RTP is at port 64.2.142.17:15014
  476. -- SIP/vitel-outbound-00000033 is making progress passing it to Local/8012096684@from-internal-6479;2
  477. -- Local/8012096684@from-internal-6479;1 is making progress passing it to SIP/vitel-inbound-00000031
  478.  
  479. <--- SIP read from UDP:10.0.0.21:5060 --->
  480. SIP/2.0 200 OK
  481. To: <sip:101@10.0.0.21:5060>;tag=c95afd287417c8d0i0
  482. From: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
  483. Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
  484. CSeq: 102 INVITE
  485. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK36e91778
  486. Contact: Allen <sip:101@10.0.0.21:5060>
  487. Server: Linksys/PAP2T-5.1.6(LS)
  488. Content-Length: 251
  489. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  490. Supported: x-sipura, replaces
  491. Content-Type: application/sdp
  492.  
  493. v=0
  494. o=- 36286809 36286809 IN IP4 10.0.0.21
  495. s=-
  496. c=IN IP4 10.0.0.21
  497. t=0 0
  498. m=audio 16408 RTP/AVP 0 100 101
  499. a=rtpmap:0 PCMU/8000
  500. a=rtpmap:100 NSE/8000
  501. a=fmtp:100 192-193
  502. a=rtpmap:101 telephone-event/8000
  503. a=fmtp:101 0-15
  504. a=ptime:10
  505. a=sendrecv
  506. <------------->
  507. --- (12 headers 13 lines) ---
  508. Found RTP audio format 0
  509. Found RTP audio format 100
  510. Found RTP audio format 101
  511. Found audio description format PCMU for ID 0
  512. Found unknown media description format NSE for ID 100
  513. Found audio description format telephone-event for ID 101
  514. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  515. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  516. Peer audio RTP is at port 10.0.0.21:16408
  517. list_route: hop: <sip:101@10.0.0.21:5060>
  518. set_destination: Parsing <sip:101@10.0.0.21:5060> for address/port to send to
  519. set_destination: set destination to 10.0.0.21:5060
  520. Transmitting (no NAT) to 10.0.0.21:5060:
  521. ACK sip:101@10.0.0.21:5060 SIP/2.0
  522. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK1312b086
  523. Max-Forwards: 70
  524. From: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
  525. To: <sip:101@10.0.0.21:5060>;tag=c95afd287417c8d0i0
  526. Contact: <sip:8014100203@24.2.66.2:5060>
  527. Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
  528. CSeq: 102 ACK
  529. User-Agent: FPBX-2.10.0(1.8.7.1)
  530. Content-Length: 0
  531.  
  532.  
  533. ---
  534. -- SIP/101-00000032 connected line has changed. Saving it until answer for SIP/vitel-inbound-00000031
  535. -- SIP/101-00000032 answered SIP/vitel-inbound-00000031
  536. Scheduling destruction of SIP dialog '73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060' in 32000 ms (Method: INVITE)
  537. Reliably Transmitting (NAT) to 64.2.142.17:5060:
  538. CANCEL sip:18012096684@outbound.vitelity.net SIP/2.0
  539. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;rport
  540. Max-Forwards: 70
  541. From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
  542. To: <sip:18012096684@outbound.vitelity.net>
  543. Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
  544. CSeq: 103 CANCEL
  545. User-Agent: FPBX-2.10.0(1.8.7.1)
  546. Content-Length: 0
  547.  
  548.  
  549. ---
  550. Scheduling destruction of SIP dialog '73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060' in 32000 ms (Method: INVITE)
  551. == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'Local/8012096684@from-internal-6479;2' in macro 'dialout-trunk'
  552. == Spawn extension (from-internal, 8012096684, 6) exited non-zero on 'Local/8012096684@from-internal-6479;2'
  553. -- Executing [s@macro-auto-blkvm:1] Set("SIP/101-00000032", "__MACRO_RESULT=") in new stack
  554. -- Executing [s@macro-auto-blkvm:2] Macro("SIP/101-00000032", "blkvm-clr,") in new stack
  555. -- Executing [h@from-internal:1] Hangup("Local/8012096684@from-internal-6479;2", "") in new stack
  556. == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/8012096684@from-internal-6479;2'
  557. -- Executing [s@macro-blkvm-clr:1] Set("SIP/101-00000032", "SHARED(BLKVM,SIP/vitel-inbound-00000031)=") in new stack
  558. -- Executing [s@macro-blkvm-clr:2] Set("SIP/101-00000032", "GOSUB_RETVAL=") in new stack
  559. -- Executing [s@macro-blkvm-clr:3] MacroExit("SIP/101-00000032", "") in new stack
  560. -- Executing [s@macro-auto-blkvm:3] ExecIf("SIP/101-00000032", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=101)") in new stack
  561. -- Executing [s@macro-auto-blkvm:4] ExecIf("SIP/101-00000032", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Allen)") in new stack
  562. -- Stopped music on hold on SIP/vitel-inbound-00000031
  563.  
  564. <--- SIP read from UDP:64.2.142.17:5060 --->
  565. SIP/2.0 487 Request Terminated
  566. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;received=24.2.66.2;rport=5060
  567. From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
  568. To: <sip:18012096684@outbound.vitelity.net>;tag=as43c26d7d
  569. Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
  570. CSeq: 103 INVITE
  571. User-Agent: Asterisk PBX
  572. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  573. Supported: replaces
  574. Content-Length: 0
  575.  
  576. <------------->
  577. --- (10 headers 0 lines) ---
  578. Transmitting (NAT) to 64.2.142.17:5060:
  579. ACK sip:18012096684@outbound.vitelity.net SIP/2.0
  580. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;rport
  581. Max-Forwards: 70
  582. From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
  583. To: <sip:18012096684@outbound.vitelity.net>;tag=as43c26d7d
  584. Contact: <sip:ccwt_pbx@24.2.66.2:5060>
  585. Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
  586. CSeq: 103 ACK
  587. User-Agent: FPBX-2.10.0(1.8.7.1)
  588. Content-Length: 0
  589.  
  590.  
  591. ---
  592.  
  593. <--- SIP read from UDP:64.2.142.17:5060 --->
  594. SIP/2.0 200 OK
  595. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;received=24.2.66.2;rport=5060
  596. From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
  597. To: <sip:18012096684@outbound.vitelity.net>;tag=as43c26d7d
  598. Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
  599. CSeq: 103 CANCEL
  600. User-Agent: Asterisk PBX
  601. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  602. Supported: replaces
  603. Content-Length: 0
  604.  
  605. <------------->
  606. --- (10 headers 0 lines) ---
  607. Really destroying SIP dialog '73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060' Method: INVITE
  608. Really destroying SIP dialog '3691bc792f8f478d2df520f95421c67c@127.0.0.1' Method: REGISTER
  609. Reliably Transmitting (no NAT) to 10.0.0.21:5060:
  610. OPTIONS sip:101@10.0.0.21:5060 SIP/2.0
  611. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK5533299c
  612. Max-Forwards: 70
  613. From: "Unknown" <sip:Unknown@24.2.66.2>;tag=as7c2b8c12
  614. To: <sip:101@10.0.0.21:5060>
  615. Contact: <sip:Unknown@24.2.66.2:5060>
  616. Call-ID: 19616e8a5d7cc8e208172ecf5eb618ea@24.2.66.2:5060
  617. CSeq: 102 OPTIONS
  618. User-Agent: FPBX-2.10.0(1.8.7.1)
  619. Date: Thu, 24 May 2012 10:45:42 GMT
  620. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  621. Supported: replaces, timer
  622. Content-Length: 0
  623.  
  624.  
  625. ---
  626.  
  627. <--- SIP read from UDP:10.0.0.21:5060 --->
  628. SIP/2.0 200 OK
  629. To: <sip:101@10.0.0.21:5060>;tag=59981087efc9843ei0
  630. From: "Unknown" <sip:Unknown@24.2.66.2>;tag=as7c2b8c12
  631. Call-ID: 19616e8a5d7cc8e208172ecf5eb618ea@24.2.66.2:5060
  632. CSeq: 102 OPTIONS
  633. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK5533299c
  634. Server: Linksys/PAP2T-5.1.6(LS)
  635. Content-Length: 0
  636. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  637. Supported: x-sipura, replaces
  638.  
  639. <------------->
  640. --- (10 headers 0 lines) ---
  641. Really destroying SIP dialog '19616e8a5d7cc8e208172ecf5eb618ea@24.2.66.2:5060' Method: OPTIONS
  642.  
  643. <--- SIP read from UDP:10.0.0.21:5060 --->
  644. BYE sip:8014100203@24.2.66.2:5060 SIP/2.0
  645. Via: SIP/2.0/UDP 10.0.0.21:5060;branch=z9hG4bK-f1c9fda4
  646. From: <sip:101@10.0.0.21:5060>;tag=c95afd287417c8d0i0
  647. To: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
  648. Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
  649. CSeq: 101 BYE
  650. Max-Forwards: 70
  651. User-Agent: Linksys/PAP2T-5.1.6(LS)
  652. Content-Length: 0
  653.  
  654. <------------->
  655. --- (9 headers 0 lines) ---
  656. Sending to 10.0.0.21:5060 (no NAT)
  657. Scheduling destruction of SIP dialog '434acf061c5563f74228d2da0d655276@24.2.66.2:5060' in 6400 ms (Method: BYE)
  658.  
  659. <--- Transmitting (no NAT) to 10.0.0.21:5060 --->
  660. SIP/2.0 200 OK
  661. Via: SIP/2.0/UDP 10.0.0.21:5060;branch=z9hG4bK-f1c9fda4;received=10.0.0.21
  662. From: <sip:101@10.0.0.21:5060>;tag=c95afd287417c8d0i0
  663. To: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
  664. Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
  665. CSeq: 101 BYE
  666. Server: FPBX-2.10.0(1.8.7.1)
  667. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  668. Supported: replaces, timer
  669. Content-Length: 0
  670.  
  671.  
  672. <------------>
  673. -- Executing [h@macro-dial:1] Macro("SIP/vitel-inbound-00000031", "hangupcall") in new stack
  674. -- Executing [s@macro-hangupcall:1] GotoIf("SIP/vitel-inbound-00000031", "1?theend") in new stack
  675. -- Goto (macro-hangupcall,s,3)
  676. -- Executing [s@macro-hangupcall:3] Hangup("SIP/vitel-inbound-00000031", "") in new stack
  677. == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/vitel-inbound-00000031' in macro 'hangupcall'
  678. == Spawn extension (macro-dial, h, 1) exited non-zero on 'SIP/vitel-inbound-00000031'
  679. == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/vitel-inbound-00000031' in macro 'dial'
  680. == Spawn extension (ext-group, 602, 15) exited non-zero on 'SIP/vitel-inbound-00000031'
  681. Scheduling destruction of SIP dialog '7c1454a27f4333035455d68e0f484ae0@66.241.96.221' in 32000 ms (Method: ACK)
  682. set_destination: Parsing <sip:8014100203@66.241.96.221> for address/port to send to
  683. set_destination: set destination to 66.241.96.221:5060
  684. Reliably Transmitting (NAT) to 66.241.96.221:5060:
  685. BYE sip:8014100203@66.241.96.221 SIP/2.0
  686. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK77471b8a;rport
  687. Max-Forwards: 70
  688. From: <sip:8017478048@24.2.66.2:5060>;tag=as34b2bc52
  689. To: "8014100203" <sip:8014100203@66.241.96.221>;tag=as6056c176
  690. Call-ID: 7c1454a27f4333035455d68e0f484ae0@66.241.96.221
  691. CSeq: 102 BYE
  692. User-Agent: FPBX-2.10.0(1.8.7.1)
  693. X-Asterisk-HangupCause: Normal Clearing
  694. X-Asterisk-HangupCauseCode: 16
  695. Content-Length: 0
  696.  
  697.  
  698. ---
  699.  
  700. <--- SIP read from UDP:66.241.96.221:5060 --->
  701. SIP/2.0 200 OK
  702. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK77471b8a;received=24.2.66.2;rport=5060
  703. From: <sip:8017478048@24.2.66.2:5060>;tag=as34b2bc52
  704. To: "8014100203" <sip:8014100203@66.241.96.221>;tag=as6056c176
  705. Call-ID: 7c1454a27f4333035455d68e0f484ae0@66.241.96.221
  706. CSeq: 102 BYE
  707. User-Agent: packetrino
  708. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  709. Supported: replaces
  710. Content-Length: 0
  711.  
  712. <------------->
  713. --- (10 headers 0 lines) ---
  714. SIP Response message for INCOMING dialog BYE arrived
  715. Really destroying SIP dialog '7c1454a27f4333035455d68e0f484ae0@66.241.96.221' Method: ACK
  716. > doing dnsmgr_lookup for 'inbound27.vitelity.net'
  717. REGISTER 11 headers, 0 lines
  718. Reliably Transmitting (NAT) to 66.241.96.221:5060:
  719. REGISTER sip:inbound27.vitelity.net SIP/2.0
  720. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK69da1905;rport
  721. Max-Forwards: 70
  722. From: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as43d1760d
  723. To: <sip:ccwt_pbx@inbound27.vitelity.net>
  724. Call-ID: 3691bc792f8f478d2df520f95421c67c@127.0.0.1
  725. CSeq: 1743 REGISTER
  726. User-Agent: FPBX-2.10.0(1.8.7.1)
  727. Authorization: Digest username="ccwt_pbx", realm="asterisk", algorithm=MD5, uri="sip:inbound27.vitelity.net", nonce="2148295e", response="b0a92dc82d45d575f4744e2979c20b85"
  728. Expires: 120
  729. Contact: <sip:s@24.2.66.2:5060>
  730. Content-Length: 0
  731.  
  732.  
  733. ---
  734.  
  735. <--- SIP read from UDP:66.241.96.221:5060 --->
  736. SIP/2.0 100 Trying
  737. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK69da1905;received=24.2.66.2;rport=5060
  738. From: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as43d1760d
  739. To: <sip:ccwt_pbx@inbound27.vitelity.net>
  740. Call-ID: 3691bc792f8f478d2df520f95421c67c@127.0.0.1
  741. CSeq: 1743 REGISTER
  742. User-Agent: packetrino
  743. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  744. Supported: replaces
  745. Content-Length: 0
  746.  
  747. <------------->
  748. --- (10 headers 0 lines) ---
  749.  
  750. <--- SIP read from UDP:66.241.96.221:5060 --->
  751. SIP/2.0 401 Unauthorized
  752. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK69da1905;received=24.2.66.2;rport=5060
  753. From: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as43d1760d
  754. To: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as1ac189f0
  755. Call-ID: 3691bc792f8f478d2df520f95421c67c@127.0.0.1
  756. CSeq: 1743 REGISTER
  757. User-Agent: packetrino
  758. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  759. Supported: replaces
  760. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66e9c3b7"
  761. Content-Length: 0
  762.  
  763. <------------->
  764. --- (11 headers 0 lines) ---
  765. Responding to challenge, registration to domain/host name inbound27.vitelity.net
  766. > doing dnsmgr_lookup for 'inbound27.vitelity.net'
  767. REGISTER 11 headers, 0 lines
  768. Reliably Transmitting (NAT) to 66.241.96.221:5060:
  769. REGISTER sip:inbound27.vitelity.net SIP/2.0
  770. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK0d572584;rport
  771. Max-Forwards: 70
  772. From: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as2d999541
  773. To: <sip:ccwt_pbx@inbound27.vitelity.net>
  774. Call-ID: 3691bc792f8f478d2df520f95421c67c@127.0.0.1
  775. CSeq: 1744 REGISTER
  776. User-Agent: FPBX-2.10.0(1.8.7.1)
  777. Authorization: Digest username="ccwt_pbx", realm="asterisk", algorithm=MD5, uri="sip:inbound27.vitelity.net", nonce="66e9c3b7", response="bc17716d8d6afa2621f68b7ede110660"
  778. Expires: 120
  779. Contact: <sip:s@24.2.66.2:5060>
  780. Content-Length: 0
  781.  
  782.  
  783. ---
  784.  
  785. <--- SIP read from UDP:66.241.96.221:5060 --->
  786. SIP/2.0 100 Trying
  787. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK0d572584;received=24.2.66.2;rport=5060
  788. From: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as2d999541
  789. To: <sip:ccwt_pbx@inbound27.vitelity.net>
  790. Call-ID: 3691bc792f8f478d2df520f95421c67c@127.0.0.1
  791. CSeq: 1744 REGISTER
  792. User-Agent: packetrino
  793. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  794. Supported: replaces
  795. Content-Length: 0
  796.  
  797. <------------->
  798. --- (10 headers 0 lines) ---
  799.  
  800. <--- SIP read from UDP:66.241.96.221:5060 --->
  801. SIP/2.0 200 OK
  802. Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK0d572584;received=24.2.66.2;rport=5060
  803. From: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as2d999541
  804. To: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as1ac189f0
  805. Call-ID: 3691bc792f8f478d2df520f95421c67c@127.0.0.1
  806. CSeq: 1744 REGISTER
  807. User-Agent: packetrino
  808. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  809. Supported: replaces
  810. Expires: 60
  811. Contact: <sip:s@24.2.66.2:5060>;expires=60
  812. Date: Thu, 24 May 2012 10:45:46 GMT
  813. Content-Length: 0
  814.  
  815. <------------->
  816. --- (13 headers 0 lines) ---
  817. Scheduling destruction of SIP dialog '3691bc792f8f478d2df520f95421c67c@127.0.0.1' in 32000 ms (Method: REGISTER)
  818. Really destroying SIP dialog '434acf061c5563f74228d2da0d655276@24.2.66.2:5060' Method: BYE
  819. pbx*CLI>
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