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  1.     -- Executing [s@macro-user-callerid:10] GotoIf("SIP/800-00000063", "1?continue") in new stack
  2.     -- Goto (macro-user-callerid,s,19)
  3.     -- Executing [s@macro-user-callerid:19] NoOp("SIP/800-00000063", "Using CallerID "Jumshaid Mubashar" <800>") in new stack
  4.     -- Executing [5145662526@from-internal:2] Set("SIP/800-00000063", "_NODEST=") in new stack
  5.     -- Executing [5145662526@from-internal:3] Macro("SIP/800-00000063", "record-enable|800|OUT|") in new stack
  6.     -- Executing [s@macro-record-enable:1] GotoIf("SIP/800-00000063", "1?check") in new stack
  7.     -- Goto (macro-record-enable,s,4)
  8.     -- Executing [s@macro-record-enable:4] AGI("SIP/800-00000063", "recordingcheck|20111205-155602|1323129362.111") in new stack
  9.     -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  10.   recordingcheck|20111205-155602|1323129362.111: Outbound recording not enabled
  11.     -- AGI Script recordingcheck completed, returning 0
  12.     -- Executing [s@macro-record-enable:5] MacroExit("SIP/800-00000063", "") in new stack
  13.     -- Executing [5145662526@from-internal:4] Macro("SIP/800-00000063", "dialout-trunk|2|5145662526||") in new stack
  14.     -- Executing [s@macro-dialout-trunk:1] Set("SIP/800-00000063", "DIAL_TRUNK=2") in new stack
  15.     -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/800-00000063", "0?sub-pincheck|s|1") in new stack
  16.     -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/800-00000063", "0?disabletrunk|1") in new stack
  17.     -- Executing [s@macro-dialout-trunk:4] Set("SIP/800-00000063", "DIAL_NUMBER=5145662526") in new stack
  18.     -- Executing [s@macro-dialout-trunk:5] Set("SIP/800-00000063", "DIAL_TRUNK_OPTIONS=tr") in new stack
  19.     -- Executing [s@macro-dialout-trunk:6] Set("SIP/800-00000063", "OUTBOUND_GROUP=OUT_2") in new stack
  20.     -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/800-00000063", "1?nomax") in new stack
  21.     -- Goto (macro-dialout-trunk,s,9)
  22.     -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/800-00000063", "0?skipoutcid") in new stack
  23.     -- Executing [s@macro-dialout-trunk:10] Set("SIP/800-00000063", "DIAL_TRUNK_OPTIONS=") in new stack
  24.     -- Executing [s@macro-dialout-trunk:11] Macro("SIP/800-00000063", "outbound-callerid|2") in new stack
  25.     -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/800-00000063", "0|SetCallerPres|") in new stack
  26.     -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/800-00000063", "0|Set|REALCALLERIDNUM=800") in new stack
  27.     -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/800-00000063", "1?normcid") in new stack
  28.     -- Goto (macro-outbound-callerid,s,6)
  29.     -- Executing [s@macro-outbound-callerid:6] Set("SIP/800-00000063", "USEROUTCID="Jumshaid Mubashar" <925335845571>") in new stack
  30.     -- Executing [s@macro-outbound-callerid:7] Set("SIP/800-00000063", "EMERGENCYCID=") in new stack
  31.     -- Executing [s@macro-outbound-callerid:8] Set("SIP/800-00000063", "TRUNKOUTCID=") in new stack
  32.     -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/800-00000063", "1?trunkcid") in new stack
  33.     -- Goto (macro-outbound-callerid,s,12)
  34.     -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/800-00000063", "0|Set|CALLERID(all)=") in new stack
  35.     -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/800-00000063", "1|Set|CALLERID(all)=Jumshaid Mubashar <925335845571>") in new stack
  36.     -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/800-00000063", "0|SetCallerPres|prohib_passed_screen") in new stack
  37.     -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/800-00000063", "1|AGI|fixlocalprefix") in new stack
  38.     -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
  39.   ==  fixlocalprefix: Dialpattern NXXNXXXXXX matched. 5145662526 -> 5145662526
  40.     -- AGI Script fixlocalprefix completed, returning 0
  41.     -- Executing [s@macro-dialout-trunk:13] Set("SIP/800-00000063", "OUTNUM=5145662526") in new stack
  42.     -- Executing [s@macro-dialout-trunk:14] Set("SIP/800-00000063", "custom=SIP/voipms") in new stack
  43.     -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/800-00000063", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
  44.     -- Executing [s@macro-dialout-trunk:16] Macro("SIP/800-00000063", "dialout-trunk-predial-hook|") in new stack
  45.     -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/800-00000063", "") in new stack
  46.     -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/800-00000063", "0?bypass|1") in new stack
  47.     -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/800-00000063", "0?customtrunk") in new stack
  48.     -- Executing [s@macro-dialout-trunk:19] Dial("SIP/800-00000063", "SIP/voipms/5145662526|300|") in new stack
  49.     -- Called voipms/5145662526
  50.     -- SIP/voipms-00000064 is making progress passing it to SIP/800-00000063
  51. Audio is at 209.159.159.229 port 12118
  52. Adding codec 0x4 (ulaw) to SDP
  53. Adding non-codec 0x1 (telephone-event) to SDP
  54.  
  55. <--- Transmitting (NAT) to 70.52.7.249:11914 --->
  56. SIP/2.0 183 Session Progress
  57. Via: SIP/2.0/UDP 192.168.1.198:11914;branch=z9hG4bK-d87543-ad75b97d49640766-1--d87543-;received=70.52.7.249;rport=11914
  58. From: "800"<sip:800@209.159.159.229>;tag=f03c5144
  59. To: "5145662526"<sip:5145662526@209.159.159.229>;tag=as0d7c5c2c
  60. Call-ID: 8b7e8a792e238c74ZDJlZWZiNzc5MDBiNjZmNzRhNjc1N2FkOTQyOWVhZWU.
  61. CSeq: 2 INVITE
  62. User-Agent: Asterisk PBX
  63. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  64. Supported: replaces
  65. Contact: <sip:5145662526@209.159.159.229>
  66. Content-Type: application/sdp
  67. Content-Length: 217
  68.  
  69. v=0
  70. o=root 1689 1689 IN IP4 209.159.159.229
  71. s=session
  72. c=IN IP4 209.159.159.229
  73. t=0 0
  74. m=audio 12118 RTP/AVP 0 101
  75. a=rtpmap:0 PCMU/8000
  76. a=rtpmap:101 telephone-event/8000
  77. a=fmtp:101 0-16
  78. a=ptime:20
  79. a=sendrecv
  80.  
  81. <------------>
  82.     -- SIP/voipms-00000064 answered SIP/800-00000063
  83. Audio is at 209.159.159.229 port 12118
  84. Adding codec 0x4 (ulaw) to SDP
  85. Adding non-codec 0x1 (telephone-event) to SDP
  86.  
  87. <--- Reliably Transmitting (NAT) to 70.52.7.249:11914 --->
  88. SIP/2.0 200 OK
  89. Via: SIP/2.0/UDP 192.168.1.198:11914;branch=z9hG4bK-d87543-ad75b97d49640766-1--d87543-;received=70.52.7.249;rport=11914
  90. From: "800"<sip:800@209.159.159.229>;tag=f03c5144
  91. To: "5145662526"<sip:5145662526@209.159.159.229>;tag=as0d7c5c2c
  92. Call-ID: 8b7e8a792e238c74ZDJlZWZiNzc5MDBiNjZmNzRhNjc1N2FkOTQyOWVhZWU.
  93. CSeq: 2 INVITE
  94. User-Agent: Asterisk PBX
  95. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  96. Supported: replaces
  97. Contact: <sip:5145662526@209.159.159.229>
  98. Content-Type: application/sdp
  99. Content-Length: 217
  100.  
  101. v=0
  102. o=root 1689 1690 IN IP4 209.159.159.229
  103. s=session
  104. c=IN IP4 209.159.159.229
  105. t=0 0
  106. m=audio 12118 RTP/AVP 0 101
  107. a=rtpmap:0 PCMU/8000
  108. a=rtpmap:101 telephone-event/8000
  109. a=fmtp:101 0-16
  110. a=ptime:20
  111. a=sendrecv
  112.  
  113. <------------>
  114.     -- Packet2Packet bridging SIP/800-00000063 and SIP/voipms-00000064
  115.  
  116. <--- SIP read from 70.52.7.249:11914 --->
  117. ACK sip:5145662526@209.159.159.229 SIP/2.0
  118. Via: SIP/2.0/UDP 192.168.1.198:11914;branch=z9hG4bK-d87543-c37a0338023b0c10-1--d87543-;rport
  119. Max-Forwards: 70
  120. Contact: <sip:800@70.52.7.249:11914>
  121. To: "5145662526"<sip:5145662526@209.159.159.229>;tag=as0d7c5c2c
  122. From: "800"<sip:800@209.159.159.229>;tag=f03c5144
  123. Call-ID: 8b7e8a792e238c74ZDJlZWZiNzc5MDBiNjZmNzRhNjc1N2FkOTQyOWVhZWU.
  124. CSeq: 2 ACK
  125. Proxy-Authorization: Digest username="800",realm="asterisk",nonce="53ff6251",uri="sip:5145662526@209.159.159.229",response="6bcc9d9a578c155e9ea7249cf8d95d94",algorithm=MD5
  126. User-Agent: eyeBeam release 1003s stamp 31159
  127. Content-Length: 0
  128.  
  129.  
  130. <------------->
  131. --- (11 headers 0 lines) ---
  132.  
  133. <--- SIP read from 70.52.7.249:11914 --->
  134.  
  135.  
  136.  
  137. <------------->
  138.  
  139. <--- SIP read from 70.52.7.249:11914 --->
  140. BYE sip:5145662526@209.159.159.229 SIP/2.0
  141. Via: SIP/2.0/UDP 192.168.1.198:11914;branch=z9hG4bK-d87543-701ba078064af306-1--d87543-;rport
  142. Max-Forwards: 70
  143. Contact: <sip:800@70.52.7.249:11914>
  144. To: "5145662526"<sip:5145662526@209.159.159.229>;tag=as0d7c5c2c
  145. From: "800"<sip:800@209.159.159.229>;tag=f03c5144
  146. Call-ID: 8b7e8a792e238c74ZDJlZWZiNzc5MDBiNjZmNzRhNjc1N2FkOTQyOWVhZWU.
  147. CSeq: 3 BYE
  148. Proxy-Authorization: Digest username="800",realm="asterisk",nonce="53ff6251",uri="sip:5145662526@209.159.159.229",response="4ee58a66a9efd73582a0657edd8c0b3d",algorithm=MD5
  149. User-Agent: eyeBeam release 1003s stamp 31159
  150. Reason: SIP;description="User Hung Up"
  151. Content-Length: 0
  152.  
  153.  
  154. <------------->
  155. --- (12 headers 0 lines) ---
  156. Sending to 70.52.7.249 : 11914 (NAT)
  157.  
  158. <--- Transmitting (NAT) to 70.52.7.249:11914 --->
  159. SIP/2.0 200 OK
  160. Via: SIP/2.0/UDP 192.168.1.198:11914;branch=z9hG4bK-d87543-701ba078064af306-1--d87543-;received=70.52.7.249;rport=11914
  161. From: "800"<sip:800@209.159.159.229>;tag=f03c5144
  162. To: "5145662526"<sip:5145662526@209.159.159.229>;tag=as0d7c5c2c
  163. Call-ID: 8b7e8a792e238c74ZDJlZWZiNzc5MDBiNjZmNzRhNjc1N2FkOTQyOWVhZWU.
  164. CSeq: 3 BYE
  165. User-Agent: Asterisk PBX
  166. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  167. Supported: replaces
  168. ontent-Length: 0
  169.  
  170.  
  171. <------------>
  172.     -- Executing [h@macro-dialout-trunk:1] Macro("SIP/800-00000063", "hangupcall|") in new stack
  173.     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/800-00000063", "1?noautomon") in new stack
  174.     -- Goto (macro-hangupcall,s,3)
  175.     -- Executing [s@macro-hangupcall:3] NoOp("SIP/800-00000063", "TOUCH_MONITOR_OUTPUT=") in new stack
  176.     -- Executing [s@macro-hangupcall:4] GotoIf("SIP/800-00000063", "1?skipblkvm") in new stack
  177.     -- Goto (macro-hangupcall,s,7)
  178.     -- Executing [s@macro-hangupcall:7] GotoIf("SIP/800-00000063", "1?theend") in new stack
  179.     -- Goto (macro-hangupcall,s,9)
  180.     -- Executing [s@macro-hangupcall:9] Hangup("SIP/800-00000063", "") in new stack
  181.   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/800-00000063' in macro 'hangupcall'
  182.   == Spawn h extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/800-00000063'
  183.   == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/800-00000063' in macro 'dialout-trunk'
  184.   == Spawn extension (from-internal, 5145662526, 4) exited non-zero on 'SIP/800-00000063'
  185. Really destroying SIP dialog '8b7e8a792e238c74ZDJlZWZiNzc5MDBiNjZmNzRhNjc1N2FkOTQyOWVhZWU.' Method: BYE
  186.  
  187. <--- SIP read from 70.52.7.249:11914 --->
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