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  1.  
  2. pbx*CLI>
  3. pbx*CLI>
  4. pbx*CLI>
  5. pbx*CLI>
  6. pbx*CLI>
  7. pbx*CLI>
  8. pbx*CLI>
  9. pbx*CLI>
  10. pbx*CLI>
  11. pbx*CLI>
  12. pbx*CLI>
  13. pbx*CLI>
  14. Really destroying SIP dialog 'NjI4Mjg4OWRjMDVhYzlkNDEyYTc0ZmUzMWY5YTg4ODM.' Method: REGISTER
  15.  
  16. <--- SIP read from UDP:192.168.6.14:64428 --->
  17. INVITE sip:19163777000@192.168.6.16:5060 SIP/2.0
  18. Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-7452910378389766-1---d8754z-;rport
  19. Max-Forwards: 70
  20. Contact: <sip:4002@10.0.0.2:64428;rinstance=a0c537b71d7010f7>
  21. To: <sip:19163777000@192.168.6.16:5060>
  22. From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
  23. Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
  24. CSeq: 1 INVITE
  25. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
  26. Content-Type: application/sdp
  27. Supported: replaces
  28. User-Agent: 3CXPhone 6.0.26523.0
  29. Content-Length: 391
  30.  
  31. v=0
  32. o=3cxVCE 344141565 37306590 IN IP4 10.0.0.2
  33. s=3cxVCE Audio Call
  34. c=IN IP4 10.0.0.2
  35. t=0 0
  36. m=audio 40012 RTP/AVP 0 8 3 101
  37. a=rtpmap:0 PCMU/8000
  38. a=rtpmap:8 PCMA/8000
  39. a=rtpmap:3 GSM/8000
  40. a=rtpmap:101 telephone-event/8000
  41. a=fmtp:101 0-15
  42. a=ptime:20
  43. a=sendrecv
  44. m=video 40010 RTP/AVP 34
  45. c=IN IP4 10.0.0.2
  46. a=rtpmap:34 H263/90000
  47. a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
  48. a=sendrecv
  49. <------------->
  50. --- (13 headers 18 lines) ---
  51. Sending to 192.168.6.14:64428 (NAT)
  52. Using INVITE request as basis request - MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
  53. Found peer '4002' for '4002' from 192.168.6.14:64428
  54.  
  55. <--- Reliably Transmitting (NAT) to 192.168.6.14:64428 --->
  56. SIP/2.0 401 Unauthorized
  57. Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-7452910378389766-1---d8754z-;received=192.168.6.14;rport=64428
  58. From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
  59. To: <sip:19163777000@192.168.6.16:5060>;tag=as4a5d64ee
  60. Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
  61. CSeq: 1 INVITE
  62. Server: FPBX-2.8.1(1.8.11.0)
  63. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  64. Supported: replaces, timer
  65. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4110095b"
  66. Content-Length: 0
  67.  
  68.  
  69. <------------>
  70. Scheduling destruction of SIP dialog 'MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.' in 6848 ms (Method: INVITE)
  71.  
  72. <--- SIP read from UDP:192.168.6.14:64428 --->
  73. ACK sip:19163777000@192.168.6.16:5060 SIP/2.0
  74. Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-7452910378389766-1---d8754z-;rport
  75. Max-Forwards: 70
  76. To: <sip:19163777000@192.168.6.16:5060>;tag=as4a5d64ee
  77. From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
  78. Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
  79. CSeq: 1 ACK
  80. Content-Length: 0
  81.  
  82. <------------->
  83. --- (8 headers 0 lines) ---
  84.  
  85. <--- SIP read from UDP:192.168.6.14:64428 --->
  86. INVITE sip:19163777000@192.168.6.16:5060 SIP/2.0
  87. Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-9f4ae525885ede12-1---d8754z-;rport
  88. Max-Forwards: 70
  89. Contact: <sip:4002@10.0.0.2:64428;rinstance=a0c537b71d7010f7>
  90. To: <sip:19163777000@192.168.6.16:5060>
  91. From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
  92. Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
  93. CSeq: 2 INVITE
  94. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
  95. Content-Type: application/sdp
  96. Supported: replaces
  97. User-Agent: 3CXPhone 6.0.26523.0
  98. Authorization: Digest username="4002",realm="asterisk",nonce="4110095b",uri="sip:19163777000@192.168.6.16:5060",response="c0ebdeed27eeef0a84c8df4ea47d5824",algorithm=MD5
  99. Content-Length: 391
  100.  
  101. v=0
  102. o=3cxVCE 344141565 37306590 IN IP4 10.0.0.2
  103. s=3cxVCE Audio Call
  104. c=IN IP4 10.0.0.2
  105. t=0 0
  106. m=audio 40012 RTP/AVP 0 8 3 101
  107. a=rtpmap:0 PCMU/8000
  108. a=rtpmap:8 PCMA/8000
  109. a=rtpmap:3 GSM/8000
  110. a=rtpmap:101 telephone-event/8000
  111. a=fmtp:101 0-15
  112. a=ptime:20
  113. a=sendrecv
  114. m=video 40010 RTP/AVP 34
  115. c=IN IP4 10.0.0.2
  116. a=rtpmap:34 H263/90000
  117. a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
  118. a=sendrecv
  119. <------------->
  120. --- (14 headers 18 lines) ---
  121. Sending to 192.168.6.14:64428 (NAT)
  122. Using INVITE request as basis request - MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
  123. Found peer '4002' for '4002' from 192.168.6.14:64428
  124. == Using SIP RTP TOS bits 184
  125. == Using SIP RTP CoS mark 5
  126. Found RTP audio format 0
  127. Found RTP audio format 8
  128. Found RTP audio format 3
  129. Found RTP audio format 101
  130. Found audio description format PCMU for ID 0
  131. Found audio description format PCMA for ID 8
  132. Found audio description format GSM for ID 3
  133. Found audio description format telephone-event for ID 101
  134. Found RTP video format 34
  135. Found video description format H263 for ID 34
  136. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
  137. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  138. Peer audio RTP is at port 10.0.0.2:40012
  139. Looking for 19163777000 in from-internal (domain 192.168.6.16)
  140. list_route: hop: <sip:4002@10.0.0.2:64428;rinstance=a0c537b71d7010f7>
  141.  
  142. <--- Transmitting (NAT) to 192.168.6.14:64428 --->
  143. SIP/2.0 100 Trying
  144. Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-9f4ae525885ede12-1---d8754z-;received=192.168.6.14;rport=64428
  145. From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
  146. To: <sip:19163777000@192.168.6.16:5060>
  147. Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
  148. CSeq: 2 INVITE
  149. Server: FPBX-2.8.1(1.8.11.0)
  150. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  151. Supported: replaces, timer
  152. Contact: <sip:19163777000@192.168.6.16:5060>
  153. Content-Length: 0
  154.  
  155.  
  156. <------------>
  157. -- Executing [19163777000@from-internal:1] Macro("SIP/4002-00000054", "user-callerid,SKIPTTL,") in new stack
  158. -- Executing [s@macro-user-callerid:1] Set("SIP/4002-00000054", "AMPUSER=4002") in new stack
  159. -- Executing [s@macro-user-callerid:2] GotoIf("SIP/4002-00000054", "0?report") in new stack
  160. -- Executing [s@macro-user-callerid:3] ExecIf("SIP/4002-00000054", "1?Set(REALCALLERIDNUM=4002)") in new stack
  161. -- Executing [s@macro-user-callerid:4] Set("SIP/4002-00000054", "AMPUSER=4002") in new stack
  162. -- Executing [s@macro-user-callerid:5] Set("SIP/4002-00000054", "AMPUSERCIDNAME=4002") in new stack
  163. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/4002-00000054", "0?report") in new stack
  164. -- Executing [s@macro-user-callerid:7] Set("SIP/4002-00000054", "AMPUSERCID=4002") in new stack
  165. -- Executing [s@macro-user-callerid:8] Set("SIP/4002-00000054", "CALLERID(all)="4002" <4002>") in new stack
  166. -- Executing [s@macro-user-callerid:9] ExecIf("SIP/4002-00000054", "0?Set(CHANNEL(language)=)") in new stack
  167. -- Executing [s@macro-user-callerid:10] GotoIf("SIP/4002-00000054", "1?continue") in new stack
  168. -- Goto (macro-user-callerid,s,19)
  169. -- Executing [s@macro-user-callerid:19] Set("SIP/4002-00000054", "CALLERID(number)=4002") in new stack
  170. -- Executing [s@macro-user-callerid:20] Set("SIP/4002-00000054", "CALLERID(name)=4002") in new stack
  171. -- Executing [s@macro-user-callerid:21] NoOp("SIP/4002-00000054", "Using CallerID "4002" <4002>") in new stack
  172. -- Executing [19163777000@from-internal:2] NoOp("SIP/4002-00000054", "Calling Out Route: nymgo") in new stack
  173. -- Executing [19163777000@from-internal:3] Set("SIP/4002-00000054", "MOHCLASS=default") in new stack
  174. -- Executing [19163777000@from-internal:4] Set("SIP/4002-00000054", "_NODEST=") in new stack
  175. -- Executing [19163777000@from-internal:5] Macro("SIP/4002-00000054", "record-enable,4002,OUT,") in new stack
  176. -- Executing [s@macro-record-enable:1] GotoIf("SIP/4002-00000054", "1?check") in new stack
  177. -- Goto (macro-record-enable,s,4)
  178. -- Executing [s@macro-record-enable:4] ExecIf("SIP/4002-00000054", "0?MacroExit()") in new stack
  179. -- Executing [s@macro-record-enable:5] GotoIf("SIP/4002-00000054", "0?Group:OUT") in new stack
  180. -- Goto (macro-record-enable,s,15)
  181. -- Executing [s@macro-record-enable:15] GotoIf("SIP/4002-00000054", "0?IN") in new stack
  182. -- Executing [s@macro-record-enable:16] ExecIf("SIP/4002-00000054", "1?MacroExit()") in new stack
  183. -- Executing [19163777000@from-internal:6] Macro("SIP/4002-00000054", "dialout-trunk,3,19163777000,") in new stack
  184. -- Executing [s@macro-dialout-trunk:1] Set("SIP/4002-00000054", "DIAL_TRUNK=3") in new stack
  185. -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/4002-00000054", "0?sub-pincheck,s,1") in new stack
  186. -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/4002-00000054", "0?disabletrunk,1") in new stack
  187. -- Executing [s@macro-dialout-trunk:4] Set("SIP/4002-00000054", "DIAL_NUMBER=19163777000") in new stack
  188. -- Executing [s@macro-dialout-trunk:5] Set("SIP/4002-00000054", "DIAL_TRUNK_OPTIONS=tr") in new stack
  189. -- Executing [s@macro-dialout-trunk:6] Set("SIP/4002-00000054", "OUTBOUND_GROUP=OUT_3") in new stack
  190. -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/4002-00000054", "1?nomax") in new stack
  191. -- Goto (macro-dialout-trunk,s,9)
  192. -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/4002-00000054", "0?skipoutcid") in new stack
  193. -- Executing [s@macro-dialout-trunk:10] Set("SIP/4002-00000054", "DIAL_TRUNK_OPTIONS=") in new stack
  194. -- Executing [s@macro-dialout-trunk:11] Macro("SIP/4002-00000054", "outbound-callerid,3") in new stack
  195. -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/4002-00000054", "0?Set(CALLERPRES()=)") in new stack
  196. -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/4002-00000054", "0?Set(REALCALLERIDNUM=4002)") in new stack
  197. -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/4002-00000054", "1?normcid") in new stack
  198. -- Goto (macro-outbound-callerid,s,6)
  199. -- Executing [s@macro-outbound-callerid:6] Set("SIP/4002-00000054", "USEROUTCID=") in new stack
  200. -- Executing [s@macro-outbound-callerid:7] Set("SIP/4002-00000054", "EMERGENCYCID=") in new stack
  201. -- Executing [s@macro-outbound-callerid:8] Set("SIP/4002-00000054", "TRUNKOUTCID=94771771229") in new stack
  202. -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/4002-00000054", "1?trunkcid") in new stack
  203. -- Goto (macro-outbound-callerid,s,12)
  204. -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/4002-00000054", "1?Set(CALLERID(all)=94771771229)") in new stack
  205. -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/4002-00000054", "0?Set(CALLERID(all)=)") in new stack
  206. -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/4002-00000054", "0?Set(CALLERID(all)=)") in new stack
  207. -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/4002-00000054", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
  208. -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/4002-00000054", "1?sub-flp-3,s,1") in new stack
  209. -- Executing [s@sub-flp-3:1] ExecIf("SIP/4002-00000054", "1?Return()") in new stack
  210. -- Executing [s@macro-dialout-trunk:13] Set("SIP/4002-00000054", "OUTNUM=19163777000") in new stack
  211. -- Executing [s@macro-dialout-trunk:14] Set("SIP/4002-00000054", "custom=SIP/com-in") in new stack
  212. -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/4002-00000054", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
  213. -- Executing [s@macro-dialout-trunk:16] Macro("SIP/4002-00000054", "dialout-trunk-predial-hook,") in new stack
  214. -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/4002-00000054", "") in new stack
  215. -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/4002-00000054", "0?bypass,1") in new stack
  216. -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/4002-00000054", "0?customtrunk") in new stack
  217. -- Executing [s@macro-dialout-trunk:19] Dial("SIP/4002-00000054", "SIP/com-in/19163777000,300,") in new stack
  218. == Using SIP RTP TOS bits 184
  219. == Using SIP RTP CoS mark 5
  220. Audio is at 17482
  221. Adding codec 0x4 (ulaw) to SDP
  222. Adding non-codec 0x1 (telephone-event) to SDP
  223. Reliably Transmitting (NAT) to 176.34.131.112:5060:
  224. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  225. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK57b4a304;rport
  226. Max-Forwards: 70
  227. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  228. To: <sip:19163777000@sip.commpeak.com>
  229. Contact: <sip:94771771229@192.168.6.16:5060>
  230. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  231. CSeq: 102 INVITE
  232. User-Agent: FPBX-2.8.1(1.8.11.0)
  233. Date: Mon, 10 Sep 2012 19:40:53 GMT
  234. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  235. Supported: replaces, timer
  236. Content-Type: application/sdp
  237. Content-Length: 235
  238.  
  239. v=0
  240. o=root 620339841 620339841 IN IP4 192.168.6.16
  241. s=Asterisk PBX 1.8.11.0
  242. c=IN IP4 192.168.6.16
  243. t=0 0
  244. m=audio 17482 RTP/AVP 0 101
  245. a=rtpmap:0 PCMU/8000
  246. a=rtpmap:101 telephone-event/8000
  247. a=fmtp:101 0-16
  248. a=ptime:20
  249. a=sendrecv
  250.  
  251. ---
  252. -- Called SIP/com-in/19163777000
  253.  
  254. <--- SIP read from UDP:176.34.131.112:5060 --->
  255. SIP/2.0 100 Trying...
  256. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK57b4a304;rport=4831
  257. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  258. To: <sip:19163777000@sip.commpeak.com>
  259. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  260. CSeq: 102 INVITE
  261. Server: CommPeak SIP Proxy
  262. Content-Length: 0
  263.  
  264. <------------->
  265. --- (8 headers 0 lines) ---
  266.  
  267. <--- SIP read from UDP:176.34.131.112:5060 --->
  268. SIP/2.0 407 Proxy Authentication Required
  269. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK57b4a304;rport=4831
  270. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  271. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.8853
  272. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  273. CSeq: 102 INVITE
  274. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e426300000083ee05bbbdd6b252cdf7c847987f9adb05"
  275. Server: CommPeak SIP Proxy
  276. Content-Length: 0
  277.  
  278. <------------->
  279. --- (9 headers 0 lines) ---
  280. set_destination: Parsing <sip:19163777000@sip.commpeak.com> for address/port to send to
  281. set_destination: set destination to 176.34.131.112:5060
  282. Transmitting (NAT) to 176.34.131.112:5060:
  283. ACK sip:19163777000@sip.commpeak.com SIP/2.0
  284. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK57b4a304;rport
  285. Max-Forwards: 70
  286. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  287. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.8853
  288. Contact: <sip:94771771229@192.168.6.16:5060>
  289. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  290. CSeq: 102 ACK
  291. User-Agent: FPBX-2.8.1(1.8.11.0)
  292. Content-Length: 0
  293.  
  294.  
  295. ---
  296. Audio is at 17482
  297. Adding codec 0x4 (ulaw) to SDP
  298. Adding non-codec 0x1 (telephone-event) to SDP
  299. Reliably Transmitting (NAT) to 176.34.131.112:5060:
  300. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  301. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK00ed031b;rport
  302. Max-Forwards: 70
  303. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  304. To: <sip:19163777000@sip.commpeak.com>
  305. Contact: <sip:94771771229@192.168.6.16:5060>
  306. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  307. CSeq: 103 INVITE
  308. User-Agent: FPBX-2.8.1(1.8.11.0)
  309. Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e426300000083ee05bbbdd6b252cdf7c847987f9adb05", response="4164d3b6a47684856459808e7f47093a"
  310. Date: Mon, 10 Sep 2012 19:40:53 GMT
  311. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  312. Supported: replaces, timer
  313. Content-Type: application/sdp
  314. Content-Length: 235
  315.  
  316. v=0
  317. o=root 620339841 620339842 IN IP4 192.168.6.16
  318. s=Asterisk PBX 1.8.11.0
  319. c=IN IP4 192.168.6.16
  320. t=0 0
  321. m=audio 17482 RTP/AVP 0 101
  322. a=rtpmap:0 PCMU/8000
  323. a=rtpmap:101 telephone-event/8000
  324. a=fmtp:101 0-16
  325. a=ptime:20
  326. a=sendrecv
  327.  
  328. ---
  329. Retransmitting #1 (NAT) to 176.34.131.112:5060:
  330. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  331. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK00ed031b;rport
  332. Max-Forwards: 70
  333. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  334. To: <sip:19163777000@sip.commpeak.com>
  335. Contact: <sip:94771771229@192.168.6.16:5060>
  336. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  337. CSeq: 103 INVITE
  338. User-Agent: FPBX-2.8.1(1.8.11.0)
  339. Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e426300000083ee05bbbdd6b252cdf7c847987f9adb05", response="4164d3b6a47684856459808e7f47093a"
  340. Date: Mon, 10 Sep 2012 19:40:53 GMT
  341. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  342. Supported: replaces, timer
  343. Content-Type: application/sdp
  344. Content-Length: 235
  345.  
  346. v=0
  347. o=root 620339841 620339842 IN IP4 192.168.6.16
  348. s=Asterisk PBX 1.8.11.0
  349. c=IN IP4 192.168.6.16
  350. t=0 0
  351. m=audio 17482 RTP/AVP 0 101
  352. a=rtpmap:0 PCMU/8000
  353. a=rtpmap:101 telephone-event/8000
  354. a=fmtp:101 0-16
  355. a=ptime:20
  356. a=sendrecv
  357.  
  358. ---
  359.  
  360. <--- SIP read from UDP:176.34.131.112:5060 --->
  361. SIP/2.0 100 Trying...
  362. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK00ed031b;rport=4831
  363. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  364. To: <sip:19163777000@sip.commpeak.com>
  365. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  366. CSeq: 103 INVITE
  367. Server: CommPeak SIP Proxy
  368. Content-Length: 0
  369.  
  370. <------------->
  371. --- (8 headers 0 lines) ---
  372.  
  373. <--- SIP read from UDP:176.34.131.112:5060 --->
  374. SIP/2.0 407 Proxy Authentication Required
  375. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK00ed031b;rport=4831
  376. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  377. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.48be
  378. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  379. CSeq: 103 INVITE
  380. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e4263000000842f1b6d8bd6345ce0b4141bcb0e2d7d46"
  381. Server: CommPeak SIP Proxy
  382. Content-Length: 0
  383.  
  384. <------------->
  385. --- (9 headers 0 lines) ---
  386. set_destination: Parsing <sip:19163777000@sip.commpeak.com> for address/port to send to
  387. set_destination: set destination to 176.34.131.112:5060
  388. Transmitting (NAT) to 176.34.131.112:5060:
  389. ACK sip:19163777000@sip.commpeak.com SIP/2.0
  390. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK00ed031b;rport
  391. Max-Forwards: 70
  392. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  393. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.48be
  394. Contact: <sip:94771771229@192.168.6.16:5060>
  395. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  396. CSeq: 103 ACK
  397. User-Agent: FPBX-2.8.1(1.8.11.0)
  398. Content-Length: 0
  399.  
  400.  
  401. ---
  402. Audio is at 17482
  403. Adding codec 0x4 (ulaw) to SDP
  404. Adding non-codec 0x1 (telephone-event) to SDP
  405. Reliably Transmitting (NAT) to 176.34.131.112:5060:
  406. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  407. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK6c464bd0;rport
  408. Max-Forwards: 70
  409. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  410. To: <sip:19163777000@sip.commpeak.com>
  411. Contact: <sip:94771771229@192.168.6.16:5060>
  412. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  413. CSeq: 104 INVITE
  414. User-Agent: FPBX-2.8.1(1.8.11.0)
  415. Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e4263000000842f1b6d8bd6345ce0b4141bcb0e2d7d46", response="cfb11196a40bdc8a55e5be2f8a0ad95f"
  416. Date: Mon, 10 Sep 2012 19:40:53 GMT
  417. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  418. Supported: replaces, timer
  419. Content-Type: application/sdp
  420. Content-Length: 235
  421.  
  422. v=0
  423. o=root 620339841 620339843 IN IP4 192.168.6.16
  424. s=Asterisk PBX 1.8.11.0
  425. c=IN IP4 192.168.6.16
  426. t=0 0
  427. m=audio 17482 RTP/AVP 0 101
  428. a=rtpmap:0 PCMU/8000
  429. a=rtpmap:101 telephone-event/8000
  430. a=fmtp:101 0-16
  431. a=ptime:20
  432. a=sendrecv
  433.  
  434. ---
  435.  
  436. <--- SIP read from UDP:176.34.131.112:5060 --->
  437. SIP/2.0 100 Trying...
  438. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK00ed031b;rport=4831
  439. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  440. To: <sip:19163777000@sip.commpeak.com>
  441. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  442. CSeq: 103 INVITE
  443. Server: CommPeak SIP Proxy
  444. Content-Length: 0
  445.  
  446. <------------->
  447. --- (8 headers 0 lines) ---
  448.  
  449. <--- SIP read from UDP:176.34.131.112:5060 --->
  450. SIP/2.0 407 Proxy Authentication Required
  451. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK00ed031b;rport=4831
  452. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  453. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.48be
  454. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  455. CSeq: 103 INVITE
  456. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e426400000085bafbe6e7191759dc8717a00903df722e"
  457. Server: CommPeak SIP Proxy
  458. Content-Length: 0
  459.  
  460. <------------->
  461. --- (9 headers 0 lines) ---
  462. Transmitting (NAT) to 176.34.131.112:5060:
  463. ACK sip:19163777000@sip.commpeak.com SIP/2.0
  464. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK6c464bd0;rport
  465. Max-Forwards: 70
  466. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  467. To: <sip:19163777000@sip.commpeak.com>
  468. Contact: <sip:94771771229@192.168.6.16:5060>
  469. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  470. CSeq: 103 ACK
  471. User-Agent: FPBX-2.8.1(1.8.11.0)
  472. Content-Length: 0
  473.  
  474.  
  475. ---
  476. Retransmitting #1 (NAT) to 176.34.131.112:5060:
  477. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  478. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK6c464bd0;rport
  479. Max-Forwards: 70
  480. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  481. To: <sip:19163777000@sip.commpeak.com>
  482. Contact: <sip:94771771229@192.168.6.16:5060>
  483. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  484. CSeq: 104 INVITE
  485. User-Agent: FPBX-2.8.1(1.8.11.0)
  486. Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e4263000000842f1b6d8bd6345ce0b4141bcb0e2d7d46", response="cfb11196a40bdc8a55e5be2f8a0ad95f"
  487. Date: Mon, 10 Sep 2012 19:40:53 GMT
  488. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  489. Supported: replaces, timer
  490. Content-Type: application/sdp
  491. Content-Length: 235
  492.  
  493. v=0
  494. o=root 620339841 620339843 IN IP4 192.168.6.16
  495. s=Asterisk PBX 1.8.11.0
  496. c=IN IP4 192.168.6.16
  497. t=0 0
  498. m=audio 17482 RTP/AVP 0 101
  499. a=rtpmap:0 PCMU/8000
  500. a=rtpmap:101 telephone-event/8000
  501. a=fmtp:101 0-16
  502. a=ptime:20
  503. a=sendrecv
  504.  
  505. ---
  506.  
  507. <--- SIP read from UDP:176.34.131.112:5060 --->
  508. SIP/2.0 100 Trying...
  509. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK6c464bd0;rport=4831
  510. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  511. To: <sip:19163777000@sip.commpeak.com>
  512. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  513. CSeq: 104 INVITE
  514. Server: CommPeak SIP Proxy
  515. Content-Length: 0
  516.  
  517. <------------->
  518. --- (8 headers 0 lines) ---
  519.  
  520. <--- SIP read from UDP:176.34.131.112:5060 --->
  521. SIP/2.0 407 Proxy Authentication Required
  522. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK6c464bd0;rport=4831
  523. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  524. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.f3c7
  525. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  526. CSeq: 104 INVITE
  527. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e4264000000867504ae8f83abb84f8a735fffd2382c7a"
  528. Server: CommPeak SIP Proxy
  529. Content-Length: 0
  530.  
  531. <------------->
  532. --- (9 headers 0 lines) ---
  533. Transmitting (NAT) to 176.34.131.112:5060:
  534. ACK sip:19163777000@sip.commpeak.com SIP/2.0
  535. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK6c464bd0;rport
  536. Max-Forwards: 70
  537. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  538. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.f3c7
  539. Contact: <sip:94771771229@192.168.6.16:5060>
  540. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  541. CSeq: 104 ACK
  542. User-Agent: FPBX-2.8.1(1.8.11.0)
  543. Content-Length: 0
  544.  
  545.  
  546. ---
  547. Audio is at 17482
  548. Adding codec 0x4 (ulaw) to SDP
  549. Adding non-codec 0x1 (telephone-event) to SDP
  550. Reliably Transmitting (NAT) to 176.34.131.112:5060:
  551. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  552. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK5648f608;rport
  553. Max-Forwards: 70
  554. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  555. To: <sip:19163777000@sip.commpeak.com>
  556. Contact: <sip:94771771229@192.168.6.16:5060>
  557. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  558. CSeq: 105 INVITE
  559. User-Agent: FPBX-2.8.1(1.8.11.0)
  560. Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e4264000000867504ae8f83abb84f8a735fffd2382c7a", response="6931a37ce53b7d79e79e84e61f5c74d3"
  561. Date: Mon, 10 Sep 2012 19:40:54 GMT
  562. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  563. Supported: replaces, timer
  564. Content-Type: application/sdp
  565. Content-Length: 235
  566.  
  567. v=0
  568. o=root 620339841 620339844 IN IP4 192.168.6.16
  569. s=Asterisk PBX 1.8.11.0
  570. c=IN IP4 192.168.6.16
  571. t=0 0
  572. m=audio 17482 RTP/AVP 0 101
  573. a=rtpmap:0 PCMU/8000
  574. a=rtpmap:101 telephone-event/8000
  575. a=fmtp:101 0-16
  576. a=ptime:20
  577. a=sendrecv
  578.  
  579. ---
  580.  
  581. <--- SIP read from UDP:176.34.131.112:5060 --->
  582. SIP/2.0 100 Trying...
  583. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK6c464bd0;rport=4831
  584. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  585. To: <sip:19163777000@sip.commpeak.com>
  586. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  587. CSeq: 104 INVITE
  588. Server: CommPeak SIP Proxy
  589. Content-Length: 0
  590.  
  591. <------------->
  592. --- (8 headers 0 lines) ---
  593.  
  594. <--- SIP read from UDP:176.34.131.112:5060 --->
  595. SIP/2.0 407 Proxy Authentication Required
  596. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK6c464bd0;rport=4831
  597. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  598. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.f3c7
  599. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  600. CSeq: 104 INVITE
  601. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e426400000087eda82a7a2bb10253eb730e3b40451d7a"
  602. Server: CommPeak SIP Proxy
  603. Content-Length: 0
  604.  
  605. <------------->
  606. --- (9 headers 0 lines) ---
  607. Transmitting (NAT) to 176.34.131.112:5060:
  608. ACK sip:19163777000@sip.commpeak.com SIP/2.0
  609. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK5648f608;rport
  610. Max-Forwards: 70
  611. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  612. To: <sip:19163777000@sip.commpeak.com>
  613. Contact: <sip:94771771229@192.168.6.16:5060>
  614. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  615. CSeq: 104 ACK
  616. User-Agent: FPBX-2.8.1(1.8.11.0)
  617. Content-Length: 0
  618.  
  619.  
  620. ---
  621. Retransmitting #1 (NAT) to 176.34.131.112:5060:
  622. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  623. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK5648f608;rport
  624. Max-Forwards: 70
  625. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  626. To: <sip:19163777000@sip.commpeak.com>
  627. Contact: <sip:94771771229@192.168.6.16:5060>
  628. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  629. CSeq: 105 INVITE
  630. User-Agent: FPBX-2.8.1(1.8.11.0)
  631. Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e4264000000867504ae8f83abb84f8a735fffd2382c7a", response="6931a37ce53b7d79e79e84e61f5c74d3"
  632. Date: Mon, 10 Sep 2012 19:40:54 GMT
  633. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  634. Supported: replaces, timer
  635. Content-Type: application/sdp
  636. Content-Length: 235
  637.  
  638. v=0
  639. o=root 620339841 620339844 IN IP4 192.168.6.16
  640. s=Asterisk PBX 1.8.11.0
  641. c=IN IP4 192.168.6.16
  642. t=0 0
  643. m=audio 17482 RTP/AVP 0 101
  644. a=rtpmap:0 PCMU/8000
  645. a=rtpmap:101 telephone-event/8000
  646. a=fmtp:101 0-16
  647. a=ptime:20
  648. a=sendrecv
  649.  
  650. ---
  651.  
  652. <--- SIP read from UDP:176.34.131.112:5060 --->
  653. SIP/2.0 100 Trying...
  654. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK5648f608;rport=4831
  655. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  656. To: <sip:19163777000@sip.commpeak.com>
  657. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  658. CSeq: 105 INVITE
  659. Server: CommPeak SIP Proxy
  660. Content-Length: 0
  661.  
  662. <------------->
  663. --- (8 headers 0 lines) ---
  664.  
  665. <--- SIP read from UDP:176.34.131.112:5060 --->
  666. SIP/2.0 407 Proxy Authentication Required
  667. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK5648f608;rport=4831
  668. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  669. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.6012
  670. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  671. CSeq: 105 INVITE
  672. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e42640000008868309f249a1d20138aaae9bd40454093"
  673. Server: CommPeak SIP Proxy
  674. Content-Length: 0
  675.  
  676. <------------->
  677. --- (9 headers 0 lines) ---
  678. Transmitting (NAT) to 176.34.131.112:5060:
  679. ACK sip:19163777000@sip.commpeak.com SIP/2.0
  680. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK5648f608;rport
  681. Max-Forwards: 70
  682. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  683. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.6012
  684. Contact: <sip:94771771229@192.168.6.16:5060>
  685. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  686. CSeq: 105 ACK
  687. User-Agent: FPBX-2.8.1(1.8.11.0)
  688. Content-Length: 0
  689.  
  690.  
  691. ---
  692. -- SIP/com-in-00000055 is circuit-busy
  693. == Everyone is busy/congested at this time (1:0/1/0)
  694. -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/4002-00000054", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21") in new stack
  695. -- Executing [s@macro-dialout-trunk:21] Goto("SIP/4002-00000054", "s-CONGESTION,1") in new stack
  696. -- Goto (macro-dialout-trunk,s-CONGESTION,1)
  697. -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/4002-00000054", "RC=21") in new stack
  698. -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/4002-00000054", "21,1") in new stack
  699. -- Goto (macro-dialout-trunk,21,1)
  700. -- Executing [21@macro-dialout-trunk:1] Goto("SIP/4002-00000054", "continue,1") in new stack
  701. -- Goto (macro-dialout-trunk,continue,1)
  702. -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/4002-00000054", "1?noreport") in new stack
  703. -- Goto (macro-dialout-trunk,continue,3)
  704. -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/4002-00000054", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks") in new stack
  705. -- Executing [continue@macro-dialout-trunk:4] Set("SIP/4002-00000054", "CALLERID(number)=4002") in new stack
  706. -- Executing [19163777000@from-internal:7] Macro("SIP/4002-00000054", "outisbusy,") in new stack
  707. -- Executing [s@macro-outisbusy:1] Progress("SIP/4002-00000054", "") in new stack
  708. Audio is at 18978
  709. Adding codec 0x4 (ulaw) to SDP
  710. Adding codec 0x8 (alaw) to SDP
  711. Adding codec 0x2 (gsm) to SDP
  712. Adding non-codec 0x1 (telephone-event) to SDP
  713.  
  714. <--- Transmitting (NAT) to 192.168.6.14:64428 --->
  715. SIP/2.0 183 Session Progress
  716. Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-9f4ae525885ede12-1---d8754z-;received=192.168.6.14;rport=64428
  717. From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
  718. To: <sip:19163777000@192.168.6.16:5060>;tag=as53a48ffe
  719. Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
  720. CSeq: 2 INVITE
  721. Server: FPBX-2.8.1(1.8.11.0)
  722. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  723. Supported: replaces, timer
  724. Contact: <sip:19163777000@192.168.6.16:5060>
  725. Content-Type: application/sdp
  726. Content-Length: 302
  727.  
  728. v=0
  729. o=root 66937535 66937535 IN IP4 192.168.6.16
  730. s=Asterisk PBX 1.8.11.0
  731. c=IN IP4 192.168.6.16
  732. t=0 0
  733. m=audio 18978 RTP/AVP 0 8 3 101
  734. a=rtpmap:0 PCMU/8000
  735. a=rtpmap:8 PCMA/8000
  736. a=rtpmap:3 GSM/8000
  737. a=rtpmap:101 telephone-event/8000
  738. a=fmtp:101 0-16
  739. a=ptime:20
  740. a=sendrecv
  741. m=video 0 RTP/AVP 34
  742.  
  743. <------------>
  744. -- Executing [s@macro-outisbusy:2] GotoIf("SIP/4002-00000054", "0?emergency,1") in new stack
  745. -- Executing [s@macro-outisbusy:3] GotoIf("SIP/4002-00000054", "0?intracompany,1") in new stack
  746. -- Executing [s@macro-outisbusy:4] Playback("SIP/4002-00000054", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
  747. -- <SIP/4002-00000054> Playing 'all-circuits-busy-now.gsm' (language 'en')
  748.  
  749. <--- SIP read from UDP:176.34.131.112:5060 --->
  750. SIP/2.0 100 Trying...
  751. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK5648f608;rport=4831
  752. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  753. To: <sip:19163777000@sip.commpeak.com>
  754. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  755. CSeq: 105 INVITE
  756. Server: CommPeak SIP Proxy
  757. Content-Length: 0
  758.  
  759. <------------->
  760. --- (8 headers 0 lines) ---
  761. Really destroying SIP dialog '783561206945f8c3671514bc125c60e1@192.168.6.16:5060' Method: INVITE
  762.  
  763. <--- SIP read from UDP:176.34.131.112:5060 --->
  764. SIP/2.0 407 Proxy Authentication Required
  765. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK5648f608;rport=4831
  766. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as3118ea55
  767. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.6012
  768. Call-ID: 783561206945f8c3671514bc125c60e1@192.168.6.16:5060
  769. CSeq: 105 INVITE
  770. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e42640000008991566720346b100b76969387973b0cb9"
  771. Server: CommPeak SIP Proxy
  772. Content-Length: 0
  773.  
  774. <------------->
  775. --- (9 headers 0 lines) ---
  776. -- <SIP/4002-00000054> Playing 'pls-try-call-later.gsm' (language 'en')
  777.  
  778. <--- SIP read from UDP:192.168.6.14:64428 --->
  779.  
  780.  
  781. <------------->
  782. -- Executing [s@macro-outisbusy:5] Congestion("SIP/4002-00000054", "20") in new stack
  783.  
  784. <--- Reliably Transmitting (NAT) to 192.168.6.14:64428 --->
  785. SIP/2.0 503 Service Unavailable
  786. Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-9f4ae525885ede12-1---d8754z-;received=192.168.6.14;rport=64428
  787. From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
  788. To: <sip:19163777000@192.168.6.16:5060>;tag=as53a48ffe
  789. Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
  790. CSeq: 2 INVITE
  791. Server: FPBX-2.8.1(1.8.11.0)
  792. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  793. Supported: replaces, timer
  794. X-Asterisk-HangupCause: Call Rejected
  795. X-Asterisk-HangupCauseCode: 21
  796. Content-Length: 0
  797.  
  798.  
  799. <------------>
  800. == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/4002-00000054' in macro 'outisbusy'
  801. == Spawn extension (from-internal, 19163777000, 7) exited non-zero on 'SIP/4002-00000054'
  802. -- Executing [h@from-internal:1] Macro("SIP/4002-00000054", "hangupcall") in new stack
  803. -- Executing [s@macro-hangupcall:1] GotoIf("SIP/4002-00000054", "1?endmixmoncheck") in new stack
  804. -- Goto (macro-hangupcall,s,9)
  805. -- Executing [s@macro-hangupcall:9] NoOp("SIP/4002-00000054", "End of MIXMON check") in new stack
  806. -- Executing [s@macro-hangupcall:10] GotoIf("SIP/4002-00000054", "1?nomeetmemon") in new stack
  807. -- Goto (macro-hangupcall,s,28)
  808. -- Executing [s@macro-hangupcall:28] NoOp("SIP/4002-00000054", "End of MEETME check") in new stack
  809. -- Executing [s@macro-hangupcall:29] GotoIf("SIP/4002-00000054", "1?noautomon") in new stack
  810. -- Goto (macro-hangupcall,s,34)
  811. -- Executing [s@macro-hangupcall:34] NoOp("SIP/4002-00000054", "TOUCH_MONITOR_OUTPUT=") in new stack
  812. -- Executing [s@macro-hangupcall:35] GotoIf("SIP/4002-00000054", "1?noautomon2") in new stack
  813. -- Goto (macro-hangupcall,s,41)
  814. -- Executing [s@macro-hangupcall:41] NoOp("SIP/4002-00000054", "MONITOR_FILENAME=") in new stack
  815. -- Executing [s@macro-hangupcall:42] GotoIf("SIP/4002-00000054", "1?skiprg") in new stack
  816. -- Goto (macro-hangupcall,s,45)
  817. -- Executing [s@macro-hangupcall:45] GotoIf("SIP/4002-00000054", "1?skipblkvm") in new stack
  818. -- Goto (macro-hangupcall,s,48)
  819. -- Executing [s@macro-hangupcall:48] GotoIf("SIP/4002-00000054", "1?theend") in new stack
  820. -- Goto (macro-hangupcall,s,50)
  821. -- Executing [s@macro-hangupcall:50] AGI("SIP/4002-00000054", "hangup.agi") in new stack
  822. -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
  823.  
  824. <--- SIP read from UDP:192.168.6.14:64428 --->
  825. ACK sip:19163777000@192.168.6.16:5060 SIP/2.0
  826. Via: SIP/2.0/UDP 10.0.0.2:64428;branch=z9hG4bK-d8754z-9f4ae525885ede12-1---d8754z-;rport
  827. Max-Forwards: 70
  828. To: <sip:19163777000@192.168.6.16:5060>;tag=as53a48ffe
  829. From: "4002"<sip:4002@192.168.6.16:5060>;tag=8e6e8671
  830. Call-ID: MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.
  831. CSeq: 2 ACK
  832. Content-Length: 0
  833.  
  834. <------------->
  835. --- (8 headers 0 lines) ---
  836. -- <SIP/4002-00000054>AGI Script hangup.agi completed, returning 0
  837. -- Executing [s@macro-hangupcall:51] Hangup("SIP/4002-00000054", "") in new stack
  838. == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/4002-00000054' in macro 'hangupcall'
  839. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/4002-00000054'
  840. Really destroying SIP dialog 'MWFkODhjMDBhODhkMTkwN2YxYTc3YjZhYmY4NDZiMDQ.' Method: ACK
  841.  
  842. <--- SIP read from UDP:61.245.172.12:7035 --->
  843.  
  844.  
  845. <------------->
  846. pbx*CLI> sip set debug off
  847. SIP Debugging Disabled
  848. pbx*CLI>
  849. pbx*CLI> pbx*CLI>
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