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  1.  
  2. pbx*CLI>
  3. pbx*CLI>
  4. pbx*CLI>
  5.  
  6. <--- SIP read from UDP:61.245.172.12:7035 --->
  7. INVITE sip:19163777000@192.168.6.16:5060 SIP/2.0
  8. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-4b179f68465b7a41-1---d8754z-;rport
  9. Max-Forwards: 70
  10. Contact: <sip:4001@10.225.20.57:7035;rinstance=b58773ece739e901>
  11. To: <sip:19163777000@213.204.79.112:5060>
  12. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  13. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  14. CSeq: 1 INVITE
  15. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
  16. Content-Type: application/sdp
  17. Supported: replaces
  18. User-Agent: 3CXPhone 6.0.26523.0
  19. Content-Length: 405
  20.  
  21. v=0
  22. o=3cxVCE 236616615 370732695 IN IP4 10.225.20.57
  23. s=3cxVCE Audio Call
  24. c=IN IP4 10.225.20.57
  25. t=0 0
  26. m=audio 40046 RTP/AVP 0 8 3 101
  27. a=rtpmap:0 PCMU/8000
  28. a=rtpmap:8 PCMA/8000
  29. a=rtpmap:3 GSM/8000
  30. a=rtpmap:101 telephone-event/8000
  31. a=fmtp:101 0-15
  32. a=ptime:20
  33. a=sendrecv
  34. m=video 10818 RTP/AVP 34
  35. c=IN IP4 61.245.172.12
  36. a=rtpmap:34 H263/90000
  37. a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
  38. a=sendrecv
  39. <------------->
  40. --- (13 headers 18 lines) ---
  41. Sending to 61.245.172.12:7035 (NAT)
  42. Using INVITE request as basis request - ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  43. Found peer '4001' for '4001' from 61.245.172.12:7035
  44.  
  45. <--- Reliably Transmitting (NAT) to 61.245.172.12:7035 --->
  46. SIP/2.0 401 Unauthorized
  47. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-4b179f68465b7a41-1---d8754z-;received=61.245.172.12;rport=7035
  48. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  49. To: <sip:19163777000@213.204.79.112:5060>;tag=as7267b889
  50. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  51. CSeq: 1 INVITE
  52. Server: FPBX-2.8.1(1.8.11.0)
  53. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  54. Supported: replaces, timer
  55. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="003f9774"
  56. Content-Length: 0
  57.  
  58.  
  59. <------------>
  60. Scheduling destruction of SIP dialog 'ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.' in 32000 ms (Method: INVITE)
  61. Retransmitting #1 (NAT) to 61.245.172.12:7035:
  62. SIP/2.0 401 Unauthorized
  63. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-4b179f68465b7a41-1---d8754z-;received=61.245.172.12;rport=7035
  64. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  65. To: <sip:19163777000@213.204.79.112:5060>;tag=as7267b889
  66. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  67. CSeq: 1 INVITE
  68. Server: FPBX-2.8.1(1.8.11.0)
  69. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  70. Supported: replaces, timer
  71. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="003f9774"
  72. Content-Length: 0
  73.  
  74.  
  75. ---
  76.  
  77. <--- SIP read from UDP:61.245.172.12:7035 --->
  78. INVITE sip:19163777000@192.168.6.16:5060 SIP/2.0
  79. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-4b179f68465b7a41-1---d8754z-;rport
  80. Max-Forwards: 70
  81. Contact: <sip:4001@10.225.20.57:7035;rinstance=b58773ece739e901>
  82. To: <sip:19163777000@213.204.79.112:5060>
  83. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  84. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  85. CSeq: 1 INVITE
  86. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
  87. Content-Type: application/sdp
  88. Supported: replaces
  89. User-Agent: 3CXPhone 6.0.26523.0
  90. Content-Length: 405
  91.  
  92. v=0
  93. o=3cxVCE 236616615 370732695 IN IP4 10.225.20.57
  94. s=3cxVCE Audio Call
  95. c=IN IP4 10.225.20.57
  96. t=0 0
  97. m=audio 40046 RTP/AVP 0 8 3 101
  98. a=rtpmap:0 PCMU/8000
  99. a=rtpmap:8 PCMA/8000
  100. a=rtpmap:3 GSM/8000
  101. a=rtpmap:101 telephone-event/8000
  102. a=fmtp:101 0-15
  103. a=ptime:20
  104. a=sendrecv
  105. m=video 10818 RTP/AVP 34
  106. c=IN IP4 61.245.172.12
  107. a=rtpmap:34 H263/90000
  108. a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
  109. a=sendrecv
  110. <------------->
  111. --- (13 headers 18 lines) ---
  112. Ignoring this INVITE request
  113.  
  114. <--- SIP read from UDP:61.245.172.12:7035 --->
  115. ACK sip:19163777000@192.168.6.16:5060 SIP/2.0
  116. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-4b179f68465b7a41-1---d8754z-;rport
  117. Max-Forwards: 70
  118. To: <sip:19163777000@213.204.79.112:5060>;tag=as7267b889
  119. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  120. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  121. CSeq: 1 ACK
  122. Content-Length: 0
  123.  
  124. <------------->
  125. --- (8 headers 0 lines) ---
  126.  
  127. <--- SIP read from UDP:61.245.172.12:7035 --->
  128. INVITE sip:19163777000@192.168.6.16:5060 SIP/2.0
  129. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-d5290454d867f370-1---d8754z-;rport
  130. Max-Forwards: 70
  131. Contact: <sip:4001@10.225.20.57:7035;rinstance=b58773ece739e901>
  132. To: <sip:19163777000@213.204.79.112:5060>
  133. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  134. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  135. CSeq: 2 INVITE
  136. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
  137. Content-Type: application/sdp
  138. Supported: replaces
  139. User-Agent: 3CXPhone 6.0.26523.0
  140. Authorization: Digest username="4001",realm="asterisk",nonce="003f9774",uri="sip:19163777000@213.204.79.112:5060",response="50fea0ccd03da711070e9e7b65479ed4",algorithm=MD5
  141. Content-Length: 405
  142.  
  143. v=0
  144. o=3cxVCE 236616615 370732695 IN IP4 10.225.20.57
  145. s=3cxVCE Audio Call
  146. c=IN IP4 10.225.20.57
  147. t=0 0
  148. m=audio 40046 RTP/AVP 0 8 3 101
  149. a=rtpmap:0 PCMU/8000
  150. a=rtpmap:8 PCMA/8000
  151. a=rtpmap:3 GSM/8000
  152. a=rtpmap:101 telephone-event/8000
  153. a=fmtp:101 0-15
  154. a=ptime:20
  155. a=sendrecv
  156. m=video 10818 RTP/AVP 34
  157. c=IN IP4 61.245.172.12
  158. a=rtpmap:34 H263/90000
  159. a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
  160. a=sendrecv
  161. <------------->
  162. --- (14 headers 18 lines) ---
  163. Sending to 61.245.172.12:7035 (NAT)
  164. Using INVITE request as basis request - ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  165. Found peer '4001' for '4001' from 61.245.172.12:7035
  166. == Using SIP RTP TOS bits 184
  167. == Using SIP RTP CoS mark 5
  168. Found RTP audio format 0
  169. Found RTP audio format 8
  170. Found RTP audio format 3
  171. Found RTP audio format 101
  172. Found audio description format PCMU for ID 0
  173. Found audio description format PCMA for ID 8
  174. Found audio description format GSM for ID 3
  175. Found audio description format telephone-event for ID 101
  176. Found RTP video format 34
  177. Found video description format H263 for ID 34
  178. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
  179. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  180. Peer audio RTP is at port 10.225.20.57:40046
  181. Looking for 19163777000 in from-internal (domain 192.168.6.16)
  182. list_route: hop: <sip:4001@10.225.20.57:7035;rinstance=b58773ece739e901>
  183.  
  184. <--- Transmitting (NAT) to 61.245.172.12:7035 --->
  185. SIP/2.0 100 Trying
  186. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-d5290454d867f370-1---d8754z-;received=61.245.172.12;rport=7035
  187. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  188. To: <sip:19163777000@213.204.79.112:5060>
  189. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  190. CSeq: 2 INVITE
  191. Server: FPBX-2.8.1(1.8.11.0)
  192. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  193. Supported: replaces, timer
  194. Contact: <sip:19163777000@192.168.6.16:5060>
  195. Content-Length: 0
  196.  
  197.  
  198. <------------>
  199. -- Executing [19163777000@from-internal:1] Macro("SIP/4001-00000050", "user-callerid,SKIPTTL,") in new stack
  200. -- Executing [s@macro-user-callerid:1] Set("SIP/4001-00000050", "AMPUSER=4001") in new stack
  201. -- Executing [s@macro-user-callerid:2] GotoIf("SIP/4001-00000050", "0?report") in new stack
  202. -- Executing [s@macro-user-callerid:3] ExecIf("SIP/4001-00000050", "1?Set(REALCALLERIDNUM=4001)") in new stack
  203. -- Executing [s@macro-user-callerid:4] Set("SIP/4001-00000050", "AMPUSER=4001") in new stack
  204. -- Executing [s@macro-user-callerid:5] Set("SIP/4001-00000050", "AMPUSERCIDNAME=4001") in new stack
  205. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/4001-00000050", "0?report") in new stack
  206. -- Executing [s@macro-user-callerid:7] Set("SIP/4001-00000050", "AMPUSERCID=4001") in new stack
  207. -- Executing [s@macro-user-callerid:8] Set("SIP/4001-00000050", "CALLERID(all)="4001" <4001>") in new stack
  208. -- Executing [s@macro-user-callerid:9] ExecIf("SIP/4001-00000050", "0?Set(CHANNEL(language)=)") in new stack
  209. -- Executing [s@macro-user-callerid:10] GotoIf("SIP/4001-00000050", "1?continue") in new stack
  210. -- Goto (macro-user-callerid,s,19)
  211. -- Executing [s@macro-user-callerid:19] Set("SIP/4001-00000050", "CALLERID(number)=4001") in new stack
  212. -- Executing [s@macro-user-callerid:20] Set("SIP/4001-00000050", "CALLERID(name)=4001") in new stack
  213. -- Executing [s@macro-user-callerid:21] NoOp("SIP/4001-00000050", "Using CallerID "4001" <4001>") in new stack
  214. -- Executing [19163777000@from-internal:2] NoOp("SIP/4001-00000050", "Calling Out Route: nymgo") in new stack
  215. -- Executing [19163777000@from-internal:3] Set("SIP/4001-00000050", "MOHCLASS=default") in new stack
  216. -- Executing [19163777000@from-internal:4] Set("SIP/4001-00000050", "_NODEST=") in new stack
  217. -- Executing [19163777000@from-internal:5] Macro("SIP/4001-00000050", "record-enable,4001,OUT,") in new stack
  218. -- Executing [s@macro-record-enable:1] GotoIf("SIP/4001-00000050", "1?check") in new stack
  219. -- Goto (macro-record-enable,s,4)
  220. -- Executing [s@macro-record-enable:4] ExecIf("SIP/4001-00000050", "0?MacroExit()") in new stack
  221. -- Executing [s@macro-record-enable:5] GotoIf("SIP/4001-00000050", "0?Group:OUT") in new stack
  222. -- Goto (macro-record-enable,s,15)
  223. -- Executing [s@macro-record-enable:15] GotoIf("SIP/4001-00000050", "0?IN") in new stack
  224. -- Executing [s@macro-record-enable:16] ExecIf("SIP/4001-00000050", "1?MacroExit()") in new stack
  225. -- Executing [19163777000@from-internal:6] Macro("SIP/4001-00000050", "dialout-trunk,3,19163777000,") in new stack
  226. -- Executing [s@macro-dialout-trunk:1] Set("SIP/4001-00000050", "DIAL_TRUNK=3") in new stack
  227. -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/4001-00000050", "0?sub-pincheck,s,1") in new stack
  228. -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/4001-00000050", "0?disabletrunk,1") in new stack
  229. -- Executing [s@macro-dialout-trunk:4] Set("SIP/4001-00000050", "DIAL_NUMBER=19163777000") in new stack
  230. -- Executing [s@macro-dialout-trunk:5] Set("SIP/4001-00000050", "DIAL_TRUNK_OPTIONS=tr") in new stack
  231. -- Executing [s@macro-dialout-trunk:6] Set("SIP/4001-00000050", "OUTBOUND_GROUP=OUT_3") in new stack
  232. -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/4001-00000050", "1?nomax") in new stack
  233. -- Goto (macro-dialout-trunk,s,9)
  234. -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/4001-00000050", "0?skipoutcid") in new stack
  235. -- Executing [s@macro-dialout-trunk:10] Set("SIP/4001-00000050", "DIAL_TRUNK_OPTIONS=") in new stack
  236. -- Executing [s@macro-dialout-trunk:11] Macro("SIP/4001-00000050", "outbound-callerid,3") in new stack
  237. -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/4001-00000050", "0?Set(CALLERPRES()=)") in new stack
  238. -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/4001-00000050", "0?Set(REALCALLERIDNUM=4001)") in new stack
  239. -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/4001-00000050", "1?normcid") in new stack
  240. -- Goto (macro-outbound-callerid,s,6)
  241. -- Executing [s@macro-outbound-callerid:6] Set("SIP/4001-00000050", "USEROUTCID=") in new stack
  242. -- Executing [s@macro-outbound-callerid:7] Set("SIP/4001-00000050", "EMERGENCYCID=") in new stack
  243. -- Executing [s@macro-outbound-callerid:8] Set("SIP/4001-00000050", "TRUNKOUTCID=94771771229") in new stack
  244. -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/4001-00000050", "1?trunkcid") in new stack
  245. -- Goto (macro-outbound-callerid,s,12)
  246. -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/4001-00000050", "1?Set(CALLERID(all)=94771771229)") in new stack
  247. -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/4001-00000050", "0?Set(CALLERID(all)=)") in new stack
  248. -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/4001-00000050", "0?Set(CALLERID(all)=)") in new stack
  249. -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/4001-00000050", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
  250. -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/4001-00000050", "1?sub-flp-3,s,1") in new stack
  251. -- Executing [s@sub-flp-3:1] ExecIf("SIP/4001-00000050", "1?Return()") in new stack
  252. -- Executing [s@macro-dialout-trunk:13] Set("SIP/4001-00000050", "OUTNUM=19163777000") in new stack
  253. -- Executing [s@macro-dialout-trunk:14] Set("SIP/4001-00000050", "custom=SIP/com-in") in new stack
  254. -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/4001-00000050", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
  255. -- Executing [s@macro-dialout-trunk:16] Macro("SIP/4001-00000050", "dialout-trunk-predial-hook,") in new stack
  256. -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/4001-00000050", "") in new stack
  257. -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/4001-00000050", "0?bypass,1") in new stack
  258. -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/4001-00000050", "0?customtrunk") in new stack
  259. -- Executing [s@macro-dialout-trunk:19] Dial("SIP/4001-00000050", "SIP/com-in/19163777000,300,") in new stack
  260. == Using SIP RTP TOS bits 184
  261. == Using SIP RTP CoS mark 5
  262. Audio is at 13978
  263. Adding codec 0x4 (ulaw) to SDP
  264. Adding non-codec 0x1 (telephone-event) to SDP
  265. Reliably Transmitting (NAT) to 176.34.131.112:5060:
  266. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  267. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK442e257c;rport
  268. Max-Forwards: 70
  269. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  270. To: <sip:19163777000@sip.commpeak.com>
  271. Contact: <sip:94771771229@192.168.6.16:5060>
  272. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  273. CSeq: 102 INVITE
  274. User-Agent: FPBX-2.8.1(1.8.11.0)
  275. Date: Mon, 10 Sep 2012 19:24:22 GMT
  276. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  277. Supported: replaces, timer
  278. Content-Type: application/sdp
  279. Content-Length: 235
  280.  
  281. v=0
  282. o=root 291704732 291704732 IN IP4 192.168.6.16
  283. s=Asterisk PBX 1.8.11.0
  284. c=IN IP4 192.168.6.16
  285. t=0 0
  286. m=audio 13978 RTP/AVP 0 101
  287. a=rtpmap:0 PCMU/8000
  288. a=rtpmap:101 telephone-event/8000
  289. a=fmtp:101 0-16
  290. a=ptime:20
  291. a=sendrecv
  292.  
  293. ---
  294. -- Called SIP/com-in/19163777000
  295.  
  296. <--- SIP read from UDP:61.245.172.12:7035 --->
  297. ACK sip:19163777000@192.168.6.16:5060 SIP/2.0
  298. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-4b179f68465b7a41-1---d8754z-;rport
  299. Max-Forwards: 70
  300. To: <sip:19163777000@213.204.79.112:5060>;tag=as7267b889
  301. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  302. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  303. CSeq: 1 ACK
  304. Content-Length: 0
  305.  
  306. <------------->
  307. --- (8 headers 0 lines) ---
  308.  
  309. <--- SIP read from UDP:176.34.131.112:5060 --->
  310. SIP/2.0 100 Trying...
  311. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK442e257c;rport=4831
  312. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  313. To: <sip:19163777000@sip.commpeak.com>
  314. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  315. CSeq: 102 INVITE
  316. Server: CommPeak SIP Proxy
  317. Content-Length: 0
  318.  
  319. <------------->
  320. --- (8 headers 0 lines) ---
  321.  
  322. <--- SIP read from UDP:176.34.131.112:5060 --->
  323. SIP/2.0 407 Proxy Authentication Required
  324. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK442e257c;rport=4831
  325. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  326. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.1431
  327. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  328. CSeq: 102 INVITE
  329. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e3e8400000074350b2724d33b9608a9a59aab70fadb5d"
  330. Server: CommPeak SIP Proxy
  331. Content-Length: 0
  332.  
  333. <------------->
  334. --- (9 headers 0 lines) ---
  335. set_destination: Parsing <sip:19163777000@sip.commpeak.com> for address/port to send to
  336. set_destination: set destination to 176.34.131.112:5060
  337. Transmitting (NAT) to 176.34.131.112:5060:
  338. ACK sip:19163777000@sip.commpeak.com SIP/2.0
  339. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK442e257c;rport
  340. Max-Forwards: 70
  341. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  342. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.1431
  343. Contact: <sip:94771771229@192.168.6.16:5060>
  344. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  345. CSeq: 102 ACK
  346. User-Agent: FPBX-2.8.1(1.8.11.0)
  347. Content-Length: 0
  348.  
  349.  
  350. ---
  351. Audio is at 13978
  352. Adding codec 0x4 (ulaw) to SDP
  353. Adding non-codec 0x1 (telephone-event) to SDP
  354. Reliably Transmitting (NAT) to 176.34.131.112:5060:
  355. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  356. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK2b753533;rport
  357. Max-Forwards: 70
  358. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  359. To: <sip:19163777000@sip.commpeak.com>
  360. Contact: <sip:94771771229@192.168.6.16:5060>
  361. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  362. CSeq: 103 INVITE
  363. User-Agent: FPBX-2.8.1(1.8.11.0)
  364. Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e3e8400000074350b2724d33b9608a9a59aab70fadb5d", response="72af0221f960fb1d2194c51edcc6fee2"
  365. Date: Mon, 10 Sep 2012 19:24:22 GMT
  366. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  367. Supported: replaces, timer
  368. Content-Type: application/sdp
  369. Content-Length: 235
  370.  
  371. v=0
  372. o=root 291704732 291704733 IN IP4 192.168.6.16
  373. s=Asterisk PBX 1.8.11.0
  374. c=IN IP4 192.168.6.16
  375. t=0 0
  376. m=audio 13978 RTP/AVP 0 101
  377. a=rtpmap:0 PCMU/8000
  378. a=rtpmap:101 telephone-event/8000
  379. a=fmtp:101 0-16
  380. a=ptime:20
  381. a=sendrecv
  382.  
  383. ---
  384.  
  385. <--- SIP read from UDP:61.245.172.12:7035 --->
  386. INVITE sip:19163777000@192.168.6.16:5060 SIP/2.0
  387. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-d5290454d867f370-1---d8754z-;rport
  388. Max-Forwards: 70
  389. Contact: <sip:4001@10.225.20.57:7035;rinstance=b58773ece739e901>
  390. To: <sip:19163777000@213.204.79.112:5060>
  391. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  392. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  393. CSeq: 2 INVITE
  394. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
  395. Content-Type: application/sdp
  396. Supported: replaces
  397. User-Agent: 3CXPhone 6.0.26523.0
  398. Authorization: Digest username="4001",realm="asterisk",nonce="003f9774",uri="sip:19163777000@213.204.79.112:5060",response="50fea0ccd03da711070e9e7b65479ed4",algorithm=MD5
  399. Content-Length: 405
  400.  
  401. v=0
  402. o=3cxVCE 236616615 370732695 IN IP4 10.225.20.57
  403. s=3cxVCE Audio Call
  404. c=IN IP4 10.225.20.57
  405. t=0 0
  406. m=audio 40046 RTP/AVP 0 8 3 101
  407. a=rtpmap:0 PCMU/8000
  408. a=rtpmap:8 PCMA/8000
  409. a=rtpmap:3 GSM/8000
  410. a=rtpmap:101 telephone-event/8000
  411. a=fmtp:101 0-15
  412. a=ptime:20
  413. a=sendrecv
  414. m=video 10818 RTP/AVP 34
  415. c=IN IP4 61.245.172.12
  416. a=rtpmap:34 H263/90000
  417. a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
  418. a=sendrecv
  419. <------------->
  420. --- (14 headers 18 lines) ---
  421. Ignoring this INVITE request
  422.  
  423. <--- Transmitting (NAT) to 61.245.172.12:7035 --->
  424. SIP/2.0 100 Trying
  425. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-d5290454d867f370-1---d8754z-;received=61.245.172.12;rport=7035
  426. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  427. To: <sip:19163777000@213.204.79.112:5060>
  428. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  429. CSeq: 2 INVITE
  430. Server: FPBX-2.8.1(1.8.11.0)
  431. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  432. Supported: replaces, timer
  433. Contact: <sip:19163777000@192.168.6.16:5060>
  434. Content-Length: 0
  435.  
  436.  
  437. <------------>
  438. Retransmitting #1 (NAT) to 176.34.131.112:5060:
  439. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  440. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK2b753533;rport
  441. Max-Forwards: 70
  442. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  443. To: <sip:19163777000@sip.commpeak.com>
  444. Contact: <sip:94771771229@192.168.6.16:5060>
  445. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  446. CSeq: 103 INVITE
  447. User-Agent: FPBX-2.8.1(1.8.11.0)
  448. Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e3e8400000074350b2724d33b9608a9a59aab70fadb5d", response="72af0221f960fb1d2194c51edcc6fee2"
  449. Date: Mon, 10 Sep 2012 19:24:22 GMT
  450. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  451. Supported: replaces, timer
  452. Content-Type: application/sdp
  453. Content-Length: 235
  454.  
  455. v=0
  456. o=root 291704732 291704733 IN IP4 192.168.6.16
  457. s=Asterisk PBX 1.8.11.0
  458. c=IN IP4 192.168.6.16
  459. t=0 0
  460. m=audio 13978 RTP/AVP 0 101
  461. a=rtpmap:0 PCMU/8000
  462. a=rtpmap:101 telephone-event/8000
  463. a=fmtp:101 0-16
  464. a=ptime:20
  465. a=sendrecv
  466.  
  467. ---
  468.  
  469. <--- SIP read from UDP:176.34.131.112:5060 --->
  470. SIP/2.0 100 Trying...
  471. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK2b753533;rport=4831
  472. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  473. To: <sip:19163777000@sip.commpeak.com>
  474. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  475. CSeq: 103 INVITE
  476. Server: CommPeak SIP Proxy
  477. Content-Length: 0
  478.  
  479. <------------->
  480. --- (8 headers 0 lines) ---
  481.  
  482. <--- SIP read from UDP:176.34.131.112:5060 --->
  483. SIP/2.0 407 Proxy Authentication Required
  484. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK2b753533;rport=4831
  485. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  486. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.94c6
  487. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  488. CSeq: 103 INVITE
  489. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e3e84000000751babb5dfce37be9e2023c5c477b9f9fe"
  490. Server: CommPeak SIP Proxy
  491. Content-Length: 0
  492.  
  493. <------------->
  494. --- (9 headers 0 lines) ---
  495. set_destination: Parsing <sip:19163777000@sip.commpeak.com> for address/port to send to
  496. set_destination: set destination to 176.34.131.112:5060
  497. Transmitting (NAT) to 176.34.131.112:5060:
  498. ACK sip:19163777000@sip.commpeak.com SIP/2.0
  499. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK2b753533;rport
  500. Max-Forwards: 70
  501. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  502. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.94c6
  503. Contact: <sip:94771771229@192.168.6.16:5060>
  504. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  505. CSeq: 103 ACK
  506. User-Agent: FPBX-2.8.1(1.8.11.0)
  507. Content-Length: 0
  508.  
  509.  
  510. ---
  511. Audio is at 13978
  512. Adding codec 0x4 (ulaw) to SDP
  513. Adding non-codec 0x1 (telephone-event) to SDP
  514. Reliably Transmitting (NAT) to 176.34.131.112:5060:
  515. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  516. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK54335007;rport
  517. Max-Forwards: 70
  518. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  519. To: <sip:19163777000@sip.commpeak.com>
  520. Contact: <sip:94771771229@192.168.6.16:5060>
  521. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  522. CSeq: 104 INVITE
  523. User-Agent: FPBX-2.8.1(1.8.11.0)
  524. Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e3e84000000751babb5dfce37be9e2023c5c477b9f9fe", response="36cb3db684b2806b1334c710782db80c"
  525. Date: Mon, 10 Sep 2012 19:24:23 GMT
  526. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  527. Supported: replaces, timer
  528. Content-Type: application/sdp
  529. Content-Length: 235
  530.  
  531. v=0
  532. o=root 291704732 291704734 IN IP4 192.168.6.16
  533. s=Asterisk PBX 1.8.11.0
  534. c=IN IP4 192.168.6.16
  535. t=0 0
  536. m=audio 13978 RTP/AVP 0 101
  537. a=rtpmap:0 PCMU/8000
  538. a=rtpmap:101 telephone-event/8000
  539. a=fmtp:101 0-16
  540. a=ptime:20
  541. a=sendrecv
  542.  
  543. ---
  544.  
  545. <--- SIP read from UDP:176.34.131.112:5060 --->
  546. SIP/2.0 100 Trying...
  547. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK2b753533;rport=4831
  548. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  549. To: <sip:19163777000@sip.commpeak.com>
  550. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  551. CSeq: 103 INVITE
  552. Server: CommPeak SIP Proxy
  553. Content-Length: 0
  554.  
  555. <------------->
  556. --- (8 headers 0 lines) ---
  557.  
  558. <--- SIP read from UDP:176.34.131.112:5060 --->
  559. SIP/2.0 407 Proxy Authentication Required
  560. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK2b753533;rport=4831
  561. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  562. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.94c6
  563. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  564. CSeq: 103 INVITE
  565. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e3e850000007695524b15c825a07cedf9eaed4580b599"
  566. Server: CommPeak SIP Proxy
  567. Content-Length: 0
  568.  
  569. <------------->
  570. --- (9 headers 0 lines) ---
  571. Transmitting (NAT) to 176.34.131.112:5060:
  572. ACK sip:19163777000@sip.commpeak.com SIP/2.0
  573. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK54335007;rport
  574. Max-Forwards: 70
  575. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  576. To: <sip:19163777000@sip.commpeak.com>
  577. Contact: <sip:94771771229@192.168.6.16:5060>
  578. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  579. CSeq: 103 ACK
  580. User-Agent: FPBX-2.8.1(1.8.11.0)
  581. Content-Length: 0
  582.  
  583.  
  584. ---
  585. Retransmitting #1 (NAT) to 176.34.131.112:5060:
  586. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  587. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK54335007;rport
  588. Max-Forwards: 70
  589. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  590. To: <sip:19163777000@sip.commpeak.com>
  591. Contact: <sip:94771771229@192.168.6.16:5060>
  592. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  593. CSeq: 104 INVITE
  594. User-Agent: FPBX-2.8.1(1.8.11.0)
  595. Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e3e84000000751babb5dfce37be9e2023c5c477b9f9fe", response="36cb3db684b2806b1334c710782db80c"
  596. Date: Mon, 10 Sep 2012 19:24:23 GMT
  597. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  598. Supported: replaces, timer
  599. Content-Type: application/sdp
  600. Content-Length: 235
  601.  
  602. v=0
  603. o=root 291704732 291704734 IN IP4 192.168.6.16
  604. s=Asterisk PBX 1.8.11.0
  605. c=IN IP4 192.168.6.16
  606. t=0 0
  607. m=audio 13978 RTP/AVP 0 101
  608. a=rtpmap:0 PCMU/8000
  609. a=rtpmap:101 telephone-event/8000
  610. a=fmtp:101 0-16
  611. a=ptime:20
  612. a=sendrecv
  613.  
  614. ---
  615.  
  616. <--- SIP read from UDP:176.34.131.112:5060 --->
  617. SIP/2.0 100 Trying...
  618. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK54335007;rport=4831
  619. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  620. To: <sip:19163777000@sip.commpeak.com>
  621. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  622. CSeq: 104 INVITE
  623. Server: CommPeak SIP Proxy
  624. Content-Length: 0
  625.  
  626. <------------->
  627. --- (8 headers 0 lines) ---
  628.  
  629. <--- SIP read from UDP:176.34.131.112:5060 --->
  630. SIP/2.0 407 Proxy Authentication Required
  631. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK54335007;rport=4831
  632. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  633. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.a1fe
  634. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  635. CSeq: 104 INVITE
  636. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e3e85000000774a097b7deed3d91698a5cc7fb8bc4c11"
  637. Server: CommPeak SIP Proxy
  638. Content-Length: 0
  639.  
  640. <------------->
  641. --- (9 headers 0 lines) ---
  642. Transmitting (NAT) to 176.34.131.112:5060:
  643. ACK sip:19163777000@sip.commpeak.com SIP/2.0
  644. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK54335007;rport
  645. Max-Forwards: 70
  646. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  647. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.a1fe
  648. Contact: <sip:94771771229@192.168.6.16:5060>
  649. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  650. CSeq: 104 ACK
  651. User-Agent: FPBX-2.8.1(1.8.11.0)
  652. Content-Length: 0
  653.  
  654.  
  655. ---
  656. Audio is at 13978
  657. Adding codec 0x4 (ulaw) to SDP
  658. Adding non-codec 0x1 (telephone-event) to SDP
  659. Reliably Transmitting (NAT) to 176.34.131.112:5060:
  660. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  661. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK44f1b054;rport
  662. Max-Forwards: 70
  663. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  664. To: <sip:19163777000@sip.commpeak.com>
  665. Contact: <sip:94771771229@192.168.6.16:5060>
  666. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  667. CSeq: 105 INVITE
  668. User-Agent: FPBX-2.8.1(1.8.11.0)
  669. Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e3e85000000774a097b7deed3d91698a5cc7fb8bc4c11", response="d1136b69ff92ac608ea65cb90abc2e8b"
  670. Date: Mon, 10 Sep 2012 19:24:23 GMT
  671. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  672. Supported: replaces, timer
  673. Content-Type: application/sdp
  674. Content-Length: 235
  675.  
  676. v=0
  677. o=root 291704732 291704735 IN IP4 192.168.6.16
  678. s=Asterisk PBX 1.8.11.0
  679. c=IN IP4 192.168.6.16
  680. t=0 0
  681. m=audio 13978 RTP/AVP 0 101
  682. a=rtpmap:0 PCMU/8000
  683. a=rtpmap:101 telephone-event/8000
  684. a=fmtp:101 0-16
  685. a=ptime:20
  686. a=sendrecv
  687.  
  688. ---
  689.  
  690. <--- SIP read from UDP:176.34.131.112:5060 --->
  691. SIP/2.0 100 Trying...
  692. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK54335007;rport=4831
  693. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  694. To: <sip:19163777000@sip.commpeak.com>
  695. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  696. CSeq: 104 INVITE
  697. Server: CommPeak SIP Proxy
  698. Content-Length: 0
  699.  
  700. <------------->
  701. --- (8 headers 0 lines) ---
  702.  
  703. <--- SIP read from UDP:176.34.131.112:5060 --->
  704. SIP/2.0 407 Proxy Authentication Required
  705. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK54335007;rport=4831
  706. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  707. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.a1fe
  708. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  709. CSeq: 104 INVITE
  710. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e3e8500000078e4b5d12c3bb8e52441fe9ebe0227ff6c"
  711. Server: CommPeak SIP Proxy
  712. Content-Length: 0
  713.  
  714. <------------->
  715. --- (9 headers 0 lines) ---
  716. Transmitting (NAT) to 176.34.131.112:5060:
  717. ACK sip:19163777000@sip.commpeak.com SIP/2.0
  718. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK44f1b054;rport
  719. Max-Forwards: 70
  720. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  721. To: <sip:19163777000@sip.commpeak.com>
  722. Contact: <sip:94771771229@192.168.6.16:5060>
  723. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  724. CSeq: 104 ACK
  725. User-Agent: FPBX-2.8.1(1.8.11.0)
  726. Content-Length: 0
  727.  
  728.  
  729. ---
  730. Retransmitting #1 (NAT) to 176.34.131.112:5060:
  731. INVITE sip:19163777000@sip.commpeak.com SIP/2.0
  732. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK44f1b054;rport
  733. Max-Forwards: 70
  734. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  735. To: <sip:19163777000@sip.commpeak.com>
  736. Contact: <sip:94771771229@192.168.6.16:5060>
  737. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  738. CSeq: 105 INVITE
  739. User-Agent: FPBX-2.8.1(1.8.11.0)
  740. Proxy-Authorization: Digest username="lasantha", realm="192.168.6.16", algorithm=MD5, uri="sip:19163777000@sip.commpeak.com", nonce="504e3e85000000774a097b7deed3d91698a5cc7fb8bc4c11", response="d1136b69ff92ac608ea65cb90abc2e8b"
  741. Date: Mon, 10 Sep 2012 19:24:23 GMT
  742. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  743. Supported: replaces, timer
  744. Content-Type: application/sdp
  745. Content-Length: 235
  746.  
  747. v=0
  748. o=root 291704732 291704735 IN IP4 192.168.6.16
  749. s=Asterisk PBX 1.8.11.0
  750. c=IN IP4 192.168.6.16
  751. t=0 0
  752. m=audio 13978 RTP/AVP 0 101
  753. a=rtpmap:0 PCMU/8000
  754. a=rtpmap:101 telephone-event/8000
  755. a=fmtp:101 0-16
  756. a=ptime:20
  757. a=sendrecv
  758.  
  759. ---
  760.  
  761. <--- SIP read from UDP:176.34.131.112:5060 --->
  762. SIP/2.0 100 Trying...
  763. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK44f1b054;rport=4831
  764. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  765. To: <sip:19163777000@sip.commpeak.com>
  766. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  767. CSeq: 105 INVITE
  768. Server: CommPeak SIP Proxy
  769. Content-Length: 0
  770.  
  771. <------------->
  772. --- (8 headers 0 lines) ---
  773.  
  774. <--- SIP read from UDP:176.34.131.112:5060 --->
  775. SIP/2.0 407 Proxy Authentication Required
  776. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK44f1b054;rport=4831
  777. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  778. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.fd2e
  779. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  780. CSeq: 105 INVITE
  781. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e3e85000000797f5563e8b70bb39d0c9094dd0223eda9"
  782. Server: CommPeak SIP Proxy
  783. Content-Length: 0
  784.  
  785. <------------->
  786. --- (9 headers 0 lines) ---
  787. Transmitting (NAT) to 176.34.131.112:5060:
  788. ACK sip:19163777000@sip.commpeak.com SIP/2.0
  789. Via: SIP/2.0/UDP 192.168.6.16:5060;branch=z9hG4bK44f1b054;rport
  790. Max-Forwards: 70
  791. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  792. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.fd2e
  793. Contact: <sip:94771771229@192.168.6.16:5060>
  794. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  795. CSeq: 105 ACK
  796. User-Agent: FPBX-2.8.1(1.8.11.0)
  797. Content-Length: 0
  798.  
  799.  
  800. ---
  801. -- SIP/com-in-00000051 is circuit-busy
  802. == Everyone is busy/congested at this time (1:0/1/0)
  803. -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/4001-00000050", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21") in new stack
  804. -- Executing [s@macro-dialout-trunk:21] Goto("SIP/4001-00000050", "s-CONGESTION,1") in new stack
  805. -- Goto (macro-dialout-trunk,s-CONGESTION,1)
  806. -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/4001-00000050", "RC=21") in new stack
  807. -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/4001-00000050", "21,1") in new stack
  808. -- Goto (macro-dialout-trunk,21,1)
  809. -- Executing [21@macro-dialout-trunk:1] Goto("SIP/4001-00000050", "continue,1") in new stack
  810. -- Goto (macro-dialout-trunk,continue,1)
  811. -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/4001-00000050", "1?noreport") in new stack
  812. -- Goto (macro-dialout-trunk,continue,3)
  813. -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/4001-00000050", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks") in new stack
  814. -- Executing [continue@macro-dialout-trunk:4] Set("SIP/4001-00000050", "CALLERID(number)=4001") in new stack
  815. -- Executing [19163777000@from-internal:7] Macro("SIP/4001-00000050", "outisbusy,") in new stack
  816. -- Executing [s@macro-outisbusy:1] Progress("SIP/4001-00000050", "") in new stack
  817. Audio is at 18518
  818. Adding codec 0x4 (ulaw) to SDP
  819. Adding codec 0x8 (alaw) to SDP
  820. Adding codec 0x2 (gsm) to SDP
  821. Adding non-codec 0x1 (telephone-event) to SDP
  822.  
  823. <--- Transmitting (NAT) to 61.245.172.12:7035 --->
  824. SIP/2.0 183 Session Progress
  825. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-d5290454d867f370-1---d8754z-;received=61.245.172.12;rport=7035
  826. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  827. To: <sip:19163777000@213.204.79.112:5060>;tag=as721d1c71
  828. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  829. CSeq: 2 INVITE
  830. Server: FPBX-2.8.1(1.8.11.0)
  831. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  832. Supported: replaces, timer
  833. Contact: <sip:19163777000@192.168.6.16:5060>
  834. Content-Type: application/sdp
  835. Content-Length: 304
  836.  
  837. v=0
  838. o=root 136742573 136742573 IN IP4 192.168.6.16
  839. s=Asterisk PBX 1.8.11.0
  840. c=IN IP4 192.168.6.16
  841. t=0 0
  842. m=audio 18518 RTP/AVP 0 8 3 101
  843. a=rtpmap:0 PCMU/8000
  844. a=rtpmap:8 PCMA/8000
  845. a=rtpmap:3 GSM/8000
  846. a=rtpmap:101 telephone-event/8000
  847. a=fmtp:101 0-16
  848. a=ptime:20
  849. a=sendrecv
  850. m=video 0 RTP/AVP 34
  851.  
  852. <------------>
  853. -- Executing [s@macro-outisbusy:2] GotoIf("SIP/4001-00000050", "0?emergency,1") in new stack
  854. -- Executing [s@macro-outisbusy:3] GotoIf("SIP/4001-00000050", "0?intracompany,1") in new stack
  855. -- Executing [s@macro-outisbusy:4] Playback("SIP/4001-00000050", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
  856. -- <SIP/4001-00000050> Playing 'all-circuits-busy-now.gsm' (language 'en')
  857.  
  858. <--- SIP read from UDP:176.34.131.112:5060 --->
  859. SIP/2.0 100 Trying...
  860. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK44f1b054;rport=4831
  861. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  862. To: <sip:19163777000@sip.commpeak.com>
  863. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  864. CSeq: 105 INVITE
  865. Server: CommPeak SIP Proxy
  866. Content-Length: 0
  867.  
  868. <------------->
  869. --- (8 headers 0 lines) ---
  870. Really destroying SIP dialog '1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060' Method: INVITE
  871.  
  872. <--- SIP read from UDP:176.34.131.112:5060 --->
  873. SIP/2.0 407 Proxy Authentication Required
  874. Via: SIP/2.0/UDP 192.168.6.16:5060;received=213.204.79.112;branch=z9hG4bK44f1b054;rport=4831
  875. From: "94771771229" <sip:94771771229@192.168.6.16>;tag=as739b7715
  876. To: <sip:19163777000@sip.commpeak.com>;tag=49d06d22b8ff758f6dbceab7ca02b15d.fd2e
  877. Call-ID: 1eec82a21ae31aac04fb5f4a612f1956@192.168.6.16:5060
  878. CSeq: 105 INVITE
  879. Proxy-Authenticate: Digest realm="192.168.6.16", nonce="504e3e850000007aebf1e512d7da67c75b708ecf6fee5c57"
  880. Server: CommPeak SIP Proxy
  881. Content-Length: 0
  882.  
  883. <------------->
  884. --- (9 headers 0 lines) ---
  885. -- <SIP/4001-00000050> Playing 'pls-try-call-later.gsm' (language 'en')
  886. -- Executing [s@macro-outisbusy:5] Congestion("SIP/4001-00000050", "20") in new stack
  887.  
  888. <--- Reliably Transmitting (NAT) to 61.245.172.12:7035 --->
  889. SIP/2.0 503 Service Unavailable
  890. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-d5290454d867f370-1---d8754z-;received=61.245.172.12;rport=7035
  891. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  892. To: <sip:19163777000@213.204.79.112:5060>;tag=as721d1c71
  893. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  894. CSeq: 2 INVITE
  895. Server: FPBX-2.8.1(1.8.11.0)
  896. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  897. Supported: replaces, timer
  898. X-Asterisk-HangupCause: Call Rejected
  899. X-Asterisk-HangupCauseCode: 21
  900. Content-Length: 0
  901.  
  902.  
  903. <------------>
  904. == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/4001-00000050' in macro 'outisbusy'
  905. == Spawn extension (from-internal, 19163777000, 7) exited non-zero on 'SIP/4001-00000050'
  906. -- Executing [h@from-internal:1] Macro("SIP/4001-00000050", "hangupcall") in new stack
  907. -- Executing [s@macro-hangupcall:1] GotoIf("SIP/4001-00000050", "1?endmixmoncheck") in new stack
  908. -- Goto (macro-hangupcall,s,9)
  909. -- Executing [s@macro-hangupcall:9] NoOp("SIP/4001-00000050", "End of MIXMON check") in new stack
  910. -- Executing [s@macro-hangupcall:10] GotoIf("SIP/4001-00000050", "1?nomeetmemon") in new stack
  911. -- Goto (macro-hangupcall,s,28)
  912. -- Executing [s@macro-hangupcall:28] NoOp("SIP/4001-00000050", "End of MEETME check") in new stack
  913. -- Executing [s@macro-hangupcall:29] GotoIf("SIP/4001-00000050", "1?noautomon") in new stack
  914. -- Goto (macro-hangupcall,s,34)
  915. -- Executing [s@macro-hangupcall:34] NoOp("SIP/4001-00000050", "TOUCH_MONITOR_OUTPUT=") in new stack
  916. -- Executing [s@macro-hangupcall:35] GotoIf("SIP/4001-00000050", "1?noautomon2") in new stack
  917. -- Goto (macro-hangupcall,s,41)
  918. -- Executing [s@macro-hangupcall:41] NoOp("SIP/4001-00000050", "MONITOR_FILENAME=") in new stack
  919. -- Executing [s@macro-hangupcall:42] GotoIf("SIP/4001-00000050", "1?skiprg") in new stack
  920. -- Goto (macro-hangupcall,s,45)
  921. -- Executing [s@macro-hangupcall:45] GotoIf("SIP/4001-00000050", "1?skipblkvm") in new stack
  922. -- Goto (macro-hangupcall,s,48)
  923. -- Executing [s@macro-hangupcall:48] GotoIf("SIP/4001-00000050", "1?theend") in new stack
  924. -- Goto (macro-hangupcall,s,50)
  925. -- Executing [s@macro-hangupcall:50] AGI("SIP/4001-00000050", "hangup.agi") in new stack
  926. -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
  927. -- <SIP/4001-00000050>AGI Script hangup.agi completed, returning 0
  928. -- Executing [s@macro-hangupcall:51] Hangup("SIP/4001-00000050", "") in new stack
  929. == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/4001-00000050' in macro 'hangupcall'
  930. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/4001-00000050'
  931. Retransmitting #1 (NAT) to 61.245.172.12:7035:
  932. SIP/2.0 503 Service Unavailable
  933. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-d5290454d867f370-1---d8754z-;received=61.245.172.12;rport=7035
  934. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  935. To: <sip:19163777000@213.204.79.112:5060>;tag=as721d1c71
  936. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  937. CSeq: 2 INVITE
  938. Server: FPBX-2.8.1(1.8.11.0)
  939. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  940. Supported: replaces, timer
  941. X-Asterisk-HangupCause: Call Rejected
  942. X-Asterisk-HangupCauseCode: 21
  943. Content-Length: 0
  944.  
  945.  
  946. ---
  947.  
  948. <--- SIP read from UDP:61.245.172.12:7035 --->
  949. ACK sip:19163777000@192.168.6.16:5060 SIP/2.0
  950. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-d5290454d867f370-1---d8754z-;rport
  951. Max-Forwards: 70
  952. To: <sip:19163777000@213.204.79.112:5060>;tag=as721d1c71
  953. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  954. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  955. CSeq: 2 ACK
  956. Content-Length: 0
  957.  
  958. <------------->
  959. --- (8 headers 0 lines) ---
  960. Really destroying SIP dialog 'ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.' Method: ACK
  961.  
  962. <--- SIP read from UDP:61.245.172.12:7035 --->
  963. ACK sip:19163777000@192.168.6.16:5060 SIP/2.0
  964. Via: SIP/2.0/UDP 10.225.20.57:64939;branch=z9hG4bK-d8754z-d5290454d867f370-1---d8754z-;rport
  965. Max-Forwards: 70
  966. To: <sip:19163777000@213.204.79.112:5060>;tag=as721d1c71
  967. From: "4001"<sip:4001@213.204.79.112:5060>;tag=8d7bed74
  968. Call-ID: ZTQ0ZmEzN2ZjNTBhNTU3ZGZiNWY3YzQyMjRlNzA1NWM.
  969. CSeq: 2 ACK
  970. Content-Length: 0
  971.  
  972. <------------->
  973. --- (8 headers 0 lines) ---
  974.  
  975. <--- SIP read from UDP:61.245.172.12:7035 --->
  976.  
  977.  
  978. <------------->
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