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  1. -- ast_get_srv: SRV lookup for '_sip._UDP.sipgate.com' mapped to host sipgate.com, port 5060
  2. REGISTER 13 headers, 0 lines
  3. Reliably Transmitting (NAT) to 204.155.28.10:5060:
  4. REGISTER sip:sipgate.com SIP/2.0
  5. Via: SIP/2.0/UDP 192.168.15.230:5060;branch=z9hG4bK7f0e5c8c;rport
  6. Max-Forwards: 70
  7. From: <sip:1082258e1@sipgate.com>;tag=as715c08a3
  8. To: <sip:1082258e1@sipgate.com>
  9. Call-ID: 584803a17e58d5b972c0f99476772e14@127.0.0.1
  10. CSeq: 109 REGISTER
  11. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  12. Authorization: Digest username="1082258e1", realm="sipgate.com", algorithm=MD5, uri="sip:sipgate.com", nonce="4c9bec277f14fcaa74f3ef8fe574481dc42b3d8a", response="a5ecdbc12313d8e0bee668fc055b02cf"
  13. Expires: 120
  14. Contact: <sip:1082258e1@192.168.15.230>
  15. Event: registration
  16. Content-Length: 0
  17.  
  18.  
  19. ---
  20. Husky*CLI>
  21. Husky*CLI>
  22. Husky*CLI>
  23. Husky*CLI>
  24. Husky*CLI>
  25. <--- SIP read from UDP://204.155.28.10:5060 --->
  26. SIP/2.0 200 OK
  27. Via: SIP/2.0/UDP 192.168.15.230:5060;received=192.168.15.230;branch=z9hG4bK7f0e5c8c;rport=1268
  28. From: <sip:1082258e1@sipgate.com>;tag=as715c08a3
  29. To: <sip:1082258e1@sipgate.com>;tag=ddf9a9f0de4595ec12441b9d8d3ee250.cfa6
  30. Call-ID: 584803a17e58d5b972c0f99476772e14@127.0.0.1
  31. CSeq: 109 REGISTER
  32. Contact: <sip:1082258e1@192.168.15.230:5060>;expires=120;received="sip:100.50.58.105:1268"
  33. Content-Length: 0
  34.  
  35.  
  36. <------------->
  37. --- (8 headers 0 lines) ---
  38. Scheduling destruction of SIP dialog '584803a17e58d5b972c0f99476772e14@127.0.0.1' in 32000 ms (Method: REGISTER)
  39. Husky*CLI>
  40. Husky*CLI>
  41. Husky*CLI>
  42. Husky*CLI>
  43. Husky*CLI>
  44. Husky*CLI>
  45. Husky*CLI>
  46. Husky*CLI>
  47. <--- SIP read from UDP://204.155.28.10:5060 --->
  48.  
  49. <------------->
  50. Husky*CLI>
  51. <--- SIP read from UDP://204.155.28.10:5060 --->
  52. INVITE sip:1082258e1@192.168.15.230:5060 SIP/2.0
  53. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  54. Record-Route: <sip:172.30.20.2;lr=on>
  55. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  56. Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0
  57. Via: SIP/2.0/UDP 172.30.20.2;branch=z9hG4bKd7cc.24486865.0
  58. Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.21;branch=z9hG4bK159d242b
  59. Via: SIP/2.0/UDP 204.155.29.57:5060;received=204.155.29.57;branch=z9hG4bK159d242b;rport=5060
  60. Max-Forwards: 67
  61. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  62. To: <sip:1082258e1@sipgate.com>
  63. Contact: <sip:3048205742@204.155.29.57>
  64. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  65. CSeq: 103 INVITE
  66. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  67. Supported: replaces, timer
  68. Content-Type: application/sdp
  69. Content-Length: 431
  70.  
  71. v=0
  72. o=root 1399517662 1399517663 IN IP4 204.155.29.57
  73. s=sipgate VoIP GW
  74. c=IN IP4 204.155.29.56
  75. t=0 0
  76. m=audio 64620 RTP/AVP 0 8 3 97 18 112 101
  77. a=rtpmap:0 PCMU/8000
  78. a=rtpmap:8 PCMA/8000
  79. a=rtpmap:3 GSM/8000
  80. a=rtpmap:97 iLBC/8000
  81. a=rtpmap:18 G729/8000
  82. a=fmtp:18 annexb=no
  83. a=rtpmap:112 G726-32/8000
  84. a=rtpmap:101 telephone-event/8000
  85. a=fmtp:101 0-16
  86. a=silenceSupp:off - - - -
  87. a=ptime:20
  88. a=sendrecv
  89. a=direction:active
  90.  
  91. <------------->
  92. --- (18 headers 19 lines) ---
  93. == Using SIP RTP TOS bits 184
  94. == Using SIP RTP CoS mark 5
  95. == Using SIP VRTP TOS bits 136
  96. == Using SIP VRTP CoS mark 6
  97. Sending to 204.155.28.10 : 5060 (NAT)
  98. Using INVITE request as basis request - 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  99. No user '3048205742' in SIP users list
  100. Found peer 'SipGate' for '3048205742' from 204.155.28.10:5060
  101. Found RTP audio format 0
  102. Found RTP audio format 8
  103. Found RTP audio format 3
  104. Found RTP audio format 97
  105. Found RTP audio format 18
  106. Found RTP audio format 112
  107. Found RTP audio format 101
  108. Found audio description format PCMU for ID 0
  109. Found audio description format PCMA for ID 8
  110. Found audio description format GSM for ID 3
  111. Found audio description format iLBC for ID 97
  112. Found audio description format G729 for ID 18
  113. Found audio description format G726-32 for ID 112
  114. Found audio description format telephone-event for ID 101
  115. Capabilities: us - 0x104 (ulaw|g729), peer - audio=0xd0e (gsm|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
  116. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  117. Peer audio RTP is at port 204.155.29.56:64620
  118. Looking for 1082258e1 in from-pstn (domain 192.168.15.230)
  119. list_route: hop: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  120. list_route: hop: <sip:172.30.20.2;lr=on>
  121. list_route: hop: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  122.  
  123. <--- Transmitting (NAT) to 204.155.28.10:5060 --->
  124. SIP/2.0 100 Trying
  125. Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0;received=204.155.28.10
  126. Via: SIP/2.0/UDP 172.30.20.2;branch=z9hG4bKd7cc.24486865.0
  127. Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.21;branch=z9hG4bK159d242b
  128. Via: SIP/2.0/UDP 204.155.29.57:5060;received=204.155.29.57;branch=z9hG4bK159d242b;rport=5060
  129. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  130. Record-Route: <sip:172.30.20.2;lr=on>
  131. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  132. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  133. To: <sip:1082258e1@sipgate.com>
  134. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  135. CSeq: 103 INVITE
  136. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  137. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  138. Supported: replaces, timer
  139. Require: timer
  140. Session-Expires: 1800;refresher=uas
  141. Contact: <sip:1082258e1@192.168.15.230>
  142. Content-Length: 0
  143.  
  144.  
  145. <------------>
  146. -- Executing [1082258e1@from-pstn:1] NoOp("SIP/SipGate-00000037", "Catch-All DID Match - Found 1082258e1 - You probably want a DID for this.") in new stack
  147. -- Executing [1082258e1@from-pstn:2] Goto("SIP/SipGate-00000037", "ext-did,s,1") in new stack
  148. -- Goto (ext-did,s,1)
  149. -- Executing [s@ext-did:1] Set("SIP/SipGate-00000037", "__FROM_DID=s") in new stack
  150. -- Executing [s@ext-did:2] Gosub("SIP/SipGate-00000037", "app-blacklist-check,s,1") in new stack
  151. -- Executing [s@app-blacklist-check:1] GotoIf("SIP/SipGate-00000037", "0?blacklisted") in new stack
  152. -- Executing [s@app-blacklist-check:2] Return("SIP/SipGate-00000037", "") in new stack
  153. -- Executing [s@ext-did:3] ExecIf("SIP/SipGate-00000037", "0 ?Set(CALLERID(name)=3048205742)") in new stack
  154. -- Executing [s@ext-did:4] Set("SIP/SipGate-00000037", "__CALLINGPRES_SV=allowed_not_screened") in new stack
  155. -- Executing [s@ext-did:5] Set("SIP/SipGate-00000037", "CALLERPRES()=allowed_not_screened") in new stack
  156. -- Executing [s@ext-did:6] Goto("SIP/SipGate-00000037", "ext-group,601,1") in new stack
  157. -- Goto (ext-group,601,1)
  158. -- Executing [601@ext-group:1] Macro("SIP/SipGate-00000037", "user-callerid,") in new stack
  159. -- Executing [s@macro-user-callerid:1] Set("SIP/SipGate-00000037", "AMPUSER=3048205742") in new stack
  160. -- Executing [s@macro-user-callerid:2] GotoIf("SIP/SipGate-00000037", "0?report") in new stack
  161. -- Executing [s@macro-user-callerid:3] ExecIf("SIP/SipGate-00000037", "1?Set(REALCALLERIDNUM=3048205742)") in new stack
  162. -- Executing [s@macro-user-callerid:4] Set("SIP/SipGate-00000037", "AMPUSER=") in new stack
  163. -- Executing [s@macro-user-callerid:5] Set("SIP/SipGate-00000037", "AMPUSERCIDNAME=") in new stack
  164. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/SipGate-00000037", "1?report") in new stack
  165. -- Goto (macro-user-callerid,s,10)
  166. -- Executing [s@macro-user-callerid:10] GotoIf("SIP/SipGate-00000037", "0?continue") in new stack
  167. -- Executing [s@macro-user-callerid:11] Set("SIP/SipGate-00000037", "__TTL=64") in new stack
  168. -- Executing [s@macro-user-callerid:12] GotoIf("SIP/SipGate-00000037", "1?continue") in new stack
  169. -- Goto (macro-user-callerid,s,19)
  170. -- Executing [s@macro-user-callerid:19] NoOp("SIP/SipGate-00000037", "Using CallerID "4157844987" <3048205742>") in new stack
  171. -- Executing [601@ext-group:2] GotoIf("SIP/SipGate-00000037", "1?skipdb") in new stack
  172. -- Goto (ext-group,601,4)
  173. -- Executing [601@ext-group:4] Set("SIP/SipGate-00000037", "__NODEST=") in new stack
  174. -- Executing [601@ext-group:5] Set("SIP/SipGate-00000037", "__BLKVM_OVERRIDE=BLKVM/601/SIP/SipGate-00000037") in new stack
  175. -- Executing [601@ext-group:6] Set("SIP/SipGate-00000037", "__BLKVM_BASE=601") in new stack
  176. -- Executing [601@ext-group:7] Set("SIP/SipGate-00000037", "DB(BLKVM/601/SIP/SipGate-00000037)=TRUE") in new stack
  177. -- Executing [601@ext-group:8] Set("SIP/SipGate-00000037", "RRNODEST=") in new stack
  178. -- Executing [601@ext-group:9] Set("SIP/SipGate-00000037", "__NODEST=601") in new stack
  179. -- Executing [601@ext-group:10] Set("SIP/SipGate-00000037", "RecordMethod=Group") in new stack
  180. -- Executing [601@ext-group:11] Macro("SIP/SipGate-00000037", "record-enable,2101-2001,Group") in new stack
  181. -- Executing [s@macro-record-enable:1] GotoIf("SIP/SipGate-00000037", "1?check") in new stack
  182. -- Goto (macro-record-enable,s,4)
  183. -- Executing [s@macro-record-enable:4] AGI("SIP/SipGate-00000037", "recordingcheck,20100923-200745,1285286865.56") in new stack
  184. -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  185. -- <SIP/SipGate-00000037>AGI Script recordingcheck completed, returning 0
  186. -- Executing [s@macro-record-enable:5] MacroExit("SIP/SipGate-00000037", "") in new stack
  187. -- Executing [601@ext-group:12] Set("SIP/SipGate-00000037", "RingGroupMethod=ringall") in new stack
  188. -- Executing [601@ext-group:13] Macro("SIP/SipGate-00000037", "dial,20,m(default)t,2101-2001") in new stack
  189. -- Executing [s@macro-dial:1] GotoIf("SIP/SipGate-00000037", "1?dial") in new stack
  190. -- Goto (macro-dial,s,3)
  191. -- Executing [s@macro-dial:3] AGI("SIP/SipGate-00000037", "dialparties.agi") in new stack
  192. -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  193. dialparties.agi: Starting New Dialparties.agi
  194. dialparties.agi: Caller ID name is '4157844987' number is '3048205742'
  195. dialparties.agi: Methodology of ring is 'ringall'
  196. -- dialparties.agi: Added extension 2101 to extension map
  197. -- dialparties.agi: Added extension 2001 to extension map
  198. -- dialparties.agi: Extension 2101 cf is disabled
  199. -- dialparties.agi: Extension 2001 cf is disabled
  200. -- dialparties.agi: Extension 2101 do not disturb is disabled
  201. -- dialparties.agi: Extension 2001 do not disturb is disabled
  202. -- dialparties.agi: dbset CALLTRACE/2101 to 3048205742
  203. -- dialparties.agi: dbset CALLTRACE/2001 to 3048205742
  204. -- dialparties.agi: Filtered ARG3: 2101-2001
  205. -- <SIP/SipGate-00000037>AGI Script dialparties.agi completed, returning 0
  206. -- Executing [s@macro-dial:7] Dial("SIP/SipGate-00000037", "IAX2/2101&SIP/2001,20,m(default)tM(auto-blkvm)") in new stack
  207. == Using SIP RTP TOS bits 184
  208. == Using SIP RTP CoS mark 5
  209. == Using SIP VRTP TOS bits 136
  210. == Using SIP VRTP CoS mark 6
  211. Audio is at 192.168.15.230 port 16468
  212. Video is at 192.168.15.230 port 17852
  213. Adding codec 0x100 (g729) to SDP
  214. Adding video codec 0x80000 (h263) to SDP
  215. Adding video codec 0x200000 (h264) to SDP
  216. Adding non-codec 0x1 (telephone-event) to SDP
  217. Reliably Transmitting (no NAT) to 192.168.15.111:5060:
  218. INVITE sip:2001@192.168.15.111:5060;user=phone;transport=udp SIP/2.0
  219. Via: SIP/2.0/UDP 192.168.15.230:5060;branch=z9hG4bK20e53aaf;rport
  220. Max-Forwards: 70
  221. From: "4157844987" <sip:3048205742@192.168.15.230>;tag=as33f09ba3
  222. To: <sip:2001@192.168.15.111:5060;user=phone;transport=udp>
  223. Contact: <sip:3048205742@192.168.15.230>
  224. Call-ID: 12b43cfa4123b27a1478bf512a38d5a9@192.168.15.230
  225. CSeq: 102 INVITE
  226. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  227. Date: Fri, 24 Sep 2010 00:07:45 GMT
  228. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  229. Supported: replaces, timer
  230. Content-Type: application/sdp
  231. Content-Length: 402
  232.  
  233. v=0
  234. o=root 1879432836 1879432836 IN IP4 192.168.15.230
  235. s=Asterisk PBX 1.6.0.26-FONCORE-r78
  236. c=IN IP4 192.168.15.230
  237. b=CT:384
  238. t=0 0
  239. m=audio 16468 RTP/AVP 18 101
  240. a=rtpmap:18 G729/8000
  241. a=fmtp:18 annexb=no
  242. a=rtpmap:101 telephone-event/8000
  243. a=fmtp:101 0-16
  244. a=silenceSupp:off - - - -
  245. a=ptime:20
  246. a=sendrecv
  247. m=video 17852 RTP/AVP 34 99
  248. a=rtpmap:34 H263/90000
  249. a=rtpmap:99 H264/90000
  250. a=sendrecv
  251.  
  252. ---
  253. -- Called 2001
  254. Audio is at 192.168.15.230 port 15474
  255. Adding codec 0x100 (g729) to SDP
  256. Adding codec 0x4 (ulaw) to SDP
  257. Adding non-codec 0x1 (telephone-event) to SDP
  258.  
  259. <--- Transmitting (NAT) to 204.155.28.10:5060 --->
  260. SIP/2.0 183 Session Progress
  261. Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0;received=204.155.28.10
  262. Via: SIP/2.0/UDP 172.30.20.2;branch=z9hG4bKd7cc.24486865.0
  263. Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.21;branch=z9hG4bK159d242b
  264. Via: SIP/2.0/UDP 204.155.29.57:5060;received=204.155.29.57;branch=z9hG4bK159d242b;rport=5060
  265. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  266. Record-Route: <sip:172.30.20.2;lr=on>
  267. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  268. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  269. To: <sip:1082258e1@sipgate.com>;tag=as34b57f31
  270. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  271. CSeq: 103 INVITE
  272. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  273. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  274. Supported: replaces, timer
  275. Require: timer
  276. Session-Expires: 1800;refresher=uas
  277. Contact: <sip:1082258e1@192.168.15.230>
  278. Content-Type: application/sdp
  279. Content-Length: 327
  280.  
  281. v=0
  282. o=root 1989648169 1989648169 IN IP4 192.168.15.230
  283. s=Asterisk PBX 1.6.0.26-FONCORE-r78
  284. c=IN IP4 192.168.15.230
  285. t=0 0
  286. m=audio 15474 RTP/AVP 18 0 101
  287. a=rtpmap:18 G729/8000
  288. a=fmtp:18 annexb=no
  289. a=rtpmap:0 PCMU/8000
  290. a=rtpmap:101 telephone-event/8000
  291. a=fmtp:101 0-16
  292. a=silenceSupp:off - - - -
  293. a=ptime:20
  294. a=sendrecv
  295.  
  296. <------------>
  297. -- Started music on hold, class 'default', on SIP/SipGate-00000037
  298. Husky*CLI>
  299. <--- SIP read from UDP://204.155.28.10:5060 --->
  300. INVITE sip:1082258e1@192.168.15.230:5060 SIP/2.0
  301. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  302. Record-Route: <sip:24.127.76.4:1345;lr=on>
  303. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  304. Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0
  305. Via: SIP/2.0/UDP 24.127.76.4:1345;branch=z9hG4bKd7cc.24486865.0
  306. Via: SIP/2.0/UDP 204.155.28.10:5060;received=24.127.76.4;branch=z9hG4bK159d242b
  307. Via: SIP/2.0/UDP 24.127.76.4:1347;received=24.127.76.4;branch=z9hG4bK159d242b;rport=5060
  308. Max-Forwards: 67
  309. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  310. To: <sip:1082258e1@sipgate.com>
  311. Contact: <sip:3048205742@24.127.76.4:1347>
  312. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  313. CSeq: 103 INVITE
  314. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  315. Supported: replaces, timer
  316. Content-Type: application/sdp
  317. Content-Length: 431
  318.  
  319. v=0
  320. o=root 1399517662 1399517663 IN IP4 204.155.29.57
  321. s=sipgate VoIP GW
  322. c=IN IP4 204.155.29.56
  323. t=0 0
  324. m=audio 64620 RTP/AVP 0 8 3 97 18 112 101
  325. a=rtpmap:0 PCMU/8000
  326. a=rtpmap:8 PCMA/8000
  327. a=rtpmap:3 GSM/8000
  328. a=rtpmap:97 iLBC/8000
  329. a=rtpmap:18 G729/8000
  330. a=fmtp:18 annexb=no
  331. a=rtpmap:112 G726-32/8000
  332. a=rtpmap:101 telephone-event/8000
  333. a=fmtp:101 0-16
  334. a=silenceSupp:off - - - -
  335. a=ptime:20
  336. a=sendrecv
  337. a=direction:active
  338.  
  339. <------------->
  340. --- (18 headers 19 lines) ---
  341. Ignoring this INVITE request
  342.  
  343. <--- Transmitting (NAT) to 204.155.28.10:5060 --->
  344. SIP/2.0 100 Trying
  345. ia: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0;received=204.155.28.10
  346. Via: SIP/2.0/UDP 24.127.76.4:1345;branch=z9hG4bKd7cc.24486865.0
  347. Via: SIP/2.0/UDP 204.155.28.10:5060;received=24.127.76.4;branch=z9hG4bK159d242b
  348. Via: SIP/2.0/UDP 24.127.76.4:1347;received=24.127.76.4;branch=z9hG4bK159d242b;rport=5060
  349. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  350. Record-Route: <sip:24.127.76.4:1345;lr=on>
  351. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  352. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  353. To: <sip:1082258e1@sipgate.com>
  354. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  355. CSeq: 103 INVITE
  356. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  357. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  358. Supported: replaces, timer
  359. Require: timer
  360. Session-Expires: 1800;refresher=uas
  361. Contact: <sip:1082258e1@192.168.15.230>
  362. Content-Length: 0
  363.  
  364.  
  365. <------------>
  366. -- Stopped music on hold on SIP/SipGate-00000037
  367. Husky*CLI>
  368. <--- SIP read from UDP://192.168.15.111:53006 --->
  369. SIP/2.0 100 Trying
  370. Via: SIP/2.0/UDP 192.168.15.230:5060;branch=z9hG4bK20e53aaf;rport
  371. From: "4157844987" <sip:3048205742@192.168.15.230>;tag=as33f09ba3
  372. To: <sip:2001@192.168.15.111:5060;user=phone;transport=udp>
  373. Call-ID: 12b43cfa4123b27a1478bf512a38d5a9@192.168.15.230
  374. Date: Fri, 24 Sep 2010 00:07:45 GMT
  375. CSeq: 102 INVITE
  376. Server: Cisco-CP7940G/8.0
  377. Contact: <sip:2001@192.168.15.111:5060;user=phone;transport=udp>
  378. Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
  379. Content-Length: 0
  380.  
  381.  
  382. <------------->
  383. --- (11 headers 0 lines) ---
  384. Husky*CLI>
  385. <--- SIP read from UDP://192.168.15.111:53007 --->
  386. SIP/2.0 180 Ringing
  387. Via: SIP/2.0/UDP 192.168.15.230:5060;branch=z9hG4bK20e53aaf;rport
  388. From: "4157844987" <sip:3048205742@192.168.15.230>;tag=as33f09ba3
  389. To: <sip:2001@192.168.15.111:5060;user=phone;transport=udp>;tag=000d29d8106a094a558d8080-38aeb0f3
  390. Call-ID: 12b43cfa4123b27a1478bf512a38d5a9@192.168.15.230
  391. Date: Fri, 24 Sep 2010 00:07:45 GMT
  392. CSeq: 102 INVITE
  393. Server: Cisco-CP7940G/8.0
  394. Contact: <sip:2001@192.168.15.111:5060;user=phone;transport=udp>
  395. Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
  396. Remote-Party-ID: "Tommy Huff" <sip:2001@192.168.15.230>;party=called;id-type=subscriber;privacy=off;screen=yes
  397. Content-Length: 0
  398.  
  399.  
  400. <------------->
  401. --- (12 headers 0 lines) ---
  402. -- SIP/2001-00000038 is ringing
  403. Husky*CLI>
  404. <--- SIP read from UDP://192.168.15.111:53008 --->
  405. SIP/2.0 200 OK
  406. Via: SIP/2.0/UDP 192.168.15.230:5060;branch=z9hG4bK20e53aaf;rport
  407. From: "4157844987" <sip:3048205742@192.168.15.230>;tag=as33f09ba3
  408. To: <sip:2001@192.168.15.111:5060;user=phone;transport=udp>;tag=000d29d8106a094a558d8080-38aeb0f3
  409. Call-ID: 12b43cfa4123b27a1478bf512a38d5a9@192.168.15.230
  410. Date: Fri, 24 Sep 2010 00:07:46 GMT
  411. CSeq: 102 INVITE
  412. Server: Cisco-CP7940G/8.0
  413. Contact: <sip:2001@192.168.15.111:5060;user=phone;transport=udp>
  414. Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
  415. Remote-Party-ID: "Tommy Huff" <sip:2001@192.168.15.230>;party=called;id-type=subscriber;privacy=off;screen=yes
  416. Supported: replaces,join,norefersub
  417. Content-Length: 279
  418. Content-Type: application/sdp
  419. Content-Disposition: session;handling=optional
  420.  
  421. v=0
  422. o=Cisco-SIPUA 25942 0 IN IP4 192.168.15.111
  423. s=SIP Call
  424. t=0 0
  425. m=audio 30650 RTP/AVP 18 101
  426. c=IN IP4 192.168.15.111
  427. a=rtpmap:18 G729/8000
  428. a=fmtp:18 annexb=no
  429. a=rtpmap:101 telephone-event/8000
  430. a=fmtp:101 0-15
  431. a=sendrecv
  432. m=video 0 RTP/AVP 34
  433. c=IN IP4 192.168.15.111
  434.  
  435. <------------->
  436. --- (15 headers 13 lines) ---
  437. Found RTP audio format 18
  438. Found RTP audio format 101
  439. Found audio description format G729 for ID 18
  440. Found audio description format telephone-event for ID 101
  441. Found RTP video format 34
  442. Capabilities: us - 0x280108 (alaw|g729|h263|h264), peer - audio=0x100 (g729)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x80100 (g729|h263)
  443. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  444. Peer audio RTP is at port 192.168.15.111:30650
  445. Peer doesn't provide video
  446. list_route: hop: <sip:2001@192.168.15.111:5060;user=phone;transport=udp>
  447. set_destination: Parsing <sip:2001@192.168.15.111:5060;user=phone;transport=udp> for address/port to send to
  448. set_destination: set destination to 192.168.15.111, port 5060
  449. Transmitting (no NAT) to 192.168.15.111:5060:
  450. ACK sip:2001@192.168.15.111:5060;user=phone;transport=udp SIP/2.0
  451. Via: SIP/2.0/UDP 192.168.15.230:5060;branch=z9hG4bK6e40a57b;rport
  452. Max-Forwards: 70
  453. From: "4157844987" <sip:3048205742@192.168.15.230>;tag=as33f09ba3
  454. To: <sip:2001@192.168.15.111:5060;user=phone;transport=udp>;tag=000d29d8106a094a558d8080-38aeb0f3
  455. Contact: <sip:3048205742@192.168.15.230>
  456. Call-ID: 12b43cfa4123b27a1478bf512a38d5a9@192.168.15.230
  457. CSeq: 102 ACK
  458. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  459. Content-Length: 0
  460.  
  461.  
  462. ---
  463. -- SIP/2001-00000038 answered SIP/SipGate-00000037
  464. -- Executing [s@macro-auto-blkvm:1] Set("SIP/2001-00000038", "__MACRO_RESULT=") in new stack
  465. -- Executing [s@macro-auto-blkvm:2] DBdel("SIP/2001-00000038", "BLKVM/601/SIP/SipGate-00000037") in new stack
  466. -- DBdel: family=BLKVM, key=601/SIP/SipGate-00000037
  467. Audio is at 192.168.15.230 port 15474
  468. Adding codec 0x100 (g729) to SDP
  469. Adding codec 0x4 (ulaw) to SDP
  470. Adding non-codec 0x1 (telephone-event) to SDP
  471.  
  472. <--- Reliably Transmitting (NAT) to 204.155.28.10:5060 --->
  473. SIP/2.0 200 OK
  474. Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0;received=204.155.28.10
  475. Via: SIP/2.0/UDP 172.30.20.2;branch=z9hG4bKd7cc.24486865.0
  476. Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.21;branch=z9hG4bK159d242b
  477. Via: SIP/2.0/UDP 204.155.29.57:5060;received=204.155.29.57;branch=z9hG4bK159d242b;rport=5060
  478. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  479. Record-Route: <sip:172.30.20.2;lr=on>
  480. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  481. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  482. To: <sip:1082258e1@sipgate.com>;tag=as34b57f31
  483. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  484. CSeq: 103 INVITE
  485. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  486. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  487. Supported: replaces, timer
  488. Require: timer
  489. Session-Expires: 1800;refresher=uas
  490. Contact: <sip:1082258e1@192.168.15.230>
  491. Content-Type: application/sdp
  492. Content-Length: 327
  493.  
  494. v=0
  495. o=root 1989648169 1989648170 IN IP4 192.168.15.230
  496. s=Asterisk PBX 1.6.0.26-FONCORE-r78
  497. c=IN IP4 192.168.15.230
  498. t=0 0
  499. m=audio 15474 RTP/AVP 18 0 101
  500. a=rtpmap:18 G729/8000
  501. a=fmtp:18 annexb=no
  502. a=rtpmap:0 PCMU/8000
  503. a=rtpmap:101 telephone-event/8000
  504. a=fmtp:101 0-16
  505. a=silenceSupp:off - - - -
  506. a=ptime:20
  507. a=sendrecv
  508.  
  509. <------------>
  510. Retransmitting #1 (NAT) to 204.155.28.10:5060:
  511. SIP/2.0 200 OK
  512. Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0;received=204.155.28.10
  513. Via: SIP/2.0/UDP 172.30.20.2;branch=z9hG4bKd7cc.24486865.0
  514. Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.21;branch=z9hG4bK159d242b
  515. Via: SIP/2.0/UDP 204.155.29.57:5060;received=204.155.29.57;branch=z9hG4bK159d242b;rport=5060
  516. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  517. Record-Route: <sip:172.30.20.2;lr=on>
  518. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  519. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  520. To: <sip:1082258e1@sipgate.com>;tag=as34b57f31
  521. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  522. CSeq: 103 INVITE
  523. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  524. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  525. Supported: replaces, timer
  526. Require: timer
  527. Session-Expires: 1800;refresher=uas
  528. Contact: <sip:1082258e1@192.168.15.230>
  529. Content-Type: application/sdp
  530. Content-Length: 327
  531.  
  532. v=0
  533. o=root 1989648169 1989648170 IN IP4 192.168.15.230
  534. s=Asterisk PBX 1.6.0.26-FONCORE-r78
  535. c=IN IP4 192.168.15.230
  536. t=0 0
  537. m=audio 15474 RTP/AVP 18 0 101
  538. a=rtpmap:18 G729/8000
  539. a=fmtp:18 annexb=no
  540. a=rtpmap:0 PCMU/8000
  541. a=rtpmap:101 telephone-event/8000
  542. a=fmtp:101 0-16
  543. a=silenceSupp:off - - - -
  544. a=ptime:20
  545. a=sendrecv
  546.  
  547. ---
  548. Retransmitting #2 (NAT) to 204.155.28.10:5060:
  549. SIP/2.0 200 OK
  550. Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0;received=204.155.28.10
  551. Via: SIP/2.0/UDP 172.30.20.2;branch=z9hG4bKd7cc.24486865.0
  552. Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.21;branch=z9hG4bK159d242b
  553. Via: SIP/2.0/UDP 204.155.29.57:5060;received=204.155.29.57;branch=z9hG4bK159d242b;rport=5060
  554. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  555. Record-Route: <sip:172.30.20.2;lr=on>
  556. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  557. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  558. To: <sip:1082258e1@sipgate.com>;tag=as34b57f31
  559. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  560. CSeq: 103 INVITE
  561. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  562. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  563. Supported: replaces, timer
  564. Require: timer
  565. Session-Expires: 1800;refresher=uas
  566. Contact: <sip:1082258e1@192.168.15.230>
  567. Content-Type: application/sdp
  568. Content-Length: 327
  569.  
  570. v=0
  571. o=root 1989648169 1989648170 IN IP4 192.168.15.230
  572. s=Asterisk PBX 1.6.0.26-FONCORE-r78
  573. c=IN IP4 192.168.15.230
  574. t=0 0
  575. m=audio 15474 RTP/AVP 18 0 101
  576. a=rtpmap:18 G729/8000
  577. a=fmtp:18 annexb=no
  578. a=rtpmap:0 PCMU/8000
  579. a=rtpmap:101 telephone-event/8000
  580. a=fmtp:101 0-16
  581. a=silenceSupp:off - - - -
  582. a=ptime:20
  583. a=sendrecv
  584.  
  585. ---
  586. Retransmitting #3 (NAT) to 204.155.28.10:5060:
  587. SIP/2.0 200 OK
  588. Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0;received=204.155.28.10
  589. Via: SIP/2.0/UDP 172.30.20.2;branch=z9hG4bKd7cc.24486865.0
  590. Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.21;branch=z9hG4bK159d242b
  591. Via: SIP/2.0/UDP 204.155.29.57:5060;received=204.155.29.57;branch=z9hG4bK159d242b;rport=5060
  592. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  593. Record-Route: <sip:172.30.20.2;lr=on>
  594. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  595. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  596. To: <sip:1082258e1@sipgate.com>;tag=as34b57f31
  597. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  598. CSeq: 103 INVITE
  599. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  601. Supported: replaces, timer
  602. Require: timer
  603. Session-Expires: 1800;refresher=uas
  604. Contact: <sip:1082258e1@192.168.15.230>
  605. Content-Type: application/sdp
  606. Content-Length: 327
  607.  
  608. v=0
  609. o=root 1989648169 1989648170 IN IP4 192.168.15.230
  610. s=Asterisk PBX 1.6.0.26-FONCORE-r78
  611. c=IN IP4 192.168.15.230
  612. t=0 0
  613. m=audio 15474 RTP/AVP 18 0 101
  614. a=rtpmap:18 G729/8000
  615. a=fmtp:18 annexb=no
  616. a=rtpmap:0 PCMU/8000
  617. a=rtpmap:101 telephone-event/8000
  618. a=fmtp:101 0-16
  619. a=silenceSupp:off - - - -
  620. a=ptime:20
  621. a=sendrecv
  622.  
  623. ---
  624. Husky*CLI>
  625. <--- SIP read from UDP://192.168.15.111:53009 --->
  626. BYE sip:3048205742@192.168.15.230 SIP/2.0
  627. Via: SIP/2.0/UDP 192.168.15.111:5060;branch=z9hG4bK2cfdd7be
  628. From: <sip:2001@192.168.15.111:5060;user=phone;transport=udp>;tag=000d29d8106a094a558d8080-38aeb0f3
  629. To: "4157844987" <sip:3048205742@192.168.15.230>;tag=as33f09ba3
  630. Call-ID: 12b43cfa4123b27a1478bf512a38d5a9@192.168.15.230
  631. Max-Forwards: 70
  632. Date: Fri, 24 Sep 2010 00:07:50 GMT
  633. CSeq: 101 BYE
  634. User-Agent: Cisco-CP7940G/8.0
  635. Content-Length: 0
  636.  
  637.  
  638. <------------->
  639. --- (10 headers 0 lines) ---
  640. Sending to 192.168.15.111 : 5060 (no NAT)
  641.  
  642. <--- Transmitting (no NAT) to 192.168.15.111:5060 --->
  643. SIP/2.0 200 OK
  644. Via: SIP/2.0/UDP 192.168.15.111:5060;branch=z9hG4bK2cfdd7be;received=192.168.15.111
  645. From: <sip:2001@192.168.15.111:5060;user=phone;transport=udp>;tag=000d29d8106a094a558d8080-38aeb0f3
  646. To: "4157844987" <sip:3048205742@192.168.15.230>;tag=as33f09ba3
  647. Call-ID: 12b43cfa4123b27a1478bf512a38d5a9@192.168.15.230
  648. CSeq: 101 BYE
  649. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  650. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  651. Supported: replaces, timer
  652. Content-Length: 0
  653.  
  654.  
  655. <------------>
  656. -- Executing [h@macro-dial:1] Macro("SIP/SipGate-00000037", "hangupcall") in new stack
  657. -- Executing [s@macro-hangupcall:1] GotoIf("SIP/SipGate-00000037", "1?skiprg") in new stack
  658. -- Goto (macro-hangupcall,s,4)
  659. -- Executing [s@macro-hangupcall:4] GotoIf("SIP/SipGate-00000037", "0?skipblkvm") in new stack
  660. -- Executing [s@macro-hangupcall:5] NoOp("SIP/SipGate-00000037", "Cleaning Up Block VM Flag: BLKVM/601/SIP/SipGate-00000037") in new stack
  661. -- Executing [s@macro-hangupcall:6] DBdel("SIP/SipGate-00000037", "BLKVM/601/SIP/SipGate-00000037") in new stack
  662. -- DBdel: family=BLKVM, key=601/SIP/SipGate-00000037
  663. -- DBdel: Error deleting key from database.
  664. -- Executing [s@macro-hangupcall:7] GotoIf("SIP/SipGate-00000037", "1?theend") in new stack
  665. -- Goto (macro-hangupcall,s,9)
  666. -- Executing [s@macro-hangupcall:9] Hangup("SIP/SipGate-00000037", "") in new stack
  667. == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/SipGate-00000037' in macro 'hangupcall'
  668. == Spawn extension (macro-dial, h, 1) exited non-zero on 'SIP/SipGate-00000037'
  669. == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/SipGate-00000037' in macro 'dial'
  670. == Spawn extension (ext-group, 601, 13) exited non-zero on 'SIP/SipGate-00000037'
  671. Scheduling destruction of SIP dialog '6da3ee7f15851e9715ccf105765d6ffc@sipgate.com' in 32000 ms (Method: INVITE)
  672. Really destroying SIP dialog '12b43cfa4123b27a1478bf512a38d5a9@192.168.15.230' Method: BYE
  673. Husky*CLI>
  674. <--- SIP read from UDP://204.155.28.10:5060 --->
  675. CANCEL sip:1082258e1@192.168.15.230:5060 SIP/2.0
  676. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  677. Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0
  678. Via: SIP/2.0/UDP 24.127.76.4:1345;branch=z9hG4bKd7cc.24486865.0
  679. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  680. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  681. To: <sip:1082258e1@sipgate.com>
  682. CSeq: 103 CANCEL
  683. Max-Forwards: 69
  684. Content-Length: 0
  685.  
  686.  
  687. <------------->
  688. --- (10 headers 0 lines) ---
  689. Sending to 204.155.28.10 : 5060 (NAT)
  690. Scheduling destruction of SIP dialog '6da3ee7f15851e9715ccf105765d6ffc@sipgate.com' in 32000 ms (Method: CANCEL)
  691.  
  692. <--- Reliably Transmitting (NAT) to 204.155.28.10:5060 --->
  693. SIP/2.0 487 Request Terminated
  694. Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0;received=204.155.28.10
  695. Via: SIP/2.0/UDP 172.30.20.2;branch=z9hG4bKd7cc.24486865.0
  696. Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.21;branch=z9hG4bK159d242b
  697. Via: SIP/2.0/UDP 204.155.29.57:5060;received=204.155.29.57;branch=z9hG4bK159d242b;rport=5060
  698. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  699. To: <sip:1082258e1@sipgate.com>;tag=as34b57f31
  700. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  701. CSeq: 103 INVITE
  702. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  703. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  704. Supported: replaces, timer
  705. Content-Length: 0
  706.  
  707.  
  708. <------------>
  709.  
  710. <--- Transmitting (NAT) to 204.155.28.10:5060 --->
  711. SIP/2.0 200 OK
  712. Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0;received=204.155.28.10
  713. Via: SIP/2.0/UDP 24.127.76.4:1345;branch=z9hG4bKd7cc.24486865.0
  714. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  715. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  716. To: <sip:1082258e1@sipgate.com>;tag=as34b57f31
  717. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  718. CSeq: 103 CANCEL
  719. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  720. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  721. Supported: replaces, timer
  722. Content-Length: 0
  723.  
  724.  
  725. <------------>
  726. Retransmitting #1 (NAT) to 204.155.28.10:5060:
  727. SIP/2.0 487 Request Terminated
  728. Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0;received=204.155.28.10
  729. Via: SIP/2.0/UDP 172.30.20.2;branch=z9hG4bKd7cc.24486865.0
  730. Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.21;branch=z9hG4bK159d242b
  731. Via: SIP/2.0/UDP 204.155.29.57:5060;received=204.155.29.57;branch=z9hG4bK159d242b;rport=5060
  732. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  733. To: <sip:1082258e1@sipgate.com>;tag=as34b57f31
  734. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  735. CSeq: 103 INVITE
  736. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  737. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  738. Supported: replaces, timer
  739. Content-Length: 0
  740.  
  741.  
  742. ---
  743. Retransmitting #4 (NAT) to 204.155.28.10:5060:
  744. SIP/2.0 200 OK
  745. Via: SIP/2.0/UDP 204.155.28.10:5060;branch=z9hG4bKd7cc.24486865.0;received=204.155.28.10
  746. Via: SIP/2.0/UDP 172.30.20.2;branch=z9hG4bKd7cc.24486865.0
  747. Via: SIP/2.0/UDP 204.155.28.10:5060;received=204.155.28.21;branch=z9hG4bK159d242b
  748. Via: SIP/2.0/UDP 204.155.29.57:5060;received=204.155.29.57;branch=z9hG4bK159d242b;rport=5060
  749. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  750. Record-Route: <sip:172.30.20.2;lr=on>
  751. Record-Route: <sip:204.155.28.10;lr=on;ftag=as077f532d>
  752. From: "4157844987" <sip:3048205742@sipgate.com>;tag=as077f532d
  753. To: <sip:1082258e1@sipgate.com>;tag=as34b57f31
  754. Call-ID: 6da3ee7f15851e9715ccf105765d6ffc@sipgate.com
  755. CSeq: 103 INVITE
  756. User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
  757. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  758. Supported: replaces, timer
  759. Require: timer
  760. Session-Expires: 1800;refresher=uas
  761. Contact: <sip:1082258e1@192.168.15.230>
  762. Content-Type: application/sdp
  763. Content-Length: 327
  764.  
  765. v=0
  766. o=root 1989648169 1989648170 IN IP4 192.168.15.230
  767. s=Asterisk PBX 1.6.0.26-FONCORE-r78
  768. c=IN IP4 192.168.15.230
  769. t=0 0
  770. m=audio 15474 RTP/AVP 18 0 101
  771. a=rtpmap:18 G729/8000
  772. a=fmtp:18 annexb=no
  773. a=rtpmap:0 PCMU/8000
  774. a=rtpmap:101 telephone-event/8000
  775. a=fmtp:101 0-16
  776. a=silenceSupp:off - - - -
  777. a=ptime:20
  778. a=sendrecv
  779.  
  780. ---
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