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  1. * Name : Trunk
  2. Secret : <Not set>
  3. MD5Secret : <Not set>
  4. Remote Secret: <Not set>
  5. Context : from-trunk-sip-Trunk
  6. Subscr.Cont. : <Not set>
  7. Language :
  8. AMA flags : Unknown
  9. Transfer mode: open
  10. CallingPres : Presentation Allowed, Not Screened
  11. Callgroup :
  12. Pickupgroup :
  13. MOH Suggest :
  14. Mailbox :
  15. VM Extension : *97
  16. LastMsgsSent : 32767/65535
  17. Call limit : 0
  18. Max forwards : 0
  19. Dynamic : No
  20. Callerid : "" <>
  21. MaxCallBR : 384 kbps
  22. Expire : -1
  23. Insecure : invite
  24. Force rport : No
  25. ACL : No
  26. DirectMedACL : No
  27. T.38 support : No
  28. T.38 EC mode : Unknown
  29. T.38 MaxDtgrm: -1
  30. DirectMedia : Yes
  31. PromiscRedir : No
  32. User=Phone : No
  33. Video Support: No
  34. Text Support : No
  35. Ign SDP ver : No
  36. Trust RPID : No
  37. Send RPID : No
  38. Subscriptions: Yes
  39. Overlap dial : Yes
  40. DTMFmode : rfc2833
  41. Timer T1 : 500
  42. Timer B : 32000
  43. ToHost : 10.100.210.254
  44. Addr->IP : 10.100.210.254:5060
  45. Defaddr->IP : (null)
  46. Prim.Transp. : UDP
  47. Allowed.Trsp : UDP
  48. Def. Username:
  49. SIP Options : (none)
  50. Codecs : 0x10c (ulaw|alaw|g729)
  51. Codec Order : (ulaw:20,alaw:20,g729:40)
  52. Auto-Framing : No
  53. Status : Unmonitored
  54. Useragent :
  55. Reg. Contact :
  56. Qualify Freq : 60000 ms
  57. Sess-Timers : Accept
  58. Sess-Refresh : uas
  59. Sess-Expires : 1800 secs
  60. Min-Sess : 90 secs
  61. RTP Engine : asterisk
  62. Parkinglot :
  63. Use Reason : No
  64. Encryption : No
  65. DND : No
  66. CallFwd Num. :
  67.  
  68.  
  69.  
  70. * Name : 3217
  71. Secret : <Set>
  72. MD5Secret : <Not set>
  73. Remote Secret: <Not set>
  74. Context : forall
  75. Subscr.Cont. : <Not set>
  76. Language :
  77. AMA flags : Unknown
  78. Transfer mode: open
  79. CallingPres : Presentation Allowed, Not Screened
  80. Callgroup :
  81. Pickupgroup :
  82. MOH Suggest :
  83. Mailbox : 3217@device
  84. VM Extension : *97
  85. LastMsgsSent : 0/0
  86. Call limit : 2147483647
  87. Max forwards : 0
  88. Dynamic : Yes
  89. Callerid : "Test" <3217>
  90. MaxCallBR : 384 kbps
  91. Expire : 3192
  92. Insecure : no
  93. Force rport : No
  94. ACL : Yes
  95. DirectMedACL : No
  96. T.38 support : No
  97. T.38 EC mode : Unknown
  98. T.38 MaxDtgrm: -1
  99. DirectMedia : Yes
  100. PromiscRedir : No
  101. User=Phone : No
  102. Video Support: No
  103. Text Support : No
  104. Ign SDP ver : No
  105. Trust RPID : Yes
  106. Send RPID : Yes
  107. Subscriptions: Yes
  108. Overlap dial : Yes
  109. DTMFmode : rfc2833
  110. Timer T1 : 500
  111. Timer B : 32000
  112. ToHost :
  113. Addr->IP : 10.200.210.51:6491
  114. Defaddr->IP : (null)
  115. Prim.Transp. : UDP
  116. Allowed.Trsp : UDP
  117. Def. Username: 3217
  118. SIP Options : (none)
  119. Codecs : 0x100 (g729)
  120. Codec Order : (g729:40)
  121. Auto-Framing : No
  122. Status : OK (388 ms)
  123. Useragent : eyeBeam release 3006o stamp 17551
  124. Reg. Contact : sip:3217@10.200.210.51:6491
  125. Qualify Freq : 60000 ms
  126. Sess-Timers : Accept
  127. Sess-Refresh : uas
  128. Sess-Expires : 1800 secs
  129. Min-Sess : 90 secs
  130. RTP Engine : asterisk
  131. Parkinglot :
  132. Use Reason : No
  133. Encryption : No
  134. DND : No
  135. CallFwd Num. :
  136.  
  137.  
  138.  
  139. <--- SIP read from UDP:10.200.210.51:6491 --->
  140.  
  141. <------------->
  142. Really destroying SIP dialog '4719b99b1f3d4cf54239e4174ea54b11@212.21.52.68:5060' Method: OPTIONS
  143.  
  144. <--- SIP read from UDP:10.200.210.51:6491 --->
  145. INVITE sip:9050555555@172.20.255.50 SIP/2.0
  146. To: <sip:9050555555@172.20.255.50>
  147. From: h<sip:3217@172.20.255.50>;tag=ef31bf1d
  148. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-715938653-1--d87543-;rport
  149. Call-ID: f15355682030a66d
  150. CSeq: 1 INVITE
  151. Contact: <sip:3217@10.200.210.51:6491>
  152. Max-Forwards: 70
  153. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  154. Content-Type: application/sdp
  155. User-Agent: eyeBeam release 3006o stamp 17551
  156. Content-Length: 240
  157.  
  158. v=0
  159. o=- 636137314 636137321 IN IP4 10.200.210.51
  160. s=eyeBeam
  161. c=IN IP4 10.200.210.51
  162. t=0 0
  163. m=audio 6038 RTP/AVP 18 0 8 101
  164. a=alt:1 1 : FCC51B9B 000000FD 10.200.210.51 6038
  165. a=fmtp:101 0-15
  166. a=rtpmap:101 telephone-event/8000
  167. a=sendrecv
  168. <------------->
  169. --- (12 headers 10 lines) ---
  170. Sending to 10.200.210.51:6491 (NAT)
  171. Using INVITE request as basis request - f15355682030a66d
  172. Found peer '3217' for '3217' from 10.200.210.51:6491
  173.  
  174. <--- Reliably Transmitting (no NAT) to 10.200.210.51:6491 --->
  175. SIP/2.0 401 Unauthorized
  176. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-715938653-1--d87543-;received=10.200.210.51;rport=6491
  177. From: h<sip:3217@172.20.255.50>;tag=ef31bf1d
  178. To: <sip:9050555555@172.20.255.50>;tag=as415b40b8
  179. Call-ID: f15355682030a66d
  180. CSeq: 1 INVITE
  181. Server: Asterisk
  182. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  183. Supported: replaces, timer
  184. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2dff0cbb"
  185. Content-Length: 0
  186.  
  187.  
  188. <------------>
  189. Scheduling destruction of SIP dialog 'f15355682030a66d' in 25152 ms (Method: INVITE)
  190.  
  191. <--- SIP read from UDP:10.200.210.51:6491 --->
  192. ACK sip:9050555555@172.20.255.50 SIP/2.0
  193. To: <sip:9050555555@172.20.255.50>;tag=as415b40b8
  194. From: h<sip:3217@172.20.255.50>;tag=ef31bf1d
  195. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-715938653-1--d87543-;rport
  196. Call-ID: f15355682030a66d
  197. CSeq: 1 ACK
  198. Content-Length: 0
  199.  
  200. <------------->
  201. --- (7 headers 0 lines) ---
  202.  
  203. <--- SIP read from UDP:10.200.210.51:6491 --->
  204. INVITE sip:9050555555@172.20.255.50 SIP/2.0
  205. To: <sip:9050555555@172.20.255.50>
  206. From: h<sip:3217@172.20.255.50>;tag=ef31bf1d
  207. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-149506423-1--d87543-;rport
  208. Call-ID: f15355682030a66d
  209. CSeq: 2 INVITE
  210. Contact: <sip:3217@10.200.210.51:6491>
  211. Max-Forwards: 70
  212. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  213. Content-Type: application/sdp
  214. User-Agent: eyeBeam release 3006o stamp 17551
  215. Authorization: Digest username="3217",realm="asterisk",nonce="2dff0cbb",uri="sip:9050555555@172.20.255.50",response="5426822667bb878569ee583c80f153bc",algorithm=MD5
  216. Content-Length: 240
  217.  
  218. v=0
  219. o=- 636137314 636137321 IN IP4 10.200.210.51
  220. s=eyeBeam
  221. c=IN IP4 10.200.210.51
  222. t=0 0
  223. m=audio 6038 RTP/AVP 18 0 8 101
  224. a=alt:1 1 : FCC51B9B 000000FD 10.200.210.51 6038
  225. a=fmtp:101 0-15
  226. a=rtpmap:101 telephone-event/8000
  227. a=sendrecv
  228. <------------->
  229. --- (13 headers 10 lines) ---
  230. Sending to 10.200.210.51:6491 (no NAT)
  231. Using INVITE request as basis request - f15355682030a66d
  232. Found peer '3217' for '3217' from 10.200.210.51:6491
  233. == Using SIP RTP TOS bits 184
  234. == Using SIP RTP CoS mark 5
  235. Found RTP audio format 18
  236. Found RTP audio format 0
  237. Found RTP audio format 8
  238. Found RTP audio format 101
  239. Found audio description format telephone-event for ID 101
  240. Capabilities: us - 0x100 (g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
  241. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  242. Peer audio RTP is at port 10.200.210.51:6038
  243. Looking for 9050555555 in forall (domain 172.20.255.50)
  244. list_route: hop: <sip:3217@10.200.210.51:6491>
  245.  
  246. <--- Transmitting (no NAT) to 10.200.210.51:6491 --->
  247. SIP/2.0 100 Trying
  248. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-149506423-1--d87543-;received=10.200.210.51;rport=6491
  249. From: h<sip:3217@172.20.255.50>;tag=ef31bf1d
  250. To: <sip:9050555555@172.20.255.50>
  251. Call-ID: f15355682030a66d
  252. CSeq: 2 INVITE
  253. Server: Asterisk
  254. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  255. Supported: replaces, timer
  256. Contact: <sip:9050555555@172.20.255.50:5060>
  257. Content-Length: 0
  258.  
  259.  
  260. <------------>
  261. -- Executing [9050555555@forall:1] Macro("SIP/3217-0000000c", "user-callerid,LIMIT,") in new stack
  262. -- Executing [s@macro-user-callerid:1] Set("SIP/3217-0000000c", "AMPUSER=3217") in new stack
  263. -- Executing [s@macro-user-callerid:2] GotoIf("SIP/3217-0000000c", "0?report") in new stack
  264. -- Executing [s@macro-user-callerid:3] ExecIf("SIP/3217-0000000c", "1?Set(REALCALLERIDNUM=3217)") in new stack
  265. -- Executing [s@macro-user-callerid:4] Set("SIP/3217-0000000c", "AMPUSER=3217") in new stack
  266. -- Executing [s@macro-user-callerid:5] Set("SIP/3217-0000000c", "AMPUSERCIDNAME=Test") in new stack
  267. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/3217-0000000c", "0?report") in new stack
  268. -- Executing [s@macro-user-callerid:7] Set("SIP/3217-0000000c", "AMPUSERCID=3217") in new stack
  269. -- Executing [s@macro-user-callerid:8] Set("SIP/3217-0000000c", "CALLERID(all)="Test" <3217>") in new stack
  270. -- Executing [s@macro-user-callerid:9] GotoIf("SIP/3217-0000000c", "0?limit") in new stack
  271. -- Executing [s@macro-user-callerid:10] ExecIf("SIP/3217-0000000c", "1?Set(GROUP(concurrency_limit)=3217)") in new stack
  272. -- Executing [s@macro-user-callerid:11] GosubIf("SIP/3217-0000000c", "7?sub-ccss,s,1(forall,9050555555)") in new stack
  273. -- Executing [s@sub-ccss:1] ExecIf("SIP/3217-0000000c", "0?Return()") in new stack
  274. -- Executing [s@sub-ccss:2] Set("SIP/3217-0000000c", "CCSS_SETUP=TRUE") in new stack
  275. -- Executing [s@sub-ccss:3] GosubIf("SIP/3217-0000000c", "0?monitor_config,1(forall,9050555555):monitor_default,1(forall,9050555555)") in new stack
  276. -- Executing [monitor_default@sub-ccss:1] GotoIf("SIP/3217-0000000c", "0?is_exten") in new stack
  277. -- Executing [monitor_default@sub-ccss:2] StackPop("SIP/3217-0000000c", "") in new stack
  278. -- Executing [monitor_default@sub-ccss:3] Return("SIP/3217-0000000c", "FALSE") in new stack
  279. -- Executing [s@macro-user-callerid:12] GotoIf("SIP/3217-0000000c", "1?continue") in new stack
  280. -- Goto (macro-user-callerid,s,25)
  281. -- Executing [s@macro-user-callerid:25] Set("SIP/3217-0000000c", "CALLERID(number)=3217") in new stack
  282. -- Executing [s@macro-user-callerid:26] Set("SIP/3217-0000000c", "CALLERID(name)=Test") in new stack
  283. -- Executing [s@macro-user-callerid:27] Set("SIP/3217-0000000c", "CHANNEL(language)=en") in new stack
  284. -- Executing [9050555555@forall:2] Set("SIP/3217-0000000c", "MOHCLASS=default") in new stack
  285. -- Executing [9050555555@forall:3] Set("SIP/3217-0000000c", "_NODEST=") in new stack
  286. -- Executing [9050555555@forall:4] Gosub("SIP/3217-0000000c", "sub-record-check,s,1(out,9050555555,)") in new stack
  287. -- Executing [s@sub-record-check:1] GotoIf("SIP/3217-0000000c", "1?check") in new stack
  288. -- Goto (sub-record-check,s,6)
  289. -- Executing [s@sub-record-check:6] Set("SIP/3217-0000000c", "__MON_FMT=g729") in new stack
  290. -- Executing [s@sub-record-check:7] GotoIf("SIP/3217-0000000c", "1?next") in new stack
  291. -- Goto (sub-record-check,s,10)
  292. -- Executing [s@sub-record-check:10] ExecIf("SIP/3217-0000000c", "0?Return()") in new stack
  293. -- Executing [s@sub-record-check:11] GotoIf("SIP/3217-0000000c", "0?out,1") in new stack
  294. -- Executing [s@sub-record-check:12] Set("SIP/3217-0000000c", "__REC_STATUS=INITIALIZED") in new stack
  295. -- Executing [s@sub-record-check:13] ExecIf("SIP/3217-0000000c", "0?Set(__REC_POLICY_MODE=)") in new stack
  296. -- Executing [s@sub-record-check:14] Set("SIP/3217-0000000c", "NOW=1362774299") in new stack
  297. -- Executing [s@sub-record-check:15] Set("SIP/3217-0000000c", "__DAY=08") in new stack
  298. -- Executing [s@sub-record-check:16] Set("SIP/3217-0000000c", "__MONTH=03") in new stack
  299. -- Executing [s@sub-record-check:17] Set("SIP/3217-0000000c", "__YEAR=2013") in new stack
  300. -- Executing [s@sub-record-check:18] Set("SIP/3217-0000000c", "__TIMESTR=20130308-152459") in new stack
  301. -- Executing [s@sub-record-check:19] Set("SIP/3217-0000000c", "__FROMEXTEN=3217") in new stack
  302. -- Executing [s@sub-record-check:20] Set("SIP/3217-0000000c", "__CALLFILENAME=out-9050555555-3217-20130308-152459-1362774299.12") in new stack
  303. -- Executing [s@sub-record-check:21] Goto("SIP/3217-0000000c", "out,1") in new stack
  304. -- Goto (sub-record-check,out,1)
  305. -- Executing [out@sub-record-check:1] ExecIf("SIP/3217-0000000c", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
  306. -- Executing [out@sub-record-check:2] GosubIf("SIP/3217-0000000c", "0?record,1(exten,9050555555,3217)") in new stack
  307. -- Executing [out@sub-record-check:3] Return("SIP/3217-0000000c", "") in new stack
  308. -- Executing [9050555555@forall:5] Macro("SIP/3217-0000000c", "dialout-trunk,3,050555555,") in new stack
  309. -- Executing [s@macro-dialout-trunk:1] Set("SIP/3217-0000000c", "DIAL_TRUNK=3") in new stack
  310. -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/3217-0000000c", "0?sub-pincheck,s,1()") in new stack
  311. -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/3217-0000000c", "0?disabletrunk,1") in new stack
  312. -- Executing [s@macro-dialout-trunk:4] Set("SIP/3217-0000000c", "DIAL_NUMBER=050555555") in new stack
  313. -- Executing [s@macro-dialout-trunk:5] Set("SIP/3217-0000000c", "DIAL_TRUNK_OPTIONS=r") in new stack
  314. -- Executing [s@macro-dialout-trunk:6] Set("SIP/3217-0000000c", "OUTBOUND_GROUP=OUT_3") in new stack
  315. -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/3217-0000000c", "1?nomax") in new stack
  316. -- Goto (macro-dialout-trunk,s,9)
  317. -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/3217-0000000c", "0?skipoutcid") in new stack
  318. -- Executing [s@macro-dialout-trunk:10] Set("SIP/3217-0000000c", "DIAL_TRUNK_OPTIONS=") in new stack
  319. -- Executing [s@macro-dialout-trunk:11] Macro("SIP/3217-0000000c", "outbound-callerid,3") in new stack
  320. -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/3217-0000000c", "0?Set(CALLERPRES()=)") in new stack
  321. -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/3217-0000000c", "0?Set(REALCALLERIDNUM=3217)") in new stack
  322. -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/3217-0000000c", "1?normcid") in new stack
  323. -- Goto (macro-outbound-callerid,s,6)
  324. -- Executing [s@macro-outbound-callerid:6] Set("SIP/3217-0000000c", "USEROUTCID=01111") in new stack
  325. -- Executing [s@macro-outbound-callerid:7] Set("SIP/3217-0000000c", "EMERGENCYCID=") in new stack
  326. -- Executing [s@macro-outbound-callerid:8] Set("SIP/3217-0000000c", "TRUNKOUTCID=") in new stack
  327. -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/3217-0000000c", "1?trunkcid") in new stack
  328. -- Goto (macro-outbound-callerid,s,12)
  329. -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/3217-0000000c", "0?Set(CALLERID(all)=)") in new stack
  330. -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/3217-0000000c", "1?Set(CALLERID(all)=01111)") in new stack
  331. -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/3217-0000000c", "0?Set(CALLERID(all)=)") in new stack
  332. -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/3217-0000000c", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
  333. -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/3217-0000000c", "0?sub-flp-3,s,1()") in new stack
  334. -- Executing [s@macro-dialout-trunk:13] Set("SIP/3217-0000000c", "OUTNUM=9050555555") in new stack
  335. -- Executing [s@macro-dialout-trunk:14] Set("SIP/3217-0000000c", "custom=SIP/Trunk") in new stack
  336. -- Executing [s@macro-dialout-trunk:15] Macro("SIP/3217-0000000c", "dialout-trunk-predial-hook,") in new stack
  337. -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/3217-0000000c", "") in new stack
  338. -- Executing [s@macro-dialout-trunk:16] GotoIf("SIP/3217-0000000c", "0?bypass,1") in new stack
  339. -- Executing [s@macro-dialout-trunk:17] ExecIf("SIP/3217-0000000c", "1?Set(CONNECTEDLINE(num,i)=050555555)") in new stack
  340. -- Executing [s@macro-dialout-trunk:18] ExecIf("SIP/3217-0000000c", "1?Set(CONNECTEDLINE(name,i)=CID:01111)") in new stack
  341. -- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/3217-0000000c", "0?customtrunk") in new stack
  342. -- Executing [s@macro-dialout-trunk:20] Dial("SIP/3217-0000000c", "SIP/Trunk/9050555555,300,") in new stack
  343. == Using SIP RTP TOS bits 184
  344. == Using SIP RTP CoS mark 5
  345. Audio is at 25750
  346. Adding codec 0x100 (g729) to SDP
  347. Adding codec 0x4 (ulaw) to SDP
  348. Adding codec 0x8 (alaw) to SDP
  349. Adding non-codec 0x1 (telephone-event) to SDP
  350. Reliably Transmitting (no NAT) to 10.100.210.254:5060:
  351. INVITE sip:9050555555@10.100.210.254 SIP/2.0
  352. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK726a0e8a
  353. Max-Forwards: 70
  354. From: "01111" <sip:01111@172.20.255.50>;tag=as0c91354b
  355. To: <sip:9050555555@10.100.210.254>
  356. Contact: <sip:01111@172.20.255.50:5060>
  357. Call-ID: 65f3176c1b2981b91c46dc9136da2623@172.20.255.50:5060
  358. CSeq: 101 INVITE
  359. User-Agent: Asterisk
  360. Date: Fri, 08 Mar 2013 20:24:59 GMT
  361. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  362. Supported: replaces, timer
  363. Content-Type: application/sdp
  364. Content-Length: 310
  365.  
  366. v=0
  367. o=root 1739091036 1739091036 IN IP4 172.20.255.50
  368. s=Asterisk PBX 1.8.14.0
  369. c=IN IP4 172.20.255.50
  370. t=0 0
  371. m=audio 25750 RTP/AVP 18 0 8 101
  372. a=rtpmap:18 G729/8000
  373. a=fmtp:18 annexb=no
  374. a=rtpmap:0 PCMU/8000
  375. a=rtpmap:8 PCMA/8000
  376. a=rtpmap:101 telephone-event/8000
  377. a=fmtp:101 0-16
  378. a=ptime:20
  379. a=sendrecv
  380.  
  381. ---
  382. -- Called SIP/Trunk/9050555555
  383.  
  384. <--- SIP read from UDP:10.100.210.254:50510 --->
  385. SIP/2.0 100 Trying
  386. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK726a0e8a
  387. From: "01111" <sip:01111@172.20.255.50>;tag=as0c91354b
  388. To: <sip:9050555555@10.100.210.254>
  389. Date: Fri, 08 Mar 2013 20:25:00 GMT
  390. Call-ID: 65f3176c1b2981b91c46dc9136da2623@172.20.255.50:5060
  391. CSeq: 101 INVITE
  392. Allow-Events: telephone-event
  393. Server: Cisco-SIPGateway/IOS-12.x
  394. Content-Length: 0
  395.  
  396. <------------->
  397. --- (10 headers 0 lines) ---
  398.  
  399. <--- SIP read from UDP:10.100.210.254:50510 --->
  400. SIP/2.0 183 Session Progress
  401. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK726a0e8a
  402. From: "01111" <sip:01111@172.20.255.50>;tag=as0c91354b
  403. To: <sip:9050555555@10.100.210.254>;tag=26399E4C-EB0
  404. Date: Fri, 08 Mar 2013 20:25:00 GMT
  405. Call-ID: 65f3176c1b2981b91c46dc9136da2623@172.20.255.50:5060
  406. CSeq: 101 INVITE
  407. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
  408. Allow-Events: telephone-event
  409. Remote-Party-ID: <sip:050555555@10.100.210.254>;party=called;screen=no;privacy=off
  410. Contact: <sip:9050555555@10.100.210.254:5060>
  411. Supported: sdp-anat
  412. Server: Cisco-SIPGateway/IOS-12.x
  413. Content-Type: application/sdp
  414. Content-Disposition: session;handling=required
  415. Content-Length: 220
  416.  
  417. v=0
  418. o=CiscoSystemsSIP-GW-UserAgent 3635 7391 IN IP4 10.100.210.254
  419. s=SIP Call
  420. c=IN IP4 10.100.210.254
  421. t=0 0
  422. m=audio 17302 RTP/AVP 18
  423. c=IN IP4 10.100.210.254
  424. a=rtpmap:18 G729/8000
  425. a=fmtp:18 annexb=no
  426. a=ptime:20
  427. <------------->
  428. --- (16 headers 10 lines) ---
  429. list_route: hop: <sip:9050555555@10.100.210.254:5060>
  430. Found RTP audio format 18
  431. Found audio description format G729 for ID 18
  432. Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
  433. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
  434. Peer audio RTP is at port 10.100.210.254:17302
  435. -- SIP/Trunk-0000000d is making progress passing it to SIP/3217-0000000c
  436. Audio is at 16414
  437. Adding codec 0x100 (g729) to SDP
  438. Adding non-codec 0x1 (telephone-event) to SDP
  439.  
  440. <--- Transmitting (no NAT) to 10.200.210.51:6491 --->
  441. SIP/2.0 183 Session Progress
  442. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-149506423-1--d87543-;received=10.200.210.51;rport=6491
  443. From: h<sip:3217@172.20.255.50>;tag=ef31bf1d
  444. To: <sip:9050555555@172.20.255.50>;tag=as447a99fe
  445. Call-ID: f15355682030a66d
  446. CSeq: 2 INVITE
  447. Server: Asterisk
  448. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  449. Supported: replaces, timer
  450. Contact: <sip:9050555555@172.20.255.50:5060>
  451. Content-Type: application/sdp
  452. Content-Length: 262
  453.  
  454. v=0
  455. o=root 1943296419 1943296419 IN IP4 172.20.255.50
  456. s=Asterisk PBX 1.8.14.0
  457. c=IN IP4 172.20.255.50
  458. t=0 0
  459. m=audio 16414 RTP/AVP 18 101
  460. a=rtpmap:18 G729/8000
  461. a=fmtp:18 annexb=no
  462. a=rtpmap:101 telephone-event/8000
  463. a=fmtp:101 0-16
  464. a=ptime:40
  465. a=sendrecv
  466.  
  467. <------------>
  468.  
  469. <--- SIP read from UDP:10.100.210.254:50510 --->
  470. SIP/2.0 200 OK
  471. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK726a0e8a
  472. From: "01111" <sip:01111@172.20.255.50>;tag=as0c91354b
  473. To: <sip:9050555555@10.100.210.254>;tag=26399E4C-EB0
  474. Date: Fri, 08 Mar 2013 20:25:00 GMT
  475. Call-ID: 65f3176c1b2981b91c46dc9136da2623@172.20.255.50:5060
  476. CSeq: 101 INVITE
  477. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
  478. Allow-Events: telephone-event
  479. Remote-Party-ID: <sip:050555555@10.100.210.254>;party=called;screen=no;privacy=off
  480. Contact: <sip:9050555555@10.100.210.254:5060>
  481. Supported: replaces
  482. Supported: sdp-anat
  483. Server: Cisco-SIPGateway/IOS-12.x
  484. Supported: timer
  485. Content-Type: application/sdp
  486. Content-Disposition: session;handling=required
  487. Content-Length: 220
  488.  
  489. v=0
  490. o=CiscoSystemsSIP-GW-UserAgent 3635 7391 IN IP4 10.100.210.254
  491. s=SIP Call
  492. c=IN IP4 10.100.210.254
  493. t=0 0
  494. m=audio 17302 RTP/AVP 18
  495. c=IN IP4 10.100.210.254
  496. a=rtpmap:18 G729/8000
  497. a=fmtp:18 annexb=no
  498. a=ptime:20
  499. <------------->
  500. --- (18 headers 10 lines) ---
  501. list_route: hop: <sip:9050555555@10.100.210.254:5060>
  502. set_destination: Parsing <sip:9050555555@10.100.210.254:5060> for address/port to send to
  503. set_destination: set destination to 10.100.210.254:5060
  504. Transmitting (no NAT) to 10.100.210.254:5060:
  505. ACK sip:9050555555@10.100.210.254:5060 SIP/2.0
  506. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK46af4eff
  507. Max-Forwards: 70
  508. From: "01111" <sip:01111@172.20.255.50>;tag=as0c91354b
  509. To: <sip:9050555555@10.100.210.254>;tag=26399E4C-EB0
  510. Contact: <sip:01111@172.20.255.50:5060>
  511. Call-ID: 65f3176c1b2981b91c46dc9136da2623@172.20.255.50:5060
  512. CSeq: 101 ACK
  513. User-Agent: Asterisk
  514. Content-Length: 0
  515.  
  516.  
  517. ---
  518. -- SIP/Trunk-0000000d answered SIP/3217-0000000c
  519. Audio is at 16414
  520. Adding codec 0x100 (g729) to SDP
  521. Adding non-codec 0x1 (telephone-event) to SDP
  522.  
  523. <--- Reliably Transmitting (no NAT) to 10.200.210.51:6491 --->
  524. SIP/2.0 200 OK
  525. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-149506423-1--d87543-;received=10.200.210.51;rport=6491
  526. From: h<sip:3217@172.20.255.50>;tag=ef31bf1d
  527. To: <sip:9050555555@172.20.255.50>;tag=as447a99fe
  528. Call-ID: f15355682030a66d
  529. CSeq: 2 INVITE
  530. Server: Asterisk
  531. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  532. Supported: replaces, timer
  533. Contact: <sip:9050555555@172.20.255.50:5060>
  534. Remote-Party-ID: "CID:01111" <sip:050555555@172.20.255.50>;party=called;privacy=off;screen=no
  535. Content-Type: application/sdp
  536. Content-Length: 262
  537.  
  538. v=0
  539. o=root 1943296419 1943296420 IN IP4 172.20.255.50
  540. s=Asterisk PBX 1.8.14.0
  541. c=IN IP4 172.20.255.50
  542. t=0 0
  543. m=audio 16414 RTP/AVP 18 101
  544. a=rtpmap:18 G729/8000
  545. a=fmtp:18 annexb=no
  546. a=rtpmap:101 telephone-event/8000
  547. a=fmtp:101 0-16
  548. a=ptime:40
  549. a=sendrecv
  550.  
  551. <------------>
  552.  
  553. <--- SIP read from UDP:10.200.210.51:6491 --->
  554. ACK sip:9050555555@172.20.255.50:5060 SIP/2.0
  555. To: <sip:9050555555@172.20.255.50>;tag=as447a99fe
  556. From: h<sip:3217@172.20.255.50>;tag=ef31bf1d
  557. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-322595709-1--d87543-;rport
  558. Call-ID: f15355682030a66d
  559. CSeq: 2 ACK
  560. Contact: <sip:3217@10.200.210.51:6491>
  561. Max-Forwards: 70
  562. User-Agent: eyeBeam release 3006o stamp 17551
  563. Authorization: Digest username="3217",realm="asterisk",nonce="2dff0cbb",uri="sip:9050555555@172.20.255.50",response="5426822667bb878569ee583c80f153bc",algorithm=MD5
  564. Content-Length: 0
  565.  
  566. <------------->
  567. --- (11 headers 0 lines) ---
  568.  
  569. <--- SIP read from UDP:10.200.210.51:6491 --->
  570.  
  571. <------------->
  572.  
  573. <--- SIP read from UDP:10.200.210.51:6491 --->
  574. BYE sip:9050555555@172.20.255.50:5060 SIP/2.0
  575. To: <sip:9050555555@172.20.255.50>;tag=as447a99fe
  576. From: h<sip:3217@172.20.255.50>;tag=ef31bf1d
  577. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-321263389-1--d87543-;rport
  578. Call-ID: f15355682030a66d
  579. CSeq: 3 BYE
  580. Contact: <sip:3217@10.200.210.51:6491>
  581. Max-Forwards: 70
  582. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  583. User-Agent: eyeBeam release 3006o stamp 17551
  584. Authorization: Digest username="3217",realm="asterisk",nonce="2dff0cbb",uri="sip:9050555555@172.20.255.50:5060",response="835a7806ada2978a68f9e7e2f3fd1022",algorithm=MD5
  585. Content-Length: 0
  586.  
  587. <------------->
  588. --- (12 headers 0 lines) ---
  589. Sending to 10.200.210.51:6491 (no NAT)
  590. Scheduling destruction of SIP dialog 'f15355682030a66d' in 25152 ms (Method: BYE)
  591.  
  592. <--- Transmitting (no NAT) to 10.200.210.51:6491 --->
  593. SIP/2.0 200 OK
  594. Via: SIP/2.0/UDP 10.200.210.51:6491;branch=z9hG4bK-d87543-321263389-1--d87543-;received=10.200.210.51;rport=6491
  595. From: h<sip:3217@172.20.255.50>;tag=ef31bf1d
  596. To: <sip:9050555555@172.20.255.50>;tag=as447a99fe
  597. Call-ID: f15355682030a66d
  598. CSeq: 3 BYE
  599. Server: Asterisk
  600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  601. Supported: replaces, timer
  602. Content-Length: 0
  603.  
  604.  
  605. <------------>
  606. -- Executing [h@macro-dialout-trunk:1] Macro("SIP/3217-0000000c", "hangupcall,") in new stack
  607. -- Executing [s@macro-hangupcall:1] GotoIf("SIP/3217-0000000c", "1?theend") in new stack
  608. -- Goto (macro-hangupcall,s,3)
  609. -- Executing [s@macro-hangupcall:3] ExecIf("SIP/3217-0000000c", "0?Set(CDR(recordingfile)=)") in new stack
  610. -- Executing [s@macro-hangupcall:4] Hangup("SIP/3217-0000000c", "") in new stack
  611. == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/3217-0000000c' in macro 'hangupcall'
  612. == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/3217-0000000c'
  613. Scheduling destruction of SIP dialog '65f3176c1b2981b91c46dc9136da2623@172.20.255.50:5060' in 32000 ms (Method: INVITE)
  614. set_destination: Parsing <sip:9050555555@10.100.210.254:5060> for address/port to send to
  615. set_destination: set destination to 10.100.210.254:5060
  616. Reliably Transmitting (no NAT) to 10.100.210.254:5060:
  617. BYE sip:9050555555@10.100.210.254:5060 SIP/2.0
  618. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK42b8c1cd
  619. Max-Forwards: 70
  620. From: "01111" <sip:01111@172.20.255.50>;tag=as0c91354b
  621. To: <sip:9050555555@10.100.210.254>;tag=26399E4C-EB0
  622. Call-ID: 65f3176c1b2981b91c46dc9136da2623@172.20.255.50:5060
  623. CSeq: 102 BYE
  624. User-Agent: Asterisk
  625. X-Asterisk-HangupCause: Normal Clearing
  626. X-Asterisk-HangupCauseCode: 16
  627. Content-Length: 0
  628.  
  629.  
  630. ---
  631. == Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/3217-0000000c' in macro 'dialout-trunk'
  632. == Spawn extension (forall, 9050555555, 5) exited non-zero on 'SIP/3217-0000000c'
  633.  
  634. <--- SIP read from UDP:10.100.210.254:50510 --->
  635. SIP/2.0 200 OK
  636. Via: SIP/2.0/UDP 172.20.255.50:5060;branch=z9hG4bK42b8c1cd
  637. From: "01111" <sip:01111@172.20.255.50>;tag=as0c91354b
  638. To: <sip:9050555555@10.100.210.254>;tag=26399E4C-EB0
  639. Date: Fri, 08 Mar 2013 20:25:09 GMT
  640. Call-ID: 65f3176c1b2981b91c46dc9136da2623@172.20.255.50:5060
  641. Server: Cisco-SIPGateway/IOS-12.x
  642. CSeq: 102 BYE
  643. Reason: Q.850;cause=16
  644. P-RTP-Stat: PS=134,OS=2680,PR=78,OR=3120,PL=0,JI=1,LA=348,DU=6
  645. Content-Length: 0
  646.  
  647. <------------->
  648. --- (11 headers 0 lines) ---
  649. Really destroying SIP dialog '65f3176c1b2981b91c46dc9136da2623@172.20.255.50:5060' Method: INVITE
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