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  1.  
  2. Mytaxi*CLI> sip set debug
  3. SIP Debugging enabled
  4.  
  5. <--- SIP read from 10.200.7.157:5060 --->
  6. OPTIONS sip:172.29.44.242:5060 SIP/2.0
  7. Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKoah2d7uph7stebakstdtaafoeT35305
  8. Call-ID: isbca7tepfdkdf7ftuse2oka4fpphtf2kc44@SoftX3000
  9. From: <sip:172.29.44.242:5060>;tag=sbc0805ekub4c4c
  10. To: <sip:172.29.44.242>
  11. CSeq: 1 OPTIONS
  12. Max-Forwards: 70
  13. Content-Length: 0
  14.  
  15.  
  16. <------------->
  17. --- (8 headers 0 lines) ---
  18. Looking for s in default (domain 172.29.44.242)
  19.  
  20. <--- Transmitting (no NAT) to 10.200.7.157:5060 --->
  21. SIP/2.0 200 OK
  22. Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKoah2d7uph7stebakstdtaafoeT35305;received=10.200.7.157
  23. From: <sip:172.29.44.242:5060>;tag=sbc0805ekub4c4c
  24. To: <sip:172.29.44.242>;tag=as0e803207
  25. Call-ID: isbca7tepfdkdf7ftuse2oka4fpphtf2kc44@SoftX3000
  26. CSeq: 1 OPTIONS
  27. User-Agent: Asterisk PBX
  28. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  29. Supported: replaces
  30. Contact: <sip:172.29.44.242>
  31. Accept: application/sdp
  32. Content-Length: 0
  33.  
  34.  
  35. <------------>
  36. Scheduling destruction of SIP dialog 'isbca7tepfdkdf7ftuse2oka4fpphtf2kc44@SoftX3000' in 32000 ms (Method: OPTIONS)
  37.  
  38. <--- SIP read from 192.168.1.75:20018 --->
  39. INVITE sip:0500302910@192.168.1.200 SIP/2.0
  40. Via: SIP/2.0/UDP 192.168.1.75:20018;branch=z9hG4bK-d8754z-2e752a1d8466b83a-1---d8754z-;rport
  41. Max-Forwards: 70
  42. Contact: <sip:803@192.168.1.75:20018>
  43. To: "0500302910"<sip:0500302910@192.168.1.200>
  44. From: "803"<sip:803@192.168.1.200>;tag=8439d733
  45. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  46. CSeq: 1 INVITE
  47. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  48. Content-Type: application/sdp
  49. User-Agent: X-Lite release 1103k stamp 53621
  50. Content-Length: 237
  51.  
  52. v=0
  53. o=- 0 2 IN IP4 192.168.1.75
  54. s=CounterPath X-Lite 3.0
  55. c=IN IP4 192.168.1.75
  56. t=0 0
  57. m=audio 61860 RTP/AVP 0 8 3 101
  58. a=alt:1 1 : BVbXU9Pp Zg3cM3LK 192.168.1.75 61860
  59. a=fmtp:101 0-15
  60. a=rtpmap:101 telephone-event/8000
  61. a=sendrecv
  62.  
  63. <------------->
  64. --- (12 headers 10 lines) ---
  65. Sending to 192.168.1.75 : 20018 (no NAT)
  66. Using INVITE request as basis request - NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  67.  
  68. <--- Reliably Transmitting (no NAT) to 192.168.1.75:20018 --->
  69. SIP/2.0 407 Proxy Authentication Required
  70. Via: SIP/2.0/UDP 192.168.1.75:20018;branch=z9hG4bK-d8754z-2e752a1d8466b83a-1---d8754z-;received=192.168.1.75;rport=20018
  71. From: "803"<sip:803@192.168.1.200>;tag=8439d733
  72. To: "0500302910"<sip:0500302910@192.168.1.200>;tag=as0e2bd3ef
  73. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  74. CSeq: 1 INVITE
  75. User-Agent: Asterisk PBX
  76. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  77. Supported: replaces
  78. Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="40f78b7b"
  79. Content-Length: 0
  80.  
  81.  
  82. <------------>
  83. Scheduling destruction of SIP dialog 'NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.' in 32000 ms (Method: INVITE)
  84. Found user '803'
  85.  
  86. <--- SIP read from 192.168.1.75:20018 --->
  87. ACK sip:0500302910@192.168.1.200 SIP/2.0
  88. Via: SIP/2.0/UDP 192.168.1.75:20018;branch=z9hG4bK-d8754z-2e752a1d8466b83a-1---d8754z-;rport
  89. To: "0500302910"<sip:0500302910@192.168.1.200>;tag=as0e2bd3ef
  90. From: "803"<sip:803@192.168.1.200>;tag=8439d733
  91. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  92. CSeq: 1 ACK
  93. Content-Length: 0
  94.  
  95.  
  96. <------------->
  97. --- (7 headers 0 lines) ---
  98.  
  99. <--- SIP read from 192.168.1.75:20018 --->
  100. INVITE sip:0500302910@192.168.1.200 SIP/2.0
  101. Via: SIP/2.0/UDP 192.168.1.75:20018;branch=z9hG4bK-d8754z-943305026329a510-1---d8754z-;rport
  102. Max-Forwards: 70
  103. Contact: <sip:803@192.168.1.75:20018>
  104. To: "0500302910"<sip:0500302910@192.168.1.200>
  105. From: "803"<sip:803@192.168.1.200>;tag=8439d733
  106. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  107. CSeq: 2 INVITE
  108. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  109. Content-Type: application/sdp
  110. Proxy-Authorization: Digest username="803",realm="asterisk",nonce="40f78b7b",uri="sip:0500302910@192.168.1.200",response="e7a192a20b7e528e5f635287dcb75932",algorithm=MD5
  111. User-Agent: X-Lite release 1103k stamp 53621
  112. Content-Length: 237
  113.  
  114. v=0
  115. o=- 0 2 IN IP4 192.168.1.75
  116. s=CounterPath X-Lite 3.0
  117. c=IN IP4 192.168.1.75
  118. t=0 0
  119. m=audio 61860 RTP/AVP 0 8 3 101
  120. a=alt:1 1 : BVbXU9Pp Zg3cM3LK 192.168.1.75 61860
  121. a=fmtp:101 0-15
  122. a=rtpmap:101 telephone-event/8000
  123. a=sendrecv
  124.  
  125. <------------->
  126. --- (13 headers 10 lines) ---
  127. Sending to 192.168.1.75 : 20018 (NAT)
  128. Using INVITE request as basis request - NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  129. Found user '803'
  130. Found RTP audio format 0
  131. Found RTP audio format 8
  132. Found RTP audio format 3
  133. Found RTP audio format 101
  134. Found audio description format telephone-event for ID 101
  135. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
  136. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  137. Peer audio RTP is at port 192.168.1.75:61860
  138. Looking for 0500302910 in internal (domain 192.168.1.200)
  139. list_route: hop: <sip:803@192.168.1.75:20018>
  140.  
  141. <--- Transmitting (NAT) to 192.168.1.75:20018 --->
  142. SIP/2.0 100 Trying
  143. Via: SIP/2.0/UDP 192.168.1.75:20018;branch=z9hG4bK-d8754z-943305026329a510-1---d8754z-;received=192.168.1.75;rport=20018
  144. From: "803"<sip:803@192.168.1.200>;tag=8439d733
  145. To: "0500302910"<sip:0500302910@192.168.1.200>
  146. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  147. CSeq: 2 INVITE
  148. User-Agent: Asterisk PBX
  149. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  150. Supported: replaces
  151. Contact: <sip:0500302910@192.168.1.200>
  152. Content-Length: 0
  153.  
  154.  
  155. <------------>
  156. -- Executing [0500302910@internal:1] Set("SIP/803-00000009", "CALLERID(num)=2847803") in new stack
  157. -- Executing [0500302910@internal:2] Dial("SIP/803-00000009", "SIP/STC-Outbound/0500302910") in new stack
  158. Audio is at 172.29.44.242 port 16414
  159. Adding codec 0x2 (gsm) to SDP
  160. Adding codec 0x4 (ulaw) to SDP
  161. Adding codec 0x8 (alaw) to SDP
  162. Adding non-codec 0x1 (telephone-event) to SDP
  163. Reliably Transmitting (no NAT) to 10.200.7.157:5060:
  164. INVITE sip:0500302910@10.200.7.157 SIP/2.0
  165. Via: SIP/2.0/UDP 172.29.44.242:5060;branch=z9hG4bK41a124ce;rport
  166. From: "803" <sip:2847803@172.29.44.242>;tag=as2d02b582
  167. To: <sip:0500302910@10.200.7.157>
  168. Contact: <sip:2847803@172.29.44.242>
  169. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  170. CSeq: 102 INVITE
  171. User-Agent: Asterisk PBX
  172. Max-Forwards: 70
  173. Date: Thu, 20 Mar 2014 19:49:54 GMT
  174. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  175. Supported: replaces
  176. Content-Type: application/sdp
  177. Content-Length: 262
  178.  
  179. v=0
  180. o=root 25106 25106 IN IP4 172.29.44.242
  181. s=session
  182. c=IN IP4 172.29.44.242
  183. t=0 0
  184. m=audio 16414 RTP/AVP 3 0 8 101
  185. a=rtpmap:3 GSM/8000
  186. a=rtpmap:0 PCMU/8000
  187. a=rtpmap:8 PCMA/8000
  188. a=rtpmap:101 telephone-event/8000
  189. a=fmtp:101 0-16
  190. a=ptime:20
  191. a=sendrecv
  192.  
  193. ---
  194. -- Called STC-Outbound/0500302910
  195.  
  196. <--- SIP read from 10.200.7.157:5060 --->
  197. SIP/2.0 100 Trying
  198. Via: SIP/2.0/UDP 172.29.44.242:5060;branch=z9hG4bK41a124ce;rport
  199. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  200. From: "803"<sip:2847803@172.29.44.242>;tag=as2d02b582
  201. To: <sip:0500302910@10.200.7.157>
  202. CSeq: 102 INVITE
  203. Content-Length: 0
  204.  
  205.  
  206. <------------->
  207. --- (7 headers 0 lines) ---
  208.  
  209. <--- SIP read from 10.200.7.157:5060 --->
  210. SIP/2.0 180 Ringing
  211. Via: SIP/2.0/UDP 172.29.44.242:5060;branch=z9hG4bK41a124ce;rport=5060
  212. Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>
  213. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  214. From: "803"<sip:2847803@172.29.44.242>;tag=as2d02b582
  215. To: <sip:0500302910@10.200.7.157>;tag=sbc0805ockk2ceu-CC-40
  216. CSeq: 102 INVITE
  217. Contact: <sip:0500302910@10.200.7.157:5060;user=phone>
  218. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
  219. Content-Length: 191
  220. Content-Type: application/sdp
  221.  
  222. v=0
  223. o=- 9614269 9614269 IN IP4 10.200.7.157
  224. s=SBC call
  225. c=IN IP4 10.200.7.157
  226. t=0 0
  227. m=audio 49688 RTP/AVP 8 101
  228. a=rtpmap:8 PCMA/8000
  229. a=rtpmap:101 telephone-event/8000
  230. a=fmtp:101 0-15
  231.  
  232. <------------->
  233. --- (11 headers 9 lines) ---
  234. Found RTP audio format 8
  235. Found RTP audio format 101
  236. Found audio description format PCMA for ID 8
  237. Found audio description format telephone-event for ID 101
  238. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
  239. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  240. Peer audio RTP is at port 10.200.7.157:49688
  241. -- SIP/STC-Outbound-0000000a is ringing
  242.  
  243. <--- Transmitting (NAT) to 192.168.1.75:20018 --->
  244. SIP/2.0 180 Ringing
  245. Via: SIP/2.0/UDP 192.168.1.75:20018;branch=z9hG4bK-d8754z-943305026329a510-1---d8754z-;received=192.168.1.75;rport=20018
  246. From: "803"<sip:803@192.168.1.200>;tag=8439d733
  247. To: "0500302910"<sip:0500302910@192.168.1.200>;tag=as62d0ca82
  248. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  249. CSeq: 2 INVITE
  250. User-Agent: Asterisk PBX
  251. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  252. Supported: replaces
  253. Contact: <sip:0500302910@192.168.1.200>
  254. Content-Length: 0
  255.  
  256.  
  257. <------------>
  258. -- SIP/STC-Outbound-0000000a is making progress passing it to SIP/803-00000009
  259. Audio is at 192.168.1.200 port 19582
  260. Adding codec 0x2 (gsm) to SDP
  261. Adding codec 0x4 (ulaw) to SDP
  262. Adding codec 0x8 (alaw) to SDP
  263. Adding non-codec 0x1 (telephone-event) to SDP
  264.  
  265. <--- Transmitting (NAT) to 192.168.1.75:20018 --->
  266. SIP/2.0 183 Session Progress
  267. Via: SIP/2.0/UDP 192.168.1.75:20018;branch=z9hG4bK-d8754z-943305026329a510-1---d8754z-;received=192.168.1.75;rport=20018
  268. From: "803"<sip:803@192.168.1.200>;tag=8439d733
  269. To: "0500302910"<sip:0500302910@192.168.1.200>;tag=as62d0ca82
  270. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  271. CSeq: 2 INVITE
  272. User-Agent: Asterisk PBX
  273. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  274. Supported: replaces
  275. Contact: <sip:0500302910@192.168.1.200>
  276. Content-Type: application/sdp
  277. Content-Length: 262
  278.  
  279. v=0
  280. o=root 25106 25106 IN IP4 192.168.1.200
  281. s=session
  282. c=IN IP4 192.168.1.200
  283. t=0 0
  284. m=audio 19582 RTP/AVP 3 0 8 101
  285. a=rtpmap:3 GSM/8000
  286. a=rtpmap:0 PCMU/8000
  287. a=rtpmap:8 PCMA/8000
  288. a=rtpmap:101 telephone-event/8000
  289. a=fmtp:101 0-16
  290. a=ptime:20
  291. a=sendrecv
  292.  
  293. <------------>
  294.  
  295. <--- SIP read from 10.200.7.157:5060 --->
  296. SIP/2.0 200 OK
  297. Via: SIP/2.0/UDP 172.29.44.242:5060;branch=z9hG4bK41a124ce;rport=5060
  298. Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>
  299. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  300. From: "803"<sip:2847803@172.29.44.242>;tag=as2d02b582
  301. To: <sip:0500302910@10.200.7.157>;tag=sbc0805ockk2ceu-CC-40
  302. CSeq: 102 INVITE
  303. Contact: <sip:0500302910@10.200.7.157:5060;user=phone>
  304. Content-Length: 191
  305. Content-Type: application/sdp
  306.  
  307. v=0
  308. o=- 9614269 9614270 IN IP4 10.200.7.157
  309. s=SBC call
  310. c=IN IP4 10.200.7.157
  311. t=0 0
  312. m=audio 49688 RTP/AVP 8 101
  313. a=rtpmap:8 PCMA/8000
  314. a=rtpmap:101 telephone-event/8000
  315. a=fmtp:101 0-15
  316.  
  317. <------------->
  318. --- (10 headers 9 lines) ---
  319. Found RTP audio format 8
  320. Found RTP audio format 101
  321. Found audio description format PCMA for ID 8
  322. Found audio description format telephone-event for ID 101
  323. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
  324. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  325. Peer audio RTP is at port 10.200.7.157:49688
  326. list_route: hop: <sip:10.200.7.157:5060;transport=udp;lr>
  327. set_destination: Parsing <sip:10.200.7.157:5060;transport=udp;lr> for address/port to send to
  328. set_destination: set destination to 10.200.7.157, port 5060
  329. Transmitting (no NAT) to 10.200.7.157:5060:
  330. ACK sip:0500302910@10.200.7.157:5060;user=phone SIP/2.0
  331. Via: SIP/2.0/UDP 172.29.44.242:5060;branch=z9hG4bK0b2faebe;rport
  332. Route: <sip:10.200.7.157:5060;transport=udp;lr>
  333. From: "803" <sip:2847803@172.29.44.242>;tag=as2d02b582
  334. To: <sip:0500302910@10.200.7.157>;tag=sbc0805ockk2ceu-CC-40
  335. Contact: <sip:2847803@172.29.44.242>
  336. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  337. CSeq: 102 ACK
  338. User-Agent: Asterisk PBX
  339. Max-Forwards: 70
  340. Content-Length: 0
  341.  
  342.  
  343. --
  344. -- SIP/STC-Outbound-0000000a answered SIP/803-00000009
  345. Audio is at 192.168.1.200 port 19582
  346. Adding codec 0x2 (gsm) 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.  
  351. <--- Reliably Transmitting (NAT) to 192.168.1.75:20018 --->
  352. SIP/2.0 200 OK
  353. Via: SIP/2.0/UDP 192.168.1.75:20018;branch=z9hG4bK-d8754z-943305026329a510-1---d8754z-;received=192.168.1.75;rport=20018
  354. From: "803"<sip:803@192.168.1.200>;tag=8439d733
  355. To: "0500302910"<sip:0500302910@192.168.1.200>;tag=as62d0ca82
  356. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  357. CSeq: 2 INVITE
  358. User-Agent: Asterisk PBX
  359. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  360. Supported: replaces
  361. Contact: <sip:0500302910@192.168.1.200>
  362. Content-Type: application/sdp
  363. Content-Length: 262
  364.  
  365. v=0
  366. o=root 25106 25107 IN IP4 192.168.1.200
  367. s=session
  368. c=IN IP4 192.168.1.200
  369. t=0 0
  370. m=audio 19582 RTP/AVP 3 0 8 101
  371. a=rtpmap:3 GSM/8000
  372. a=rtpmap:0 PCMU/8000
  373. a=rtpmap:8 PCMA/8000
  374. a=rtpmap:101 telephone-event/8000
  375. a=fmtp:101 0-16
  376. a=ptime:20
  377. a=sendrecv
  378.  
  379. <------------>
  380. -- Native bridging SIP/803-00000009 and SIP/STC-Outbound-0000000a
  381. set_destination: Parsing <sip:10.200.7.157:5060;transport=udp;lr> for address/port to send to
  382. set_destination: set destination to 10.200.7.157, port 5060
  383. Audio is at 172.29.44.242 port 16414
  384. Adding codec 0x8 (alaw) to SDP
  385. Adding non-codec 0x1 (telephone-event) to SDP
  386. Reliably Transmitting (no NAT) to 10.200.7.157:5060:
  387. INVITE sip:0500302910@10.200.7.157:5060;user=phone SIP/2.0
  388. Via: SIP/2.0/UDP 172.29.44.242:5060;branch=z9hG4bK5fc20520;rport
  389. Route: <sip:10.200.7.157:5060;transport=udp;lr>
  390. From: "803" <sip:2847803@172.29.44.242>;tag=as2d02b582
  391. To: <sip:0500302910@10.200.7.157>;tag=sbc0805ockk2ceu-CC-40
  392. Contact: <sip:2847803@172.29.44.242>
  393. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  394. CSeq: 103 INVITE
  395. User-Agent: Asterisk PBX
  396. Max-Forwards: 70
  397. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  398. Supported: replaces
  399. X-asterisk-Info: SIP re-invite (External RTP bridge)
  400. Content-Type: application/sdp
  401. Content-Length: 213
  402.  
  403. v=0
  404. o=root 25106 25107 IN IP4 192.168.1.75
  405. s=session
  406. c=IN IP4 192.168.1.75
  407. t=0 0
  408. m=audio 61860 RTP/AVP 8 101
  409. a=rtpmap:8 PCMA/8000
  410. a=rtpmap:101 telephone-event/8000
  411. a=fmtp:101 0-16
  412. a=ptime:20
  413. a=sendrecv
  414.  
  415. ---
  416.  
  417. <--- SIP read from 10.200.7.157:5060 --->
  418. SIP/2.0 100 Trying
  419. Via: SIP/2.0/UDP 172.29.44.242:5060;branch=z9hG4bK5fc20520;rport
  420. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  421. From: "803"<sip:2847803@172.29.44.242>;tag=as2d02b582
  422. To: <sip:0500302910@10.200.7.157>;tag=sbc0805ockk2ceu-CC-40
  423. CSeq: 103 INVITE
  424. Content-Length: 0
  425.  
  426.  
  427. <------------->
  428. --- (7 headers 0 lines) ---
  429.  
  430. <--- SIP read from 10.200.7.157:5060 --->
  431. OPTIONS sip:2847803@172.29.44.242 SIP/2.0
  432. Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bK7h42kacdao2hud2u27cuba2ofT35396
  433. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  434. From: <sip:0500302910@10.200.7.157>;tag=sbc0805ockk2ceu-CC-40
  435. To: "803"<sip:2847803@172.29.44.242>;tag=as2d02b582
  436. CSeq: 1 OPTIONS
  437. Accept: application/sdp
  438. Max-Forwards: 70
  439. Content-Length: 0
  440.  
  441.  
  442. <------------->
  443. --- (9 headers 0 lines) ---
  444.  
  445. <--- Transmitting (no NAT) to 10.200.7.157:5060 --->
  446. SIP/2.0 200 OK
  447. Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bK7h42kacdao2hud2u27cuba2ofT35396;received=10.200.7.157
  448. From: <sip:0500302910@10.200.7.157>;tag=sbc0805ockk2ceu-CC-40
  449. To: "803"<sip:2847803@172.29.44.242>;tag=as2d02b582
  450. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  451. CSeq: 1 OPTIONS
  452. User-Agent: Asterisk PBX
  453. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  454. Supported: replaces
  455. Contact: <sip:2847803@172.29.44.242>
  456. Accept: application/sdp
  457. Content-Length: 0
  458.  
  459.  
  460. <------------>
  461.  
  462. <--- SIP read from 192.168.1.75:20018 --->
  463. ACK sip:0500302910@192.168.1.200 SIP/2.0
  464. Via: SIP/2.0/UDP 192.168.1.75:20018;branch=z9hG4bK-d8754z-99559158d50c3b71-1---d8754z-;rport
  465. Max-Forwards: 70
  466. Contact: <sip:803@192.168.1.75:20018>
  467. To: "0500302910"<sip:0500302910@192.168.1.200>;tag=as62d0ca82
  468. From: "803"<sip:803@192.168.1.200>;tag=8439d733
  469. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  470. CSeq: 2 ACK
  471. Proxy-Authorization: Digest username="803",realm="asterisk",nonce="40f78b7b",uri="sip:0500302910@192.168.1.200",response="e7a192a20b7e528e5f635287dcb75932",algorithm=MD5
  472. User-Agent: X-Lite release 1103k stamp 53621
  473. Content-Length: 0
  474.  
  475.  
  476. <------------->
  477. --- (11 headers 0 lines) ---
  478. set_destination: Parsing <sip:803@192.168.1.75:20018> for address/port to send to
  479. set_destination: set destination to 192.168.1.75, port 20018
  480. Audio is at 192.168.1.200 port 19582
  481. Adding codec 0x8 (alaw) to SDP
  482. Adding non-codec 0x1 (telephone-event) to SDP
  483. Reliably Transmitting (NAT) to 192.168.1.75:20018:
  484. INVITE sip:803@192.168.1.75:20018 SIP/2.0
  485. Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK582564c4;rport
  486. From: "0500302910"<sip:0500302910@192.168.1.200>;tag=as62d0ca82
  487. To: "803"<sip:803@192.168.1.200>;tag=8439d733
  488. Contact: <sip:0500302910@192.168.1.200>
  489. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  490. CSeq: 102 INVITE
  491. User-Agent: Asterisk PBX
  492. Max-Forwards: 70
  493. llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  494. Supported: replaces
  495. X-asterisk-Info: SIP re-invite (External RTP bridge)
  496. Content-Type: application/sdp
  497. Content-Length: 213
  498.  
  499. v=0
  500. o=root 25106 25108 IN IP4 10.200.7.157
  501. s=session
  502. c=IN IP4 10.200.7.157
  503. t=0 0
  504. m=audio 49688 RTP/AVP 8 101
  505. a=rtpmap:8 PCMA/8000
  506. a=rtpmap:101 telephone-event/8000
  507. a=fmtp:101 0-16
  508. a=ptime:20
  509. a=sendrecv
  510.  
  511. ---
  512.  
  513. <--- SIP read from 10.200.7.157:5060 --->
  514. SIP/2.0 200 OK
  515. Via: SIP/2.0/UDP 172.29.44.242:5060;branch=z9hG4bK5fc20520;rport=5060
  516. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  517. From: "803"<sip:2847803@172.29.44.242>;tag=as2d02b582
  518. To: <sip:0500302910@10.200.7.157>;tag=sbc0805ockk2ceu-CC-40
  519. CSeq: 103 INVITE
  520. Contact: <sip:0500302910@10.200.7.157:5060;user=phone>
  521. Content-Length: 191
  522. Content-Type: application/sdp
  523.  
  524. v=0
  525. o=- 9614269 9614271 IN IP4 10.200.7.157
  526. s=SBC call
  527. c=IN IP4 10.200.7.157
  528. t=0 0
  529. m=audio 49688 RTP/AVP 8 101
  530. a=rtpmap:8 PCMA/8000
  531. a=rtpmap:101 telephone-event/8000
  532. a=fmtp:101 0-15
  533.  
  534. <------------->
  535. --- (9 headers 9 lines) ---
  536. Found RTP audio format 8
  537. Found RTP audio format 101
  538. Found audio description format PCMA for ID 8
  539. Found audio description format telephone-event for ID 101
  540. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
  541. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  542. Peer audio RTP is at port 10.200.7.157:49688
  543. set_destination: Parsing <sip:10.200.7.157:5060;transport=udp;lr> for address/port to send to
  544. set_destination: set destination to 10.200.7.157, port 5060
  545. Transmitting (no NAT) to 10.200.7.157:5060:
  546. ACK sip:0500302910@10.200.7.157:5060;user=phone SIP/2.0
  547. Via: SIP/2.0/UDP 172.29.44.242:5060;branch=z9hG4bK76e1b761;rport
  548. Route: <sip:10.200.7.157:5060;transport=udp;lr>
  549. From: "803" <sip:2847803@172.29.44.242>;tag=as2d02b582
  550. To: <sip:0500302910@10.200.7.157>;tag=sbc0805ockk2ceu-CC-40
  551. Contact: <sip:2847803@172.29.44.242>
  552. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  553. CSeq: 103 ACK
  554. User-Agent: Asterisk PBX
  555. Max-Forwards: 70
  556. Content-Length: 0
  557.  
  558.  
  559. --
  560.  
  561. <--- SIP read from 10.200.7.157:5060 --->
  562. OPTIONS sip:2847803@172.29.44.242 SIP/2.0
  563. Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKfkckspaekdteaabufcoutsp4bT35398
  564. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  565. From: <sip:0500302910@10.200.7.157>;tag=sbc0805ockk2ceu-CC-40
  566. To: "803"<sip:2847803@172.29.44.242>;tag=as2d02b582
  567. CSeq: 2 OPTIONS
  568. Accept: application/sdp
  569. Max-Forwards: 70
  570. Content-Length: 0
  571.  
  572.  
  573. <------------->
  574. --- (9 headers 0 lines) ---
  575.  
  576. <--- Transmitting (no NAT) to 10.200.7.157:5060 --->
  577. SIP/2.0 200 OK
  578. Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKfkckspaekdteaabufcoutsp4bT35398;received=10.200.7.157
  579. From: <sip:0500302910@10.200.7.157>;tag=sbc0805ockk2ceu-CC-40
  580. To: "803"<sip:2847803@172.29.44.242>;tag=as2d02b582
  581. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  582. CSeq: 2 OPTIONS
  583. User-Agent: Asterisk PBX
  584. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  585. Supported: replaces
  586. Contact: <sip:2847803@172.29.44.242>
  587. Accept: application/sdp
  588. Content-Length: 0
  589.  
  590.  
  591. <------------>
  592.  
  593. <--- SIP read from 192.168.1.75:20018 --->
  594. SIP/2.0 200 OK
  595. Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK582564c4;rport=5060
  596. Contact: <sip:803@192.168.1.75:20018>
  597. To: "803"<sip:803@192.168.1.200>;tag=8439d733
  598. From: "0500302910"<sip:0500302910@192.168.1.200>;tag=as62d0ca82
  599. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  600. CSeq: 102 INVITE
  601. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  602. Content-Type: application/sdp
  603. User-Agent: X-Lite release 1103k stamp 53621
  604. Content-Length: 183
  605.  
  606. v=0
  607. o=- 0 3 IN IP4 192.168.1.75
  608. s=CounterPath X-Lite 3.0
  609. c=IN IP4 192.168.1.75
  610. t=0 0
  611. m=audio 61860 RTP/AVP 8 101
  612. a=fmtp:101 0-15
  613. a=rtpmap:101 telephone-event/8000
  614. a=sendrecv
  615.  
  616. <------------->
  617. --- (11 headers 9 lines) ---
  618. Found RTP audio format 8
  619. Found RTP audio format 101
  620. Found audio description format telephone-event for ID 101
  621. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
  622. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  623. Peer audio RTP is at port 192.168.1.75:61860
  624. set_destination: Parsing <sip:803@192.168.1.75:20018> for address/port to send to
  625. set_destination: set destination to 192.168.1.75, port 20018
  626. Transmitting (NAT) to 192.168.1.75:20018:
  627. ACK sip:803@192.168.1.75:20018 SIP/2.0
  628. Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK53f791c4;rport
  629. From: "0500302910"<sip:0500302910@192.168.1.200>;tag=as62d0ca82
  630. To: "803"<sip:803@192.168.1.200>;tag=8439d733
  631. Contact: <sip:0500302910@192.168.1.200>
  632. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  633. CSeq: 102 ACK
  634. User-Agent: Asterisk PBX
  635. Max-Forwards: 70
  636. Content-Length: 0
  637.  
  638.  
  639. ---
  640. Really destroying SIP dialog 'isbcbut4edod4f4bfc2cfepshuehhpuhu7tc@SoftX3000' Method: OPTIONS
  641.  
  642. <--- SIP read from 192.168.1.75:20018 --->
  643.  
  644.  
  645.  
  646. <------------->
  647.  
  648. <--- SIP read from 10.200.7.157:5060 --->
  649. BYE sip:2847803@172.29.44.242 SIP/2.0
  650. Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKhbebscp724ssuocu7asokd4cpT35493
  651. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  652. From: <sip:0500302910@10.200.7.157>;tag=sbc0805ockk2ceu-CC-40
  653. To: "803"<sip:2847803@172.29.44.242>;tag=as2d02b582
  654. CSeq: 3 BYE
  655. Reason: Q.850;cause=16;text="normal call clearing"
  656. Max-Forwards: 70
  657. Content-Length: 0
  658.  
  659.  
  660. <------------->
  661. --- (9 headers 0 lines) ---
  662. Sending to 10.200.7.157 : 5060 (no NAT)
  663.  
  664. <--- Transmitting (no NAT) to 10.200.7.157:5060 --->
  665. SIP/2.0 200 OK
  666. Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKhbebscp724ssuocu7asokd4cpT35493;received=10.200.7.157
  667. From: <sip:0500302910@10.200.7.157>;tag=sbc0805ockk2ceu-CC-40
  668. To: "803"<sip:2847803@172.29.44.242>;tag=as2d02b582
  669. Call-ID: 26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242
  670. CSeq: 3 BYE
  671. User-Agent: Asterisk PBX
  672. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  673. Supported: replaces
  674. Content-Length: 0
  675.  
  676.  
  677. <------------>
  678. set_destination: Parsing <sip:803@192.168.1.75:20018> for address/port to send to
  679. set_destination: set destination to 192.168.1.75, port 20018
  680. Audio is at 192.168.1.200 port 19582
  681. Adding codec 0x8 (alaw) to SDP
  682. Adding non-codec 0x1 (telephone-event) to SDP
  683. Reliably Transmitting (NAT) to 192.168.1.75:20018:
  684. INVITE sip:803@192.168.1.75:20018 SIP/2.0
  685. Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK7fd13116;rport
  686. From: "0500302910"<sip:0500302910@192.168.1.200>;tag=as62d0ca82
  687. To: "803"<sip:803@192.168.1.200>;tag=8439d733
  688. Contact: <sip:0500302910@192.168.1.200>
  689. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  690. CSeq: 103 INVITE
  691. User-Agent: Asterisk PBX
  692. Max-Forwards: 70
  693. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  694. Supported: replaces
  695. X-asterisk-Info: SIP re-invite (External RTP bridge)
  696. Content-Type: application/sdp
  697. Content-Length: 215
  698.  
  699. v=0
  700. o=root 25106 25109 IN IP4 192.168.1.200
  701. s=session
  702. c=IN IP4 192.168.1.200
  703. t=0 0
  704. m=audio 19582 RTP/AVP 8 101
  705. a=rtpmap:8 PCMA/8000
  706. a=rtpmap:101 telephone-event/8000
  707. a=fmtp:101 0-16
  708. a=ptime:20
  709. a=sendrecv
  710.  
  711. ---
  712. == Spawn extension (internal, 0500302910, 2) exited non-zero on 'SIP/803-00000009'
  713. Scheduling destruction of SIP dialog 'NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.' in 32000 ms (Method: ACK)
  714. Really destroying SIP dialog '26cd7e184e3d37ab4b95518b0bab4e9a@172.29.44.242' Method: BYE
  715.  
  716. <--- SIP read from 192.168.1.75:20018 --->
  717. SIP/2.0 200 OK
  718. Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK7fd13116;rport=5060
  719. Contact: <sip:803@192.168.1.75:20018>
  720. To: "803"<sip:803@192.168.1.200>;tag=8439d733
  721. From: "0500302910"<sip:0500302910@192.168.1.200>;tag=as62d0ca82
  722. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  723. CSeq: 103 INVITE
  724. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  725. Content-Type: application/sdp
  726. User-Agent: X-Lite release 1103k stamp 53621
  727. Content-Length: 183
  728.  
  729. v=0
  730. o=- 0 3 IN IP4 192.168.1.75
  731. s=CounterPath X-Lite 3.0
  732. c=IN IP4 192.168.1.75
  733. t=0 0
  734. m=audio 61860 RTP/AVP 8 101
  735. a=fmtp:101 0-15
  736. a=rtpmap:101 telephone-event/8000
  737. a=sendrecv
  738.  
  739. <------------->
  740. --- (11 headers 9 lines) ---
  741. Found RTP audio format 8
  742. Found RTP audio format 101
  743. Found audio description format telephone-event for ID 101
  744. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
  745. Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  746. Peer audio RTP is at port 192.168.1.75:61860
  747. set_destination: Parsing <sip:803@192.168.1.75:20018> for address/port to send to
  748. set_destination: set destination to 192.168.1.75, port 20018
  749. Transmitting (NAT) to 192.168.1.75:20018:
  750. ACK sip:803@192.168.1.75:20018 SIP/2.0
  751. Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK7387d0c9;rport
  752. From: "0500302910"<sip:0500302910@192.168.1.200>;tag=as62d0ca82
  753. To: "803"<sip:803@192.168.1.200>;tag=8439d733
  754. Contact: <sip:0500302910@192.168.1.200>
  755. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  756. CSeq: 103 ACK
  757. User-Agent: Asterisk PBX
  758. Max-Forwards: 70
  759. Content-Length: 0
  760.  
  761.  
  762. ---
  763. set_destination: Parsing <sip:803@192.168.1.75:20018> for address/port to send to
  764. set_destination: set destination to 192.168.1.75, port 20018
  765. Reliably Transmitting (NAT) to 192.168.1.75:20018:
  766. BYE sip:803@192.168.1.75:20018 SIP/2.0
  767. Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK15a3910a;rport
  768. From: "0500302910"<sip:0500302910@192.168.1.200>;tag=as62d0ca82
  769. To: "803"<sip:803@192.168.1.200>;tag=8439d733
  770. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  771. CSeq: 104 BYE
  772. User-Agent: Asterisk PBX
  773. Max-Forwards: 70
  774. X-Asterisk-HangupCause: Normal Clearing
  775. X-Asterisk-HangupCauseCode: 16
  776. Content-Length: 0
  777.  
  778.  
  779. ---
  780. Scheduling destruction of SIP dialog 'NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.' in 32000 ms (Method: ACK)
  781.  
  782. <--- SIP read from 192.168.1.75:20018 --->
  783. SIP/2.0 200 OK
  784. Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK15a3910a;rport=5060
  785. Contact: <sip:803@192.168.1.75:20018>
  786. To: "803"<sip:803@192.168.1.200>;tag=8439d733
  787. From: "0500302910"<sip:0500302910@192.168.1.200>;tag=as62d0ca82
  788. Call-ID: NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.
  789. CSeq: 104 BYE
  790. User-Agent: X-Lite release 1103k stamp 53621
  791. Content-Length: 0
  792.  
  793.  
  794. <------------->
  795. --- (9 headers 0 lines) ---
  796. Really destroying SIP dialog 'NWVjODI4ZGVmN2QxMDc5ZDgzMmI2MmQ5ZjljNDUxOGE.' Method: ACK
  797.  
  798. <--- SIP read from 10.200.7.157:5060 --->
  799. OPTIONS sip:172.29.44.242:5060 SIP/2.0
  800. Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKt4eeu24k742etbe42ekhpeeatT21994
  801. Call-ID: isbcfeafdcp4pbpfpubhsduukshdc2foufh7@SoftX3000
  802. From: <sip:172.29.44.242:5060>;tag=sbc0804dos4hucp
  803. To: <sip:172.29.44.242>
  804. CSeq: 1 OPTIONS
  805. Max-Forwards: 70
  806. Content-Length: 0
  807.  
  808.  
  809. <------------->
  810. --- (8 headers 0 lines) ---
  811. Looking for s in default (domain 172.29.44.242)
  812.  
  813. <--- Transmitting (no NAT) to 10.200.7.157:5060 --->
  814. SIP/2.0 200 OK
  815. Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKt4eeu24k742etbe42ekhpeeatT21994;received=10.200.7.157
  816. From: <sip:172.29.44.242:5060>;tag=sbc0804dos4hucp
  817. To: <sip:172.29.44.242>;tag=as107e451f
  818. Call-ID: isbcfeafdcp4pbpfpubhsduukshdc2foufh7@SoftX3000
  819. CSeq: 1 OPTIONS
  820. User-Agent: Asterisk PBX
  821. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  822. Supported: replaces
  823. Contact: <sip:172.29.44.242>
  824. Accept: application/sdp
  825. Content-Length: 0
  826.  
  827.  
  828. <------------>
  829. Scheduling destruction of SIP dialog 'isbcfeafdcp4pbpfpubhsduukshdc2foufh7@SoftX3000' in 32000 ms (Method: OPTIONS)
  830. Mytaxi*CLI> sip set debug off
  831. SIP Debugging Disabled
  832. Mytaxi*CLI>
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