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  1. ------------>
  2. Audio is at 5060
  3. Adding codec 0x8 (alaw) to SDP
  4. Adding non-codec 0x1 (telephone-event) to SDP
  5. Reliably Transmitting (no NAT) to 192.168.1.11:5060:
  6. INVITE sip:elartey@192.168.1.11 SIP/2.0
  7. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK74c76cd1
  8. Max-Forwards: 70
  9. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  10. To: <sip:elartey@192.168.1.11>
  11. Contact: <sip:emma@192.168.1.155:5060>
  12. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  13. CSeq: 102 INVITE
  14. User-Agent: Asterisk PBX 1.8.5.0
  15. Date: Thu, 14 Jul 2011 18:45:59 GMT
  16. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  17. Supported: replaces, timer
  18. Content-Type: application/sdp
  19. Content-Length: 263
  20.  
  21. v=0
  22. o=root 542604981 542604981 IN IP4 192.168.1.155
  23. s=Asterisk PBX 1.8.5.0
  24. c=IN IP4 192.168.1.155
  25. t=0 0
  26. m=audio 16700 RTP/AVP 8 101
  27. a=rtpmap:8 PCMA/8000
  28. a=rtpmap:101 telephone-event/8000
  29. a=fmtp:101 0-16
  30. a=silenceSupp:off - - - -
  31. a=ptime:20
  32. a=sendrecv
  33.  
  34. ---
  35.  
  36. <--- SIP read from UDP:192.168.1.11:5060 --->
  37. SIP/2.0 100 Trying
  38. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK74c76cd1
  39. To: <sip:elartey@192.168.1.11>
  40. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  41. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  42. CSeq: 102 INVITE
  43. Server: Twinkle/1.4.2
  44. Content-Length: 0
  45.  
  46. <------------->
  47. --- (8 headers 0 lines) ---
  48.  
  49. <--- SIP read from UDP:192.168.1.11:5060 --->
  50. SIP/2.0 180 Ringing
  51. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK74c76cd1
  52. To: <sip:elartey@192.168.1.11>;tag=vjawn
  53. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  54. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  55. CSeq: 102 INVITE
  56. Contact: <sip:elartey@192.168.1.11>
  57. Server: Twinkle/1.4.2
  58. Content-Length: 0
  59.  
  60. <------------->
  61. --- (9 headers 0 lines) ---
  62.  
  63. <--- Transmitting (no NAT) to 127.0.0.1:5063 --->
  64. SIP/2.0 180 Ringing
  65. Via: SIP/2.0/UDP 127.0.0.1:5063;branch=z9hG4bKzltfvink;received=127.0.0.1;rport=5063
  66. From: "emma" <sip:emma@127.0.0.1>;tag=fzhus
  67. To: <sip:1002@127.0.0.1>;tag=as6b10a823
  68. Call-ID: bhiszntdfujazks@dove
  69. CSeq: 597 INVITE
  70. Server: Asterisk PBX 1.8.5.0
  71. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  72. Supported: replaces, timer
  73. Contact: <sip:1002@127.0.0.1:5060>
  74. Content-Length: 0
  75.  
  76.  
  77. <------------>
  78.  
  79. <--- SIP read from UDP:192.168.1.11:5060 --->
  80. SIP/2.0 200 OK
  81. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK74c76cd1
  82. To: <sip:elartey@192.168.1.11>;tag=vjawn
  83. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  84. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  85. CSeq: 102 INVITE
  86. Contact: <sip:elartey@192.168.1.11>
  87. Content-Type: application/sdp
  88. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  89. Server: Twinkle/1.4.2
  90. Supported: replaces,norefersub
  91. Content-Length: 206
  92.  
  93. v=0
  94. o=twinkle 1946095611 702177621 IN IP4 192.168.1.11
  95. s=-
  96. c=IN IP4 192.168.1.11
  97. t=0 0
  98. m=audio 8000 RTP/AVP 8 101
  99. a=rtpmap:8 PCMA/8000
  100. a=rtpmap:101 telephone-event/8000
  101. a=fmtp:101 0-15
  102. a=ptime:20
  103. <------------->
  104. --- (12 headers 10 lines) ---
  105. Found RTP audio format 8
  106. Found RTP audio format 101
  107. Found audio description format PCMA for ID 8
  108. Found audio description format telephone-event for ID 101
  109. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  110. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  111. Peer audio RTP is at port 192.168.1.11:8000
  112. list_route: hop: <sip:elartey@192.168.1.11>
  113. set_destination: Parsing <sip:elartey@192.168.1.11> for address/port to send to
  114. set_destination: set destination to 192.168.1.11:5060
  115. Transmitting (no NAT) to 192.168.1.11:5060:
  116. ACK sip:elartey@192.168.1.11 SIP/2.0
  117. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK26af192e
  118. Max-Forwards: 70
  119. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  120. To: <sip:elartey@192.168.1.11>;tag=vjawn
  121. Contact: <sip:emma@192.168.1.155:5060>
  122. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  123. CSeq: 102 ACK
  124. User-Agent: Asterisk PBX 1.8.5.0
  125. Content-Length: 0
  126.  
  127.  
  128. ---
  129. Audio is at 5060
  130. Adding codec 0x8 (alaw) to SDP
  131. Adding non-codec 0x1 (telephone-event) to SDP
  132.  
  133. <--- Reliably Transmitting (no NAT) to 127.0.0.1:5063 --->
  134. SIP/2.0 200 OK
  135. Via: SIP/2.0/UDP 127.0.0.1:5063;branch=z9hG4bKzltfvink;received=127.0.0.1;rport=5063
  136. From: "emma" <sip:emma@127.0.0.1>;tag=fzhus
  137. To: <sip:1002@127.0.0.1>;tag=as6b10a823
  138. Call-ID: bhiszntdfujazks@dove
  139. CSeq: 597 INVITE
  140. Server: Asterisk PBX 1.8.5.0
  141. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  142. Supported: replaces, timer
  143. Contact: <sip:1002@127.0.0.1:5060>
  144. Content-Type: application/sdp
  145. Content-Length: 257
  146.  
  147. v=0
  148. o=root 1364756848 1364756848 IN IP4 127.0.0.1
  149. s=Asterisk PBX 1.8.5.0
  150. c=IN IP4 127.0.0.1
  151. t=0 0
  152. m=audio 12444 RTP/AVP 8 101
  153. a=rtpmap:8 PCMA/8000
  154. a=rtpmap:101 telephone-event/8000
  155. a=fmtp:101 0-16
  156. a=silenceSupp:off - - - -
  157. a=ptime:20
  158. a=sendrecv
  159.  
  160. <------------>
  161. -- Remotely bridging SIP/emma-00000005 and SIP/elartey-00000006
  162. set_destination: Parsing <sip:elartey@192.168.1.11> for address/port to send to
  163. set_destination: set destination to 192.168.1.11:5060
  164. Audio is at 5060
  165. Adding codec 0x8 (alaw) to SDP
  166. Adding non-codec 0x1 (telephone-event) to SDP
  167. Reliably Transmitting (no NAT) to 192.168.1.11:5060:
  168. INVITE sip:elartey@192.168.1.11 SIP/2.0
  169. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK22e76237
  170. Max-Forwards: 70
  171. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  172. To: <sip:elartey@192.168.1.11>;tag=vjawn
  173. Contact: <sip:emma@192.168.1.155:5060>
  174. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  175. CSeq: 103 INVITE
  176. User-Agent: Asterisk PBX 1.8.5.0
  177. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  178. Supported: replaces, timer
  179. X-asterisk-Info: SIP re-invite (External RTP bridge)
  180. Content-Type: application/sdp
  181. Content-Length: 254
  182.  
  183. v=0
  184. o=root 542604981 542604982 IN IP4 127.0.0.1
  185. s=Asterisk PBX 1.8.5.0
  186. c=IN IP4 127.0.0.1
  187. t=0 0
  188. m=audio 8000 RTP/AVP 8 101
  189. a=rtpmap:8 PCMA/8000
  190. a=rtpmap:101 telephone-event/8000
  191. a=fmtp:101 0-16
  192. a=silenceSupp:off - - - -
  193. a=ptime:20
  194. a=sendrecv
  195.  
  196. ---
  197.  
  198. <--- SIP read from UDP:192.168.1.11:5060 --->
  199. SIP/2.0 200 OK
  200. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK22e76237
  201. To: <sip:elartey@192.168.1.11>;tag=vjawn
  202. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  203. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  204. CSeq: 103 INVITE
  205. Contact: <sip:elartey@192.168.1.11>
  206. Content-Type: application/sdp
  207. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  208. Server: Twinkle/1.4.2
  209. Supported: replaces,norefersub
  210. Content-Length: 206
  211.  
  212. v=0
  213. o=twinkle 1946095611 702177622 IN IP4 192.168.1.11
  214. s=-
  215. c=IN IP4 192.168.1.11
  216. t=0 0
  217. m=audio 8000 RTP/AVP 8 101
  218. a=rtpmap:8 PCMA/8000
  219. a=rtpmap:101 telephone-event/8000
  220. a=fmtp:101 0-15
  221. a=ptime:20
  222. <------------->
  223. --- (12 headers 10 lines) ---
  224. Found RTP audio format 8
  225. Found RTP audio format 101
  226. Found audio description format PCMA for ID 8
  227. Found audio description format telephone-event for ID 101
  228. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  229. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  230. Peer audio RTP is at port 192.168.1.11:8000
  231. set_destination: Parsing <sip:elartey@192.168.1.11> for address/port to send to
  232. set_destination: set destination to 192.168.1.11:5060
  233. Transmitting (no NAT) to 192.168.1.11:5060:
  234. ACK sip:elartey@192.168.1.11 SIP/2.0
  235. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK4a051eec
  236. Max-Forwards: 70
  237. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  238. To: <sip:elartey@192.168.1.11>;tag=vjawn
  239. Contact: <sip:emma@192.168.1.155:5060>
  240. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  241. CSeq: 103 ACK
  242. User-Agent: Asterisk PBX 1.8.5.0
  243. Content-Length: 0
  244.  
  245.  
  246. ---
  247.  
  248. <--- SIP read from UDP:127.0.0.1:5063 --->
  249. ACK sip:1002@127.0.0.1:5060 SIP/2.0
  250. Via: SIP/2.0/UDP 127.0.0.1:5063;rport;branch=z9hG4bKgolnyeml
  251. Max-Forwards: 70
  252. To: <sip:1002@127.0.0.1>;tag=as6b10a823
  253. From: "emma" <sip:emma@127.0.0.1>;tag=fzhus
  254. Call-ID: bhiszntdfujazks@dove
  255. CSeq: 597 ACK
  256. User-Agent: Twinkle/1.4.2
  257. Content-Length: 0
  258.  
  259. <------------->
  260. --- (9 headers 0 lines) ---
  261. set_destination: Parsing <sip:emma@127.0.0.1:5063> for address/port to send to
  262. set_destination: set destination to 127.0.0.1:5063
  263. Audio is at 5060
  264. Adding codec 0x8 (alaw) to SDP
  265. Adding non-codec 0x1 (telephone-event) to SDP
  266. Reliably Transmitting (no NAT) to 127.0.0.1:5063:
  267. INVITE sip:emma@127.0.0.1:5063 SIP/2.0
  268. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK20ff81d2;rport
  269. Max-Forwards: 70
  270. From: <sip:1002@127.0.0.1>;tag=as6b10a823
  271. To: "emma" <sip:emma@127.0.0.1>;tag=fzhus
  272. Contact: <sip:1002@127.0.0.1:5060>
  273. Call-ID: bhiszntdfujazks@dove
  274. CSeq: 102 INVITE
  275. User-Agent: Asterisk PBX 1.8.5.0
  276. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  277. Supported: replaces, timer
  278. X-asterisk-Info: SIP re-invite (External RTP bridge)
  279. Content-Type: application/sdp
  280. Content-Length: 262
  281.  
  282. v=0
  283. o=root 1364756848 1364756849 IN IP4 192.168.1.11
  284. s=Asterisk PBX 1.8.5.0
  285. c=IN IP4 192.168.1.11
  286. t=0 0
  287. m=audio 8000 RTP/AVP 8 101
  288. a=rtpmap:8 PCMA/8000
  289. a=rtpmap:101 telephone-event/8000
  290. a=fmtp:101 0-16
  291. a=silenceSupp:off - - - -
  292. a=ptime:20
  293. a=sendrecv
  294.  
  295. ---
  296.  
  297. <--- SIP read from UDP:127.0.0.1:5063 --->
  298. SIP/2.0 200 OK
  299. Via: SIP/2.0/UDP 127.0.0.1:5060;received=127.0.0.1;rport=5060;branch=z9hG4bK20ff81d2
  300. To: "emma" <sip:emma@127.0.0.1>;tag=fzhus
  301. From: <sip:1002@127.0.0.1>;tag=as6b10a823
  302. Call-ID: bhiszntdfujazks@dove
  303. CSeq: 102 INVITE
  304. Contact: <sip:emma@127.0.0.1:5063>
  305. Content-Type: application/sdp
  306. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  307. Server: Twinkle/1.4.2
  308. Supported: replaces,norefersub
  309. Content-Length: 200
  310.  
  311. v=0
  312. o=twinkle 242437781 1978171153 IN IP4 127.0.0.1
  313. s=-
  314. c=IN IP4 127.0.0.1
  315. t=0 0
  316. m=audio 8000 RTP/AVP 8 101
  317. a=rtpmap:8 PCMA/8000
  318. a=rtpmap:101 telephone-event/8000
  319. a=fmtp:101 0-15
  320. a=ptime:20
  321. <------------->
  322. --- (12 headers 10 lines) ---
  323. Found RTP audio format 8
  324. Found RTP audio format 101
  325. Found audio description format PCMA for ID 8
  326. Found audio description format telephone-event for ID 101
  327. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  328. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  329. Peer audio RTP is at port 127.0.0.1:8000
  330. set_destination: Parsing <sip:emma@127.0.0.1:5063> for address/port to send to
  331. set_destination: set destination to 127.0.0.1:5063
  332. Transmitting (no NAT) to 127.0.0.1:5063:
  333. ACK sip:emma@127.0.0.1:5063 SIP/2.0
  334. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK38355d13;rport
  335. Max-Forwards: 70
  336. From: <sip:1002@127.0.0.1>;tag=as6b10a823
  337. To: "emma" <sip:emma@127.0.0.1>;tag=fzhus
  338. Contact: <sip:1002@127.0.0.1:5060>
  339. Call-ID: bhiszntdfujazks@dove
  340. CSeq: 102 ACK
  341. User-Agent: Asterisk PBX 1.8.5.0
  342. Content-Length: 0
  343.  
  344.  
  345. ---
  346.  
  347. <--- SIP read from UDP:127.0.0.1:5063 --->
  348. BYE sip:1002@127.0.0.1:5060 SIP/2.0
  349. Via: SIP/2.0/UDP 127.0.0.1:5063;rport;branch=z9hG4bKqtlazboo
  350. Max-Forwards: 70
  351. To: <sip:1002@127.0.0.1>;tag=as6b10a823
  352. From: "emma" <sip:emma@127.0.0.1>;tag=fzhus
  353. Call-ID: bhiszntdfujazks@dove
  354. CSeq: 598 BYE
  355. User-Agent: Twinkle/1.4.2
  356. Content-Length: 0
  357.  
  358. <------------->
  359. --- (9 headers 0 lines) ---
  360. Sending to 127.0.0.1:5063 (no NAT)
  361. Scheduling destruction of SIP dialog 'bhiszntdfujazks@dove' in 32000 ms (Method: BYE)
  362.  
  363. <--- Transmitting (no NAT) to 127.0.0.1:5063 --->
  364. SIP/2.0 200 OK
  365. Via: SIP/2.0/UDP 127.0.0.1:5063;branch=z9hG4bKqtlazboo;received=127.0.0.1;rport=5063
  366. From: "emma" <sip:emma@127.0.0.1>;tag=fzhus
  367. To: <sip:1002@127.0.0.1>;tag=as6b10a823
  368. Call-ID: bhiszntdfujazks@dove
  369. CSeq: 598 BYE
  370. Server: Asterisk PBX 1.8.5.0
  371. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  372. Supported: replaces, timer
  373. Content-Length: 0
  374.  
  375.  
  376. <------------>
  377. set_destination: Parsing <sip:elartey@192.168.1.11> for address/port to send to
  378. set_destination: set destination to 192.168.1.11:5060
  379. Audio is at 5060
  380. Adding codec 0x8 (alaw) to SDP
  381. Adding non-codec 0x1 (telephone-event) to SDP
  382. Reliably Transmitting (no NAT) to 192.168.1.11:5060:
  383. INVITE sip:elartey@192.168.1.11 SIP/2.0
  384. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK5b590eea
  385. Max-Forwards: 70
  386. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  387. To: <sip:elartey@192.168.1.11>;tag=vjawn
  388. Contact: <sip:emma@192.168.1.155:5060>
  389. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  390. CSeq: 104 INVITE
  391. User-Agent: Asterisk PBX 1.8.5.0
  392. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  393. Supported: replaces, timer
  394. X-asterisk-Info: SIP re-invite (External RTP bridge)
  395. Content-Type: application/sdp
  396. Content-Length: 263
  397.  
  398. v=0
  399. o=root 542604981 542604983 IN IP4 192.168.1.155
  400. s=Asterisk PBX 1.8.5.0
  401. c=IN IP4 192.168.1.155
  402. t=0 0
  403. m=audio 16700 RTP/AVP 8 101
  404. a=rtpmap:8 PCMA/8000
  405. a=rtpmap:101 telephone-event/8000
  406. a=fmtp:101 0-16
  407. a=silenceSupp:off - - - -
  408. a=ptime:20
  409. a=sendrecv
  410.  
  411. ---
  412. [Jul 14 18:46:44] ERROR[3638]: cdr_csv.c:314 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied
  413. Scheduling destruction of SIP dialog '24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060' in 32000 ms (Method: INVITE)
  414.  
  415. <--- SIP read from UDP:192.168.1.11:5060 --->
  416. SIP/2.0 200 OK
  417. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK5b590eea
  418. To: <sip:elartey@192.168.1.11>;tag=vjawn
  419. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  420. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  421. CSeq: 104 INVITE
  422. Contact: <sip:elartey@192.168.1.11>
  423. Content-Type: application/sdp
  424. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  425. Server: Twinkle/1.4.2
  426. Supported: replaces,norefersub
  427. Content-Length: 206
  428.  
  429. v=0
  430. o=twinkle 1946095611 702177623 IN IP4 192.168.1.11
  431. s=-
  432. c=IN IP4 192.168.1.11
  433. t=0 0
  434. m=audio 8000 RTP/AVP 8 101
  435. a=rtpmap:8 PCMA/8000
  436. a=rtpmap:101 telephone-event/8000
  437. a=fmtp:101 0-15
  438. a=ptime:20
  439. <------------->
  440. --- (12 headers 10 lines) ---
  441. Found RTP audio format 8
  442. Found RTP audio format 101
  443. Found audio description format PCMA for ID 8
  444. Found audio description format telephone-event for ID 101
  445. Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
  446. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  447. Peer audio RTP is at port 192.168.1.11:8000
  448. set_destination: Parsing <sip:elartey@192.168.1.11> for address/port to send to
  449. set_destination: set destination to 192.168.1.11:5060
  450. Transmitting (no NAT) to 192.168.1.11:5060:
  451. ACK sip:elartey@192.168.1.11 SIP/2.0
  452. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK3ad47218
  453. Max-Forwards: 70
  454. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  455. To: <sip:elartey@192.168.1.11>;tag=vjawn
  456. Contact: <sip:emma@192.168.1.155:5060>
  457. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  458. CSeq: 104 ACK
  459. User-Agent: Asterisk PBX 1.8.5.0
  460. Content-Length: 0
  461.  
  462.  
  463. ---
  464. set_destination: Parsing <sip:elartey@192.168.1.11> for address/port to send to
  465. set_destination: set destination to 192.168.1.11:5060
  466. Reliably Transmitting (no NAT) to 192.168.1.11:5060:
  467. BYE sip:elartey@192.168.1.11 SIP/2.0
  468. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK00d57c67
  469. Max-Forwards: 70
  470. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  471. To: <sip:elartey@192.168.1.11>;tag=vjawn
  472. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  473. CSeq: 105 BYE
  474. User-Agent: Asterisk PBX 1.8.5.0
  475. X-Asterisk-HangupCause: Normal Clearing
  476. X-Asterisk-HangupCauseCode: 16
  477. Content-Length: 0
  478.  
  479.  
  480. ---
  481. Scheduling destruction of SIP dialog '24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060' in 32000 ms (Method: INVITE)
  482. Retransmitting #1 (no NAT) to 192.168.1.11:5060:
  483. BYE sip:elartey@192.168.1.11 SIP/2.0
  484. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK00d57c67
  485. Max-Forwards: 70
  486. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  487. To: <sip:elartey@192.168.1.11>;tag=vjawn
  488. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  489. CSeq: 105 BYE
  490. User-Agent: Asterisk PBX 1.8.5.0
  491. X-Asterisk-HangupCause: Normal Clearing
  492. X-Asterisk-HangupCauseCode: 16
  493. Content-Length: 0
  494.  
  495.  
  496. ---
  497.  
  498. <--- SIP read from UDP:192.168.1.11:5060 --->
  499. SIP/2.0 200 OK
  500. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK00d57c67
  501. To: <sip:elartey@192.168.1.11>;tag=vjawn
  502. From: "emma" <sip:emma@192.168.1.155>;tag=as4e26c625
  503. Call-ID: 24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060
  504. CSeq: 105 BYE
  505. Server: Twinkle/1.4.2
  506. Content-Length: 0
  507.  
  508. <------------->
  509. --- (8 headers 0 lines) ---
  510. Really destroying SIP dialog '24eec20a0f3c9edb7de8ad5469cc0bf9@192.168.1.155:5060' Method: INVITE
  511. Really destroying SIP dialog 'bhiszntdfujazks@dove' Method: BYE
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