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  1. <--- SIP read from UDP:192.168.1.11:5060 --->
  2. INVITE sip:1000@192.168.1.155 SIP/2.0
  3. Via: SIP/2.0/UDP 192.168.1.11;rport;branch=z9hG4bKawtswoqo
  4. Max-Forwards: 70
  5. To: <sip:1000@192.168.1.155>
  6. From: <sip:elartey@192.168.1.155>;tag=xaepv
  7. Call-ID: vqcxgioasemuiqj@elartey-HP-Pavilion-dv7-Notebook-PC.rancardsolutions.com
  8. CSeq: 484 INVITE
  9. Contact: <sip:elartey@192.168.1.11>
  10. Content-Type: application/sdp
  11. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  12. Supported: replaces,norefersub,100rel
  13. User-Agent: Twinkle/1.4.2
  14. Content-Length: 457
  15.  
  16. v=0
  17. o=twinkle 632228344 9187692 IN IP4 192.168.1.11
  18. s=-
  19. c=IN IP4 192.168.1.11
  20. t=0 0
  21. m=audio 8000 RTP/AVP 3 8 0 97 98 99 102 103 104 105 101
  22. a=rtpmap:3 GSM/8000
  23. a=rtpmap:8 PCMA/8000
  24. a=rtpmap:0 PCMU/8000
  25. a=rtpmap:97 speex/8000
  26. a=rtpmap:98 speex/16000
  27. a=rtpmap:99 speex/32000
  28. a=rtpmap:102 G726-16/8000
  29. a=rtpmap:103 G726-24/8000
  30. a=rtpmap:104 G726-32/8000
  31. a=rtpmap:105 G726-40/8000
  32. a=rtpmap:101 telephone-event/8000
  33. a=fmtp:101 0-15
  34. a=ptime:20
  35. <------------->
  36. --- (13 headers 19 lines) ---
  37. Sending to 192.168.1.11:5060 (no NAT)
  38. Using INVITE request as basis request - vqcxgioasemuiqj@elartey-HP-Pavilion-dv7-Notebook-PC.rancardsolutions.com
  39. Found peer 'elartey' for 'elartey' from 192.168.1.11:5060
  40. Found RTP audio format 3
  41. Found RTP audio format 8
  42. Found RTP audio format 0
  43. Found RTP audio format 97
  44. Found RTP audio format 98
  45. Found RTP audio format 99
  46. Found RTP audio format 102
  47. Found RTP audio format 103
  48. Found RTP audio format 104
  49. Found RTP audio format 105
  50. Found RTP audio format 101
  51. Found audio description format GSM for ID 3
  52. Found audio description format PCMA for ID 8
  53. Found audio description format PCMU for ID 0
  54. Found audio description format speex for ID 97
  55. Found audio description format speex for ID 98
  56. Found unknown media description format speex for ID 99
  57. Found unknown media description format G726-16 for ID 102
  58. Found unknown media description format G726-24 for ID 103
  59. Found audio description format G726-32 for ID 104
  60. Found unknown media description format G726-40 for ID 105
  61. Found audio description format telephone-event for ID 101
  62. Capabilities: us - 0x2 (gsm), peer - audio=0x200000a0e (gsm|ulaw|alaw|g726|speex|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
  63. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  64. Peer audio RTP is at port 192.168.1.11:8000
  65. Looking for 1000 in phones (domain 192.168.1.155)
  66. list_route: hop: <sip:elartey@192.168.1.11>
  67.  
  68. <--- Transmitting (no NAT) to 192.168.1.11:5060 --->
  69. SIP/2.0 100 Trying
  70. Via: SIP/2.0/UDP 192.168.1.11;branch=z9hG4bKawtswoqo;received=192.168.1.11;rport=5060
  71. From: <sip:elartey@192.168.1.155>;tag=xaepv
  72. To: <sip:1000@192.168.1.155>
  73. Call-ID: vqcxgioasemuiqj@elartey-HP-Pavilion-dv7-Notebook-PC.rancardsolutions.com
  74. CSeq: 484 INVITE
  75. Server: Asterisk PBX 1.8.5.0
  76. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  77. Supported: replaces, timer
  78. Contact: <sip:1000@192.168.1.155:5060>
  79. Content-Length: 0
  80.  
  81.  
  82. <------------>
  83. Audio is at 5060
  84. Adding codec 0x2 (gsm) to SDP
  85. Adding non-codec 0x1 (telephone-event) to SDP
  86. Reliably Transmitting (no NAT) to 192.168.1.124:5060:
  87. INVITE sip:emma@192.168.1.124 SIP/2.0
  88. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK68880890
  89. Max-Forwards: 70
  90. From: "elartey" <sip:elartey@192.168.1.155>;tag=as226fb3e2
  91. To: <sip:emma@192.168.1.124>
  92. Contact: <sip:elartey@192.168.1.155:5060>
  93. Call-ID: 211786636eb10746681c1faf0496feac@192.168.1.155:5060
  94. CSeq: 102 INVITE
  95. User-Agent: Asterisk PBX 1.8.5.0
  96. Date: Fri, 15 Jul 2011 18:13:49 GMT
  97. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  98. Supported: replaces, timer
  99. Content-Type: application/sdp
  100. Content-Length: 264
  101.  
  102. v=0
  103. o=root 1574970781 1574970781 IN IP4 192.168.1.155
  104. s=Asterisk PBX 1.8.5.0
  105. c=IN IP4 192.168.1.155
  106. t=0 0
  107. m=audio 21906 RTP/AVP 3 101
  108. a=rtpmap:3 GSM/8000
  109. a=rtpmap:101 telephone-event/8000
  110. a=fmtp:101 0-16
  111. a=silenceSupp:off - - - -
  112. a=ptime:20
  113. a=sendrecv
  114.  
  115. ---
  116.  
  117. <--- SIP read from UDP:192.168.1.124:5060 --->
  118. SIP/2.0 100 Trying
  119. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK68880890
  120. To: <sip:emma@192.168.1.124>
  121. From: "elartey" <sip:elartey@192.168.1.155>;tag=as226fb3e2
  122. Call-ID: 211786636eb10746681c1faf0496feac@192.168.1.155:5060
  123. CSeq: 102 INVITE
  124. Server: Twinkle/1.4.2
  125. Content-Length: 0
  126.  
  127. <------------->
  128. --- (8 headers 0 lines) ---
  129.  
  130. <--- SIP read from UDP:192.168.1.124:5060 --->
  131. SIP/2.0 180 Ringing
  132. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK68880890
  133. To: <sip:emma@192.168.1.124>;tag=jcych
  134. From: "elartey" <sip:elartey@192.168.1.155>;tag=as226fb3e2
  135. Call-ID: 211786636eb10746681c1faf0496feac@192.168.1.155:5060
  136. CSeq: 102 INVITE
  137. Contact: <sip:emma@192.168.1.124>
  138. Server: Twinkle/1.4.2
  139. Content-Length: 0
  140.  
  141. <------------->
  142. --- (9 headers 0 lines) ---
  143.  
  144. <--- Transmitting (no NAT) to 192.168.1.11:5060 --->
  145. SIP/2.0 180 Ringing
  146. Via: SIP/2.0/UDP 192.168.1.11;branch=z9hG4bKawtswoqo;received=192.168.1.11;rport=5060
  147. From: <sip:elartey@192.168.1.155>;tag=xaepv
  148. To: <sip:1000@192.168.1.155>;tag=as156b30d4
  149. Call-ID: vqcxgioasemuiqj@elartey-HP-Pavilion-dv7-Notebook-PC.rancardsolutions.com
  150. CSeq: 484 INVITE
  151. Server: Asterisk PBX 1.8.5.0
  152. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  153. Supported: replaces, timer
  154. Contact: <sip:1000@192.168.1.155:5060>
  155. Content-Length: 0
  156.  
  157.  
  158. <------------>
  159.  
  160. <--- SIP read from UDP:192.168.1.124:5060 --->
  161. SIP/2.0 200 OK
  162. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK68880890
  163. To: <sip:emma@192.168.1.124>;tag=jcych
  164. From: "elartey" <sip:elartey@192.168.1.155>;tag=as226fb3e2
  165. Call-ID: 211786636eb10746681c1faf0496feac@192.168.1.155:5060
  166. CSeq: 102 INVITE
  167. Contact: <sip:emma@192.168.1.124>
  168. Content-Type: application/sdp
  169. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  170. Server: Twinkle/1.4.2
  171. Supported: replaces,norefersub
  172. Content-Length: 196
  173.  
  174. v=0
  175. o=twinkle 1425742287 1864420431 IN IP4 192.168.1.124
  176. s=-
  177. c=IN IP4 192.168.1.124
  178. t=0 0
  179. m=audio 8000 RTP/AVP 3 101
  180. a=rtpmap:3 GSM/8000
  181. a=rtpmap:101 telephone-event/8000
  182. a=fmtp:101 0-15
  183. <------------->
  184. --- (12 headers 9 lines) ---
  185. Found RTP audio format 3
  186. Found RTP audio format 101
  187. Found audio description format GSM for ID 3
  188. Found audio description format telephone-event for ID 101
  189. Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
  190. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  191. Peer audio RTP is at port 192.168.1.124:8000
  192. list_route: hop: <sip:emma@192.168.1.124>
  193. set_destination: Parsing <sip:emma@192.168.1.124> for address/port to send to
  194. set_destination: set destination to 192.168.1.124:5060
  195. Transmitting (no NAT) to 192.168.1.124:5060:
  196. ACK sip:emma@192.168.1.124 SIP/2.0
  197. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK615e9f7d
  198. Max-Forwards: 70
  199. rom: "elartey" <sip:elartey@192.168.1.155>;tag=as226fb3e2
  200. To: <sip:emma@192.168.1.124>;tag=jcych
  201. Contact: <sip:elartey@192.168.1.155:5060>
  202. Call-ID: 211786636eb10746681c1faf0496feac@192.168.1.155:5060
  203. CSeq: 102 ACK
  204. User-Agent: Asterisk PBX 1.8.5.0
  205. Content-Length: 0
  206.  
  207.  
  208. ---
  209. Audio is at 5060
  210. Adding codec 0x2 (gsm) to SDP
  211. Adding non-codec 0x1 (telephone-event) to SDP
  212.  
  213. <--- Reliably Transmitting (no NAT) to 192.168.1.11:5060 --->
  214. SIP/2.0 200 OK
  215. Via: SIP/2.0/UDP 192.168.1.11;branch=z9hG4bKawtswoqo;received=192.168.1.11;rport=5060
  216. From: <sip:elartey@192.168.1.155>;tag=xaepv
  217. To: <sip:1000@192.168.1.155>;tag=as156b30d4
  218. Call-ID: vqcxgioasemuiqj@elartey-HP-Pavilion-dv7-Notebook-PC.rancardsolutions.com
  219. CSeq: 484 INVITE
  220. Server: Asterisk PBX 1.8.5.0
  221. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  222. Supported: replaces, timer
  223. Contact: <sip:1000@192.168.1.155:5060>
  224. Content-Type: application/sdp
  225. Content-Length: 262
  226.  
  227. v=0
  228. o=root 699177414 699177414 IN IP4 192.168.1.155
  229. s=Asterisk PBX 1.8.5.0
  230. c=IN IP4 192.168.1.155
  231. t=0 0
  232. m=audio 20106 RTP/AVP 3 101
  233. a=rtpmap:3 GSM/8000
  234. a=rtpmap:101 telephone-event/8000
  235. a=fmtp:101 0-16
  236. a=silenceSupp:off - - - -
  237. a=ptime:20
  238. a=sendrecv
  239.  
  240. <------------>
  241. -- Locally bridging SIP/elartey-0000002e and SIP/emma-0000002f
  242.  
  243. <--- SIP read from UDP:192.168.1.11:5060 --->
  244. ACK sip:1000@192.168.1.155:5060 SIP/2.0
  245. Via: SIP/2.0/UDP 192.168.1.11;rport;branch=z9hG4bKjnkzpjhx
  246. Max-Forwards: 70
  247. To: <sip:1000@192.168.1.155>;tag=as156b30d4
  248. From: <sip:elartey@192.168.1.155>;tag=xaepv
  249. Call-ID: vqcxgioasemuiqj@elartey-HP-Pavilion-dv7-Notebook-PC.rancardsolutions.com
  250. CSeq: 484 ACK
  251. User-Agent: Twinkle/1.4.2
  252. Content-Length: 0
  253.  
  254. <------------->
  255. --- (9 headers 0 lines) ---
  256.  
  257. <--- SIP read from UDP:192.168.1.11:5060 --->
  258. BYE sip:1000@192.168.1.155:5060 SIP/2.0
  259. Via: SIP/2.0/UDP 192.168.1.11;rport;branch=z9hG4bKtrcugfwy
  260. Max-Forwards: 70
  261. To: <sip:1000@192.168.1.155>;tag=as156b30d4
  262. From: <sip:elartey@192.168.1.155>;tag=xaepv
  263. Call-ID: vqcxgioasemuiqj@elartey-HP-Pavilion-dv7-Notebook-PC.rancardsolutions.com
  264. CSeq: 485 BYE
  265. User-Agent: Twinkle/1.4.2
  266. Content-Length: 0
  267.  
  268. <------------->
  269. --- (9 headers 0 lines) ---
  270. Sending to 192.168.1.11:5060 (no NAT)
  271. Scheduling destruction of SIP dialog 'vqcxgioasemuiqj@elartey-HP-Pavilion-dv7-Notebook-PC.rancardsolutions.com' in 32000 ms (Method: BYE)
  272.  
  273. <--- Transmitting (no NAT) to 192.168.1.11:5060 --->
  274. SIP/2.0 200 OK
  275. Via: SIP/2.0/UDP 192.168.1.11;branch=z9hG4bKtrcugfwy;received=192.168.1.11;rport=5060
  276. From: <sip:elartey@192.168.1.155>;tag=xaepv
  277. To: <sip:1000@192.168.1.155>;tag=as156b30d4
  278. Call-ID: vqcxgioasemuiqj@elartey-HP-Pavilion-dv7-Notebook-PC.rancardsolutions.com
  279. CSeq: 485 BYE
  280. Server: Asterisk PBX 1.8.5.0
  281. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  282. Supported: replaces, timer
  283. Content-Length: 0
  284.  
  285.  
  286. <------------>
  287. [Jul 15 18:16:49] ERROR[10141]: cdr_csv.c:314 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied
  288. Scheduling destruction of SIP dialog '211786636eb10746681c1faf0496feac@192.168.1.155:5060' in 32000 ms (Method: INVITE)
  289. set_destination: Parsing <sip:emma@192.168.1.124> for address/port to send to
  290. set_destination: set destination to 192.168.1.124:5060
  291. Reliably Transmitting (no NAT) to 192.168.1.124:5060:
  292. BYE sip:emma@192.168.1.124 SIP/2.0
  293. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK46bee3ce
  294. Max-Forwards: 70
  295. From: "elartey" <sip:elartey@192.168.1.155>;tag=as226fb3e2
  296. To: <sip:emma@192.168.1.124>;tag=jcych
  297. Call-ID: 211786636eb10746681c1faf0496feac@192.168.1.155:5060
  298. CSeq: 103 BYE
  299. User-Agent: Asterisk PBX 1.8.5.0
  300. X-Asterisk-HangupCause: Normal Clearing
  301. X-Asterisk-HangupCauseCode: 16
  302. Content-Length: 0
  303.  
  304.  
  305. ---
  306.  
  307. <--- SIP read from UDP:192.168.1.124:5060 --->
  308. SIP/2.0 200 OK
  309. Via: SIP/2.0/UDP 192.168.1.155:5060;branch=z9hG4bK46bee3ce
  310. To: <sip:emma@192.168.1.124>;tag=jcych
  311. From: "elartey" <sip:elartey@192.168.1.155>;tag=as226fb3e2
  312. Call-ID: 211786636eb10746681c1faf0496feac@192.168.1.155:5060
  313. CSeq: 103 BYE
  314. Server: Twinkle/1.4.2
  315. Content-Length: 0
  316.  
  317. <------------->
  318. --- (8 headers 0 lines) ---
  319. Really destroying SIP dialog '211786636eb10746681c1faf0496feac@192.168.1.155:5060' Method: INVITE
  320. Really destroying SIP dialog 'vqcxgioasemuiqj@elartey-HP-Pavilion-dv7-Notebook-PC.rancardsolutions.com' Method: BYE
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