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  1. --- SIP read from UDP:127.0.0.1:5063 --->
  2. INVITE sip:1002@127.0.0.1 SIP/2.0
  3. Via: SIP/2.0/UDP 127.0.0.1:5063;rport;branch=z9hG4bKufpejwgi
  4. Max-Forwards: 70
  5. To: <sip:1002@127.0.0.1>
  6. From: "emma" <sip:emma@127.0.0.1>;tag=bndji
  7. Call-ID: bpsnpyhvmseazxb@dove.dyndns-at-home.com
  8. CSeq: 65 INVITE
  9. Contact: <sip:emma@127.0.0.1:5063>
  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: 454
  15.  
  16. v=0
  17. o=twinkle 1702280581 463102200 IN IP4 127.0.0.1
  18. s=-
  19. c=IN IP4 127.0.0.1
  20. t=0 0
  21. m=audio 8000 RTP/AVP 3 8 97 98 99 102 103 104 105 0 101
  22. a=rtpmap:3 GSM/8000
  23. a=rtpmap:8 PCMA/8000
  24. a=rtpmap:97 speex/8000
  25. a=rtpmap:98 speex/16000
  26. a=rtpmap:99 speex/32000
  27. a=rtpmap:102 G726-16/8000
  28. a=rtpmap:103 G726-24/8000
  29. a=rtpmap:104 G726-32/8000
  30. a=rtpmap:105 G726-40/8000
  31. a=rtpmap:0 PCMU/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 127.0.0.1:5063 (no NAT)
  38. Using INVITE request as basis request - bpsnpyhvmseazxb@dove.dyndns-at-home.com
  39. Found peer 'emma' for 'emma' from 127.0.0.1:5063
  40. Found RTP audio format 3
  41. Found RTP audio format 8
  42. Found RTP audio format 97
  43. Found RTP audio format 98
  44. Found RTP audio format 99
  45. Found RTP audio format 102
  46. Found RTP audio format 103
  47. Found RTP audio format 104
  48. Found RTP audio format 105
  49. Found RTP audio format 0
  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 speex for ID 97
  54. Found audio description format speex for ID 98
  55. Found unknown media description format speex for ID 99
  56. Found unknown media description format G726-16 for ID 102
  57. Found unknown media description format G726-24 for ID 103
  58. Found audio description format G726-32 for ID 104
  59. Found unknown media description format G726-40 for ID 105
  60. Found audio description format PCMU for ID 0
  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 127.0.0.1:8000
  65. Looking for 1002 in phones (domain 127.0.0.1)
  66. list_route: hop: <sip:emma@127.0.0.1:5063>
  67.  
  68. <--- Transmitting (no NAT) to 127.0.0.1:5063 --->
  69. SIP/2.0 100 Trying
  70. Via: SIP/2.0/UDP 127.0.0.1:5063;branch=z9hG4bKufpejwgi;received=127.0.0.1;rport=5063
  71. From: "emma" <sip:emma@127.0.0.1>;tag=bndji
  72. To: <sip:1002@127.0.0.1>
  73. Call-ID: bpsnpyhvmseazxb@dove.dyndns-at-home.com
  74. CSeq: 65 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:1002@127.0.0.1:5060>
  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 127.0.0.1:5062:
  87. INVITE sip:elartey@127.0.0.1:5062 SIP/2.0
  88. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4176a640
  89. Max-Forwards: 70
  90. From: "emma" <sip:emma@127.0.0.1>;tag=as70ba8b3f
  91. To: <sip:elartey@127.0.0.1:5062>
  92. Contact: <sip:emma@127.0.0.1:5060>
  93. Call-ID: 21257def5268f4e617b1c8b459943de8@127.0.0.1:5060
  94. CSeq: 102 INVITE
  95. User-Agent: Asterisk PBX 1.8.5.0
  96. Date: Mon, 18 Jul 2011 12:05:06 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: 256
  101.  
  102. v=0
  103. o=root 1536944079 1536944079 IN IP4 127.0.0.1
  104. s=Asterisk PBX 1.8.5.0
  105. c=IN IP4 127.0.0.1
  106. t=0 0
  107. m=audio 17976 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:127.0.0.1:5062 --->
  118. SIP/2.0 100 Trying
  119. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4176a640
  120. To: <sip:elartey@127.0.0.1:5062>
  121. From: "emma" <sip:emma@127.0.0.1>;tag=as70ba8b3f
  122. Call-ID: 21257def5268f4e617b1c8b459943de8@127.0.0.1:5060
  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:127.0.0.1:5062 --->
  131. SIP/2.0 180 Ringing
  132. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4176a640
  133. To: <sip:elartey@127.0.0.1:5062>;tag=aeqzw
  134. From: "emma" <sip:emma@127.0.0.1>;tag=as70ba8b3f
  135. Call-ID: 21257def5268f4e617b1c8b459943de8@127.0.0.1:5060
  136. CSeq: 102 INVITE
  137. Contact: <sip:elartey@127.0.0.1:5062>
  138. Server: Twinkle/1.4.2
  139. Content-Length: 0
  140.  
  141. <------------->
  142. --- (9 headers 0 lines) ---
  143.  
  144. <--- Transmitting (no NAT) to 127.0.0.1:5063 --->
  145. SIP/2.0 180 Ringing
  146. Via: SIP/2.0/UDP 127.0.0.1:5063;branch=z9hG4bKufpejwgi;received=127.0.0.1;rport=5063
  147. From: "emma" <sip:emma@127.0.0.1>;tag=bndji
  148. To: <sip:1002@127.0.0.1>;tag=as3db982f3
  149. Call-ID: bpsnpyhvmseazxb@dove.dyndns-at-home.com
  150. CSeq: 65 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:1002@127.0.0.1:5060>
  155. Content-Length: 0
  156.  
  157.  
  158. <------------>
  159. Really destroying SIP dialog 'gsnkinfgjpfwwgn@dove.dyndns-at-home.com' Method: REGISTER
  160.  
  161. <--- SIP read from UDP:127.0.0.1:5062 --->
  162. SIP/2.0 200 OK
  163. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK4176a640
  164. To: <sip:elartey@127.0.0.1:5062>;tag=aeqzw
  165. From: "emma" <sip:emma@127.0.0.1>;tag=as70ba8b3f
  166. Call-ID: 21257def5268f4e617b1c8b459943de8@127.0.0.1:5060
  167. CSeq: 102 INVITE
  168. Contact: <sip:elartey@127.0.0.1:5062>
  169. Content-Type: application/sdp
  170. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
  171. Server: Twinkle/1.4.2
  172. Supported: replaces,norefersub
  173. Content-Length: 188
  174.  
  175. v=0
  176. o=twinkle 2019235702 1963404372 IN IP4 127.0.0.1
  177. s=-
  178. c=IN IP4 127.0.0.1
  179. t=0 0
  180. m=audio 8000 RTP/AVP 3 101
  181. a=rtpmap:3 GSM/8000
  182. a=rtpmap:101 telephone-event/8000
  183. a=fmtp:101 0-15
  184. <------------->
  185. --- (12 headers 9 lines) ---
  186. Found RTP audio format 3
  187. Found RTP audio format 101
  188. Found audio description format GSM for ID 3
  189. Found audio description format telephone-event for ID 101
  190. Capabilities: us - 0x2 (gsm), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)
  191. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  192. Peer audio RTP is at port 127.0.0.1:8000
  193. list_route: hop: <sip:elartey@127.0.0.1:5062>
  194. set_destination: Parsing <sip:elartey@127.0.0.1:5062> for address/port to send to
  195. set_destination: set destination to 127.0.0.1:5062
  196. Transmitting (no NAT) to 127.0.0.1:5062:
  197. ACK sip:elartey@127.0.0.1:5062 SIP/2.0
  198. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK3d9502ee
  199. Max-Forwards: 70
  200. From: "emma" <sip:emma@127.0.0.1>;tag=as70ba8b3f
  201. To: <sip:elartey@127.0.0.1:5062>;tag=aeqzw
  202. Contact: <sip:emma@127.0.0.1:5060>
  203. Call-ID: 21257def5268f4e617b1c8b459943de8@127.0.0.1:5060
  204. CSeq: 102 ACK
  205. User-Agent: Asterisk PBX 1.8.5.0
  206. Content-Length: 0
  207.  
  208.  
  209. ---
  210. Audio is at 5060
  211. Adding codec 0x2 (gsm) to SDP
  212. Adding non-codec 0x1 (telephone-event) to SDP
  213.  
  214. <--- Reliably Transmitting (no NAT) to 127.0.0.1:5063 --->
  215. SIP/2.0 200 OK
  216. Via: SIP/2.0/UDP 127.0.0.1:5063;branch=z9hG4bKufpejwgi;received=127.0.0.1;rport=5063
  217. From: "emma" <sip:emma@127.0.0.1>;tag=bndji
  218. To: <sip:1002@127.0.0.1>;tag=as3db982f3
  219. Call-ID: bpsnpyhvmseazxb@dove.dyndns-at-home.com
  220. CSeq: 65 INVITE
  221. Server: Asterisk PBX 1.8.5.0
  222. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  223. Supported: replaces, timer
  224. Contact: <sip:1002@127.0.0.1:5060>
  225. Content-Type: application/sdp
  226. Content-Length: 256
  227.  
  228. v=0
  229. o=root 1926225515 1926225515 IN IP4 127.0.0.1
  230. s=Asterisk PBX 1.8.5.0
  231. c=IN IP4 127.0.0.1
  232. t=0 0
  233. m=audio 13566 RTP/AVP 3 101
  234. a=rtpmap:3 GSM/8000
  235. a=rtpmap:101 telephone-event/8000
  236. a=fmtp:101 0-16
  237. a=silenceSupp:off - - - -
  238. a=ptime:20
  239. a=sendrecv
  240.  
  241. <------------>
  242. -- Locally bridging SIP/emma-00000000 and SIP/elartey-00000001
  243.  
  244. <--- SIP read from UDP:127.0.0.1:5063 --->
  245. ACK sip:1002@127.0.0.1:5060 SIP/2.0
  246. Via: SIP/2.0/UDP 127.0.0.1:5063;rport;branch=z9hG4bKhjpcyhah
  247. Max-Forwards: 70
  248. To: <sip:1002@127.0.0.1>;tag=as3db982f3
  249. From: "emma" <sip:emma@127.0.0.1>;tag=bndji
  250. Call-ID: bpsnpyhvmseazxb@dove.dyndns-at-home.com
  251. CSeq: 65 ACK
  252. User-Agent: Twinkle/1.4.2
  253. Content-Length: 0
  254.  
  255. <------------->
  256. --- (9 headers 0 lines) ---
  257.  
  258. <--- SIP read from UDP:127.0.0.1:5062 --->
  259. BYE sip:emma@127.0.0.1:5060 SIP/2.0
  260. Via: SIP/2.0/UDP 127.0.0.1:5062;rport;branch=z9hG4bKzvdsyikx
  261. Max-Forwards: 70
  262. To: "emma" <sip:emma@127.0.0.1>;tag=as70ba8b3f
  263. From: <sip:elartey@127.0.0.1:5062>;tag=aeqzw
  264. Call-ID: 21257def5268f4e617b1c8b459943de8@127.0.0.1:5060
  265. CSeq: 255 BYE
  266. User-Agent: Twinkle/1.4.2
  267. Content-Length: 0
  268.  
  269. <------------->
  270. --- (9 headers 0 lines) ---
  271. Sending to 127.0.0.1:5062 (no NAT)
  272. Scheduling destruction of SIP dialog '21257def5268f4e617b1c8b459943de8@127.0.0.1:5060' in 32000 ms (Method: BYE)
  273.  
  274. <--- Transmitting (no NAT) to 127.0.0.1:5062 --->
  275. SIP/2.0 200 OK
  276. Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bKzvdsyikx;received=127.0.0.1;rport=5062
  277. From: <sip:elartey@127.0.0.1:5062>;tag=aeqzw
  278. To: "emma" <sip:emma@127.0.0.1>;tag=as70ba8b3f
  279. Call-ID: 21257def5268f4e617b1c8b459943de8@127.0.0.1:5060
  280. CSeq: 255 BYE
  281. Server: Asterisk PBX 1.8.5.0
  282. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  283. Supported: replaces, timer
  284. Content-Length: 0
  285.  
  286.  
  287. <------------>
  288. [Jul 18 12:05:36] ERROR[4429]: cdr_csv.c:314 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied
  289. Scheduling destruction of SIP dialog 'bpsnpyhvmseazxb@dove.dyndns-at-home.com' in 32000 ms (Method: ACK)
  290. set_destination: Parsing <sip:emma@127.0.0.1:5063> for address/port to send to
  291. set_destination: set destination to 127.0.0.1:5063
  292. Reliably Transmitting (no NAT) to 127.0.0.1:5063:
  293. BYE sip:emma@127.0.0.1:5063 SIP/2.0
  294. Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5da5a536;rport
  295. Max-Forwards: 70
  296. From: <sip:1002@127.0.0.1>;tag=as3db982f3
  297. To: "emma" <sip:emma@127.0.0.1>;tag=bndji
  298. Call-ID: bpsnpyhvmseazxb@dove.dyndns-at-home.com
  299. CSeq: 102 BYE
  300. User-Agent: Asterisk PBX 1.8.5.0
  301. X-Asterisk-HangupCause: Normal Clearing
  302. X-Asterisk-HangupCauseCode: 16
  303. Content-Length: 0
  304.  
  305.  
  306. ---
  307.  
  308. <--- SIP read from UDP:127.0.0.1:5063 --->
  309. SIP/2.0 200 OK
  310. Via: SIP/2.0/UDP 127.0.0.1:5060;received=127.0.0.1;rport=5060;branch=z9hG4bK5da5a536
  311. To: "emma" <sip:emma@127.0.0.1>;tag=bndji
  312. From: <sip:1002@127.0.0.1>;tag=as3db982f3
  313. Call-ID: bpsnpyhvmseazxb@dove.dyndns-at-home.com
  314. CSeq: 102 BYE
  315. Server: Twinkle/1.4.2
  316. Content-Length: 0
  317.  
  318. <------------->
  319. --- (8 headers 0 lines) ---
  320. SIP Response message for INCOMING dialog BYE arrived
  321. Really destroying SIP dialog 'bpsnpyhvmseazxb@dove.dyndns-at-home.com' Method: ACK
  322. Really destroying SIP dialog '21257def5268f4e617b1c8b459943de8@127.0.0.1:5060' Method: BYE
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