API tools faq
paste
Advertisement
Guest User

Untitled

a guest
May 13th, 2019
89
0
Never
Add comment
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
text 13.78 KB | None | 0 0
raw download clone embed print report
  1. -- Executing [284@odoo:5] Dial("SIP/1064-0000001c", "SIP/284@25") in new stack
  2. == Using SIP RTP CoS mark 5
  3. We think we can do text
  4. Audio is at 15696
  5. Adding codec ulaw to SDP
  6. Adding codec alaw to SDP
  7. Adding codec gsm to SDP
  8. Adding codec amr to SDP
  9. Adding codec amrwb to SDP
  10. Adding codec g723 to SDP
  11. Adding codec g726 to SDP
  12. Adding codec g726aal2 to SDP
  13. Adding codec adpcm to SDP
  14. Adding codec slin to SDP
  15. Adding codec slin to SDP
  16. Adding codec slin to SDP
  17. Adding codec slin to SDP
  18. Adding codec slin to SDP
  19. Adding codec slin to SDP
  20. Adding codec slin to SDP
  21. Adding codec slin to SDP
  22. Adding codec slin to SDP
  23. Adding codec lpc10 to SDP
  24. Adding codec g729 to SDP
  25. Adding codec speex to SDP
  26. Adding codec speex to SDP
  27. Adding codec speex to SDP
  28. Adding codec ilbc to SDP
  29. Adding codec g722 to SDP
  30. Adding codec siren7 to SDP
  31. Adding codec siren14 to SDP
  32. Adding codec testlaw to SDP
  33. Adding codec g719 to SDP
  34. Adding codec opus to SDP
  35. Adding codec silk to SDP
  36. Adding codec silk to SDP
  37. Adding codec silk to SDP
  38. Adding codec silk to SDP
  39. Adding non-codec 0x1 (telephone-event) to SDP
  40. Reliably Transmitting (no NAT) to 192.168.18.18:5060:
  41. INVITE sip:284@192.168.18.18 SIP/2.0
  42. Via: SIP/2.0/UDP 192.168.130.30:5060;branch=z9hG4bK21389493
  43. Max-Forwards: 70
  44. From: "25" <sip:25@192.168.18.18>;tag=as2ca087db
  45. To: <sip:284@192.168.18.18>
  46. Contact: <sip:25@192.168.130.30:5060>
  47. Call-ID: 264b0d9d1e2021ab21a0ae4a7ed35b19@192.168.18.18
  48. CSeq: 102 INVITE
  49. User-Agent: Asterisk PBX 13.18.3~dfsg-1ubuntu4
  50. Date: Mon, 13 May 2019 16:19:29 GMT
  51. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  52. Supported: replaces, timer
  53. Content-Type: application/sdp
  54. Content-Length: 1480
  55.  
  56. v=0
  57. o=root 1348916619 1348916619 IN IP4 192.168.130.30
  58. s=Asterisk PBX 13.18.3~dfsg-1ubuntu4
  59. c=IN IP4 192.168.130.30
  60. t=0 0
  61. m=audio 15696 RTP/SAVP 0 8 3 108 109 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
  62. a=rtpmap:0 PCMU/8000
  63. a=rtpmap:8 PCMA/8000
  64. a=rtpmap:3 GSM/8000
  65. a=rtpmap:108 AMR/8000
  66. a=rtpmap:109 AMR-WB/16000
  67. a=rtpmap:4 G723/8000
  68. a=fmtp:4 annexa=no
  69. a=rtpmap:111 G726-32/8000
  70. a=rtpmap:112 AAL2-G726-32/8000
  71. a=rtpmap:5 DVI4/8000
  72. a=rtpmap:10 L16/8000
  73. a=rtpmap:118 L16/16000
  74. a=rtpmap:7 LPC/8000
  75. a=rtpmap:18 G729/8000
  76. a=fmtp:18 annexb=no
  77. a=rtpmap:110 speex/8000
  78. a=rtpmap:117 speex/16000
  79. a=rtpmap:119 speex/32000
  80. a=rtpmap:97 iLBC/8000
  81. a=rtpmap:9 G722/8000
  82. a=rtpmap:102 G7221/16000
  83. a=fmtp:102 bitrate=32000
  84. a=rtpmap:115 G7221/32000
  85. a=fmtp:115 bitrate=48000
  86. a=rtpmap:116 G719/48000
  87. a=fmtp:116 bitrate=64000
  88. a=rtpmap:107 opus/48000/2
  89. a=rtpmap:101 telephone-event/8000
  90. a=fmtp:101 0-16
  91. a=maxptime:20
  92. a=ice-ufrag:3e71cde669250ed06f4b3e9b39368fbf
  93. a=ice-pwd:6fe535bf27f256c4247985d815ed8a8f
  94. a=candidate:Hc0a8821e 1 UDP 2130706431 192.168.130.30 15696 typ host
  95. a=candidate:Sd46fef52 1 UDP 1694498815 212.111.239.82 15696 typ srflx raddr 192.168.130.30 rport 15696
  96. a=candidate:Hc0a8821e 2 UDP 2130706430 192.168.130.30 15697 typ host
  97. a=candidate:Sd46fef52 2 UDP 1694498814 212.111.239.82 15697 typ srflx raddr 192.168.130.30 rport 15697
  98. a=sendrecv
  99. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qfzOLVdVSf/Z+ROpcA/njkyORmgDAgAUNzuw51RG
  100.  
  101. ---
  102. -- Called SIP/284@25
  103.  
  104. <--- SIP read from UDP:192.168.18.18:5060 --->
  105. SIP/2.0 401 Unauthorized
  106. Via: SIP/2.0/UDP 192.168.130.30:5060;branch=z9hG4bK21389493
  107. From: "25" <sip:25@192.168.18.18>;tag=as2ca087db
  108. To: <sip:284@192.168.18.18>;tag=42155826
  109. Call-ID: 264b0d9d1e2021ab21a0ae4a7ed35b19@192.168.18.18
  110. CSeq: 102 INVITE
  111. Server: OpenScape Business M5T SIP Stack/4.2.20.35
  112. WWW-Authenticate: Digest realm="SMO-SIP",nonce="9667ad8c978bb620e3420926aa52f109e5a83530C132F2E8CD76",algorithm=MD5
  113. Content-Length: 0
  114.  
  115. <------------->
  116. --- (9 headers 0 lines) ---
  117. Transmitting (no NAT) to 192.168.18.18:5060:
  118. ACK sip:284@192.168.18.18 SIP/2.0
  119. Via: SIP/2.0/UDP 192.168.130.30:5060;branch=z9hG4bK21389493
  120. Max-Forwards: 70
  121. From: "25" <sip:25@192.168.18.18>;tag=as2ca087db
  122. To: <sip:284@192.168.18.18>;tag=42155826
  123. Contact: <sip:25@192.168.130.30:5060>
  124. Call-ID: 264b0d9d1e2021ab21a0ae4a7ed35b19@192.168.18.18
  125. CSeq: 102 ACK
  126. User-Agent: Asterisk PBX 13.18.3~dfsg-1ubuntu4
  127. Content-Length: 0
  128.  
  129.  
  130. ---
  131. We think we can do text
  132. Audio is at 15696
  133. Adding codec ulaw to SDP
  134. Adding codec alaw to SDP
  135. Adding codec gsm to SDP
  136. Adding codec amr to SDP
  137. Adding codec amrwb to SDP
  138. Adding codec g723 to SDP
  139. Adding codec g726 to SDP
  140. Adding codec g726aal2 to SDP
  141. Adding codec adpcm to SDP
  142. Adding codec slin to SDP
  143. Adding codec slin to SDP
  144. Adding codec slin to SDP
  145. Adding codec slin to SDP
  146. Adding codec slin to SDP
  147. Adding codec slin to SDP
  148. Adding codec slin to SDP
  149. Adding codec slin to SDP
  150. Adding codec slin to SDP
  151. Adding codec lpc10 to SDP
  152. Adding codec g729 to SDP
  153. Adding codec speex to SDP
  154. Adding codec speex to SDP
  155. Adding codec speex to SDP
  156. Adding codec ilbc to SDP
  157. Adding codec g722 to SDP
  158. Adding codec siren7 to SDP
  159. Adding codec siren14 to SDP
  160. Adding codec testlaw to SDP
  161. Adding codec g719 to SDP
  162. Adding codec opus to SDP
  163. Adding codec silk to SDP
  164. Adding codec silk to SDP
  165. Adding codec silk to SDP
  166. Adding codec silk to SDP
  167. Adding non-codec 0x1 (telephone-event) to SDP
  168. Reliably Transmitting (no NAT) to 192.168.18.18:5060:
  169. INVITE sip:284@192.168.18.18 SIP/2.0
  170. Via: SIP/2.0/UDP 192.168.130.30:5060;branch=z9hG4bK0e6fd475
  171. Max-Forwards: 70
  172. From: "25" <sip:25@192.168.18.18>;tag=as2ca087db
  173. To: <sip:284@192.168.18.18>
  174. Contact: <sip:25@192.168.130.30:5060>
  175. Call-ID: 264b0d9d1e2021ab21a0ae4a7ed35b19@192.168.18.18
  176. CSeq: 103 INVITE
  177. User-Agent: Asterisk PBX 13.18.3~dfsg-1ubuntu4
  178. Authorization: Digest username="25", realm="SMO-SIP", algorithm=MD5, uri="sip:284@192.168.18.18", nonce="9667ad8c978bb620e3420926aa52f109e5a83530C132F2E8CD76", response="1a2eecca8c9d615fffffba8e7c80737b"
  179. Date: Mon, 13 May 2019 16:19:29 GMT
  180. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  181. Supported: replaces, timer
  182. Content-Type: application/sdp
  183. Content-Length: 1480
  184.  
  185. v=0
  186. o=root 1348916619 1348916620 IN IP4 192.168.130.30
  187. s=Asterisk PBX 13.18.3~dfsg-1ubuntu4
  188. c=IN IP4 192.168.130.30
  189. t=0 0
  190. m=audio 15696 RTP/SAVP 0 8 3 108 109 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
  191. a=rtpmap:0 PCMU/8000
  192. a=rtpmap:8 PCMA/8000
  193. a=rtpmap:3 GSM/8000
  194. a=rtpmap:108 AMR/8000
  195. a=rtpmap:109 AMR-WB/16000
  196. a=rtpmap:4 G723/8000
  197. a=fmtp:4 annexa=no
  198. a=rtpmap:111 G726-32/8000
  199. a=rtpmap:112 AAL2-G726-32/8000
  200. a=rtpmap:5 DVI4/8000
  201. a=rtpmap:10 L16/8000
  202. a=rtpmap:118 L16/16000
  203. a=rtpmap:7 LPC/8000
  204. a=rtpmap:18 G729/8000
  205. a=fmtp:18 annexb=no
  206. a=rtpmap:110 speex/8000
  207. a=rtpmap:117 speex/16000
  208. a=rtpmap:119 speex/32000
  209. a=rtpmap:97 iLBC/8000
  210. a=rtpmap:9 G722/8000
  211. a=rtpmap:102 G7221/16000
  212. a=fmtp:102 bitrate=32000
  213. a=rtpmap:115 G7221/32000
  214. a=fmtp:115 bitrate=48000
  215. a=rtpmap:116 G719/48000
  216. a=fmtp:116 bitrate=64000
  217. a=rtpmap:107 opus/48000/2
  218. a=rtpmap:101 telephone-event/8000
  219. a=fmtp:101 0-16
  220. a=maxptime:20
  221. a=ice-ufrag:3e71cde669250ed06f4b3e9b39368fbf
  222. a=ice-pwd:6fe535bf27f256c4247985d815ed8a8f
  223. a=candidate:Hc0a8821e 1 UDP 2130706431 192.168.130.30 15696 typ host
  224. a=candidate:Sd46fef52 1 UDP 1694498815 212.111.239.82 15696 typ srflx raddr 192.168.130.30 rport 15696
  225. a=candidate:Hc0a8821e 2 UDP 2130706430 192.168.130.30 15697 typ host
  226. a=candidate:Sd46fef52 2 UDP 1694498814 212.111.239.82 15697 typ srflx raddr 192.168.130.30 rport 15697
  227. a=sendrecv
  228. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qfzOLVdVSf/Z+ROpcA/njkyORmgDAgAUNzuw51RG
  229.  
  230. ---
  231. Retransmitting #1 (no NAT) to 192.168.18.18:5060:
  232. INVITE sip:284@192.168.18.18 SIP/2.0
  233. Via: SIP/2.0/UDP 192.168.130.30:5060;branch=z9hG4bK0e6fd475
  234. Max-Forwards: 70
  235. From: "25" <sip:25@192.168.18.18>;tag=as2ca087db
  236. To: <sip:284@192.168.18.18>
  237. Contact: <sip:25@192.168.130.30:5060>
  238. Call-ID: 264b0d9d1e2021ab21a0ae4a7ed35b19@192.168.18.18
  239. CSeq: 103 INVITE
  240. User-Agent: Asterisk PBX 13.18.3~dfsg-1ubuntu4
  241. Authorization: Digest username="25", realm="SMO-SIP", algorithm=MD5, uri="sip:284@192.168.18.18", nonce="9667ad8c978bb620e3420926aa52f109e5a83530C132F2E8CD76", response="1a2eecca8c9d615fffffba8e7c80737b"
  242. Date: Mon, 13 May 2019 16:19:29 GMT
  243. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  244. Supported: replaces, timer
  245. Content-Type: application/sdp
  246. Content-Length: 1480
  247.  
  248. v=0
  249. o=root 1348916619 1348916620 IN IP4 192.168.130.30
  250. s=Asterisk PBX 13.18.3~dfsg-1ubuntu4
  251. c=IN IP4 192.168.130.30
  252. t=0 0
  253. m=audio 15696 RTP/SAVP 0 8 3 108 109 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
  254. a=rtpmap:0 PCMU/8000
  255. a=rtpmap:8 PCMA/8000
  256. a=rtpmap:3 GSM/8000
  257. a=rtpmap:108 AMR/8000
  258. a=rtpmap:109 AMR-WB/16000
  259. a=rtpmap:4 G723/8000
  260. a=fmtp:4 annexa=no
  261. a=rtpmap:111 G726-32/8000
  262. a=rtpmap:112 AAL2-G726-32/8000
  263. a=rtpmap:5 DVI4/8000
  264. a=rtpmap:10 L16/8000
  265. a=rtpmap:118 L16/16000
  266. a=rtpmap:7 LPC/8000
  267. a=rtpmap:18 G729/8000
  268. a=fmtp:18 annexb=no
  269. a=rtpmap:110 speex/8000
  270. a=rtpmap:117 speex/16000
  271. a=rtpmap:119 speex/32000
  272. a=rtpmap:97 iLBC/8000
  273. a=rtpmap:9 G722/8000
  274. a=rtpmap:102 G7221/16000
  275. a=fmtp:102 bitrate=32000
  276. a=rtpmap:115 G7221/32000
  277. a=fmtp:115 bitrate=48000
  278. a=rtpmap:116 G719/48000
  279. a=fmtp:116 bitrate=64000
  280. a=rtpmap:107 opus/48000/2
  281. a=rtpmap:101 telephone-event/8000
  282. a=fmtp:101 0-16
  283. a=maxptime:20
  284. a=ice-ufrag:3e71cde669250ed06f4b3e9b39368fbf
  285. a=ice-pwd:6fe535bf27f256c4247985d815ed8a8f
  286. a=candidate:Hc0a8821e 1 UDP 2130706431 192.168.130.30 15696 typ host
  287. a=candidate:Sd46fef52 1 UDP 1694498815 212.111.239.82 15696 typ srflx raddr 192.168.130.30 rport 15696
  288. a=candidate:Hc0a8821e 2 UDP 2130706430 192.168.130.30 15697 typ host
  289. a=candidate:Sd46fef52 2 UDP 1694498814 212.111.239.82 15697 typ srflx raddr 192.168.130.30 rport 15697
  290. a=sendrecv
  291. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:qfzOLVdVSf/Z+ROpcA/njkyORmgDAgAUNzuw51RG
  292.  
  293. ---
  294.  
  295. <--- SIP read from UDP:192.168.18.18:5060 --->
  296. SIP/2.0 180 Ringing
  297. Accept: application/sdp
  298. Via: SIP/2.0/UDP 192.168.130.30:5060;branch=z9hG4bK0e6fd475
  299. From: "25" <sip:25@192.168.18.18>;tag=as2ca087db
  300. To: <sip:284@192.168.18.18>;tag=42855568
  301. Call-ID: 264b0d9d1e2021ab21a0ae4a7ed35b19@192.168.18.18
  302. CSeq: 103 INVITE
  303. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, PRACK, UPDATE
  304. Contact: <sip:284@192.168.18.18:5060;transport=udp>
  305. P-Asserted-Identity: "Mobile OG5" <sip:284@192.168.18.18>
  306. Server: OpenScape Business M5T SIP Stack/4.2.20.35
  307. Content-Length: 0
  308.  
  309. <------------->
  310. --- (12 headers 0 lines) ---
  311. sip_route_dump: route/path hop: <sip:284@192.168.18.18:5060;transport=udp>
  312. -- SIP/25-0000001d is ringing
  313.  
  314. <--- Transmitting (no NAT) to 192.168.130.30:5065 --->
  315. SIP/2.0 180 Ringing
  316. Via: SIP/2.0/UDP 192.168.130.30:5065;branch=z9hG4bKvpcqgeyj;received=192.168.130.30;rport=5065
  317. From: <sip:1064@192.168.130.30>;tag=bxlrf
  318. To: <sip:284@192.168.130.30>;tag=as07e52e46
  319. Call-ID: palqvznrnsnvqbg@manaws0233
  320. CSeq: 430 INVITE
  321. Server: Asterisk PBX 13.18.3~dfsg-1ubuntu4
  322. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  323. Supported: replaces, timer
  324. Contact: <sip:284@192.168.130.30:5060>
  325. Content-Length: 0
  326.  
  327.  
  328. <------------>
  329.  
  330. <--- SIP read from UDP:192.168.18.18:5060 --->
  331. SIP/2.0 200 OK
  332. Accept: application/sdp
  333. Via: SIP/2.0/UDP 192.168.130.30:5060;branch=z9hG4bK0e6fd475
  334. From: "25" <sip:25@192.168.18.18>;tag=as2ca087db
  335. To: <sip:284@192.168.18.18>;tag=42855568
  336. Call-ID: 264b0d9d1e2021ab21a0ae4a7ed35b19@192.168.18.18
  337. CSeq: 103 INVITE
  338. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, PRACK, UPDATE
  339. Contact: <sip:284@192.168.18.18:5060;transport=udp>
  340. P-Asserted-Identity: "Mobile OG5" <sip:284@192.168.18.18>
  341. Server: OpenScape Business M5T SIP Stack/4.2.20.35
  342. Supported: replaces
  343. X-Siemens-Call-Type: ST-insecure
  344. Content-Type: application/sdp
  345. Content-Length: 125
  346.  
  347. v=0
  348. c=IN IP4 0.0.0.0
  349. t=0 0
  350. m=audio 0 RTP/SAVP 0 8 3 108 109 4 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
  351. <------------->
  352. --- (15 headers 4 lines) ---
  353. [May 13 18:19:36] WARNING[8786][C-0000000e]: chan_sip.c:10828 process_sdp: Failing due to no acceptable offer found
  354. sip_route_dump: route/path hop: <sip:284@192.168.18.18:5060;transport=udp>
  355. set_destination: Parsing <sip:284@192.168.18.18:5060;transport=udp> for address/port to send to
  356. set_destination: set destination to 192.168.18.18:5060
  357. Transmitting (no NAT) to 192.168.18.18:5060:
  358. ACK sip:284@192.168.18.18:5060;transport=udp SIP/2.0
  359. Via: SIP/2.0/UDP 192.168.130.30:5060;branch=z9hG4bK4dff6493
  360. Max-Forwards: 70
  361. From: "25" <sip:25@192.168.18.18>;tag=as2ca087db
  362. To: <sip:284@192.168.18.18>;tag=42855568
  363. Contact: <sip:25@192.168.130.30:5060>
  364. Call-ID: 264b0d9d1e2021ab21a0ae4a7ed35b19@192.168.18.18
  365. CSeq: 103 ACK
  366. User-Agent: Asterisk PBX 13.18.3~dfsg-1ubuntu4
  367. Content-Length: 0
  368.  
  369.  
  370. ---
  371. set_destination: Parsing <sip:284@192.168.18.18:5060;transport=udp> for address/port to send to
  372. set_destination: set destination to 192.168.18.18:5060
  373. Reliably Transmitting (no NAT) to 192.168.18.18:5060:
  374. BYE sip:284@192.168.18.18:5060;transport=udp SIP/2.0
  375. Via: SIP/2.0/UDP 192.168.130.30:5060;branch=z9hG4bK3a68e9f7
  376. Max-Forwards: 70
  377. From: "25" <sip:25@192.168.18.18>;tag=as2ca087db
  378. To: <sip:284@192.168.18.18>;tag=42855568
  379. Call-ID: 264b0d9d1e2021ab21a0ae4a7ed35b19@192.168.18.18
  380. CSeq: 104 BYE
  381. User-Agent: Asterisk PBX 13.18.3~dfsg-1ubuntu4
  382. Authorization: Digest username="25", realm="SMO-SIP", algorithm=MD5, uri="sip:284@192.168.18.18:5060", nonce="9667ad8c978bb620e3420926aa52f109e5a83530C132F2E8CD76", response="24075c32cd03eaa30811318d83332cb0"
  383. X-Asterisk-HangupCause: Bearer capability not available
  384. X-Asterisk-HangupCauseCode: 58
  385. Content-Length: 0
  386.  
  387.  
  388. ---
  389. Scheduling destruction of SIP dialog '264b0d9d1e2021ab21a0ae4a7ed35b19@192.168.18.18' in 6400 ms (Method: INVITE)
  390. Scheduling destruction of SIP dialog '264b0d9d1e2021ab21a0ae4a7ed35b19@192.168.18.18' in 6400 ms (Method: INVITE)
  391. == Everyone is busy/congested at this time (1:0/0/1)
  392. -- Executing [284@odoo:6] Hangup("SIP/1064-0000001c", "") in new stack
  393. == Spawn extension (odoo, 284, 6) exited non-zero on 'SIP/1064-0000001c'
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement
Public Pastes
Advertisement
We use cookies for various purposes including analytics. By continuing to use Pastebin, you agree to our use of cookies as described in the Cookies Policy.  OK, I Understand
Not a member of Pastebin yet?
Sign Up, it unlocks many cool features!