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  1.  
  2. Appel d'un sip interne vers un portable externe via trunk OVH,
  3.  
  4. sonnerie KO poste appellant, appelé sonnerie OK, conversation KO, pas de raccroché détecté.
  5.  
  6. Le poste SIP raccroche manuellement à la fin.
  7.  
  8. SIP Debugging enabled
  9.  
  10. <--- SIP read from UDP:192.168.1.150:5060 --->
  11. INVITE sip:[email protected] SIP/2.0
  12. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKa67cf3240a443c02
  13. From: "Florent TOTOR" <sip:[email protected]>;tag=ba6dba2dbe613f58
  14. To: <sip:[email protected]>
  15. Contact: <sip:[email protected]:5060;transport=udp>
  16. Supported: replaces, timer, path
  17. X-Grandstream-PBX: true
  18. P-Early-Media: Supported
  19. Call-ID: [email protected]
  20. CSeq: 33404 INVITE
  21. User-Agent: Grandstream GXP1200 1.2.5.3
  22. Max-Forwards: 70
  23. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  24. Content-Type: application/sdp
  25. Content-Length: 313
  26.  
  27. v=0
  28. o=101 8000 8000 IN IP4 192.168.1.150
  29. s=SIP Call
  30. c=IN IP4 192.168.1.150
  31. t=0 0
  32. m=audio 5014 RTP/AVP 9 18 8 0 2 101
  33. a=sendrecv
  34. a=rtpmap:9 G722/8000
  35. a=rtpmap:18 G729/8000
  36. a=rtpmap:8 PCMA/8000
  37. a=rtpmap:0 PCMU/8000
  38. a=rtpmap:2 G726-32/8000
  39. a=ptime:20
  40. a=rtpmap:101 telephone-event/8000
  41. a=fmtp:101 0-11
  42. <------------->
  43. --- (15 headers 15 lines) ---
  44. Sending to 192.168.1.150:5060 (no NAT)
  45. Using INVITE request as basis request - [email protected]
  46. Found peer '101' for '101' from 192.168.1.150:5060
  47.  
  48. <--- Reliably Transmitting (NAT) to 192.168.1.150:5060 --->
  49. SIP/2.0 401 Unauthorized
  50. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKa67cf3240a443c02;received=192.168.1.150;rport=5060
  51. From: "Florent TOTOR" <sip:[email protected]>;tag=ba6dba2dbe613f58
  52. To: <sip:[email protected]>;tag=as4c3d30a9
  53. Call-ID: [email protected]
  54. CSeq: 33404 INVITE
  55. Server: AskoziaPBX
  56. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  57. Supported: replaces, timer
  58. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75a6cfa2"
  59. Content-Length: 0
  60.  
  61.  
  62. <------------>
  63. Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
  64.  
  65. <--- SIP read from UDP:192.168.1.150:5060 --->
  66. ACK sip:[email protected] SIP/2.0
  67. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKa67cf3240a443c02
  68. From: "Florent TOTOR" <sip:[email protected]>;tag=ba6dba2dbe613f58
  69. To: <sip:[email protected]>;tag=as4c3d30a9
  70. Contact: <sip:[email protected]:5060;transport=udp>
  71. Supported: path
  72. X-Grandstream-PBX: true
  73. Call-ID: [email protected]
  74. CSeq: 33404 ACK
  75. User-Agent: Grandstream GXP1200 1.2.5.3
  76. Max-Forwards: 70
  77. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  78. Content-Length: 0
  79.  
  80. <------------->
  81. --- (13 headers 0 lines) ---
  82.  
  83. <--- SIP read from UDP:192.168.1.150:5060 --->
  84. INVITE sip:[email protected] SIP/2.0
  85. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194
  86. From: "Florent TOTOR" <sip:[email protected]>;tag=ba6dba2dbe613f58
  87. To: <sip:[email protected]>
  88. Contact: <sip:[email protected]:5060;transport=udp>
  89. Supported: replaces, timer, path
  90. X-Grandstream-PBX: true
  91. P-Early-Media: Supported
  92. Authorization: Digest username="101", realm="asterisk", algorithm=MD5, uri="sip:[email protected]", nonce="75a6cfa2", response="56ea22283ad59cd23789e02653c31800"
  93. Call-ID: [email protected]
  94. CSeq: 33405 INVITE
  95. User-Agent: Grandstream GXP1200 1.2.5.3
  96. Max-Forwards: 70
  97. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  98. Content-Type: application/sdp
  99. Content-Length: 313
  100.  
  101. v=0
  102. o=101 8000 8001 IN IP4 192.168.1.150
  103. s=SIP Call
  104. c=IN IP4 192.168.1.150
  105. t=0 0
  106. m=audio 5014 RTP/AVP 9 18 8 0 2 101
  107. a=sendrecv
  108. a=rtpmap:9 G722/8000
  109. a=rtpmap:18 G729/8000
  110. a=rtpmap:8 PCMA/8000
  111. a=rtpmap:0 PCMU/8000
  112. a=rtpmap:2 G726-32/8000
  113. a=ptime:20
  114. a=rtpmap:101 telephone-event/8000
  115. a=fmtp:101 0-11
  116. <------------->
  117. --- (16 headers 15 lines) ---
  118. Sending to 192.168.1.150:5060 (NAT)
  119. Using INVITE request as basis request - [email protected]
  120. Found peer '101' for '101' from 192.168.1.150:5060
  121. Found RTP audio format 9
  122. Found RTP audio format 18
  123. Found RTP audio format 8
  124. Found RTP audio format 0
  125. Found RTP audio format 2
  126. Found RTP audio format 101
  127. Found audio description format G722 for ID 9
  128. Found audio description format G729 for ID 18
  129. Found audio description format PCMA for ID 8
  130. Found audio description format PCMU for ID 0
  131. Found audio description format G726-32 for ID 2
  132. Found audio description format telephone-event for ID 101
  133. Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
  134. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  135. Peer audio RTP is at port 192.168.1.150:5014
  136. Looking for 06XXXXXXXX in SIP-PHONE-5171585574ee5e5a8ba8f6 (domain 192.168.1.200)
  137. list_route: hop: <sip:[email protected]:5060;transport=udp>
  138.  
  139. <--- Transmitting (NAT) to 192.168.1.150:5060 --->
  140. SIP/2.0 100 Trying
  141. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194;received=192.168.1.150;rport=5060
  142. From: "Florent TOTOR" <sip:[email protected]>;tag=ba6dba2dbe613f58
  143. To: <sip:[email protected]>
  144. Call-ID: [email protected]
  145. CSeq: 33405 INVITE
  146. Server: AskoziaPBX
  147. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  148. Supported: replaces, timer
  149. Contact: <sip:[email protected]:5060>
  150. Content-Length: 0
  151.  
  152.  
  153. <------------>
  154. Audio is at 5060
  155. Adding codec 0x4 (ulaw) to SDP
  156. Adding codec 0x8 (alaw) to SDP
  157. Adding codec 0x2 (gsm) to SDP
  158. Adding non-codec 0x1 (telephone-event) to SDP
  159. Reliably Transmitting (NAT) to 91.121.129.17:5060:
  160. INVITE sip:[email protected] SIP/2.0
  161. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK3e7cd865;rport
  162. Max-Forwards: 70
  163. From: "Default Extension" <sip:[email protected]>;tag=as4523e5ab
  164. To: <sip:[email protected]>
  165. Contact: <sip:[email protected]:5060>
  166. Call-ID: [email protected]
  167. CSeq: 102 INVITE
  168. User-Agent: AskoziaPBX
  169. Date: Mon, 12 Dec 2011 11:48:06 GMT
  170. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  171. Supported: replaces, timer
  172. Content-Type: application/sdp
  173. Content-Length: 285
  174.  
  175. v=0
  176. o=root 1595019147 1595019147 IN IP4 10.192.26.204
  177. s=Asterisk PBX 1.8.4.4
  178. c=IN IP4 10.192.26.204
  179. t=0 0
  180. m=audio 10072 RTP/AVP 0 8 3 101
  181. a=rtpmap:0 PCMU/8000
  182. a=rtpmap:8 PCMA/8000
  183. a=rtpmap:3 GSM/8000
  184. a=rtpmap:101 telephone-event/8000
  185. a=fmtp:101 0-16
  186. a=ptime:20
  187. a=sendrecv
  188.  
  189. ---
  190. Retransmitting #1 (NAT) to 91.121.129.17:5060:
  191. INVITE sip:[email protected] SIP/2.0
  192. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK3e7cd865;rport
  193. Max-Forwards: 70
  194. From: "Default Extension" <sip:[email protected]>;tag=as4523e5ab
  195. To: <sip:[email protected]>
  196. Contact: <sip:[email protected]:5060>
  197. Call-ID: [email protected]
  198. CSeq: 102 INVITE
  199. User-Agent: AskoziaPBX
  200. Date: Mon, 12 Dec 2011 11:48:06 GMT
  201. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  202. Supported: replaces, timer
  203. Content-Type: application/sdp
  204. Content-Length: 285
  205.  
  206. v=0
  207. o=root 1595019147 1595019147 IN IP4 10.192.26.204
  208. s=Asterisk PBX 1.8.4.4
  209. c=IN IP4 10.192.26.204
  210. t=0 0
  211. m=audio 10072 RTP/AVP 0 8 3 101
  212. a=rtpmap:0 PCMU/8000
  213. a=rtpmap:8 PCMA/8000
  214. a=rtpmap:3 GSM/8000
  215. a=rtpmap:101 telephone-event/8000
  216. a=fmtp:101 0-16
  217. a=ptime:20
  218. a=sendrecv
  219.  
  220. ---
  221.  
  222. <--- SIP read from UDP:91.121.129.17:5060 --->
  223. SIP/2.0 407 authentication required
  224. Allow: UPDATE,REFER,INFO
  225. Call-ID: [email protected]
  226. Contact: <sip:[email protected]:5060;user=phone>
  227. CSeq: 102 INVITE
  228. From: "Default Extension" <sip:[email protected]>;tag=as4523e5ab
  229. Proxy-Authenticate: Digest realm="sip.ovh.net",nonce="0973b5c766fddd071e1927cd6756ac7d",opaque="0972c2e8159490c",stale=false,algorithm=MD5
  230. Server: Cirpack/v4.42j (gw_sip)
  231. To: <sip:[email protected]>;tag=00-07862-0973bb13-60144c9b0
  232. Via: SIP/2.0/UDP 10.192.26.204:5060;received=95.210.161.54;rport=5060;branch=z9hG4bK3e7cd865
  233. Content-Length: 0
  234.  
  235. <------------->
  236. --- (11 headers 0 lines) ---
  237. Transmitting (NAT) to 91.121.129.17:5060:
  238. ACK sip:[email protected] SIP/2.0
  239. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK3e7cd865;rport
  240. Max-Forwards: 70
  241. From: "Default Extension" <sip:[email protected]>;tag=as4523e5ab
  242. To: <sip:[email protected]>;tag=00-07862-0973bb13-60144c9b0
  243. Contact: <sip:[email protected]:5060>
  244. Call-ID: [email protected]
  245. CSeq: 102 ACK
  246. User-Agent: AskoziaPBX
  247. Content-Length: 0
  248.  
  249.  
  250. ---
  251. Audio is at 5060
  252. Adding codec 0x4 (ulaw) to SDP
  253. Adding codec 0x8 (alaw) to SDP
  254. Adding codec 0x2 (gsm) to SDP
  255. Adding non-codec 0x1 (telephone-event) to SDP
  256. Reliably Transmitting (NAT) to 91.121.129.17:5060:
  257. INVITE sip:[email protected] SIP/2.0
  258. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK0b23a06a;rport
  259. Max-Forwards: 70
  260. From: "Default Extension" <sip:[email protected]>;tag=as4523e5ab
  261. To: <sip:[email protected]>
  262. Contact: <sip:[email protected]:5060>
  263. Call-ID: [email protected]
  264. CSeq: 103 INVITE
  265. User-Agent: AskoziaPBX
  266. Proxy-Authorization: Digest username="0033183643585", realm="sip.ovh.net", algorithm=MD5, uri="sip:[email protected]", nonce="0973b5c766fddd071e1927cd6756ac7d", response="8d1362978804f22773300724ae9822ef", opaque="0972c2e8159490c"
  267. Date: Mon, 12 Dec 2011 11:48:07 GMT
  268. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  269. Supported: replaces, timer
  270. Content-Type: application/sdp
  271. Content-Length: 285
  272.  
  273. v=0
  274. o=root 1595019147 1595019148 IN IP4 10.192.26.204
  275. s=Asterisk PBX 1.8.4.4
  276. c=IN IP4 10.192.26.204
  277. t=0 0
  278. m=audio 10072 RTP/AVP 0 8 3 101
  279. a=rtpmap:0 PCMU/8000
  280. a=rtpmap:8 PCMA/8000
  281. a=rtpmap:3 GSM/8000
  282. a=rtpmap:101 telephone-event/8000
  283. a=fmtp:101 0-16
  284. a=ptime:20
  285. a=sendrecv
  286.  
  287. ---
  288.  
  289. <--- SIP read from UDP:91.121.129.17:5060 --->
  290. SIP/2.0 407 authentication required
  291. Allow: UPDATE,REFER,INFO
  292. Call-ID: [email protected]
  293. Contact: <sip:[email protected]:5060;user=phone>
  294. CSeq: 102 INVITE
  295. From: "Default Extension" <sip:[email protected]>;tag=as4523e5ab
  296. Proxy-Authenticate: Digest realm="sip.ovh.net",nonce="0973b5c766fddd071e1927cd6756ac7d",opaque="0972c2e8159490c",stale=false,algorithm=MD5
  297. Server: Cirpack/v4.42j (gw_sip)
  298. To: <sip:[email protected]>;tag=00-07862-0973bb13-60144c9b0
  299. Via: SIP/2.0/UDP 10.192.26.204:5060;received=95.210.161.54;rport=5060;branch=z9hG4bK3e7cd865
  300. Content-Length: 0
  301.  
  302. <------------->
  303. --- (11 headers 0 lines) ---
  304. Transmitting (NAT) to 91.121.129.17:5060:
  305. ACK sip:[email protected] SIP/2.0
  306. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK0b23a06a;rport
  307. Max-Forwards: 70
  308. From: "Default Extension" <sip:[email protected]>;tag=as4523e5ab
  309. To: <sip:[email protected]>
  310. Contact: <sip:[email protected]:5060>
  311. Call-ID: [email protected]
  312. CSeq: 102 ACK
  313. User-Agent: AskoziaPBX
  314. Content-Length: 0
  315.  
  316.  
  317. ---
  318. Retransmitting #1 (NAT) to 91.121.129.17:5060:
  319. INVITE sip:[email protected] SIP/2.0
  320. Via: SIP/2.0/UDP 10.192.26.204:5060;branch=z9hG4bK0b23a06a;rport
  321. Max-Forwards: 70
  322. From: "Default Extension" <sip:[email protected]>;tag=as4523e5ab
  323. To: <sip:[email protected]>
  324. Contact: <sip:[email protected]:5060>
  325. Call-ID: [email protected]
  326. CSeq: 103 INVITE
  327. User-Agent: AskoziaPBX
  328. Proxy-Authorization: Digest username="0033183643585", realm="sip.ovh.net", algorithm=MD5, uri="sip:[email protected]", nonce="0973b5c766fddd071e1927cd6756ac7d", response="8d1362978804f22773300724ae9822ef", opaque="0972c2e8159490c"
  329. Date: Mon, 12 Dec 2011 11:48:07 GMT
  330. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  331. Supported: replaces, timer
  332. Content-Type: application/sdp
  333. Content-Length: 285
  334.  
  335. v=0
  336. o=root 1595019147 1595019148 IN IP4 10.192.26.204
  337. s=Asterisk PBX 1.8.4.4
  338. c=IN IP4 10.192.26.204
  339. t=0 0
  340. m=audio 10072 RTP/AVP 0 8 3 101
  341. a=rtpmap:0 PCMU/8000
  342. a=rtpmap:8 PCMA/8000
  343. a=rtpmap:3 GSM/8000
  344. a=rtpmap:101 telephone-event/8000
  345. a=fmtp:101 0-16
  346. a=ptime:20
  347. a=sendrecv
  348.  
  349. ---
  350.  
  351. <--- SIP read from UDP:91.121.129.17:5060 --->
  352. SIP/2.0 100 Trying
  353. Allow: UPDATE,REFER,INFO
  354. Call-ID: [email protected]
  355. Contact: <sip:91.121.129.17:5060>
  356. CSeq: 103 INVITE
  357. From: "Default Extension" <sip:[email protected]>;tag=as4523e5ab
  358. Server: Cirpack/v4.42j (gw_sip)
  359. To: <sip:[email protected]>
  360. Via: SIP/2.0/UDP 10.192.26.204:5060;received=95.210.161.54;rport=5060;branch=z9hG4bK0b23a06a
  361. Content-Length: 0
  362.  
  363. <------------->
  364. --- (10 headers 0 lines) ---
  365.  
  366. <--- SIP read from UDP:192.168.1.150:5060 --->
  367. CANCEL sip:[email protected] SIP/2.0
  368. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194
  369. From: "Florent TOTOR" <sip:[email protected]>;tag=ba6dba2dbe613f58
  370. To: <sip:[email protected]>
  371. Supported: path
  372. X-Grandstream-PBX: true
  373. Call-ID: [email protected]
  374. CSeq: 33405 CANCEL
  375. User-Agent: Grandstream GXP1200 1.2.5.3
  376. Max-Forwards: 70
  377. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  378. Content-Length: 0
  379.  
  380. <------------->
  381. --- (12 headers 0 lines) ---
  382. Sending to 192.168.1.150:5060 (NAT)
  383.  
  384. <--- Reliably Transmitting (NAT) to 192.168.1.150:5060 --->
  385. SIP/2.0 487 Request Terminated
  386. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194;received=192.168.1.150;rport=5060
  387. From: "Florent TOTOR" <sip:[email protected]>;tag=ba6dba2dbe613f58
  388. To: <sip:[email protected]>;tag=as0683db8c
  389. Call-ID: [email protected]
  390. CSeq: 33405 INVITE
  391. Server: AskoziaPBX
  392. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  393. Supported: replaces, timer
  394. Content-Length: 0
  395.  
  396.  
  397. <------------>
  398.  
  399. <--- Transmitting (NAT) to 192.168.1.150:5060 --->
  400. SIP/2.0 200 OK
  401. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194;received=192.168.1.150;rport=5060
  402. From: "Florent TOTOR" <sip:[email protected]>;tag=ba6dba2dbe613f58
  403. To: <sip:[email protected]>;tag=as0683db8c
  404. Call-ID: [email protected]
  405. CSeq: 33405 CANCEL
  406. Server: AskoziaPBX
  407. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  408. Supported: replaces, timer
  409. Content-Length: 0
  410.  
  411.  
  412. <------------>
  413. Scheduling destruction of SIP dialog '[email protected]' in 53440 ms (Method: INVITE)
  414. Retransmitting #1 (NAT) to 192.168.1.150:5060:
  415. SIP/2.0 487 Request Terminated
  416. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194;received=192.168.1.150;rport=5060
  417. From: "Florent TOTOR" <sip:[email protected]>;tag=ba6dba2dbe613f58
  418. To: <sip:[email protected]>;tag=as0683db8c
  419. Call-ID: [email protected]
  420. CSeq: 33405 INVITE
  421. Server: AskoziaPBX
  422. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  423. Supported: replaces, timer
  424. Content-Length: 0
  425.  
  426.  
  427. ---
  428.  
  429. <--- SIP read from UDP:192.168.1.150:5060 --->
  430. ACK sip:[email protected] SIP/2.0
  431. Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bKb82faf11bd292194
  432. From: "Florent TOTOR" <sip:[email protected]>;tag=ba6dba2dbe613f58
  433. To: <sip:[email protected]>;tag=as0683db8c
  434. Contact: <sip:[email protected]:5060;transport=udp>
  435. Supported: path
  436. X-Grandstream-PBX: true
  437. Authorization: Digest username="101", realm="asterisk", algorithm=MD5, uri="sip:[email protected]", nonce="75a6cfa2", response="56ea22283ad59cd23789e02653c31800"
  438. Call-ID: [email protected]
  439. CSeq: 33405 ACK
  440. User-Agent: Grandstream GXP1200 1.2.5.3
  441. Max-Forwards: 70
  442. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
  443. Content-Length: 0
  444.  
  445. <------------->
  446. --- (14 headers 0 lines) ---
  447. Really destroying SIP dialog '[email protected]' Method: ACK
  448. Really destroying SIP dialog '[email protected]' Method: REGISTER
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