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  1. Connected to Asterisk 13.3.2 currently running on almudo (pid = 8286)
  2. == Using SIP RTP CoS mark 5
  3. -- Executing [5659@from-internal:1] NoOp("SIP/enlace-trunk-00000002", "Incomming call from 5659 to "2288175116" <2288175116> to PAranoids") in new stack
  4. -- Executing [5659@from-internal:2] Set("SIP/enlace-trunk-00000002", "CALLERID(all)=9999") in new stack
  5. -- Executing [5659@from-internal:3] Dial("SIP/enlace-trunk-00000002", "SIP/paralax/5659") in new stack
  6. == Using SIP RTP CoS mark 5
  7. -- Called SIP/paralax/5659
  8. -- SIP/paralax-00000003 is ringing
  9. -- SIP/paralax-00000003 answered SIP/enlace-trunk-00000002
  10. -- Channel SIP/enlace-trunk-00000002 joined 'simple_bridge' basic-bridge <1e6f92e5-b154-43e9-a729-9cc184b6c4f2>
  11. -- Channel SIP/paralax-00000003 joined 'simple_bridge' basic-bridge <1e6f92e5-b154-43e9-a729-9cc184b6c4f2>
  12. > Bridge 1e6f92e5-b154-43e9-a729-9cc184b6c4f2: switching from simple_bridge technology to native_rtp
  13. > Remotely bridged 'SIP/paralax-00000003' and 'SIP/enlace-trunk-00000002' - media will flow directly between them
  14. > Remotely bridged 'SIP/paralax-00000003' and 'SIP/enlace-trunk-00000002' - media will flow directly between them
  15. -- Channel SIP/paralax-00000003 left 'native_rtp' basic-bridge <1e6f92e5-b154-43e9-a729-9cc184b6c4f2>
  16. -- Channel SIP/enlace-trunk-00000002 left 'native_rtp' basic-bridge <1e6f92e5-b154-43e9-a729-9cc184b6c4f2>
  17. == Spawn extension (from-internal, 5659, 3) exited non-zero on 'SIP/enlace-trunk-00000002'
  18. almudo*CLI> sip set debug on
  19. SIP Debugging enabled
  20.  
  21. <--- SIP read from UDP:10.100.5.67:5060 --->
  22. INVITE sip:[email protected]:5060;user=phone SIP/2.0
  23. Via: SIP/2.0/UDP 10.100.5.67:5060;branch=z9hG4bKibh86f200o9hv4lli6a0.1
  24. To: "1331342282375659"<sip:[email protected];transport=udp>
  25. From: "2288175116"<sip:[email protected];transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
  26. Call-ID: [email protected]
  27. CSeq: 1 INVITE
  28. Max-Forwards: 66
  29. Content-Length: 333
  30. Contact: <sip:10.100.5.67:5060;transport=udp>
  31. Content-Type: application/sdp
  32. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, INFO, PRACK, NOTIFY, MESSAGE, UPDATE
  33. Supported: 100rel, timer, histinfo, precondition
  34. P-Asserted-Identity: <sip:[email protected];SBC;SBC>
  35. Min-SE: 90
  36. Session-Expires: 1800;refresher=uac
  37. P-Early-Media: supported
  38. Route: <sip:[email protected]:5060;user=phone;lr>
  39.  
  40. v=0
  41. o=HuaweiSoftx3000 1109197089 1109197090 IN IP4 10.100.5.67
  42. s=SipCall
  43. c=IN IP4 10.100.5.67
  44. t=0 0
  45. m=audio 59534 RTP/AVP 8 116
  46. a=rtpmap:8 PCMA/8000
  47. a=rtpmap:116 telephone-event/8000
  48. a=ptime:20
  49. a=curr:qos local none
  50. a=curr:qos remote none
  51. a=des:qos mandatory local sendrecv
  52. a=des:qos optional remote sendrecv
  53. a=3gOoBTC
  54. <------------->
  55. --- (17 headers 14 lines) ---
  56. Sending to 10.100.5.67:5060 (no NAT)
  57. Sending to 10.100.5.67:5060 (no NAT)
  58. Using INVITE request as basis request - [email protected]
  59. Found peer 'enlace-trunk' for '2288175116' from 10.100.5.67:5060
  60. == Using SIP RTP CoS mark 5
  61. Found RTP audio format 8
  62. Found RTP audio format 116
  63. Found audio description format PCMA for ID 8
  64. Found audio description format telephone-event for ID 116
  65. Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
  66. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  67. Peer audio RTP is at port 10.100.5.67:59534
  68. Looking for 5659 in from-internal (domain sbcacme.ims.iusatel.com)
  69. sip_route_dump: route/path hop: <sip:10.100.5.67:5060;transport=udp>
  70.  
  71. <--- Transmitting (no NAT) to 10.100.5.67:5060 --->
  72. SIP/2.0 100 Trying
  73. Via: SIP/2.0/UDP 10.100.5.67:5060;branch=z9hG4bKibh86f200o9hv4lli6a0.1;received=10.100.5.67
  74. From: "2288175116"<sip:[email protected];transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
  75. To: "1331342282375659"<sip:[email protected];transport=udp>
  76. Call-ID: [email protected]
  77. CSeq: 1 INVITE
  78. Server: Asterisk PBX 13.3.2
  79. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  80. Supported: replaces, timer
  81. Session-Expires: 1800;refresher=uac
  82. Contact: <sip:[email protected]:5060>
  83. Content-Length: 0
  84.  
  85.  
  86. <------------>
  87. -- Executing [5659@from-internal:1] NoOp("SIP/enlace-trunk-00000004", "Incomming call from 5659 to "2288175116" <2288175116> to PAranoids") in new stack
  88. -- Executing [5659@from-internal:2] Set("SIP/enlace-trunk-00000004", "CALLERID(all)=9999") in new stack
  89. -- Executing [5659@from-internal:3] Dial("SIP/enlace-trunk-00000004", "SIP/paralax/5659") in new stack
  90. == Using SIP RTP CoS mark 5
  91. Audio is at 11086
  92. Adding codec alaw to SDP
  93. Adding codec ulaw to SDP
  94. Adding non-codec 0x1 (telephone-event) to SDP
  95. Reliably Transmitting (no NAT) to 10.0.1.251:5060:
  96. INVITE sip:[email protected] SIP/2.0
  97. Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7bec7829
  98. Max-Forwards: 70
  99. From: <sip:[email protected]>;tag=as5e212125
  100. To: <sip:[email protected]>
  101. Contact: <sip:[email protected]:5060>
  102. Call-ID: [email protected]:5060
  103. CSeq: 102 INVITE
  104. User-Agent: Asterisk PBX 13.3.2
  105. Date: Mon, 08 Feb 2016 17:06:11 GMT
  106. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  107. Supported: replaces, timer
  108. Content-Type: application/sdp
  109. Content-Length: 259
  110.  
  111. v=0
  112. o=root 1718921788 1718921788 IN IP4 10.0.1.115
  113. s=Asterisk PBX 13.3.2
  114. c=IN IP4 10.0.1.115
  115. t=0 0
  116. m=audio 11086 RTP/AVP 8 0 116
  117. a=rtpmap:8 PCMA/8000
  118. a=rtpmap:0 PCMU/8000
  119. a=rtpmap:116 telephone-event/8000
  120. a=fmtp:116 0-16
  121. a=maxptime:150
  122. a=sendrecv
  123.  
  124. ---
  125. -- Called SIP/paralax/5659
  126.  
  127. <--- SIP read from UDP:10.0.1.251:5060 --->
  128. SIP/2.0 401 Unauthorized
  129. Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7bec7829;received=10.0.1.115;rport=5060
  130. From: <sip:[email protected]>;tag=as5e212125
  131. To: <sip:[email protected]>;tag=as0b69474c
  132. Call-ID: [email protected]:5060
  133. CSeq: 102 INVITE
  134. Server: FPBX-AsteriskNOW-12.0.68(13.0.1)
  135. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  136. Supported: replaces, timer
  137. WWW-Authenticate: Digest algorithm=MD5, realm="187.188.213.132", nonce="24d8e0eb"
  138. Content-Length: 0
  139.  
  140. <------------->
  141. --- (11 headers 0 lines) ---
  142. Transmitting (no NAT) to 10.0.1.251:5060:
  143. ACK sip:[email protected] SIP/2.0
  144. Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7bec7829
  145. Max-Forwards: 70
  146. From: <sip:[email protected]>;tag=as5e212125
  147. To: <sip:[email protected]>;tag=as0b69474c
  148. Contact: <sip:[email protected]:5060>
  149. Call-ID: [email protected]:5060
  150. CSeq: 102 ACK
  151. User-Agent: Asterisk PBX 13.3.2
  152. Content-Length: 0
  153.  
  154.  
  155. ---
  156. Audio is at 11086
  157. Adding codec alaw to SDP
  158. Adding codec ulaw to SDP
  159. Adding non-codec 0x1 (telephone-event) to SDP
  160. Reliably Transmitting (no NAT) to 10.0.1.251:5060:
  161. INVITE sip:[email protected] SIP/2.0
  162. Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK57e27115
  163. Max-Forwards: 70
  164. From: <sip:[email protected]>;tag=as5e212125
  165. To: <sip:[email protected]>
  166. Contact: <sip:[email protected]:5060>
  167. Call-ID: [email protected]:5060
  168. CSeq: 103 INVITE
  169. User-Agent: Asterisk PBX 13.3.2
  170. Authorization: Digest username="9999", realm="187.188.213.132", algorithm=MD5, uri="sip:[email protected]", nonce="24d8e0eb", response="ad75953f2fb5e5512a246c493ef10fc3"
  171. Date: Mon, 08 Feb 2016 17:06:11 GMT
  172. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  173. Supported: replaces, timer
  174. Content-Type: application/sdp
  175. Content-Length: 259
  176.  
  177. v=0
  178. o=root 1718921788 1718921789 IN IP4 10.0.1.115
  179. s=Asterisk PBX 13.3.2
  180. c=IN IP4 10.0.1.115
  181. t=0 0
  182. m=audio 11086 RTP/AVP 8 0 116
  183. a=rtpmap:8 PCMA/8000
  184. a=rtpmap:0 PCMU/8000
  185. a=rtpmap:116 telephone-event/8000
  186. a=fmtp:116 0-16
  187. a=maxptime:150
  188. a=sendrecv
  189.  
  190. ---
  191.  
  192. <--- SIP read from UDP:10.0.1.251:5060 --->
  193. SIP/2.0 100 Trying
  194. Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK57e27115;received=10.0.1.115;rport=5060
  195. From: <sip:[email protected]>;tag=as5e212125
  196. To: <sip:[email protected]>
  197. Call-ID: [email protected]:5060
  198. CSeq: 103 INVITE
  199. Server: FPBX-AsteriskNOW-12.0.68(13.0.1)
  200. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  201. Supported: replaces, timer
  202. Session-Expires: 1800;refresher=uas
  203. Contact: <sip:[email protected]:5060>
  204. Content-Length: 0
  205.  
  206. <------------->
  207. --- (12 headers 0 lines) ---
  208.  
  209. <--- SIP read from UDP:10.0.1.251:5060 --->
  210. SIP/2.0 180 Ringing
  211. Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK57e27115;received=10.0.1.115;rport=5060
  212. From: <sip:[email protected]>;tag=as5e212125
  213. To: <sip:[email protected]>;tag=as4f65d3d3
  214. Call-ID: [email protected]:5060
  215. CSeq: 103 INVITE
  216. Server: FPBX-AsteriskNOW-12.0.68(13.0.1)
  217. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  218. Supported: replaces, timer
  219. Session-Expires: 1800;refresher=uas
  220. Contact: <sip:[email protected]:5060>
  221. P-Asserted-Identity: "dev8" <sip:[email protected]>
  222. Content-Length: 0
  223.  
  224. <------------->
  225. --- (13 headers 0 lines) ---
  226. sip_route_dump: route/path hop: <sip:[email protected]:5060>
  227. -- SIP/paralax-00000005 is ringing
  228.  
  229. <--- Transmitting (no NAT) to 10.100.5.67:5060 --->
  230. SIP/2.0 180 Ringing
  231. Via: SIP/2.0/UDP 10.100.5.67:5060;branch=z9hG4bKibh86f200o9hv4lli6a0.1;received=10.100.5.67
  232. From: "2288175116"<sip:[email protected];transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
  233. To: "1331342282375659"<sip:[email protected];transport=udp>;tag=as7723fa42
  234. Call-ID: [email protected]
  235. CSeq: 1 INVITE
  236. Server: Asterisk PBX 13.3.2
  237. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  238. Supported: replaces, timer
  239. Session-Expires: 1800;refresher=uac
  240. Contact: <sip:[email protected]:5060>
  241. Content-Length: 0
  242.  
  243.  
  244. <------------>
  245.  
  246. <--- SIP read from UDP:10.0.1.251:5060 --->
  247. SIP/2.0 200 OK
  248. Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK57e27115;received=10.0.1.115;rport=5060
  249. From: <sip:[email protected]>;tag=as5e212125
  250. To: <sip:[email protected]>;tag=as4f65d3d3
  251. Call-ID: [email protected]:5060
  252. CSeq: 103 INVITE
  253. Server: FPBX-AsteriskNOW-12.0.68(13.0.1)
  254. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  255. Supported: replaces, timer
  256. Session-Expires: 1800;refresher=uas
  257. Contact: <sip:[email protected]:5060>
  258. P-Asserted-Identity: "dev8" <sip:[email protected]>
  259. Content-Type: application/sdp
  260. Require: timer
  261. Content-Length: 282
  262.  
  263. v=0
  264. o=root 1402105843 1402105843 IN IP4 10.0.1.251
  265. s=Asterisk PBX 13.0.1
  266. c=IN IP4 10.0.1.251
  267. t=0 0
  268. m=audio 15450 RTP/AVP 8 18 116
  269. a=rtpmap:8 PCMA/8000
  270. a=rtpmap:18 G729/8000
  271. a=fmtp:18 annexb=no
  272. a=rtpmap:116 telephone-event/8000
  273. a=fmtp:116 0-16
  274. a=maxptime:150
  275. a=sendrecv
  276. <------------->
  277. --- (15 headers 13 lines) ---
  278. Found RTP audio format 8
  279. Found RTP audio format 18
  280. Found RTP audio format 116
  281. Found audio description format PCMA for ID 8
  282. Found audio description format G729 for ID 18
  283. Found audio description format telephone-event for ID 116
  284. Capabilities: us - (alaw|ulaw), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
  285. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  286. Peer audio RTP is at port 10.0.1.251:15450
  287. sip_route_dump: route/path hop: <sip:[email protected]:5060>
  288. set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  289. set_destination: set destination to 10.0.1.251:5060
  290. Transmitting (no NAT) to 10.0.1.251:5060:
  291. ACK sip:[email protected]:5060 SIP/2.0
  292. Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK3f3b65a1
  293. Max-Forwards: 70
  294. From: <sip:[email protected]>;tag=as5e212125
  295. To: <sip:[email protected]>;tag=as4f65d3d3
  296. Contact: <sip:[email protected]:5060>
  297. Call-ID: [email protected]:5060
  298. CSeq: 103 ACK
  299. User-Agent: Asterisk PBX 13.3.2
  300. Content-Length: 0
  301.  
  302.  
  303. ---
  304. -- SIP/paralax-00000005 answered SIP/enlace-trunk-00000004
  305. Audio is at 13642
  306. Adding codec alaw to SDP
  307. Adding codec g729 to SDP
  308. Adding non-codec 0x1 (telephone-event) to SDP
  309.  
  310. <--- Reliably Transmitting (no NAT) to 10.100.5.67:5060 --->
  311. SIP/2.0 200 OK
  312. Via: SIP/2.0/UDP 10.100.5.67:5060;branch=z9hG4bKibh86f200o9hv4lli6a0.1;received=10.100.5.67
  313. From: "2288175116"<sip:[email protected];transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
  314. To: "1331342282375659"<sip:[email protected];transport=udp>;tag=as7723fa42
  315. Call-ID: [email protected]
  316. CSeq: 1 INVITE
  317. Server: Asterisk PBX 13.3.2
  318. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  319. Supported: replaces, timer
  320. Session-Expires: 1800;refresher=uac
  321. Contact: <sip:[email protected]:5060>
  322. Content-Type: application/sdp
  323. Require: timer
  324. Content-Length: 284
  325.  
  326. v=0
  327. o=root 438711394 438711394 IN IP4 10.50.98.214
  328. s=Asterisk PBX 13.3.2
  329. c=IN IP4 10.50.98.214
  330. t=0 0
  331. m=audio 13642 RTP/AVP 8 18 116
  332. a=rtpmap:8 PCMA/8000
  333. a=rtpmap:18 G729/8000
  334. a=fmtp:18 annexb=no
  335. a=rtpmap:116 telephone-event/8000
  336. a=fmtp:116 0-16
  337. a=maxptime:150
  338. a=sendrecv
  339.  
  340. <------------>
  341. -- Channel SIP/enlace-trunk-00000004 joined 'simple_bridge' basic-bridge <041d4c74-5a1e-44d3-99da-77bffd26259b>
  342. -- Channel SIP/paralax-00000005 joined 'simple_bridge' basic-bridge <041d4c74-5a1e-44d3-99da-77bffd26259b>
  343. > Bridge 041d4c74-5a1e-44d3-99da-77bffd26259b: switching from simple_bridge technology to native_rtp
  344. set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  345. set_destination: set destination to 10.0.1.251:5060
  346. Audio is at 11086
  347. Adding codec alaw to SDP
  348. Adding codec ulaw to SDP
  349. Adding non-codec 0x1 (telephone-event) to SDP
  350. Reliably Transmitting (no NAT) to 10.0.1.251:5060:
  351. INVITE sip:[email protected]:5060 SIP/2.0
  352. Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7160ff0f
  353. Max-Forwards: 70
  354. From: <sip:[email protected]>;tag=as5e212125
  355. To: <sip:[email protected]>;tag=as4f65d3d3
  356. Contact: <sip:[email protected]:5060>
  357. Call-ID: [email protected]:5060
  358. CSeq: 104 INVITE
  359. User-Agent: Asterisk PBX 13.3.2
  360. Session-Expires: 1800;refresher=uas
  361. Min-SE: 90
  362. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  363. Supported: replaces, timer
  364. X-asterisk-Info: SIP re-invite (External RTP bridge)
  365. Content-Type: application/sdp
  366. Content-Length: 261
  367.  
  368. v=0
  369. o=root 1718921788 1718921790 IN IP4 10.100.5.67
  370. s=Asterisk PBX 13.3.2
  371. c=IN IP4 10.100.5.67
  372. t=0 0
  373. m=audio 59534 RTP/AVP 8 0 116
  374. a=rtpmap:8 PCMA/8000
  375. a=rtpmap:0 PCMU/8000
  376. a=rtpmap:116 telephone-event/8000
  377. a=fmtp:116 0-16
  378. a=maxptime:150
  379. a=sendrecv
  380.  
  381. ---
  382. > Remotely bridged 'SIP/paralax-00000005' and 'SIP/enlace-trunk-00000004' - media will flow directly between them
  383. > Remotely bridged 'SIP/paralax-00000005' and 'SIP/enlace-trunk-00000004' - media will flow directly between them
  384.  
  385. <--- SIP read from UDP:10.0.1.251:5060 --->
  386. SIP/2.0 100 Trying
  387. Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7160ff0f;received=10.0.1.115;rport=5060
  388. From: <sip:[email protected]>;tag=as5e212125
  389. To: <sip:[email protected]>;tag=as4f65d3d3
  390. Call-ID: [email protected]:5060
  391. CSeq: 104 INVITE
  392. Server: FPBX-AsteriskNOW-12.0.68(13.0.1)
  393. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  394. Supported: replaces, timer
  395. Session-Expires: 1800;refresher=uas
  396. Contact: <sip:[email protected]:5060>
  397. Content-Length: 0
  398.  
  399. <------------->
  400. --- (12 headers 0 lines) ---
  401.  
  402. <--- SIP read from UDP:10.0.1.251:5060 --->
  403. SIP/2.0 200 OK
  404. Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7160ff0f;received=10.0.1.115;rport=5060
  405. From: <sip:[email protected]>;tag=as5e212125
  406. To: <sip:[email protected]>;tag=as4f65d3d3
  407. Call-ID: [email protected]:5060
  408. CSeq: 104 INVITE
  409. Server: FPBX-AsteriskNOW-12.0.68(13.0.1)
  410. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  411. Supported: replaces, timer
  412. Session-Expires: 1800;refresher=uas
  413. Contact: <sip:[email protected]:5060>
  414. Content-Type: application/sdp
  415. Require: timer
  416. Content-Length: 282
  417.  
  418. v=0
  419. o=root 1402105843 1402105844 IN IP4 10.0.1.251
  420. s=Asterisk PBX 13.0.1
  421. c=IN IP4 10.0.1.251
  422. t=0 0
  423. m=audio 15450 RTP/AVP 8 18 116
  424. a=rtpmap:8 PCMA/8000
  425. a=rtpmap:18 G729/8000
  426. a=fmtp:18 annexb=no
  427. a=rtpmap:116 telephone-event/8000
  428. a=fmtp:116 0-16
  429. a=maxptime:150
  430. a=sendrecv
  431. <------------->
  432. --- (14 headers 13 lines) ---
  433. Found RTP audio format 8
  434. Found RTP audio format 18
  435. Found RTP audio format 116
  436. Found audio description format PCMA for ID 8
  437. Found audio description format G729 for ID 18
  438. Found audio description format telephone-event for ID 116
  439. Capabilities: us - (alaw|ulaw), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
  440. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  441. Peer audio RTP is at port 10.0.1.251:15450
  442. set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
  443. set_destination: set destination to 10.0.1.251:5060
  444. Transmitting (no NAT) to 10.0.1.251:5060:
  445. ACK sip:[email protected]:5060 SIP/2.0
  446. Via: SIP/2.0/UDP 10.0.1.115:5060;branch=z9hG4bK7ffb1e69
  447. Max-Forwards: 70
  448. From: <sip:[email protected]>;tag=as5e212125
  449. To: <sip:[email protected]>;tag=as4f65d3d3
  450. Contact: <sip:[email protected]:5060>
  451. Call-ID: [email protected]:5060
  452. CSeq: 104 ACK
  453. User-Agent: Asterisk PBX 13.3.2
  454. Content-Length: 0
  455.  
  456.  
  457. ---
  458.  
  459. <--- SIP read from UDP:10.100.5.67:5060 --->
  460. ACK sip:[email protected]:5060 SIP/2.0
  461. Via: SIP/2.0/UDP 10.100.5.67:5060;branch=z9hG4bKdstbld3008ugh3hvn100.1
  462. Call-ID: [email protected]
  463. From: "2288175116"<sip:[email protected];transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
  464. To: "1331342282375659"<sip:[email protected];transport=udp>;tag=as7723fa42
  465. CSeq: 1 ACK
  466. Max-Forwards: 68
  467. Content-Length: 0
  468.  
  469. <------------->
  470. --- (8 headers 0 lines) ---
  471. set_destination: Parsing <sip:10.100.5.67:5060;transport=udp> for address/port to send to
  472. set_destination: set destination to 10.100.5.67:5060
  473. Audio is at 13642
  474. Adding codec alaw to SDP
  475. Adding codec g729 to SDP
  476. Adding non-codec 0x1 (telephone-event) to SDP
  477. Reliably Transmitting (no NAT) to 10.100.5.67:5060:
  478. INVITE sip:10.100.5.67:5060;transport=udp SIP/2.0
  479. Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK3557de89
  480. Max-Forwards: 70
  481. From: "1331342282375659"<sip:[email protected];transport=udp>;tag=as7723fa42
  482. To: "2288175116"<sip:[email protected];transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
  483. Contact: <sip:[email protected]:5060>
  484. Call-ID: [email protected]
  485. CSeq: 102 INVITE
  486. User-Agent: Asterisk PBX 13.3.2
  487. Session-Expires: 1800;refresher=uas
  488. Min-SE: 90
  489. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  490. Supported: replaces, timer
  491. X-asterisk-Info: SIP re-invite (External RTP bridge)
  492. Content-Type: application/sdp
  493. Content-Length: 280
  494.  
  495. v=0
  496. o=root 438711394 438711395 IN IP4 10.0.1.251
  497. s=Asterisk PBX 13.3.2
  498. c=IN IP4 10.0.1.251
  499. t=0 0
  500. m=audio 15450 RTP/AVP 8 18 116
  501. a=rtpmap:8 PCMA/8000
  502. a=rtpmap:18 G729/8000
  503. a=fmtp:18 annexb=no
  504. a=rtpmap:116 telephone-event/8000
  505. a=fmtp:116 0-16
  506. a=maxptime:150
  507. a=sendrecv
  508.  
  509. ---
  510.  
  511. <--- SIP read from UDP:10.100.5.67:5060 --->
  512. SIP/2.0 200 OK
  513. Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK3557de89
  514. From: "1331342282375659"<sip:[email protected];transport=udp>;tag=as7723fa42
  515. To: "2288175116"<sip:[email protected];transport=udp>;tag=go5z55gn-CC-1014-OFC-231
  516. Call-ID: [email protected]
  517. CSeq: 102 INVITE
  518. Require: timer
  519. Session-Expires: 1800;refresher=uas
  520. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
  521. Contact: <sip:10.100.5.67:5060;transport=udp>
  522. Content-Length: 203
  523. Content-Type: application/sdp
  524.  
  525. v=0
  526. o=HuaweiSoftx3000 1109197089 1109197091 IN IP4 10.100.5.67
  527. s=SipCall
  528. c=IN IP4 10.100.5.67
  529. t=0 0
  530. m=audio 59534 RTP/AVP 8 116
  531. a=rtpmap:8 PCMA/8000
  532. a=rtpmap:116 telephone-event/8000
  533. a=ptime:20
  534. <------------->
  535. --- (12 headers 9 lines) ---
  536. Found RTP audio format 8
  537. Found RTP audio format 116
  538. Found audio description format PCMA for ID 8
  539. Found audio description format telephone-event for ID 116
  540. Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
  541. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  542. Peer audio RTP is at port 10.100.5.67:59534
  543. set_destination: Parsing <sip:10.100.5.67:5060;transport=udp> for address/port to send to
  544. set_destination: set destination to 10.100.5.67:5060
  545. Transmitting (no NAT) to 10.100.5.67:5060:
  546. ACK sip:10.100.5.67:5060;transport=udp SIP/2.0
  547. Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK56958df8
  548. Max-Forwards: 70
  549. From: "1331342282375659"<sip:[email protected];transport=udp>;tag=as7723fa42
  550. To: "2288175116"<sip:[email protected];transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
  551. Contact: <sip:[email protected]:5060>
  552. Call-ID: [email protected]
  553. CSeq: 102 ACK
  554. User-Agent: Asterisk PBX 13.3.2
  555. Content-Length: 0
  556.  
  557.  
  558. ---
  559.  
  560. <--- SIP read from UDP:10.0.1.251:5060 --->
  561. BYE sip:[email protected]:5060 SIP/2.0
  562. Via: SIP/2.0/UDP 10.0.1.251:5060;branch=z9hG4bK2cf673ec;rport
  563. Max-Forwards: 70
  564. From: <sip:[email protected]>;tag=as4f65d3d3
  565. To: <sip:[email protected]>;tag=as5e212125
  566. Call-ID: [email protected]:5060
  567. CSeq: 102 BYE
  568. User-Agent: FPBX-AsteriskNOW-12.0.68(13.0.1)
  569. Proxy-Authorization: Digest username="9999", realm="187.188.213.132", algorithm=MD5, uri="sip:10.0.1.251", nonce="24d8e0eb", response="7e7221853f1e2f04b7023a0b56faa86c"
  570. X-Asterisk-HangupCause: Normal Clearing
  571. X-Asterisk-HangupCauseCode: 16
  572. Content-Length: 0
  573.  
  574. <------------->
  575. --- (12 headers 0 lines) ---
  576. Sending to 10.0.1.251:5060 (no NAT)
  577. Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: BYE)
  578.  
  579. <--- Transmitting (no NAT) to 10.0.1.251:5060 --->
  580. SIP/2.0 200 OK
  581. Via: SIP/2.0/UDP 10.0.1.251:5060;branch=z9hG4bK2cf673ec;received=10.0.1.251;rport=5060
  582. From: <sip:[email protected]>;tag=as4f65d3d3
  583. To: <sip:[email protected]>;tag=as5e212125
  584. Call-ID: [email protected]:5060
  585. CSeq: 102 BYE
  586. Server: Asterisk PBX 13.3.2
  587. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  588. Supported: replaces, timer
  589. Content-Length: 0
  590.  
  591.  
  592. <------------>
  593. -- Channel SIP/paralax-00000005 left 'native_rtp' basic-bridge <041d4c74-5a1e-44d3-99da-77bffd26259b>
  594. set_destination: Parsing <sip:10.100.5.67:5060;transport=udp> for address/port to send to
  595. set_destination: set destination to 10.100.5.67:5060
  596. Audio is at 13642
  597. Adding codec alaw to SDP
  598. Adding codec g729 to SDP
  599. Adding non-codec 0x1 (telephone-event) to SDP
  600. Reliably Transmitting (no NAT) to 10.100.5.67:5060:
  601. INVITE sip:10.100.5.67:5060;transport=udp SIP/2.0
  602. Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK25efe9e2
  603. Max-Forwards: 70
  604. From: "1331342282375659"<sip:[email protected];transport=udp>;tag=as7723fa42
  605. To: "2288175116"<sip:[email protected];transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
  606. Contact: <sip:[email protected]:5060>
  607. Call-ID: [email protected]
  608. CSeq: 103 INVITE
  609. User-Agent: Asterisk PBX 13.3.2
  610. Session-Expires: 1800;refresher=uas
  611. Min-SE: 90
  612. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  613. Supported: replaces, timer
  614. X-asterisk-Info: SIP re-invite (External RTP bridge)
  615. Content-Type: application/sdp
  616. Content-Length: 284
  617.  
  618. v=0
  619. o=root 438711394 438711396 IN IP4 10.50.98.214
  620. s=Asterisk PBX 13.3.2
  621. c=IN IP4 10.50.98.214
  622. t=0 0
  623. m=audio 13642 RTP/AVP 8 18 116
  624. a=rtpmap:8 PCMA/8000
  625. a=rtpmap:18 G729/8000
  626. a=fmtp:18 annexb=no
  627. a=rtpmap:116 telephone-event/8000
  628. a=fmtp:116 0-16
  629. a=maxptime:150
  630. a=sendrecv
  631.  
  632. ---
  633. -- Channel SIP/enlace-trunk-00000004 left 'native_rtp' basic-bridge <041d4c74-5a1e-44d3-99da-77bffd26259b>
  634. == Spawn extension (from-internal, 5659, 3) exited non-zero on 'SIP/enlace-trunk-00000004'
  635. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: ACK)
  636.  
  637. <--- SIP read from UDP:10.100.5.67:5060 --->
  638. SIP/2.0 200 OK
  639. Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK25efe9e2
  640. From: "1331342282375659"<sip:[email protected];transport=udp>;tag=as7723fa42
  641. To: "2288175116"<sip:[email protected];transport=udp>;tag=go5z55gn-CC-1014-OFC-231
  642. Call-ID: [email protected]
  643. CSeq: 103 INVITE
  644. Require: timer
  645. Session-Expires: 1800;refresher=uas
  646. Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
  647. Contact: <sip:10.100.5.67:5060;transport=udp>
  648. Content-Length: 203
  649. Content-Type: application/sdp
  650.  
  651. v=0
  652. o=HuaweiSoftx3000 1109197089 1109197092 IN IP4 10.100.5.67
  653. s=SipCall
  654. c=IN IP4 10.100.5.67
  655. t=0 0
  656. m=audio 59534 RTP/AVP 8 116
  657. a=rtpmap:8 PCMA/8000
  658. a=rtpmap:116 telephone-event/8000
  659. a=ptime:20
  660. <------------->
  661. --- (12 headers 9 lines) ---
  662. Found RTP audio format 8
  663. Found RTP audio format 116
  664. Found audio description format PCMA for ID 8
  665. Found audio description format telephone-event for ID 116
  666. Capabilities: us - (alaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
  667. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  668. Peer audio RTP is at port 10.100.5.67:59534
  669. set_destination: Parsing <sip:10.100.5.67:5060;transport=udp> for address/port to send to
  670. set_destination: set destination to 10.100.5.67:5060
  671. Transmitting (no NAT) to 10.100.5.67:5060:
  672. ACK sip:10.100.5.67:5060;transport=udp SIP/2.0
  673. Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK60e7eadd
  674. Max-Forwards: 70
  675. From: "1331342282375659"<sip:[email protected];transport=udp>;tag=as7723fa42
  676. To: "2288175116"<sip:[email protected];transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
  677. Contact: <sip:[email protected]:5060>
  678. Call-ID: [email protected]
  679. CSeq: 103 ACK
  680. User-Agent: Asterisk PBX 13.3.2
  681. Content-Length: 0
  682.  
  683.  
  684. ---
  685. set_destination: Parsing <sip:10.100.5.67:5060;transport=udp> for address/port to send to
  686. set_destination: set destination to 10.100.5.67:5060
  687. Reliably Transmitting (no NAT) to 10.100.5.67:5060:
  688. BYE sip:10.100.5.67:5060;transport=udp SIP/2.0
  689. Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK473307f5
  690. Max-Forwards: 70
  691. From: "1331342282375659"<sip:[email protected];transport=udp>;tag=as7723fa42
  692. To: "2288175116"<sip:[email protected];transport=udp;user=phone>;tag=go5z55gn-CC-1014-OFC-231
  693. Call-ID: [email protected]
  694. CSeq: 104 BYE
  695. User-Agent: Asterisk PBX 13.3.2
  696. X-Asterisk-HangupCause: Normal Clearing
  697. X-Asterisk-HangupCauseCode: 16
  698. Content-Length: 0
  699.  
  700.  
  701. ---
  702. Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: ACK)
  703.  
  704. <--- SIP read from UDP:10.100.5.67:5060 --->
  705. SIP/2.0 200 OK
  706. Via: SIP/2.0/UDP 10.50.98.214:5060;branch=z9hG4bK473307f5
  707. From: "1331342282375659"<sip:[email protected];transport=udp>;tag=as7723fa42
  708. To: "2288175116"<sip:[email protected];transport=udp>;tag=go5z55gn-CC-1014-OFC-231
  709. Call-ID: [email protected]
  710. CSeq: 104 BYE
  711. Content-Length: 0
  712.  
  713. <------------->
  714. --- (7 headers 0 lines) ---
  715. Really destroying SIP dialog '[email protected]' Method: ACK
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