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  1. voip1*CLI> sip set debug on
  2. SIP Debugging enabled
  3. Really destroying SIP dialog '2130837723@192.168.2.253' Method: REGISTER
  4.  
  5. <--- SIP read from UDP:10.128.69.4:5060 --->
  6. INVITE sip:699@192.168.2.253 SIP/2.0
  7. Max-Forwards: 20
  8. Via: SIP/2.0/UDP 10.128.69.4:5060;rport;branch=z9hG4bK1470626021
  9. From: <sip:802@192.168.2.253>;tag=1103491194
  10. To: <sip:699@192.168.2.253>
  11. Call-ID: 749509218@192.168.2.253
  12. CSeq: 10 INVITE
  13. User-Agent: YATE/5.5.0
  14. Contact: <sip:802@10.128.69.4:5060>
  15. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO
  16. Content-Type: application/sdp
  17. Content-Length: 502
  18.  
  19. v=0
  20. o=yate 1502460068 1502460068 IN IP4 10.128.69.4
  21. s=SIP Call
  22. c=IN IP4 10.128.69.4
  23. t=0 0
  24. m=audio 20400 RTP/AVP 0 8 3 11 98 97 102 103 104 105 106 101
  25. a=rtpmap:0 PCMU/8000
  26. a=rtpmap:8 PCMA/8000
  27. a=rtpmap:3 GSM/8000
  28. a=rtpmap:11 L16/8000
  29. a=rtpmap:98 iLBC/8000
  30. a=fmtp:98 mode=20
  31. a=rtpmap:97 iLBC/8000
  32. a=fmtp:97 mode=30
  33. a=rtpmap:102 SPEEX/8000
  34. a=rtpmap:103 SPEEX/16000
  35. a=rtpmap:104 SPEEX/32000
  36. a=rtpmap:105 iSAC/16000
  37. a=rtpmap:106 iSAC/32000
  38. a=rtpmap:101 telephone-event/8000
  39. a=ptime:30
  40. <------------->
  41. --- (12 headers 21 lines) ---
  42. Sending to 10.128.69.4:5060 (NAT)
  43. Sending to 10.128.69.4:5060 (NAT)
  44. Using INVITE request as basis request - 749509218@192.168.2.253
  45. Found peer '802' for '802' from 10.128.69.4:5060
  46.  
  47. <--- Reliably Transmitting (no NAT) to 10.128.69.4:5060 --->
  48. SIP/2.0 401 Unauthorized
  49. Via: SIP/2.0/UDP 10.128.69.4:5060;branch=z9hG4bK1470626021;received=10.128.69.4;rport=5060
  50. From: <sip:802@192.168.2.253>;tag=1103491194
  51. To: <sip:699@192.168.2.253>;tag=as5511a80b
  52. Call-ID: 749509218@192.168.2.253
  53. CSeq: 10 INVITE
  54. Server: FPBX-14.0.1.1(14.6.0)
  55. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  56. Supported: replaces, timer
  57. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="25b3f291"
  58. Content-Length: 0
  59.  
  60.  
  61. <------------>
  62. Scheduling destruction of SIP dialog '749509218@192.168.2.253' in 6400 ms (Method: INVITE)
  63.  
  64. <--- SIP read from UDP:10.128.69.4:5060 --->
  65. ACK sip:699@192.168.2.253 SIP/2.0
  66. Via: SIP/2.0/UDP 10.128.69.4:5060;rport;branch=z9hG4bK1470626021
  67. From: <sip:802@192.168.2.253>;tag=1103491194
  68. To: <sip:699@192.168.2.253>;tag=as5511a80b
  69. Call-ID: 749509218@192.168.2.253
  70. CSeq: 10 ACK
  71. Max-Forwards: 20
  72. Contact: <sip:802@10.128.69.4:5060>
  73. User-Agent: YATE/5.5.0
  74. Content-Length: 0
  75.  
  76. <------------->
  77. --- (10 headers 0 lines) ---
  78.  
  79. <--- SIP read from UDP:10.128.69.4:5060 --->
  80. INVITE sip:699@192.168.2.253 SIP/2.0
  81. Max-Forwards: 20
  82. Via: SIP/2.0/UDP 10.128.69.4:5060;rport;branch=z9hG4bK536449654
  83. From: <sip:802@192.168.2.253>;tag=1103491194
  84. To: <sip:699@192.168.2.253>
  85. Call-ID: 749509218@192.168.2.253
  86. User-Agent: YATE/5.5.0
  87. Contact: <sip:802@10.128.69.4:5060>
  88. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO
  89. CSeq: 11 INVITE
  90. Authorization: Digest username="802", realm="asterisk", nonce="25b3f291", uri="sip:699@192.168.2.253", response="9c6caffa8827cd9cc4692626cbccca91", algorithm=MD5
  91. Content-Type: application/sdp
  92. Content-Length: 502
  93.  
  94. v=0
  95. o=yate 1502460068 1502460068 IN IP4 10.128.69.4
  96. s=SIP Call
  97. c=IN IP4 10.128.69.4
  98. t=0 0
  99. m=audio 20400 RTP/AVP 0 8 3 11 98 97 102 103 104 105 106 101
  100. a=rtpmap:0 PCMU/8000
  101. a=rtpmap:8 PCMA/8000
  102. a=rtpmap:3 GSM/8000
  103. a=rtpmap:11 L16/8000
  104. a=rtpmap:98 iLBC/8000
  105. a=fmtp:98 mode=20
  106. a=rtpmap:97 iLBC/8000
  107. a=fmtp:97 mode=30
  108. a=rtpmap:102 SPEEX/8000
  109. a=rtpmap:103 SPEEX/16000
  110. a=rtpmap:104 SPEEX/32000
  111. a=rtpmap:105 iSAC/16000
  112. a=rtpmap:106 iSAC/32000
  113. a=rtpmap:101 telephone-event/8000
  114. a=ptime:30
  115. <------------->
  116. --- (13 headers 21 lines) ---
  117. Sending to 10.128.69.4:5060 (no NAT)
  118. Using INVITE request as basis request - 749509218@192.168.2.253
  119. Found peer '802' for '802' from 10.128.69.4:5060
  120. == Using SIP RTP TOS bits 184
  121. == Using SIP RTP CoS mark 5
  122. Found RTP audio format 0
  123. Found RTP audio format 8
  124. Found RTP audio format 3
  125. Found RTP audio format 11
  126. Found RTP audio format 98
  127. Found RTP audio format 97
  128. Found RTP audio format 102
  129. Found RTP audio format 103
  130. Found RTP audio format 104
  131. Found RTP audio format 105
  132. Found RTP audio format 106
  133. Found RTP audio format 101
  134. Found audio description format PCMU for ID 0
  135. Found audio description format PCMA for ID 8
  136. Found audio description format GSM for ID 3
  137. Found audio description format L16 for ID 11
  138. Found audio description format iLBC for ID 98
  139. Found audio description format iLBC for ID 97
  140. Found audio description format SPEEX for ID 102
  141. Found audio description format SPEEX for ID 103
  142. Found audio description format SPEEX for ID 104
  143. Found unknown media description format iSAC for ID 105
  144. Found unknown media description format iSAC for ID 106
  145. Found audio description format telephone-event for ID 101
  146. Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|gsm|alaw|slin|ilbc|h263p|speex|speex16|speex32)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
  147. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  148. Peer audio RTP is at port 10.128.69.4:20400
  149. Looking for 699 in from-internal (domain 192.168.2.253)
  150. sip_route_dump: route/path hop: <sip:802@10.128.69.4:5060>
  151.  
  152. <--- Transmitting (no NAT) to 10.128.69.4:5060 --->
  153. SIP/2.0 100 Trying
  154. Via: SIP/2.0/UDP 10.128.69.4:5060;branch=z9hG4bK536449654;received=10.128.69.4;rport=5060
  155. From: <sip:802@192.168.2.253>;tag=1103491194
  156. To: <sip:699@192.168.2.253>
  157. Call-ID: 749509218@192.168.2.253
  158. CSeq: 11 INVITE
  159. Server: FPBX-14.0.1.1(14.6.0)
  160. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  161. Supported: replaces, timer
  162. Contact: <sip:699@192.168.2.253:5060>
  163. Content-Length: 0
  164.  
  165.  
  166. <------------>
  167. -- Executing [699@from-internal:1] Goto("SIP/802-00000000", "app-pagegroups,699,1") in new stack
  168. -- Goto (app-pagegroups,699,1)
  169. -- Executing [699@app-pagegroups:1] Macro("SIP/802-00000000", "user-callerid,") in new stack
  170. -- Executing [s@macro-user-callerid:1] Set("SIP/802-00000000", "TOUCH_MONITOR=1502460070.0") in new stack
  171. -- Executing [s@macro-user-callerid:2] Set("SIP/802-00000000", "AMPUSER=802") in new stack
  172. -- Executing [s@macro-user-callerid:3] GotoIf("SIP/802-00000000", "0?report") in new stack
  173. -- Executing [s@macro-user-callerid:4] ExecIf("SIP/802-00000000", "1?Set(__REALCALLERIDNUM=802)") in new stack
  174. -- Executing [s@macro-user-callerid:5] Set("SIP/802-00000000", "AMPUSER=802") in new stack
  175. -- Executing [s@macro-user-callerid:6] GotoIf("SIP/802-00000000", "0?limit") in new stack
  176. -- Executing [s@macro-user-callerid:7] Set("SIP/802-00000000", "AMPUSERCIDNAME=Speaker2") in new stack
  177. -- Executing [s@macro-user-callerid:8] GotoIf("SIP/802-00000000", "0?report") in new stack
  178. -- Executing [s@macro-user-callerid:9] Set("SIP/802-00000000", "AMPUSERCID=802") in new stack
  179. -- Executing [s@macro-user-callerid:10] Set("SIP/802-00000000", "__DIAL_OPTIONS=Ttr") in new stack
  180. -- Executing [s@macro-user-callerid:11] Set("SIP/802-00000000", "CALLERID(all)="Speaker2" <802>") in new stack
  181. -- Executing [s@macro-user-callerid:12] GotoIf("SIP/802-00000000", "0?limit") in new stack
  182. -- Executing [s@macro-user-callerid:13] ExecIf("SIP/802-00000000", "0?Set(GROUP(concurrency_limit)=802)") in new stack
  183. -- Executing [s@macro-user-callerid:14] ExecIf("SIP/802-00000000", "0?Set(CHANNEL(language)=)") in new stack
  184. -- Executing [s@macro-user-callerid:15] GotoIf("SIP/802-00000000", "0?continue") in new stack
  185. -- Executing [s@macro-user-callerid:16] ExecIf("SIP/802-00000000", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
  186. -- Executing [s@macro-user-callerid:17] Set("SIP/802-00000000", "__TTL=6") in new stack
  187. -- Executing [s@macro-user-callerid:18] GotoIf("SIP/802-00000000", "1?continue") in new stack
  188. -- Goto (macro-user-callerid,s,29)
  189. -- Executing [s@macro-user-callerid:29] Set("SIP/802-00000000", "CALLERID(number)=802") in new stack
  190. -- Executing [s@macro-user-callerid:30] Set("SIP/802-00000000", "CALLERID(name)=Speaker2") in new stack
  191. -- Executing [s@macro-user-callerid:31] GotoIf("SIP/802-00000000", "0?cnum") in new stack
  192. -- Executing [s@macro-user-callerid:32] Set("SIP/802-00000000", "CDR(cnam)=Speaker2") in new stack
  193. -- Executing [s@macro-user-callerid:33] Set("SIP/802-00000000", "CDR(cnum)=802") in new stack
  194. -- Executing [s@macro-user-callerid:34] Set("SIP/802-00000000", "CHANNEL(language)=en") in new stack
  195. -- Executing [699@app-pagegroups:2] Set("SIP/802-00000000", "_PAGEGROUP=699") in new stack
  196. -- Executing [699@app-pagegroups:3] GotoIf("SIP/802-00000000", "1?:busy") in new stack
  197. -- Executing [699@app-pagegroups:4] Set("SIP/802-00000000", "DEVICE_STATE(Custom:PAGE699)=INUSE") in new stack
  198. -- Executing [699@app-pagegroups:5] Gosub("SIP/802-00000000", "app-paging,ssetup,1()") in new stack
  199. -- Executing [ssetup@app-paging:1] Set("SIP/802-00000000", "_SIPURI=") in new stack
  200. -- Executing [ssetup@app-paging:2] Set("SIP/802-00000000", "_ALERTINFO=Ring Answer") in new stack
  201. -- Executing [ssetup@app-paging:3] Set("SIP/802-00000000", "_CALLINFO=<uri>;answer-after=0") in new stack
  202. -- Executing [ssetup@app-paging:4] Set("SIP/802-00000000", "_SIPURI=intercom=true") in new stack
  203. -- Executing [ssetup@app-paging:5] Set("SIP/802-00000000", "_DTIME=5") in new stack
  204. -- Executing [ssetup@app-paging:6] Set("SIP/802-00000000", "_ANSWERMACRO=") in new stack
  205. -- Executing [ssetup@app-paging:7] Set("SIP/802-00000000", "PAGE_CONF=1502460070990") in new stack
  206. -- Executing [ssetup@app-paging:8] Return("SIP/802-00000000", "") in new stack
  207. -- Executing [699@app-pagegroups:6] Set("SIP/802-00000000", "PAGEMODE=FPAGE") in new stack
  208. -- Executing [699@app-pagegroups:7] Set("SIP/802-00000000", "PAGE_MEMBERS=801-802-803") in new stack
  209. -- Executing [699@app-pagegroups:8] Set("SIP/802-00000000", "PAGE_CONF_OPTS=") in new stack
  210. -- Executing [699@app-pagegroups:9] Set("SIP/802-00000000", "ANNOUNCEMENT=custom/Testing") in new stack
  211. -- Executing [699@app-pagegroups:10] AGI("SIP/802-00000000", "page.agi") in new stack
  212. -- Launched AGI Script /var/lib/asterisk/agi-bin/page.agi
  213. -- Called s@app-page-stream
  214. -- Executing [s@app-page-stream:1] Wait("Local/s@app-page-stream-00000000;2", "1") in new stack
  215. -- Called PAGE801@app-paging/n
  216. -- Executing [PAGE801@app-paging:1] Macro("Local/PAGE801@app-paging-00000001;2", "autoanswer,801") in new stack
  217. -- Executing [s@macro-autoanswer:1] GotoIf("Local/PAGE801@app-paging-00000001;2", "1?knowndial") in new stack
  218. -- Goto (macro-autoanswer,s,19)
  219. -- Executing [s@macro-autoanswer:19] Set("Local/PAGE801@app-paging-00000001;2", "DIAL=SIP/801") in new stack
  220. -- Executing [s@macro-autoanswer:20] ExecIf("Local/PAGE801@app-paging-00000001;2", "0?Set(DIAL=DAHDI/801)") in new stack
  221. -- Executing [s@macro-autoanswer:21] GotoIf("Local/PAGE801@app-paging-00000001;2", "0?macro") in new stack
  222. -- Executing [s@macro-autoanswer:22] GotoIf("Local/PAGE801@app-paging-00000001;2", "0?pjsipua") in new stack
  223. -- Executing [s@macro-autoanswer:23] Set("Local/PAGE801@app-paging-00000001;2", "USERAGENT=") in new stack
  224. -- Executing [s@macro-autoanswer:24] Goto("Local/PAGE801@app-paging-00000001;2", "uafin") in new stack
  225. -- Goto (macro-autoanswer,s,28)
  226. -- Executing [s@macro-autoanswer:28] ExecIf("Local/PAGE801@app-paging-00000001;2", "0?Set(USERAGENT=)") in new stack
  227. -- Executing [s@macro-autoanswer:29] ExecIf("Local/PAGE801@app-paging-00000001;2", "0?Set(PAGE_VOL=;volume=)") in new stack
  228. -- Executing [s@macro-autoanswer:30] ExecIf("Local/PAGE801@app-paging-00000001;2", "0?Set(ALERTINFO=ring-answer)") in new stack
  229. -- Executing [s@macro-autoanswer:31] ExecIf("Local/PAGE801@app-paging-00000001;2", "0?Set(CALLINFO=<sip:broadworks.net>;answer-after=0)") in new stack
  230. -- Executing [s@macro-autoanswer:32] ExecIf("Local/PAGE801@app-paging-00000001;2", "0?Set(ALERTINFO=<http://example.com>;info=alert-autoanswer)") in new stack
  231. -- Executing [s@macro-autoanswer:33] ExecIf("Local/PAGE801@app-paging-00000001;2", "0?Set(ALERTINFO=Intercom)") in new stack
  232. -- Executing [s@macro-autoanswer:34] ExecIf("Local/PAGE801@app-paging-00000001;2", "0?Set(ALERTINFO=Alert-Info: Auto Answer)") in new stack
  233. -- Executing [s@macro-autoanswer:35] ExecIf("Local/PAGE801@app-paging-00000001;2", "0?Set(ALERTINFO=<http://www.sangoma.com>;info=external)") in new stack
  234. -- Executing [s@macro-autoanswer:36] ExecIf("Local/PAGE801@app-paging-00000001;2", "1?Set(__SIP_URI_OPTIONS=intercom=true)") in new stack
  235. -- Executing [PAGE801@app-paging:2] NoOp("Local/PAGE801@app-paging-00000001;2", "") in new stack
  236. -- Executing [PAGE801@app-paging:3] GotoIf("Local/PAGE801@app-paging-00000001;2", "1?doptions") in new stack
  237. -- Goto (app-paging,PAGE801,6)
  238. -- Executing [PAGE801@app-paging:6] ExecIf("Local/PAGE801@app-paging-00000001;2", "1?Set(_DOPTIONS=b(autoanswer^s^1(Ring Answer,<uri>;answer-after=0)))") in new stack
  239. -- Executing [PAGE801@app-paging:7] Dial("Local/PAGE801@app-paging-00000001;2", "SIP/801,5,A(custom/Testing)b(autoanswer^s^1(Ring Answer,<uri>;answer-after=0))") in new stack
  240. [2017-08-11 14:01:10] WARNING[3646][C-00000003]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  241. == Everyone is busy/congested at this time (1:0/0/1)
  242. -- Executing [PAGE801@app-paging:8] Hangup("Local/PAGE801@app-paging-00000001;2", "") in new stack
  243. == Spawn extension (app-paging, PAGE801, 8) exited non-zero on 'Local/PAGE801@app-paging-00000001;2'
  244. -- <SIP/802-00000000>AGI Script page.agi completed, returning 0
  245. -- Executing [699@app-pagegroups:11] Set("SIP/802-00000000", "CONFBRIDGE(user,template)=page_user_duplex") in new stack
  246. -- Executing [699@app-pagegroups:12] Set("SIP/802-00000000", "CONFBRIDGE(user,admin)=yes") in new stack
  247. -- Executing [699@app-pagegroups:13] Set("SIP/802-00000000", "CONFBRIDGE(user,marked)=yes") in new stack
  248. -- Executing [699@app-pagegroups:14] Answer("SIP/802-00000000", "") in new stack
  249. Audio is at 18236
  250. Adding codec ulaw to SDP
  251. Adding codec alaw to SDP
  252. Adding codec gsm to SDP
  253. Adding non-codec 0x1 (telephone-event) to SDP
  254.  
  255. <--- Reliably Transmitting (no NAT) to 10.128.69.4:5060 --->
  256. SIP/2.0 200 OK
  257. Via: SIP/2.0/UDP 10.128.69.4:5060;branch=z9hG4bK536449654;received=10.128.69.4;rport=5060
  258. From: <sip:802@192.168.2.253>;tag=1103491194
  259. To: <sip:699@192.168.2.253>;tag=as4d00707f
  260. Call-ID: 749509218@192.168.2.253
  261. CSeq: 11 INVITE
  262. Server: FPBX-14.0.1.1(14.6.0)
  263. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  264. Supported: replaces, timer
  265. Contact: <sip:699@192.168.2.253:5060>
  266. Content-Type: application/sdp
  267. Content-Length: 298
  268.  
  269. v=0
  270. o=root 691690305 691690305 IN IP4 192.168.2.253
  271. s=Asterisk PBX 14.6.0
  272. c=IN IP4 192.168.2.253
  273. t=0 0
  274. m=audio 18236 RTP/AVP 0 8 3 101
  275. a=rtpmap:0 PCMU/8000
  276. a=rtpmap:8 PCMA/8000
  277. a=rtpmap:3 GSM/8000
  278. a=rtpmap:101 telephone-event/8000
  279. a=fmtp:101 0-16
  280. a=ptime:20
  281. a=maxptime:150
  282. a=sendrecv
  283.  
  284. <------------>
  285. Really destroying SIP dialog '424661d40d6c7c9e76c57ca417817448@127.0.0.1:5060' Method: INVITE
  286. -- Called s@app-page-stream
  287. -- Executing [s@app-page-stream:1] Wait("Local/s@app-page-stream-00000003;2", "1") in new stack
  288. -- Called PAGE803@app-paging/n
  289. -- Executing [PAGE803@app-paging:1] Macro("Local/PAGE803@app-paging-00000002;2", "autoanswer,803") in new stack
  290. -- Executing [s@macro-autoanswer:1] GotoIf("Local/PAGE803@app-paging-00000002;2", "1?knowndial") in new stack
  291. -- Goto (macro-autoanswer,s,19)
  292. -- Executing [s@macro-autoanswer:19] Set("Local/PAGE803@app-paging-00000002;2", "DIAL=SIP/803") in new stack
  293. -- Executing [s@macro-autoanswer:20] ExecIf("Local/PAGE803@app-paging-00000002;2", "0?Set(DIAL=DAHDI/803)") in new stack
  294. -- Executing [s@macro-autoanswer:21] GotoIf("Local/PAGE803@app-paging-00000002;2", "0?macro") in new stack
  295. -- Executing [s@macro-autoanswer:22] GotoIf("Local/PAGE803@app-paging-00000002;2", "0?pjsipua") in new stack
  296. -- Executing [s@macro-autoanswer:23] Set("Local/PAGE803@app-paging-00000002;2", "USERAGENT=YATE/5.4.0") in new stack
  297. -- Executing [s@macro-autoanswer:24] Goto("Local/PAGE803@app-paging-00000002;2", "uafin") in new stack
  298. -- Goto (macro-autoanswer,s,28)
  299. -- Executing [s@macro-autoanswer:28] ExecIf("Local/PAGE803@app-paging-00000002;2", "0?Set(USERAGENT=)") in new stack
  300. -- Executing [s@macro-autoanswer:29] ExecIf("Local/PAGE803@app-paging-00000002;2", "0?Set(PAGE_VOL=;volume=)") in new stack
  301. -- Executing [s@macro-autoanswer:30] ExecIf("Local/PAGE803@app-paging-00000002;2", "0?Set(ALERTINFO=ring-answer)") in new stack
  302. -- Executing [s@macro-autoanswer:31] ExecIf("Local/PAGE803@app-paging-00000002;2", "0?Set(CALLINFO=<sip:broadworks.net>;answer-after=0)") in new stack
  303. -- Executing [s@macro-autoanswer:32] ExecIf("Local/PAGE803@app-paging-00000002;2", "0?Set(ALERTINFO=<http://example.com>;info=alert-autoanswer)") in new stack
  304. -- Executing [s@macro-autoanswer:33] ExecIf("Local/PAGE803@app-paging-00000002;2", "0?Set(ALERTINFO=Intercom)") in new stack
  305. -- Executing [s@macro-autoanswer:34] ExecIf("Local/PAGE803@app-paging-00000002;2", "0?Set(ALERTINFO=Alert-Info: Auto Answer)") in new stack
  306. -- Executing [s@macro-autoanswer:35] ExecIf("Local/PAGE803@app-paging-00000002;2", "0?Set(ALERTINFO=<http://www.sangoma.com>;info=external)") in new stack
  307. -- Executing [s@macro-autoanswer:36] ExecIf("Local/PAGE803@app-paging-00000002;2", "1?Set(__SIP_URI_OPTIONS=intercom=true)") in new stack
  308.  
  309. <--- SIP read from UDP:10.128.69.4:5060 --->
  310. ACK sip:699@192.168.2.253:5060 SIP/2.0
  311. Via: SIP/2.0/UDP 10.128.69.4:5060;rport;branch=z9hG4bK320123982
  312. From: <sip:802@192.168.2.253>;tag=1103491194
  313. To: <sip:699@192.168.2.253>;tag=as4d00707f
  314. Call-ID: 749509218@192.168.2.253
  315. CSeq: 11 ACK
  316. Max-Forwards: 20
  317. Contact: <sip:802@10.128.69.4:5060>
  318. Authorization: Digest username="802", realm="asterisk", nonce="25b3f291", uri="sip:699@192.168.2.253", response="9c6caffa8827cd9cc4692626cbccca91", algorithm=MD5
  319. User-Agent: YATE/5.5.0
  320. Content-Length: 0
  321.  
  322. <------------->
  323. --- (11 headers 0 lines) ---
  324. -- Executing [PAGE803@app-paging:2] NoOp("Local/PAGE803@app-paging-00000002;2", "") in new stack
  325. -- Executing [PAGE803@app-paging:3] GotoIf("Local/PAGE803@app-paging-00000002;2", "1?doptions") in new stack
  326. -- Goto (app-paging,PAGE803,6)
  327. -- Executing [PAGE803@app-paging:6] ExecIf("Local/PAGE803@app-paging-00000002;2", "1?Set(_DOPTIONS=b(autoanswer^s^1(Ring Answer,<uri>;answer-after=0)))") in new stack
  328. -- Executing [PAGE803@app-paging:7] Dial("Local/PAGE803@app-paging-00000002;2", "SIP/803,5,A(custom/Testing)b(autoanswer^s^1(Ring Answer,<uri>;answer-after=0))") in new stack
  329. == Using SIP RTP TOS bits 184
  330. == Using SIP RTP CoS mark 5
  331. -- SIP/803-00000001 Internal Gosub(autoanswer,s,1(Ring Answer,<uri>;answer-after=0)) start
  332. -- Executing [s@autoanswer:1] GosubIf("SIP/803-00000001", "1?func-set-sipheader,s,1(Alert-Info,Ring Answer)") in new stack
  333. -- Executing [s@func-set-sipheader:1] NoOp("SIP/803-00000001", "Sip Add Header function called. Adding Alert-Info = Ring Answer") in new stack
  334. -- Executing [s@func-set-sipheader:2] Set("SIP/803-00000001", "HASH(_SIPHEADERS,Alert-Info)=Ring Answer") in new stack
  335. -- Executing [s@func-set-sipheader:3] Return("SIP/803-00000001", "") in new stack
  336. -- Executing [s@autoanswer:2] GosubIf("SIP/803-00000001", "1?func-set-sipheader,s,1(Call-Info,<uri>;answer-after=0)") in new stack
  337. -- Executing [s@func-set-sipheader:1] NoOp("SIP/803-00000001", "Sip Add Header function called. Adding Call-Info = <uri>;answer-after=0") in new stack
  338. -- Executing [s@func-set-sipheader:2] Set("SIP/803-00000001", "HASH(_SIPHEADERS,Call-Info)=<uri>;answer-after=0") in new stack
  339. -- Executing [s@func-set-sipheader:3] Return("SIP/803-00000001", "") in new stack
  340. -- Executing [s@autoanswer:3] Gosub("SIP/803-00000001", "func-apply-sipheaders,s,1()") in new stack
  341. -- Executing [s@func-apply-sipheaders:1] ExecIf("SIP/803-00000001", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
  342. -- Executing [s@func-apply-sipheaders:2] NoOp("SIP/803-00000001", "Applying SIP Headers to channel") in new stack
  343. -- Executing [s@func-apply-sipheaders:3] Set("SIP/803-00000001", "SIPHEADERKEYS=Call-Info,Alert-Info") in new stack
  344. -- Executing [s@func-apply-sipheaders:4] While("SIP/803-00000001", "1") in new stack
  345. -- Executing [s@func-apply-sipheaders:5] Set("SIP/803-00000001", "sipheader=<uri>;answer-after=0") in new stack
  346. -- Executing [s@func-apply-sipheaders:6] SIPAddHeader("SIP/803-00000001", "Call-Info: <uri>;answer-after=0") in new stack
  347. -- Executing [s@func-apply-sipheaders:7] EndWhile("SIP/803-00000001", "") in new stack
  348. -- Executing [s@func-apply-sipheaders:4] While("SIP/803-00000001", "1") in new stack
  349. -- Executing [s@func-apply-sipheaders:5] Set("SIP/803-00000001", "sipheader=Ring Answer") in new stack
  350. -- Executing [s@func-apply-sipheaders:6] SIPAddHeader("SIP/803-00000001", "Alert-Info: Ring Answer") in new stack
  351. -- Executing [s@func-apply-sipheaders:7] EndWhile("SIP/803-00000001", "") in new stack
  352. -- Executing [s@func-apply-sipheaders:4] While("SIP/803-00000001", "0") in new stack
  353. -- Executing [s@func-apply-sipheaders:8] Return("SIP/803-00000001", "") in new stack
  354. -- Executing [s@autoanswer:4] Return("SIP/803-00000001", "") in new stack
  355. == Spawn extension (from-internal, PAGE803, 1) exited non-zero on 'SIP/803-00000001'
  356. -- SIP/803-00000001 Internal Gosub(autoanswer,s,1(Ring Answer,<uri>;answer-after=0)) complete GOSUB_RETVAL=
  357. Audio is at 17078
  358. Adding codec ulaw to SDP
  359. Adding codec alaw to SDP
  360. Adding codec gsm to SDP
  361. Adding codec g726 to SDP
  362. Adding codec g722 to SDP
  363. Adding non-codec 0x1 (telephone-event) to SDP
  364. Reliably Transmitting (no NAT) to 10.128.69.2:5060:
  365. INVITE sip:803@10.128.69.2:5060;intercom=true SIP/2.0
  366. Via: SIP/2.0/UDP 192.168.2.253:5060;branch=z9hG4bK73fcaede
  367. Max-Forwards: 70
  368. From: "Speaker2" <sip:802@192.168.2.253>;tag=as3de7edbe
  369. To: <sip:803@10.128.69.2:5060;intercom=true>
  370. Contact: <sip:802@192.168.2.253:5060>
  371. Call-ID: 576869272052e04f6aa3dd9079fcf696@192.168.2.253:5060
  372. CSeq: 102 INVITE
  373. User-Agent: FPBX-14.0.1.1(14.6.0)
  374. Date: Fri, 11 Aug 2017 14:01:10 GMT
  375. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  376. Supported: replaces, timer
  377. Alert-Info: Ring Answer
  378. Call-Info: <uri>;answer-after=0
  379. P-Asserted-Identity: "Speaker2" <sip:802@192.168.2.253>
  380. Content-Type: application/sdp
  381. Content-Length: 353
  382.  
  383. v=0
  384. o=root 933473789 933473789 IN IP4 192.168.2.253
  385. s=Asterisk PBX 14.6.0
  386. c=IN IP4 192.168.2.253
  387. t=0 0
  388. m=audio 17078 RTP/AVP 0 8 3 111 9 101
  389. a=rtpmap:0 PCMU/8000
  390. a=rtpmap:8 PCMA/8000
  391. a=rtpmap:3 GSM/8000
  392. a=rtpmap:111 G726-32/8000
  393. a=rtpmap:9 G722/8000
  394. a=rtpmap:101 telephone-event/8000
  395. a=fmtp:101 0-16
  396. a=ptime:20
  397. a=maxptime:150
  398. a=sendrecv
  399.  
  400. ---
  401. -- Called SIP/803
  402.  
  403. <--- SIP read from UDP:10.128.69.2:5060 --->
  404. SIP/2.0 100 Trying
  405. Via: SIP/2.0/UDP 192.168.2.253:5060;branch=z9hG4bK73fcaede;received=192.168.2.253
  406. From: "Speaker2" <sip:802@192.168.2.253>;tag=as3de7edbe
  407. To: <sip:803@10.128.69.2:5060;intercom=true>
  408. Call-ID: 576869272052e04f6aa3dd9079fcf696@192.168.2.253:5060
  409. CSeq: 102 INVITE
  410. Server: YATE/5.4.0
  411. Content-Length: 0
  412.  
  413. <------------->
  414. --- (8 headers 0 lines) ---
  415.  
  416. <--- SIP read from UDP:10.128.69.2:5060 --->
  417. SIP/2.0 200 OK
  418. Via: SIP/2.0/UDP 192.168.2.253:5060;branch=z9hG4bK73fcaede;received=192.168.2.253
  419. From: "Speaker2" <sip:802@192.168.2.253>;tag=as3de7edbe
  420. To: <sip:803@10.128.69.2:5060;intercom=true>;tag=1132674989
  421. Call-ID: 576869272052e04f6aa3dd9079fcf696@192.168.2.253:5060
  422. CSeq: 102 INVITE
  423. Server: YATE/5.4.0
  424. Contact: <sip:803@10.128.69.2:5060>
  425. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO
  426. Content-Type: application/sdp
  427. Content-Length: 181
  428.  
  429. v=0
  430. o=yate 1502460070 1502460070 IN IP4 10.128.69.2
  431. s=SIP Call
  432. c=IN IP4 10.128.69.2
  433. t=0 0
  434. m=audio 28320 RTP/AVP 0 101
  435. a=rtpmap:0 PCMU/8000
  436. a=rtpmap:101 telephone-event/8000
  437. <------------->
  438. --- (11 headers 8 lines) ---
  439. Found RTP audio format 0
  440. Found RTP audio format 101
  441. Found audio description format PCMU for ID 0
  442. Found audio description format telephone-event for ID 101
  443. Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
  444. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  445. Peer audio RTP is at port 10.128.69.2:28320
  446. sip_route_dump: route/path hop: <sip:803@10.128.69.2:5060>
  447. set_destination: Parsing <sip:803@10.128.69.2:5060> for address/port to send to
  448. set_destination: set destination to 10.128.69.2:5060
  449. Transmitting (no NAT) to 10.128.69.2:5060:
  450. ACK sip:803@10.128.69.2:5060 SIP/2.0
  451. Via: SIP/2.0/UDP 192.168.2.253:5060;branch=z9hG4bK661648ae
  452. Max-Forwards: 70
  453. From: "Speaker2" <sip:802@192.168.2.253>;tag=as3de7edbe
  454. To: <sip:803@10.128.69.2:5060;intercom=true>;tag=1132674989
  455. Contact: <sip:802@192.168.2.253:5060>
  456. Call-ID: 576869272052e04f6aa3dd9079fcf696@192.168.2.253:5060
  457. CSeq: 102 ACK
  458. User-Agent: FPBX-14.0.1.1(14.6.0)
  459. Content-Length: 0
  460.  
  461.  
  462. ---
  463. -- SIP/803-00000001 answered Local/PAGE803@app-paging-00000002;2
  464. -- <SIP/803-00000001> Playing 'custom/Testing.ulaw' (language 'en')
  465. > 0x7f224002e090 -- Probation passed - setting RTP source address to 10.128.69.2:28320
  466. -- Executing [699@app-pagegroups:15] ConfBridge("SIP/802-00000000", "1502460070990,,,admin_menu") in new stack
  467. -- Channel SIP/802-00000000 joined 'softmix' base-bridge <a7ff7568-439b-4e6b-a48a-c72d5cfc8c4b>
  468. -- Channel CBAnn/1502460070990-00000004;2 joined 'softmix' base-bridge <a7ff7568-439b-4e6b-a48a-c72d5cfc8c4b>
  469. -- Executing [s@app-page-stream:2] Answer("Local/s@app-page-stream-00000000;2", "") in new stack
  470. -- Local/s@app-page-stream-00000000;1 answered
  471. > Launching Wait(5) on Local/s@app-page-stream-00000000;1
  472. -- Executing [s@app-page-stream:2] Answer("Local/s@app-page-stream-00000003;2", "") in new stack
  473. -- Local/s@app-page-stream-00000003;1 answered
  474. > Launching Playback(beep) on Local/s@app-page-stream-00000003;1
  475. -- Executing [s@app-page-stream:3] Set("Local/s@app-page-stream-00000000;2", "CONFBRIDGE(user,template)=page_user_duplex") in new stack
  476. -- Executing [s@app-page-stream:4] Set("Local/s@app-page-stream-00000000;2", "CONFBRIDGE(user,marked)=yes") in new stack
  477. -- Executing [s@app-page-stream:5] ConfBridge("Local/s@app-page-stream-00000000;2", "1502460070990,,,") in new stack
  478. -- Channel Local/s@app-page-stream-00000000;2 joined 'softmix' base-bridge <a7ff7568-439b-4e6b-a48a-c72d5cfc8c4b>
  479. -- <Local/s@app-page-stream-00000003;1> Playing 'beep.slin16' (language 'en')
  480. -- Executing [s@app-page-stream:3] Set("Local/s@app-page-stream-00000003;2", "CONFBRIDGE(user,template)=page_user_duplex") in new stack
  481. -- Executing [s@app-page-stream:4] Set("Local/s@app-page-stream-00000003;2", "CONFBRIDGE(user,marked)=yes") in new stack
  482. -- Executing [s@app-page-stream:5] ConfBridge("Local/s@app-page-stream-00000003;2", "1502460070990,,,") in new stack
  483. -- Channel Local/s@app-page-stream-00000003;2 joined 'softmix' base-bridge <a7ff7568-439b-4e6b-a48a-c72d5cfc8c4b>
  484. -- Channel Local/s@app-page-stream-00000003;2 left 'softmix' base-bridge <a7ff7568-439b-4e6b-a48a-c72d5cfc8c4b>
  485.  
  486. <--- SIP read from UDP:10.128.69.2:5060 --->
  487.  
  488. <------------->
  489. -- Channel Local/s@app-page-stream-00000000;2 left 'softmix' base-bridge <a7ff7568-439b-4e6b-a48a-c72d5cfc8c4b>
  490.  
  491. <--- SIP read from UDP:10.128.69.4:5060 --->
  492.  
  493. <------------->
  494. Really destroying SIP dialog '157716510@192.168.2.253' Method: REGISTER
  495. Reliably Transmitting (no NAT) to 10.128.69.4:5060:
  496. OPTIONS sip:802@10.128.69.4:5060 SIP/2.0
  497. Via: SIP/2.0/UDP 192.168.2.253:5060;branch=z9hG4bK5c89b217
  498. Max-Forwards: 70
  499. From: "Unknown" <sip:Unknown@192.168.2.253>;tag=as0742e32f
  500. To: <sip:802@10.128.69.4:5060>
  501. Contact: <sip:Unknown@192.168.2.253:5060>
  502. Call-ID: 0c603ead1ec8536b61fad4827eaf65c7@192.168.2.253:5060
  503. CSeq: 102 OPTIONS
  504. User-Agent: FPBX-14.0.1.1(14.6.0)
  505. Date: Fri, 11 Aug 2017 14:01:25 GMT
  506. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  507. Supported: replaces, timer
  508. Content-Length: 0
  509.  
  510.  
  511. ---
  512.  
  513. <--- SIP read from UDP:10.128.69.4:5060 --->
  514. SIP/2.0 100 Trying
  515. Via: SIP/2.0/UDP 192.168.2.253:5060;branch=z9hG4bK5c89b217;received=192.168.2.253
  516. From: "Unknown" <sip:Unknown@192.168.2.253>;tag=as0742e32f
  517. To: <sip:802@10.128.69.4:5060>
  518. Call-ID: 0c603ead1ec8536b61fad4827eaf65c7@192.168.2.253:5060
  519. CSeq: 102 OPTIONS
  520. Server: YATE/5.5.0
  521. Content-Length: 0
  522.  
  523. <------------->
  524. --- (8 headers 0 lines) ---
  525.  
  526. <--- SIP read from UDP:10.128.69.4:5060 --->
  527. SIP/2.0 200 OK
  528. Via: SIP/2.0/UDP 192.168.2.253:5060;branch=z9hG4bK5c89b217;received=192.168.2.253
  529. From: "Unknown" <sip:Unknown@192.168.2.253>;tag=as0742e32f
  530. To: <sip:802@10.128.69.4:5060>;tag=1487019880
  531. Call-ID: 0c603ead1ec8536b61fad4827eaf65c7@192.168.2.253:5060
  532. CSeq: 102 OPTIONS
  533. Server: YATE/5.5.0
  534. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO
  535. Content-Length: 0
  536.  
  537. <------------->
  538. --- (9 headers 0 lines) ---
  539. Really destroying SIP dialog '0c603ead1ec8536b61fad4827eaf65c7@192.168.2.253:5060' Method: OPTIONS
  540.  
  541. <--- SIP read from UDP:10.128.69.2:5060 --->
  542.  
  543. <------------->
  544. Scheduling destruction of SIP dialog '576869272052e04f6aa3dd9079fcf696@192.168.2.253:5060' in 6400 ms (Method: INVITE)
  545. set_destination: Parsing <sip:803@10.128.69.2:5060> for address/port to send to
  546. set_destination: set destination to 10.128.69.2:5060
  547. Reliably Transmitting (no NAT) to 10.128.69.2:5060:
  548. BYE sip:803@10.128.69.2:5060 SIP/2.0
  549. Via: SIP/2.0/UDP 192.168.2.253:5060;branch=z9hG4bK5a509721
  550. Max-Forwards: 70
  551. From: "Speaker2" <sip:802@192.168.2.253>;tag=as3de7edbe
  552. To: <sip:803@10.128.69.2:5060;intercom=true>;tag=1132674989
  553. Call-ID: 576869272052e04f6aa3dd9079fcf696@192.168.2.253:5060
  554. CSeq: 103 BYE
  555. User-Agent: FPBX-14.0.1.1(14.6.0)
  556. X-Asterisk-HangupCause: Normal Clearing
  557. X-Asterisk-HangupCauseCode: 16
  558. Content-Length: 0
  559.  
  560.  
  561. ---
  562. == Spawn extension (app-paging, PAGE803, 7) exited non-zero on 'Local/PAGE803@app-paging-00000002;2'
  563.  
  564. <--- SIP read from UDP:10.128.69.2:5060 --->
  565. SIP/2.0 100 Trying
  566. Via: SIP/2.0/UDP 192.168.2.253:5060;branch=z9hG4bK5a509721;received=192.168.2.253
  567. From: "Speaker2" <sip:802@192.168.2.253>;tag=as3de7edbe
  568. To: <sip:803@10.128.69.2:5060;intercom=true>;tag=1132674989
  569. Call-ID: 576869272052e04f6aa3dd9079fcf696@192.168.2.253:5060
  570. CSeq: 103 BYE
  571. Server: YATE/5.4.0
  572. Content-Length: 0
  573.  
  574. <------------->
  575. --- (8 headers 0 lines) ---
  576.  
  577. <--- SIP read from UDP:10.128.69.2:5060 --->
  578. SIP/2.0 200 OK
  579. Via: SIP/2.0/UDP 192.168.2.253:5060;branch=z9hG4bK5a509721;received=192.168.2.253
  580. From: "Speaker2" <sip:802@192.168.2.253>;tag=as3de7edbe
  581. To: <sip:803@10.128.69.2:5060;intercom=true>;tag=1132674989
  582. Call-ID: 576869272052e04f6aa3dd9079fcf696@192.168.2.253:5060
  583. CSeq: 103 BYE
  584. P-RTP-Stat: PS=1499,OS=239840,PR=1497,OR=239520,PL=0
  585. Server: YATE/5.4.0
  586. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO
  587. Content-Length: 0
  588.  
  589. <------------->
  590. --- (10 headers 0 lines) ---
  591. Really destroying SIP dialog '576869272052e04f6aa3dd9079fcf696@192.168.2.253:5060' Method: INVITE
  592.  
  593. <--- SIP read from UDP:10.128.69.4:5060 --->
  594.  
  595. <------------->
  596. Reliably Transmitting (no NAT) to 10.128.69.2:5060:
  597. OPTIONS sip:803@10.128.69.2:5060 SIP/2.0
  598. Via: SIP/2.0/UDP 192.168.2.253:5060;branch=z9hG4bK1a6aa7a0
  599. Max-Forwards: 70
  600. From: "Unknown" <sip:Unknown@192.168.2.253>;tag=as638fa553
  601. To: <sip:803@10.128.69.2:5060>
  602. Contact: <sip:Unknown@192.168.2.253:5060>
  603. Call-ID: 0bfcc4d51b1e45464bbdc4c21fc98912@192.168.2.253:5060
  604. CSeq: 102 OPTIONS
  605. User-Agent: FPBX-14.0.1.1(14.6.0)
  606. Date: Fri, 11 Aug 2017 14:01:47 GMT
  607. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  608. Supported: replaces, timer
  609. Content-Length: 0
  610.  
  611.  
  612. ---
  613.  
  614. <--- SIP read from UDP:10.128.69.2:5060 --->
  615. SIP/2.0 100 Trying
  616. Via: SIP/2.0/UDP 192.168.2.253:5060;branch=z9hG4bK1a6aa7a0;received=192.168.2.253
  617. From: "Unknown" <sip:Unknown@192.168.2.253>;tag=as638fa553
  618. To: <sip:803@10.128.69.2:5060>
  619. Call-ID: 0bfcc4d51b1e45464bbdc4c21fc98912@192.168.2.253:5060
  620. CSeq: 102 OPTIONS
  621. Server: YATE/5.4.0
  622. Content-Length: 0
  623.  
  624. <------------->
  625. --- (8 headers 0 lines) ---
  626.  
  627. <--- SIP read from UDP:10.128.69.2:5060 --->
  628. SIP/2.0 200 OK
  629. Via: SIP/2.0/UDP 192.168.2.253:5060;branch=z9hG4bK1a6aa7a0;received=192.168.2.253
  630. From: "Unknown" <sip:Unknown@192.168.2.253>;tag=as638fa553
  631. To: <sip:803@10.128.69.2:5060>;tag=84788202
  632. Call-ID: 0bfcc4d51b1e45464bbdc4c21fc98912@192.168.2.253:5060
  633. CSeq: 102 OPTIONS
  634. Server: YATE/5.4.0
  635. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO
  636. Content-Length: 0
  637.  
  638. <------------->
  639. --- (9 headers 0 lines) ---
  640. Really destroying SIP dialog '0bfcc4d51b1e45464bbdc4c21fc98912@192.168.2.253:5060' Method: OPTIONS
  641. voip1*CLI> sip set debug off
  642. SIP Debugging Disabled
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