debug

From henry, 9 Months ago, written in Plain Text, viewed 52 times.
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  1. a=fmtp:101 0-16
  2. a=ptime:20
  3. a=maxptime:150
  4. a=sendrecv
  5.  
  6. ---
  7.  
  8. <--- SIP read from UDP:46.31.231.185:5060 --->
  9. SIP/2.0 100 Trying
  10. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK0e45325a;received=85.92.195.146;rport=5160
  11. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as2c206cd1
  12. To: <sip:07927800653@sip.voipfone.net>;tag=VFf0d744a2583f56f8497e206196d7
  13. Call-ID: 50284148397024fa345a86f94131b8e2@85.92.195.146:5160
  14. CSeq: 103 INVITE
  15. User-Agent: Voipfone Sip Network
  16. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  17. Contact: <sip:07927800653@46.31.231.185>
  18. Content-Length: 0
  19.  
  20. <------------->
  21. --- (10 headers 0 lines) ---
  22.  
  23. <--- SIP read from UDP:46.31.231.185:5060 --->
  24. SIP/2.0 183 Session Progress
  25. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK0e45325a;received=85.92.195.146;rport=5160
  26. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as2c206cd1
  27. To: <sip:07927800653@sip.voipfone.net>;tag=VFf0d744a2583f56f8497e206196d7
  28. Call-ID: 50284148397024fa345a86f94131b8e2@85.92.195.146:5160
  29. CSeq: 103 INVITE
  30. User-Agent: Voipfone Sip Network
  31. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  32. Contact: <sip:07927800653@46.31.231.185>
  33. Content-Type: application/sdp
  34. Content-Length: 333
  35.  
  36. v=0
  37. o=root 2431 2431 IN IP4 46.31.231.185
  38. s=session
  39. c=IN IP4 46.31.231.185
  40. t=0 0
  41. m=audio 49012 RTP/AVP 8 2 97 3 110 101
  42. a=sendrecv
  43. a=rtpmap:8 PCMA/8000
  44. a=rtpmap:2 G726-32/8000
  45. a=rtpmap:97 iLBC/8000
  46. a=rtpmap:3 GSM/8000
  47. a=rtpmap:110 speex/8000
  48. a=rtpmap:101 telephone-event/8000
  49. a=fmtp:101 0-16
  50. a=silenceSupp:off - - - -
  51. <------------->
  52. --- (11 headers 15 lines) ---
  53. sip_route_dump: route/path hop: <sip:07927800653@46.31.231.185>
  54. Found RTP audio format 8
  55. Found RTP audio format 2
  56. Found RTP audio format 97
  57. Found RTP audio format 3
  58. Found RTP audio format 110
  59. Found RTP audio format 101
  60. Found audio description format PCMA for ID 8
  61. Found audio description format G726-32 for ID 2
  62. Found audio description format iLBC for ID 97
  63. Found audio description format GSM for ID 3
  64. Found audio description format speex for ID 110
  65. Found audio description format telephone-event for ID 101
  66. Capabilities: us - (alaw), peer - audio=(g726|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (alaw)
  67. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  68. Peer audio RTP is at port 46.31.231.185:49012
  69.     -- SIP/voip-00000007 is making progress passing it to SIP/3006-00000006
  70. Audio is at 11020
  71. Adding codec ulaw to SDP
  72. Adding codec alaw to SDP
  73. Adding codec gsm to SDP
  74. Adding codec g722 to SDP
  75. Adding codec speex to SDP
  76. Adding non-codec 0x1 (telephone-event) to SDP
  77.  
  78. <--- Transmitting (no NAT) to 82.147.28.150:61123 --->
  79. SIP/2.0 183 Session Progress
  80. Via: SIP/2.0/UDP 172.16.103.99:61123;branch=z9hG4bKPjnZR-V4c0SpGQNzCyr.VD9qHUfb5qsFgF;received=82.147.28.150;rport=61123
  81. From: "Henry" <sip:3006@85.92.195.146>;tag=cJTKMC5V8THEhrEsK-sd7RtKTUWc1iUa
  82. To: <sip:07927800653@85.92.195.146>;tag=as4c17d22d
  83. Call-ID: j0VSztO5Z0Jq7nkvbV37M8f15topYrG-
  84. CSeq: 12598 INVITE
  85. Server: FPBX-14.0.1.1(13.16.0)
  86. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  87. Supported: replaces, timer
  88. Contact: <sip:07927800653@85.92.195.146:5160>
  89. Content-Type: application/sdp
  90. Content-Length: 353
  91.  
  92. v=0
  93. o=root 1328454054 1328454054 IN IP4 85.92.195.146
  94. s=Asterisk PBX 13.16.0
  95. c=IN IP4 85.92.195.146
  96. t=0 0
  97. m=audio 11020 RTP/AVP 0 8 3 9 102 101
  98. a=rtpmap:0 PCMU/8000
  99. a=rtpmap:8 PCMA/8000
  100. a=rtpmap:3 GSM/8000
  101. a=rtpmap:9 G722/8000
  102. a=rtpmap:102 speex/8000
  103. a=rtpmap:101 telephone-event/8000
  104. a=fmtp:101 0-16
  105. a=ptime:20
  106. a=maxptime:60
  107. a=sendrecv
  108.  
  109. <------------>
  110.        > 0x7fbf5004c9a0 -- Probation passed - setting RTP source address to 46.31.231.185:49012
  111.        > 0x7fbf4400f830 -- Probation passed - setting RTP source address to 82.147.28.150:4026
  112.  
  113. <--- SIP read from UDP:82.147.28.150:61123 --->
  114. CANCEL sip:07927800653@85.92.195.146:5160 SIP/2.0
  115. Via: SIP/2.0/UDP 172.16.103.99:61123;rport;branch=z9hG4bKPjnZR-V4c0SpGQNzCyr.VD9qHUfb5qsFgF
  116. Max-Forwards: 70
  117. From: "Henry" <sip:3006@85.92.195.146>;tag=cJTKMC5V8THEhrEsK-sd7RtKTUWc1iUa
  118. To: <sip:07927800653@85.92.195.146>
  119. Call-ID: j0VSztO5Z0Jq7nkvbV37M8f15topYrG-
  120. CSeq: 12598 CANCEL
  121. User-Agent: Telephone 1.2.6
  122. Content-Length: 0
  123.  
  124. <------------->
  125. --- (9 headers 0 lines) ---
  126. Sending to 82.147.28.150:61123 (no NAT)
  127.  
  128. <--- Reliably Transmitting (no NAT) to 82.147.28.150:61123 --->
  129. SIP/2.0 487 Request Terminated
  130. Via: SIP/2.0/UDP 172.16.103.99:61123;branch=z9hG4bKPjnZR-V4c0SpGQNzCyr.VD9qHUfb5qsFgF;received=82.147.28.150;rport=61123
  131. From: "Henry" <sip:3006@85.92.195.146>;tag=cJTKMC5V8THEhrEsK-sd7RtKTUWc1iUa
  132. To: <sip:07927800653@85.92.195.146>;tag=as4c17d22d
  133. Call-ID: j0VSztO5Z0Jq7nkvbV37M8f15topYrG-
  134. CSeq: 12598 INVITE
  135. Server: FPBX-14.0.1.1(13.16.0)
  136. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  137. Supported: replaces, timer
  138. Content-Length: 0
  139.  
  140.  
  141. <------------>
  142.  
  143. <--- Transmitting (no NAT) to 82.147.28.150:61123 --->
  144. SIP/2.0 200 OK
  145. Via: SIP/2.0/UDP 172.16.103.99:61123;branch=z9hG4bKPjnZR-V4c0SpGQNzCyr.VD9qHUfb5qsFgF;received=82.147.28.150;rport=61123
  146. From: "Henry" <sip:3006@85.92.195.146>;tag=cJTKMC5V8THEhrEsK-sd7RtKTUWc1iUa
  147. To: <sip:07927800653@85.92.195.146>;tag=as4c17d22d
  148. Call-ID: j0VSztO5Z0Jq7nkvbV37M8f15topYrG-
  149. CSeq: 12598 CANCEL
  150. Server: FPBX-14.0.1.1(13.16.0)
  151. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  152. Supported: replaces, timer
  153. Content-Length: 0
  154.  
  155.  
  156. <------------>
  157. Scheduling destruction of SIP dialog '50284148397024fa345a86f94131b8e2@85.92.195.146:5160' in 6400 ms (Method: INVITE)
  158. Reliably Transmitting (no NAT) to 46.31.231.185:5060:
  159. CANCEL sip:07927800653@sip.voipfone.net SIP/2.0
  160. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK0e45325a
  161. Max-Forwards: 70
  162. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as2c206cd1
  163. To: <sip:07927800653@sip.voipfone.net>
  164. Call-ID: 50284148397024fa345a86f94131b8e2@85.92.195.146:5160
  165. CSeq: 103 CANCEL
  166. User-Agent: FPBX-14.0.1.1(13.16.0)
  167. Content-Length: 0
  168.  
  169.  
  170. ---
  171. Scheduling destruction of SIP dialog '50284148397024fa345a86f94131b8e2@85.92.195.146:5160' in 6400 ms (Method: INVITE)
  172.   == Spawn extension (macro-dialout-trunk, s, 30) exited non-zero on 'SIP/3006-00000006' in macro 'dialout-trunk'
  173.   == Spawn extension (from-internal, 07927800653, 6) exited non-zero on 'SIP/3006-00000006'
  174.     -- Executing [h@from-internal:1] Macro("SIP/3006-00000006", "hangupcall") in new stack
  175.     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/3006-00000006", "1?theend") in new stack
  176.     -- Goto (macro-hangupcall,s,3)
  177.  
  178. <--- SIP read from UDP:82.147.28.150:61123 --->
  179. ACK sip:07927800653@85.92.195.146:5160 SIP/2.0
  180. Via: SIP/2.0/UDP 172.16.103.99:61123;rport;branch=z9hG4bKPjnZR-V4c0SpGQNzCyr.VD9qHUfb5qsFgF
  181. Max-Forwards: 70
  182. From: "Henry" <sip:3006@85.92.195.146>;tag=cJTKMC5V8THEhrEsK-sd7RtKTUWc1iUa
  183. To: <sip:07927800653@85.92.195.146>;tag=as4c17d22d
  184. Call-ID: j0VSztO5Z0Jq7nkvbV37M8f15topYrG-
  185. CSeq: 12598 ACK
  186. Content-Length: 0
  187.  
  188. <------------->
  189. --- (8 headers 0 lines) ---
  190.     -- Executing [s@macro-hangupcall:3] ExecIf("SIP/3006-00000006", "0?Set(CDR(recordingfile)=)") in new stack
  191.     -- Executing [s@macro-hangupcall:4] Hangup("SIP/3006-00000006", "") in new stack
  192.   == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/3006-00000006' in macro 'hangupcall'
  193.   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3006-00000006'
  194.     -- SIP/3006-00000006 Internal Gosub(crm-hangup,s,1) start
  195.     -- Executing [s@crm-hangup:1] NoOp("SIP/3006-00000006", "Sending Hangup to CRM") in new stack
  196.     -- Executing [s@crm-hangup:2] NoOp("SIP/3006-00000006", "HANGUP CAUSE: 16") in new stack
  197.     -- Executing [s@crm-hangup:3] ExecIf("SIP/3006-00000006", "0?Set(__CRM_VOICEMAIL=)") in new stack
  198.     -- Executing [s@crm-hangup:4] NoOp("SIP/3006-00000006", "MASTER CHANNEL: 1499960902.12 = 1499960902.12") in new stack
  199.     -- Executing [s@crm-hangup:5] GotoIf("SIP/3006-00000006", "0?return") in new stack
  200.     -- Executing [s@crm-hangup:6] Set("SIP/3006-00000006", "__CRM_HANGUP=1") in new stack
  201.     -- Executing [s@crm-hangup:7] AGI("SIP/3006-00000006", "sangomacrm.agi") in new stack
  202.     -- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
  203.     -- <SIP/3006-00000006>AGI Script sangomacrm.agi completed, returning 0
  204.     -- Executing [s@crm-hangup:8] Return("SIP/3006-00000006", "") in new stack
  205.   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3006-00000006'
  206.     -- SIP/3006-00000006 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
  207. Retransmitting #1 (no NAT) to 46.31.231.185:5060:
  208. CANCEL sip:07927800653@sip.voipfone.net SIP/2.0
  209. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK0e45325a
  210. Max-Forwards: 70
  211. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as2c206cd1
  212. To: <sip:07927800653@sip.voipfone.net>
  213. Call-ID: 50284148397024fa345a86f94131b8e2@85.92.195.146:5160
  214. CSeq: 103 CANCEL
  215. User-Agent: FPBX-14.0.1.1(13.16.0)
  216. Content-Length: 0
  217.  
  218.  
  219. ---
  220. Really destroying SIP dialog 'j0VSztO5Z0Jq7nkvbV37M8f15topYrG-' Method: ACK
  221.  
  222. <--- SIP read from UDP:46.31.231.185:5060 --->
  223. SIP/2.0 487 Request Terminated
  224. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK0e45325a;received=85.92.195.146;rport=5160
  225. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as2c206cd1
  226. To: <sip:07927800653@sip.voipfone.net>;tag=VFf0d744a2583f56f8497e206196d7
  227. Call-ID: 50284148397024fa345a86f94131b8e2@85.92.195.146:5160
  228. CSeq: 103 INVITE
  229. User-Agent: Voipfone Sip Network
  230. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  231. Contact: <sip:07927800653@46.31.231.185>
  232. Content-Length: 0
  233.  
  234. <------------->
  235. --- (10 headers 0 lines) ---
  236. Transmitting (no NAT) to 46.31.231.185:5060:
  237. ACK sip:07927800653@46.31.231.185 SIP/2.0
  238. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK0e45325a
  239. Max-Forwards: 70
  240. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as2c206cd1
  241. To: <sip:07927800653@sip.voipfone.net>;tag=VFf0d744a2583f56f8497e206196d7
  242. Contact: <sip:30178956@85.92.195.146:5160>
  243. Call-ID: 50284148397024fa345a86f94131b8e2@85.92.195.146:5160
  244. CSeq: 103 ACK
  245. User-Agent: FPBX-14.0.1.1(13.16.0)
  246. Content-Length: 0
  247.  
  248.  
  249. ---
  250. Scheduling destruction of SIP dialog '50284148397024fa345a86f94131b8e2@85.92.195.146:5160' in 6400 ms (Method: INVITE)
  251.  
  252. <--- SIP read from UDP:46.31.231.185:5060 --->
  253. SIP/2.0 200 OK
  254. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK0e45325a;received=85.92.195.146;rport=5160
  255. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as2c206cd1
  256. To: <sip:07927800653@sip.voipfone.net>;tag=VFf0d744a2583f56f8497e206196d7
  257. Call-ID: 50284148397024fa345a86f94131b8e2@85.92.195.146:5160
  258. CSeq: 103 CANCEL
  259. User-Agent: Voipfone Sip Network
  260. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  261. Contact: <sip:07927800653@46.31.231.185>
  262. Content-Length: 0
  263.  
  264. <------------->
  265. --- (10 headers 0 lines) ---
  266.  
  267. <--- SIP read from UDP:46.31.231.185:5060 --->
  268. SIP/2.0 200 OK
  269. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK0e45325a;received=85.92.195.146;rport=5160
  270. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as2c206cd1
  271. To: <sip:07927800653@sip.voipfone.net>;tag=VFf0d744a2583f56f8497e206196d7
  272. Call-ID: 50284148397024fa345a86f94131b8e2@85.92.195.146:5160
  273. CSeq: 103 CANCEL
  274. User-Agent: Voipfone Sip Network
  275. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  276. Contact: <sip:07927800653@46.31.231.185>
  277. Content-Length: 0
  278.  
  279. <------------->
  280. --- (10 headers 0 lines) ---
  281.  
  282. <--- SIP read from UDP:82.147.28.150:61123 --->
  283.  
  284. <------------->

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