debug

From henry, 9 Months ago, written in Plain Text, viewed 38 times.
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  1. [2017-07-13 15:21:42] NOTICE[12298]: chan_sip.c:15722 sip_reregister:    -- Re-registration for  30178956@sip.voipfone.net
  2. REGISTER 12 headers, 0 lines
  3. Reliably Transmitting (NAT) to 46.31.231.185:5060:
  4. REGISTER sip:sip.voipfone.net SIP/2.0
  5. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK31e5e873;rport
  6. Max-Forwards: 70
  7. From: <sip:30178956@sip.voipfone.net>;tag=as2c82c51d
  8. To: <sip:30178956@sip.voipfone.net>
  9. Call-ID: 39dcff7a7fa4dcea361f1c797f5e590e@127.0.0.1
  10. CSeq: 109 REGISTER
  11. Supported: replaces, timer
  12. User-Agent: FPBX-14.0.1.1(13.16.0)
  13. Authorization: Digest username="30178956", realm="asterisk", algorithm=MD5, uri="sip:sip.voipfone.net", nonce="bohsaylm", response="9f269255571775b7d5907af806515e28"
  14. Expires: 120
  15. Contact: <sip:30178956@85.92.195.146:5160>
  16. Content-Length: 0
  17.  
  18.  
  19. ---
  20.  
  21. <--- SIP read from UDP:46.31.231.185:5060 --->
  22. SIP/2.0 200 OK
  23. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK31e5e873;rport
  24. From: <sip:30178956@sip.voipfone.net>;tag=as2c82c51d
  25. To: <sip:30178956@sip.voipfone.net>
  26. Call-ID: 39dcff7a7fa4dcea361f1c797f5e590e@127.0.0.1
  27. CSeq: 109 REGISTER
  28. Contact: <sip:30178956@85.92.195.146:5160>;expires=60
  29. Expires: 60
  30. Date: Thu, 13 Jul 2017 15:21:42 GMT
  31. Min-Expires: 60
  32. User-Agent: Voipfone
  33. Content-Length: 0
  34.  
  35. <------------->
  36. --- (12 headers 0 lines) ---
  37. [2017-07-13 15:21:42] NOTICE[12298]: chan_sip.c:24538 handle_response_register: Outbound Registration: Expiry for sip.voipfone.net is 60 sec (Scheduling reregistration in 45 s)
  38. Really destroying SIP dialog '39dcff7a7fa4dcea361f1c797f5e590e@127.0.0.1' Method: REGISTER
  39.  
  40. <--- SIP read from UDP:82.147.28.150:61123 --->
  41. INVITE sip:07927800653@85.92.195.146:5160 SIP/2.0
  42. Via: SIP/2.0/UDP 172.16.103.99:61123;rport;branch=z9hG4bKPjeoIaCGtXlWW8X4fkZ9v-vWpC59nypC6n
  43. Max-Forwards: 70
  44. From: "Henry" <sip:3006@85.92.195.146>;tag=Q2lbioeZQfR0W7DlO6-Sr.aukx-KR-W1
  45. To: <sip:07927800653@85.92.195.146>
  46. Contact: "Henry" <sip:3006@172.16.103.99:61123;ob>
  47. Call-ID: MUeWPCzKNtDim4ZX-z-KHicPk7TcH7CF
  48. CSeq: 7662 INVITE
  49. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  50. Supported: replaces, 100rel, norefersub
  51. User-Agent: Telephone 1.2.6
  52. Content-Type: application/sdp
  53. Content-Length: 544
  54.  
  55. v=0
  56. o=- 3708948103 3708948103 IN IP4 172.16.103.99
  57. s=pjmedia
  58. b=AS:117
  59. t=0 0
  60. a=X-nat:0
  61. m=audio 4016 RTP/AVP 103 102 104 125 109 3 0 8 9 101
  62. c=IN IP4 172.16.103.99
  63. b=TIAS:96000
  64. a=rtcp:4017 IN IP4 172.16.103.99
  65. a=sendrecv
  66. a=rtpmap:103 speex/16000
  67. a=rtpmap:102 speex/8000
  68. a=rtpmap:104 speex/32000
  69. a=rtpmap:125 opus/48000/2
  70. a=fmtp:125 useinbandfec=1
  71. a=rtpmap:109 iLBC/8000
  72. a=fmtp:109 mode=30
  73. a=rtpmap:3 GSM/8000
  74. a=rtpmap:0 PCMU/8000
  75. a=rtpmap:8 PCMA/8000
  76. a=rtpmap:9 G722/8000
  77. a=rtpmap:101 telephone-event/8000
  78. a=fmtp:101 0-16
  79. <------------->
  80. --- (13 headers 24 lines) ---
  81. Sending to 82.147.28.150:61123 (NAT)
  82. Sending to 82.147.28.150:61123 (NAT)
  83. Using INVITE request as basis request - MUeWPCzKNtDim4ZX-z-KHicPk7TcH7CF
  84. Found peer '3006' for '3006' from 82.147.28.150:61123
  85.  
  86. <--- Reliably Transmitting (no NAT) to 82.147.28.150:61123 --->
  87. SIP/2.0 401 Unauthorized
  88. Via: SIP/2.0/UDP 172.16.103.99:61123;branch=z9hG4bKPjeoIaCGtXlWW8X4fkZ9v-vWpC59nypC6n;received=82.147.28.150;rport=61123
  89. From: "Henry" <sip:3006@85.92.195.146>;tag=Q2lbioeZQfR0W7DlO6-Sr.aukx-KR-W1
  90. To: <sip:07927800653@85.92.195.146>;tag=as4746837c
  91. Call-ID: MUeWPCzKNtDim4ZX-z-KHicPk7TcH7CF
  92. CSeq: 7662 INVITE
  93. Server: FPBX-14.0.1.1(13.16.0)
  94. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  95. Supported: replaces, timer
  96. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="101aefa9"
  97. Content-Length: 0
  98.  
  99.  
  100. <------------>
  101. Scheduling destruction of SIP dialog 'MUeWPCzKNtDim4ZX-z-KHicPk7TcH7CF' in 6400 ms (Method: INVITE)
  102.  
  103. <--- SIP read from UDP:82.147.28.150:61123 --->
  104. ACK sip:07927800653@85.92.195.146:5160 SIP/2.0
  105. Via: SIP/2.0/UDP 172.16.103.99:61123;rport;branch=z9hG4bKPjeoIaCGtXlWW8X4fkZ9v-vWpC59nypC6n
  106. Max-Forwards: 70
  107. From: "Henry" <sip:3006@85.92.195.146>;tag=Q2lbioeZQfR0W7DlO6-Sr.aukx-KR-W1
  108. To: <sip:07927800653@85.92.195.146>;tag=as4746837c
  109. Call-ID: MUeWPCzKNtDim4ZX-z-KHicPk7TcH7CF
  110. CSeq: 7662 ACK
  111. Content-Length: 0
  112.  
  113. <------------->
  114. --- (8 headers 0 lines) ---
  115.  
  116. <--- SIP read from UDP:82.147.28.150:61123 --->
  117. INVITE sip:07927800653@85.92.195.146:5160 SIP/2.0
  118. Via: SIP/2.0/UDP 172.16.103.99:61123;rport;branch=z9hG4bKPj6e1f2UH6tVTNW.FeOyiTlHi1NCx267qs
  119. Max-Forwards: 70
  120. From: "Henry" <sip:3006@85.92.195.146>;tag=Q2lbioeZQfR0W7DlO6-Sr.aukx-KR-W1
  121. To: <sip:07927800653@85.92.195.146>
  122. Contact: "Henry" <sip:3006@172.16.103.99:61123;ob>
  123. Call-ID: MUeWPCzKNtDim4ZX-z-KHicPk7TcH7CF
  124. CSeq: 7663 INVITE
  125. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  126. Supported: replaces, 100rel, norefersub
  127. User-Agent: Telephone 1.2.6
  128. Authorization: Digest username="3006", realm="asterisk", nonce="101aefa9", uri="sip:07927800653@85.92.195.146:5160", response="dd61909b78fa04217cada1666736ead8", algorithm=MD5
  129. Content-Type: application/sdp
  130. Content-Length: 544
  131.  
  132. v=0
  133. o=- 3708948103 3708948103 IN IP4 172.16.103.99
  134. s=pjmedia
  135. b=AS:117
  136. t=0 0
  137. a=X-nat:0
  138. m=audio 4016 RTP/AVP 103 102 104 125 109 3 0 8 9 101
  139. c=IN IP4 172.16.103.99
  140. b=TIAS:96000
  141. a=rtcp:4017 IN IP4 172.16.103.99
  142. a=sendrecv
  143. a=rtpmap:103 speex/16000
  144. a=rtpmap:102 speex/8000
  145. a=rtpmap:104 speex/32000
  146. a=rtpmap:125 opus/48000/2
  147. a=fmtp:125 useinbandfec=1
  148. a=rtpmap:109 iLBC/8000
  149. a=fmtp:109 mode=30
  150. a=rtpmap:3 GSM/8000
  151. a=rtpmap:0 PCMU/8000
  152. a=rtpmap:8 PCMA/8000
  153. a=rtpmap:9 G722/8000
  154. a=rtpmap:101 telephone-event/8000
  155. a=fmtp:101 0-16
  156. <------------->
  157. --- (14 headers 24 lines) ---
  158. Sending to 82.147.28.150:61123 (no NAT)
  159. Using INVITE request as basis request - MUeWPCzKNtDim4ZX-z-KHicPk7TcH7CF
  160. Found peer '3006' for '3006' from 82.147.28.150:61123
  161.   == Using SIP RTP TOS bits 184
  162.   == Using SIP RTP CoS mark 5
  163. Found RTP audio format 103
  164. Found RTP audio format 102
  165. Found RTP audio format 104
  166. Found RTP audio format 125
  167. Found RTP audio format 109
  168. Found RTP audio format 3
  169. Found RTP audio format 0
  170. Found RTP audio format 8
  171. Found RTP audio format 9
  172. Found RTP audio format 101
  173. Found audio description format speex for ID 103
  174. Found audio description format speex for ID 102
  175. Found audio description format speex for ID 104
  176. Found audio description format opus for ID 125
  177. Found audio description format iLBC for ID 109
  178. Found audio description format GSM for ID 3
  179. Found audio description format PCMU for ID 0
  180. Found audio description format PCMA for ID 8
  181. Found audio description format G722 for ID 9
  182. Found audio description format telephone-event for ID 101
  183. Capabilities: us - (ulaw|alaw|gsm|g726|g722|g723|speex|siren7|adpcm|g719|g729|slin), peer - audio=(ulaw|gsm|alaw|g722|speex|speex16|speex32|ilbc|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g722|speex)
  184. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  185. Peer audio RTP is at port 172.16.103.99:4016
  186. Looking for 07927800653 in from-internal (domain 85.92.195.146)
  187. sip_route_dump: route/path hop: <sip:3006@172.16.103.99:61123;ob>
  188.  
  189. <--- Transmitting (no NAT) to 82.147.28.150:61123 --->
  190. SIP/2.0 100 Trying
  191. Via: SIP/2.0/UDP 172.16.103.99:61123;branch=z9hG4bKPj6e1f2UH6tVTNW.FeOyiTlHi1NCx267qs;received=82.147.28.150;rport=61123
  192. From: "Henry" <sip:3006@85.92.195.146>;tag=Q2lbioeZQfR0W7DlO6-Sr.aukx-KR-W1
  193. To: <sip:07927800653@85.92.195.146>
  194. Call-ID: MUeWPCzKNtDim4ZX-z-KHicPk7TcH7CF
  195. CSeq: 7663 INVITE
  196. Server: FPBX-14.0.1.1(13.16.0)
  197. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  198. Supported: replaces, timer
  199. Contact: <sip:07927800653@85.92.195.146:5160>
  200. Content-Length: 0
  201.  
  202.  
  203. <------------>
  204.     -- Executing [07927800653@from-internal:1] Macro("SIP/3006-0000013f", "user-callerid,LIMIT,EXTERNAL,") in new stack
  205.     -- Executing [s@macro-user-callerid:1] Set("SIP/3006-0000013f", "TOUCH_MONITOR=1499959303.480") in new stack
  206.     -- Executing [s@macro-user-callerid:2] Set("SIP/3006-0000013f", "AMPUSER=3006") in new stack
  207.     -- Executing [s@macro-user-callerid:3] GotoIf("SIP/3006-0000013f", "0?report") in new stack
  208.     -- Executing [s@macro-user-callerid:4] ExecIf("SIP/3006-0000013f", "1?Set(__REALCALLERIDNUM=3006)") in new stack
  209.     -- Executing [s@macro-user-callerid:5] Set("SIP/3006-0000013f", "AMPUSER=3006") in new stack
  210.     -- Executing [s@macro-user-callerid:6] GotoIf("SIP/3006-0000013f", "0?limit") in new stack
  211.     -- Executing [s@macro-user-callerid:7] Set("SIP/3006-0000013f", "AMPUSERCIDNAME=Southcoast Payments") in new stack
  212.     -- Executing [s@macro-user-callerid:8] GotoIf("SIP/3006-0000013f", "0?report") in new stack
  213.     -- Executing [s@macro-user-callerid:9] Set("SIP/3006-0000013f", "AMPUSERCID=3006") in new stack
  214.     -- Executing [s@macro-user-callerid:10] Set("SIP/3006-0000013f", "__DIAL_OPTIONS=Ttr") in new stack
  215.     -- Executing [s@macro-user-callerid:11] Set("SIP/3006-0000013f", "CALLERID(all)="Southcoast Payments" <3006>") in new stack
  216.     -- Executing [s@macro-user-callerid:12] GotoIf("SIP/3006-0000013f", "0?limit") in new stack
  217.     -- Executing [s@macro-user-callerid:13] ExecIf("SIP/3006-0000013f", "1?Set(GROUP(concurrency_limit)=3006)") in new stack
  218.     -- Executing [s@macro-user-callerid:14] ExecIf("SIP/3006-0000013f", "0?Set(CHANNEL(language)=)") in new stack
  219.     -- Executing [s@macro-user-callerid:15] GotoIf("SIP/3006-0000013f", "1?continue") in new stack
  220.     -- Goto (macro-user-callerid,s,29)
  221.     -- Executing [s@macro-user-callerid:29] Set("SIP/3006-0000013f", "CALLERID(number)=3006") in new stack
  222.     -- Executing [s@macro-user-callerid:30] Set("SIP/3006-0000013f", "CALLERID(name)=Southcoast Payments") in new stack
  223.     -- Executing [s@macro-user-callerid:31] GotoIf("SIP/3006-0000013f", "0?cnum") in new stack
  224.     -- Executing [s@macro-user-callerid:32] Set("SIP/3006-0000013f", "CDR(cnam)=Southcoast Payments") in new stack
  225.     -- Executing [s@macro-user-callerid:33] Set("SIP/3006-0000013f", "CDR(cnum)=3006") in new stack
  226.     -- Executing [s@macro-user-callerid:34] Set("SIP/3006-0000013f", "CHANNEL(language)=en") in new stack
  227.     -- Executing [07927800653@from-internal:2] Gosub("SIP/3006-0000013f", "sub-record-check,s,1(out,07927800653,dontcare)") in new stack
  228.     -- Executing [s@sub-record-check:1] GotoIf("SIP/3006-0000013f", "0?initialized") in new stack
  229.     -- Executing [s@sub-record-check:2] Set("SIP/3006-0000013f", "__REC_STATUS=INITIALIZED") in new stack
  230.     -- Executing [s@sub-record-check:3] Set("SIP/3006-0000013f", "NOW=1499959303") in new stack
  231.     -- Executing [s@sub-record-check:4] Set("SIP/3006-0000013f", "__DAY=13") in new stack
  232.     -- Executing [s@sub-record-check:5] Set("SIP/3006-0000013f", "__MONTH=07") in new stack
  233.     -- Executing [s@sub-record-check:6] Set("SIP/3006-0000013f", "__YEAR=2017") in new stack
  234.     -- Executing [s@sub-record-check:7] Set("SIP/3006-0000013f", "__TIMESTR=20170713-152143") in new stack
  235.     -- Executing [s@sub-record-check:8] Set("SIP/3006-0000013f", "__FROMEXTEN=3006") in new stack
  236.     -- Executing [s@sub-record-check:9] Set("SIP/3006-0000013f", "__MON_FMT=wav") in new stack
  237.     -- Executing [s@sub-record-check:10] NoOp("SIP/3006-0000013f", "Recordings initialized") in new stack
  238.     -- Executing [s@sub-record-check:11] ExecIf("SIP/3006-0000013f", "0?Set(ARG3=dontcare)") in new stack
  239.     -- Executing [s@sub-record-check:12] Set("SIP/3006-0000013f", "REC_POLICY_MODE_SAVE=") in new stack
  240.     -- Executing [s@sub-record-check:13] ExecIf("SIP/3006-0000013f", "0?Set(REC_STATUS=NO)") in new stack
  241.     -- Executing [s@sub-record-check:14] GotoIf("SIP/3006-0000013f", "3?checkaction") in new stack
  242.     -- Goto (sub-record-check,s,17)
  243.     -- Executing [s@sub-record-check:17] GotoIf("SIP/3006-0000013f", "1?sub-record-check,out,1") in new stack
  244.     -- Goto (sub-record-check,out,1)
  245.     -- Executing [out@sub-record-check:1] NoOp("SIP/3006-0000013f", "Outbound Recording Check from 3006 to 07927800653") in new stack
  246.     -- Executing [out@sub-record-check:2] Set("SIP/3006-0000013f", "RECMODE=dontcare") in new stack
  247.     -- Executing [out@sub-record-check:3] ExecIf("SIP/3006-0000013f", "1?Goto(routewins)") in new stack
  248.     -- Goto (sub-record-check,out,7)
  249.     -- Executing [out@sub-record-check:7] Gosub("SIP/3006-0000013f", "recordcheck,1(dontcare,out,07927800653)") in new stack
  250.     -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/3006-0000013f", "Starting recording check against dontcare") in new stack
  251.     -- Executing [recordcheck@sub-record-check:2] Goto("SIP/3006-0000013f", "dontcare") in new stack
  252.     -- Goto (sub-record-check,recordcheck,3)
  253.     -- Executing [recordcheck@sub-record-check:3] Return("SIP/3006-0000013f", "") in new stack
  254.     -- Executing [out@sub-record-check:8] Return("SIP/3006-0000013f", "") in new stack
  255.     -- Executing [07927800653@from-internal:3] ExecIf("SIP/3006-0000013f", "0 ?Set(CDR(accountcode)=)") in new stack
  256.     -- Executing [07927800653@from-internal:4] Set("SIP/3006-0000013f", "MOHCLASS=default") in new stack
  257.     -- Executing [07927800653@from-internal:5] Set("SIP/3006-0000013f", "_NODEST=") in new stack
  258.     -- Executing [07927800653@from-internal:6] Macro("SIP/3006-0000013f", "dialout-trunk,1,07927800653,,off") in new stack
  259.     -- Executing [s@macro-dialout-trunk:1] Set("SIP/3006-0000013f", "DIAL_TRUNK=1") in new stack
  260.     -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/3006-0000013f", "0?sub-pincheck,s,1()") in new stack
  261.     -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/3006-0000013f", "0?disabletrunk,1") in new stack
  262.     -- Executing [s@macro-dialout-trunk:4] Set("SIP/3006-0000013f", "DIAL_NUMBER=07927800653") in new stack
  263.     -- Executing [s@macro-dialout-trunk:5] Set("SIP/3006-0000013f", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
  264.     -- Executing [s@macro-dialout-trunk:6] Set("SIP/3006-0000013f", "OUTBOUND_GROUP=OUT_1") in new stack
  265.     -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/3006-0000013f", "1?nomax") in new stack
  266.     -- Goto (macro-dialout-trunk,s,9)
  267.     -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/3006-0000013f", "0?skipoutcid") in new stack
  268.     -- Executing [s@macro-dialout-trunk:10] Set("SIP/3006-0000013f", "DIAL_TRUNK_OPTIONS=T") in new stack
  269.     -- Executing [s@macro-dialout-trunk:11] Macro("SIP/3006-0000013f", "outbound-callerid,1") in new stack
  270.     -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/3006-0000013f", "0?Set(CALLERPRES(name-pres)=)") in new stack
  271.     -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/3006-0000013f", "0?Set(CALLERPRES(num-pres)=)") in new stack
  272.     -- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/3006-0000013f", "0?Set(REALCALLERIDNUM=3006)") in new stack
  273.     -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/3006-0000013f", "1?normcid") in new stack
  274.     -- Goto (macro-outbound-callerid,s,7)
  275.     -- Executing [s@macro-outbound-callerid:7] Set("SIP/3006-0000013f", "USEROUTCID=") in new stack
  276.     -- Executing [s@macro-outbound-callerid:8] Set("SIP/3006-0000013f", "EMERGENCYCID=") in new stack
  277.     -- Executing [s@macro-outbound-callerid:9] Set("SIP/3006-0000013f", "TRUNKOUTCID=hidden") in new stack
  278.     -- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/3006-0000013f", "1?trunkcid") in new stack
  279.     -- Goto (macro-outbound-callerid,s,15)
  280.     -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/3006-0000013f", "1?Set(CALLERID(all)=hidden)") in new stack
  281.     -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/3006-0000013f", "0?Set(CALLERID(all)=)") in new stack
  282.     -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/3006-0000013f", "0?Set(CALLERID(all)=)") in new stack
  283.     -- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/3006-0000013f", "1?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
  284.     -- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/3006-0000013f", "1?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
  285.     -- Executing [s@macro-outbound-callerid:20] Set("SIP/3006-0000013f", "CDR(outbound_cnum)=") in new stack
  286.     -- Executing [s@macro-outbound-callerid:21] Set("SIP/3006-0000013f", "CDR(outbound_cnam)=hidden") in new stack
  287.     -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/3006-0000013f", "0?sub-flp-1,s,1()") in new stack
  288.     -- Executing [s@macro-dialout-trunk:13] Set("SIP/3006-0000013f", "OUTNUM=07927800653") in new stack
  289.     -- Executing [s@macro-dialout-trunk:14] Set("SIP/3006-0000013f", "custom=SIP/voipfone") in new stack
  290.     -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/3006-0000013f", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
  291.     -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/3006-0000013f", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
  292.     -- Executing [s@macro-dialout-trunk:17] Macro("SIP/3006-0000013f", "dialout-trunk-predial-hook,") in new stack
  293.     -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/3006-0000013f", "") in new stack
  294.     -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/3006-0000013f", "0?skipcrm") in new stack
  295.     -- Executing [s@macro-dialout-trunk:19] Set("SIP/3006-0000013f", "__CRM_DIRECTION=OUTBOUND") in new stack
  296.     -- Executing [s@macro-dialout-trunk:20] Set("SIP/3006-0000013f", "__CRM_DESTINATION=07927800653") in new stack
  297.     -- Executing [s@macro-dialout-trunk:21] Set("SIP/3006-0000013f", "__CRM_SOURCE=3006") in new stack
  298.     -- Executing [s@macro-dialout-trunk:22] AGI("SIP/3006-0000013f", "sangomacrm.agi") in new stack
  299.     -- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
  300.     -- <SIP/3006-0000013f>AGI Script sangomacrm.agi completed, returning 0
  301.     -- Executing [s@macro-dialout-trunk:23] Set("SIP/3006-0000013f", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
  302.     -- Executing [s@macro-dialout-trunk:24] NoOp("SIP/3006-0000013f", "CRM Finished") in new stack
  303.     -- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/3006-0000013f", "0?bypass,1") in new stack
  304.     -- Executing [s@macro-dialout-trunk:26] ExecIf("SIP/3006-0000013f", "1?Set(CONNECTEDLINE(num,i)=07927800653)") in new stack
  305.     -- Executing [s@macro-dialout-trunk:27] ExecIf("SIP/3006-0000013f", "0?Set(CONNECTEDLINE(name,i)=CID:)") in new stack
  306.     -- Executing [s@macro-dialout-trunk:28] ExecIf("SIP/3006-0000013f", "1?Set(CONNECTEDLINE(name,i)=CID:(Hidden))") in new stack
  307.     -- Executing [s@macro-dialout-trunk:29] GotoIf("SIP/3006-0000013f", "0?customtrunk") in new stack
  308.     -- Executing [s@macro-dialout-trunk:30] Dial("SIP/3006-0000013f", "SIP/voipfone/07927800653@Voipfone,300,T") in new stack
  309.   == Using SIP RTP TOS bits 184
  310.   == Using SIP RTP CoS mark 5
  311. Audio is at 13506
  312. Adding codec alaw to SDP
  313. Adding non-codec 0x1 (telephone-event) to SDP
  314. Reliably Transmitting (NAT) to 46.31.231.185:5060:
  315. INVITE sip:07927800653%40Voipfone@sip.voipfone.net SIP/2.0
  316. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK0927e1c9;rport
  317. Max-Forwards: 70
  318. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6f8c2a48
  319. To: <sip:07927800653%40Voipfone@sip.voipfone.net>
  320. Contact: <sip:30178956@85.92.195.146:5160>
  321. Call-ID: 4c6758d076a3c9f104f8b68840eaace2@85.92.195.146:5160
  322. CSeq: 102 INVITE
  323. User-Agent: FPBX-14.0.1.1(13.16.0)
  324. Date: Thu, 13 Jul 2017 15:21:43 GMT
  325. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  326. Supported: replaces, timer
  327. Content-Type: application/sdp
  328. Content-Length: 250
  329.  
  330. v=0
  331. o=root 86930962 86930962 IN IP4 85.92.195.146
  332. s=Asterisk PBX 13.16.0
  333. c=IN IP4 85.92.195.146
  334. t=0 0
  335. m=audio 13506 RTP/AVP 8 101
  336. a=rtpmap:8 PCMA/8000
  337. a=rtpmap:101 telephone-event/8000
  338. a=fmtp:101 0-16
  339. a=ptime:20
  340. a=maxptime:150
  341. a=sendrecv
  342.  
  343. ---
  344.     -- Called SIP/voipfone/07927800653@Voipfone
  345. Retransmitting #1 (NAT) to 46.31.231.185:5060:
  346. INVITE sip:07927800653%40Voipfone@sip.voipfone.net SIP/2.0
  347. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK0927e1c9;rport
  348. Max-Forwards: 70
  349. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6f8c2a48
  350. To: <sip:07927800653%40Voipfone@sip.voipfone.net>
  351. Contact: <sip:30178956@85.92.195.146:5160>
  352. Call-ID: 4c6758d076a3c9f104f8b68840eaace2@85.92.195.146:5160
  353. CSeq: 102 INVITE
  354. User-Agent: FPBX-14.0.1.1(13.16.0)
  355. Date: Thu, 13 Jul 2017 15:21:43 GMT
  356. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  357. Supported: replaces, timer
  358. Content-Type: application/sdp
  359. Content-Length: 250
  360.  
  361. v=0
  362. o=root 86930962 86930962 IN IP4 85.92.195.146
  363. s=Asterisk PBX 13.16.0
  364. c=IN IP4 85.92.195.146
  365. t=0 0
  366. m=audio 13506 RTP/AVP 8 101
  367. a=rtpmap:8 PCMA/8000
  368. a=rtpmap:101 telephone-event/8000
  369. a=fmtp:101 0-16
  370. a=ptime:20
  371. a=maxptime:150
  372. a=sendrecv
  373.  
  374. ---
  375.  
  376. <--- SIP read from UDP:46.31.231.185:5060 --->
  377. SIP/2.0 407 Proxy Authentication Required
  378. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK0927e1c9;received=85.92.195.146;rport=5160
  379. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6f8c2a48
  380. To: <sip:07927800653%40Voipfone@sip.voipfone.net>;tag=VF9aeb83e639053270837d36ddd2bb
  381. Call-ID: 4c6758d076a3c9f104f8b68840eaace2@85.92.195.146:5160
  382. CSeq: 102 INVITE
  383. User-Agent: Voipfone Sip Network
  384. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  385. Contact: <sip:07927800653%40Voipfone@46.31.231.185>
  386. Proxy-Authenticate: Digest realm="asterisk", nonce="1d624c4f"
  387. Content-Length: 0
  388.  
  389. <------------->
  390. --- (11 headers 0 lines) ---
  391. Transmitting (NAT) to 46.31.231.185:5060:
  392. ACK sip:07927800653%40Voipfone@sip.voipfone.net SIP/2.0
  393. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK0927e1c9;rport
  394. Max-Forwards: 70
  395. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6f8c2a48
  396. To: <sip:07927800653%40Voipfone@sip.voipfone.net>;tag=VF9aeb83e639053270837d36ddd2bb
  397. Contact: <sip:30178956@85.92.195.146:5160>
  398. Call-ID: 4c6758d076a3c9f104f8b68840eaace2@85.92.195.146:5160
  399. CSeq: 102 ACK
  400. User-Agent: FPBX-14.0.1.1(13.16.0)
  401. Content-Length: 0
  402.  
  403.  
  404. ---
  405. Audio is at 13506
  406. Adding codec alaw to SDP
  407. Adding non-codec 0x1 (telephone-event) to SDP
  408. Reliably Transmitting (NAT) to 46.31.231.185:5060:
  409. INVITE sip:07927800653%40Voipfone@sip.voipfone.net SIP/2.0
  410. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK15e2c66d;rport
  411. Max-Forwards: 70
  412. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6f8c2a48
  413. To: <sip:07927800653%40Voipfone@sip.voipfone.net>
  414. Contact: <sip:30178956@85.92.195.146:5160>
  415. Call-ID: 4c6758d076a3c9f104f8b68840eaace2@85.92.195.146:5160
  416. CSeq: 103 INVITE
  417. User-Agent: FPBX-14.0.1.1(13.16.0)
  418. Proxy-Authorization: Digest username="30178956", realm="asterisk", algorithm=MD5, uri="sip:07927800653%40Voipfone@sip.voipfone.net", nonce="1d624c4f", response="1ec2a3f52c731e84e2c8d470c3d35069"
  419. Date: Thu, 13 Jul 2017 15:21:43 GMT
  420. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  421. Supported: replaces, timer
  422. Content-Type: application/sdp
  423. Content-Length: 250
  424.  
  425. v=0
  426. o=root 86930962 86930963 IN IP4 85.92.195.146
  427. s=Asterisk PBX 13.16.0
  428. c=IN IP4 85.92.195.146
  429. t=0 0
  430. m=audio 13506 RTP/AVP 8 101
  431. a=rtpmap:8 PCMA/8000
  432. a=rtpmap:101 telephone-event/8000
  433. a=fmtp:101 0-16
  434. a=ptime:20
  435. a=maxptime:150
  436. a=sendrecv
  437.  
  438. ---
  439. Retransmitting #1 (NAT) to 46.31.231.185:5060:
  440. INVITE sip:07927800653%40Voipfone@sip.voipfone.net SIP/2.0
  441. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK15e2c66d;rport
  442. Max-Forwards: 70
  443. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6f8c2a48
  444. To: <sip:07927800653%40Voipfone@sip.voipfone.net>
  445. Contact: <sip:30178956@85.92.195.146:5160>
  446. Call-ID: 4c6758d076a3c9f104f8b68840eaace2@85.92.195.146:5160
  447. CSeq: 103 INVITE
  448. User-Agent: FPBX-14.0.1.1(13.16.0)
  449. Proxy-Authorization: Digest username="30178956", realm="asterisk", algorithm=MD5, uri="sip:07927800653%40Voipfone@sip.voipfone.net", nonce="1d624c4f", response="1ec2a3f52c731e84e2c8d470c3d35069"
  450. Date: Thu, 13 Jul 2017 15:21:43 GMT
  451. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  452. Supported: replaces, timer
  453. Content-Type: application/sdp
  454. Content-Length: 250
  455.  
  456. v=0
  457. o=root 86930962 86930963 IN IP4 85.92.195.146
  458. s=Asterisk PBX 13.16.0
  459. c=IN IP4 85.92.195.146
  460. t=0 0
  461. m=audio 13506 RTP/AVP 8 101
  462. a=rtpmap:8 PCMA/8000
  463. a=rtpmap:101 telephone-event/8000
  464. a=fmtp:101 0-16
  465. a=ptime:20
  466. a=maxptime:150
  467. a=sendrecv
  468.  
  469. ---
  470.  
  471. <--- SIP read from UDP:46.31.231.185:5060 --->
  472. SIP/2.0 100 Trying
  473. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK15e2c66d;received=85.92.195.146;rport=5160
  474. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6f8c2a48
  475. To: <sip:07927800653%40Voipfone@sip.voipfone.net>;tag=VF0ca38ccdaf4459db9e9f6110a4b1
  476. Call-ID: 4c6758d076a3c9f104f8b68840eaace2@85.92.195.146:5160
  477. CSeq: 103 INVITE
  478. User-Agent: Voipfone Sip Network
  479. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  480. Contact: <sip:07927800653%40Voipfone@46.31.231.185>
  481. Content-Length: 0
  482.  
  483. <------------->
  484. --- (10 headers 0 lines) ---
  485.  
  486. <--- SIP read from UDP:46.31.231.185:5060 --->
  487. SIP/2.0 100 Trying
  488. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK15e2c66d;received=85.92.195.146;rport=5160
  489. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6f8c2a48
  490. To: <sip:07927800653%40Voipfone@sip.voipfone.net>;tag=VF0ca38ccdaf4459db9e9f6110a4b1
  491. Call-ID: 4c6758d076a3c9f104f8b68840eaace2@85.92.195.146:5160
  492. CSeq: 103 INVITE
  493. User-Agent: Voipfone Sip Network
  494. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  495. Contact: <sip:07927800653%40Voipfone@46.31.231.185>
  496. Content-Length: 0
  497.  
  498. <------------->
  499. --- (10 headers 0 lines) ---
  500.  
  501. <--- SIP read from UDP:46.31.231.185:5060 --->
  502. SIP/2.0 480 Number Unavailable
  503. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK15e2c66d;received=85.92.195.146;rport=5160
  504. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6f8c2a48
  505. To: <sip:07927800653%40Voipfone@sip.voipfone.net>;tag=VF0ca38ccdaf4459db9e9f6110a4b1
  506. Call-ID: 4c6758d076a3c9f104f8b68840eaace2@85.92.195.146:5160
  507. CSeq: 103 INVITE
  508. User-Agent: Voipfone Sip Network
  509. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  510. Contact: <sip:07927800653%40Voipfone@46.31.231.185>
  511. Content-Length: 0
  512.  
  513. <------------->
  514. --- (10 headers 0 lines) ---
  515. Transmitting (NAT) to 46.31.231.185:5060:
  516. ACK sip:07927800653%40Voipfone@sip.voipfone.net SIP/2.0
  517. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK15e2c66d;rport
  518. Max-Forwards: 70
  519. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6f8c2a48
  520. To: <sip:07927800653%40Voipfone@sip.voipfone.net>;tag=VF0ca38ccdaf4459db9e9f6110a4b1
  521. Contact: <sip:30178956@85.92.195.146:5160>
  522. Call-ID: 4c6758d076a3c9f104f8b68840eaace2@85.92.195.146:5160
  523. CSeq: 103 ACK
  524. User-Agent: FPBX-14.0.1.1(13.16.0)
  525. Content-Length: 0
  526.  
  527.  
  528. ---
  529.     -- SIP/voipfone-00000140 redirecting info has changed, passing it to SIP/3006-0000013f
  530.  
  531. <--- Transmitting (no NAT) to 82.147.28.150:61123 --->
  532. SIP/2.0 181 Call is being forwarded
  533. Via: SIP/2.0/UDP 172.16.103.99:61123;branch=z9hG4bKPj6e1f2UH6tVTNW.FeOyiTlHi1NCx267qs;received=82.147.28.150;rport=61123
  534. From: "Henry" <sip:3006@85.92.195.146>;tag=Q2lbioeZQfR0W7DlO6-Sr.aukx-KR-W1
  535. To: <sip:07927800653@85.92.195.146>;tag=as6cb44d79
  536. Call-ID: MUeWPCzKNtDim4ZX-z-KHicPk7TcH7CF
  537. CSeq: 7663 INVITE
  538. Server: FPBX-14.0.1.1(13.16.0)
  539. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  540. Supported: replaces, timer
  541. Contact: <sip:07927800653@85.92.195.146:5160>
  542. Content-Length: 0
  543.  
  544.  
  545. <------------>
  546.     -- SIP/voipfone-00000140 is busy
  547. Scheduling destruction of SIP dialog '4c6758d076a3c9f104f8b68840eaace2@85.92.195.146:5160' in 6400 ms (Method: INVITE)
  548.   == Everyone is busy/congested at this time (1:1/0/0)
  549.     -- Executing [s@macro-dialout-trunk:31] NoOp("SIP/3006-0000013f", "Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 19") in new stack
  550.     -- Executing [s@macro-dialout-trunk:32] GotoIf("SIP/3006-0000013f", "0?continue,1:s-BUSY,1") in new stack
  551.     -- Goto (macro-dialout-trunk,s-BUSY,1)
  552.     -- Executing [s-BUSY@macro-dialout-trunk:1] NoOp("SIP/3006-0000013f", "Dial failed due to trunk reporting BUSY - giving up") in new stack
  553.     -- Executing [s-BUSY@macro-dialout-trunk:2] PlayTones("SIP/3006-0000013f", "busy") in new stack
  554.     -- Executing [s-BUSY@macro-dialout-trunk:3] Busy("SIP/3006-0000013f", "20") in new stack
  555.  
  556. <--- Reliably Transmitting (no NAT) to 82.147.28.150:61123 --->
  557. SIP/2.0 486 Busy Here
  558. Via: SIP/2.0/UDP 172.16.103.99:61123;branch=z9hG4bKPj6e1f2UH6tVTNW.FeOyiTlHi1NCx267qs;received=82.147.28.150;rport=61123
  559. From: "Henry" <sip:3006@85.92.195.146>;tag=Q2lbioeZQfR0W7DlO6-Sr.aukx-KR-W1
  560. To: <sip:07927800653@85.92.195.146>;tag=as6cb44d79
  561. Call-ID: MUeWPCzKNtDim4ZX-z-KHicPk7TcH7CF
  562. CSeq: 7663 INVITE
  563. Server: FPBX-14.0.1.1(13.16.0)
  564. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  565. Supported: replaces, timer
  566. X-Asterisk-HangupCause: User alerting, no answer
  567. X-Asterisk-HangupCauseCode: 19
  568. Content-Length: 0
  569.  
  570.  
  571. <------------>
  572.   == Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on 'SIP/3006-0000013f' in macro 'dialout-trunk'
  573.   == Spawn extension (from-internal, 07927800653, 6) exited non-zero on 'SIP/3006-0000013f'
  574.     -- Executing [h@from-internal:1] Macro("SIP/3006-0000013f", "hangupcall") in new stack
  575.     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/3006-0000013f", "1?theend") in new stack
  576.     -- Goto (macro-hangupcall,s,3)
  577.     -- Executing [s@macro-hangupcall:3] ExecIf("SIP/3006-0000013f", "0?Set(CDR(recordingfile)=)") in new stack
  578.     -- Executing [s@macro-hangupcall:4] Hangup("SIP/3006-0000013f", "") in new stack
  579.   == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/3006-0000013f' in macro 'hangupcall'
  580.   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3006-0000013f'
  581.     -- SIP/3006-0000013f Internal Gosub(crm-hangup,s,1) start
  582.     -- Executing [s@crm-hangup:1] NoOp("SIP/3006-0000013f", "Sending Hangup to CRM") in new stack
  583.     -- Executing [s@crm-hangup:2] NoOp("SIP/3006-0000013f", "HANGUP CAUSE: 17") in new stack
  584.     -- Executing [s@crm-hangup:3] ExecIf("SIP/3006-0000013f", "0?Set(__CRM_VOICEMAIL=)") in new stack
  585.     -- Executing [s@crm-hangup:4] NoOp("SIP/3006-0000013f", "MASTER CHANNEL: 1499959303.480 = 1499959303.480") in new stack
  586.     -- Executing [s@crm-hangup:5] GotoIf("SIP/3006-0000013f", "0?return") in new stack
  587.     -- Executing [s@crm-hangup:6] Set("SIP/3006-0000013f", "__CRM_HANGUP=1") in new stack
  588.     -- Executing [s@crm-hangup:7] AGI("SIP/3006-0000013f", "sangomacrm.agi") in new stack
  589.     -- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
  590.  
  591. <--- SIP read from UDP:82.147.28.150:61123 --->
  592. ACK sip:07927800653@85.92.195.146:5160 SIP/2.0
  593. Via: SIP/2.0/UDP 172.16.103.99:61123;rport;branch=z9hG4bKPj6e1f2UH6tVTNW.FeOyiTlHi1NCx267qs
  594. Max-Forwards: 70
  595. From: "Henry" <sip:3006@85.92.195.146>;tag=Q2lbioeZQfR0W7DlO6-Sr.aukx-KR-W1
  596. To: <sip:07927800653@85.92.195.146>;tag=as6cb44d79
  597. Call-ID: MUeWPCzKNtDim4ZX-z-KHicPk7TcH7CF
  598. CSeq: 7663 ACK
  599. Content-Length: 0
  600.  
  601. <------------->
  602. --- (8 headers 0 lines) ---
  603.     -- <SIP/3006-0000013f>AGI Script sangomacrm.agi completed, returning 0
  604.     -- Executing [s@crm-hangup:8] Return("SIP/3006-0000013f", "") in new stack
  605.   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3006-0000013f'
  606.     -- SIP/3006-0000013f Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
  607. Really destroying SIP dialog 'MUeWPCzKNtDim4ZX-z-KHicPk7TcH7CF' Method: ACK
  608. [2017-07-13 15:21:49] NOTICE[12703]: res_pjsip/pjsip_distributor.c:526 log_failed_request: Request 'OPTIONS' from '"sipvicious" <sip:100@1.1.1.1>' failed for '163.172.214.69:5144' (callid: 376365035958935404184061) - No matching endpoint found
  609. Really destroying SIP dialog '4c6758d076a3c9f104f8b68840eaace2@85.92.195.146:5160' Method: INVITE
  610.  
  611. <--- SIP read from UDP:82.147.28.150:61123 --->
  612.  
  613. <------------->
  614. Reliably Transmitting (no NAT) to 82.147.28.150:61123:
  615. OPTIONS sip:3006@172.16.103.99:61123;ob SIP/2.0
  616. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK0156daf4
  617. Max-Forwards: 70
  618. From: "Unknown" <sip:Unknown@85.92.195.146:5160>;tag=as74f8d8a7
  619. To: <sip:3006@172.16.103.99:61123;ob>
  620. Contact: <sip:Unknown@85.92.195.146:5160>
  621. Call-ID: 3fb7cf663a875af24e12c5d44d3d2c53@85.92.195.146:5160
  622. CSeq: 102 OPTIONS
  623. User-Agent: FPBX-14.0.1.1(13.16.0)
  624. Date: Thu, 13 Jul 2017 15:22:02 GMT
  625. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  626. Supported: replaces, timer
  627. Content-Length: 0
  628.  
  629.  
  630. ---
  631.  
  632. <--- SIP read from UDP:82.147.28.150:61123 --->
  633. SIP/2.0 200 OK
  634. Via: SIP/2.0/UDP 85.92.195.146:5160;received=85.92.195.146;branch=z9hG4bK0156daf4
  635. Call-ID: 3fb7cf663a875af24e12c5d44d3d2c53@85.92.195.146:5160
  636. From: "Unknown" <sip:Unknown@85.92.195.146>;tag=as74f8d8a7
  637. To: <sip:3006@172.16.103.99;ob>;tag=z9hG4bK0156daf4
  638. CSeq: 102 OPTIONS
  639. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  640. Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
  641. Supported: replaces, 100rel, timer, norefersub
  642. Allow-Events: presence, message-summary, refer
  643. User-Agent: Telephone 1.2.6
  644. Content-Length: 0
  645.  
  646. <------------->
  647. --- (12 headers 0 lines) ---
  648. Really destroying SIP dialog '3fb7cf663a875af24e12c5d44d3d2c53@85.92.195.146:5160' Method: OPTIONS
  649.  
  650. <--- SIP read from UDP:82.147.28.150:61123 --->
  651.  
  652. <------------->
  653. Reliably Transmitting (NAT) to 46.31.231.185:5060:
  654. OPTIONS sip:sip.voipfone.net SIP/2.0
  655. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK061ef63e;rport
  656. Max-Forwards: 70
  657. From: "Unknown" <sip:30178956@85.92.195.146:5160>;tag=as5c49183d
  658. To: <sip:sip.voipfone.net>
  659. Contact: <sip:30178956@85.92.195.146:5160>
  660. Call-ID: 4adb839114e0599916481f76039333bc@85.92.195.146:5160
  661. CSeq: 102 OPTIONS
  662. User-Agent: FPBX-14.0.1.1(13.16.0)
  663. Date: Thu, 13 Jul 2017 15:22:12 GMT
  664. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  665. Supported: replaces, timer
  666. Content-Length: 0
  667.  
  668.  
  669. ---
  670.  

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