debug

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  1. [root@freepbx ~]# asterisk -rvvvvvvvvv
  2. Asterisk 13.16.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  3. Created by Mark Spencer <markster@digium.com>
  4. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  5. This is free software, with components licensed under the GNU General Public
  6. License version 2 and other licenses; you are welcome to redistribute it under
  7. certain conditions. Type 'core show license' for details.
  8. =========================================================================
  9. Connected to Asterisk 13.16.0 currently running on freepbx (pid = 2808)
  10. [2017-07-13 15:52:32] NOTICE[2906]: chan_sip.c:15722 sip_reregister:    -- Re-registration for  30178956@sip.voipfone.net
  11. REGISTER 12 headers, 0 lines
  12. Reliably Transmitting (NAT) to 46.31.231.185:5060:
  13. REGISTER sip:sip.voipfone.net SIP/2.0
  14. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK3d8912fb;rport
  15. Max-Forwards: 70
  16. From: <sip:30178956@sip.voipfone.net>;tag=as6f8ec4a6
  17. To: <sip:30178956@sip.voipfone.net>
  18. Call-ID: 7b23c7c724761e89583cbc41071062a4@127.0.0.1
  19. CSeq: 105 REGISTER
  20. Supported: replaces, timer
  21. User-Agent: FPBX-14.0.1.1(13.16.0)
  22. Authorization: Digest username="30178956", realm="asterisk", algorithm=MD5, uri="sip:sip.voipfone.net", nonce="frjjriea", response="fd6f94bfeb127dd6a610408727ae3367"
  23. Expires: 120
  24. Contact: <sip:30178956@85.92.195.146:5160>
  25. Content-Length: 0
  26.  
  27.  
  28. ---
  29.  
  30. <--- SIP read from UDP:46.31.231.185:5060 --->
  31. SIP/2.0 200 OK
  32. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK3d8912fb;rport
  33. From: <sip:30178956@sip.voipfone.net>;tag=as6f8ec4a6
  34. To: <sip:30178956@sip.voipfone.net>
  35. Call-ID: 7b23c7c724761e89583cbc41071062a4@127.0.0.1
  36. CSeq: 105 REGISTER
  37. Contact: <sip:30178956@85.92.195.146:5160>;expires=60
  38. Expires: 60
  39. Date: Thu, 13 Jul 2017 15:52:32 GMT
  40. Min-Expires: 60
  41. User-Agent: Voipfone
  42. Content-Length: 0
  43.  
  44. <------------->
  45. --- (12 headers 0 lines) ---
  46. [2017-07-13 15:52:32] NOTICE[2906]: chan_sip.c:24538 handle_response_register: Outbound Registration: Expiry for sip.voipfone.net is 60 sec (Scheduling reregistration in 45 s)
  47. Really destroying SIP dialog '7b23c7c724761e89583cbc41071062a4@127.0.0.1' Method: REGISTER
  48. freepbx*CLI> sip set debug on
  49. SIP Debugging re-enabled
  50.  
  51. <--- SIP read from UDP:82.147.28.150:61123 --->
  52. INVITE sip:07927800653@85.92.195.146:5160 SIP/2.0
  53. Via: SIP/2.0/UDP 172.16.103.99:61123;rport;branch=z9hG4bKPjLM4NWeeZpgh8fQyH-YVsNYG9mKPhVYF1
  54. Max-Forwards: 70
  55. From: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  56. To: <sip:07927800653@85.92.195.146>
  57. Contact: "Henry" <sip:3006@172.16.103.99:61123;ob>
  58. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  59. CSeq: 12392 INVITE
  60. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  61. Supported: replaces, 100rel, norefersub
  62. User-Agent: Telephone 1.2.6
  63. Content-Type: application/sdp
  64. Content-Length: 544
  65.  
  66. v=0
  67. o=- 3708949954 3708949954 IN IP4 172.16.103.99
  68. s=pjmedia
  69. b=AS:117
  70. t=0 0
  71. a=X-nat:0
  72. m=audio 4028 RTP/AVP 103 102 104 125 109 3 0 8 9 101
  73. c=IN IP4 172.16.103.99
  74. b=TIAS:96000
  75. a=rtcp:4029 IN IP4 172.16.103.99
  76. a=sendrecv
  77. a=rtpmap:103 speex/16000
  78. a=rtpmap:102 speex/8000
  79. a=rtpmap:104 speex/32000
  80. a=rtpmap:125 opus/48000/2
  81. a=fmtp:125 useinbandfec=1
  82. a=rtpmap:109 iLBC/8000
  83. a=fmtp:109 mode=30
  84. a=rtpmap:3 GSM/8000
  85. a=rtpmap:0 PCMU/8000
  86. a=rtpmap:8 PCMA/8000
  87. a=rtpmap:9 G722/8000
  88. a=rtpmap:101 telephone-event/8000
  89. a=fmtp:101 0-16
  90. <------------->
  91. --- (13 headers 24 lines) ---
  92. Sending to 82.147.28.150:61123 (NAT)
  93. Sending to 82.147.28.150:61123 (NAT)
  94. Using INVITE request as basis request - bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  95. Found peer '3006' for '3006' from 82.147.28.150:61123
  96.  
  97. <--- Reliably Transmitting (no NAT) to 82.147.28.150:61123 --->
  98. SIP/2.0 401 Unauthorized
  99. Via: SIP/2.0/UDP 172.16.103.99:61123;branch=z9hG4bKPjLM4NWeeZpgh8fQyH-YVsNYG9mKPhVYF1;received=82.147.28.150;rport=61123
  100. From: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  101. To: <sip:07927800653@85.92.195.146>;tag=as65335e5f
  102. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  103. CSeq: 12392 INVITE
  104. Server: FPBX-14.0.1.1(13.16.0)
  105. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  106. Supported: replaces, timer
  107. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ad8e6e2"
  108. Content-Length: 0
  109.  
  110.  
  111. <------------>
  112. Scheduling destruction of SIP dialog 'bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO' in 6400 ms (Method: INVITE)
  113.  
  114. <--- SIP read from UDP:82.147.28.150:61123 --->
  115. ACK sip:07927800653@85.92.195.146:5160 SIP/2.0
  116. Via: SIP/2.0/UDP 172.16.103.99:61123;rport;branch=z9hG4bKPjLM4NWeeZpgh8fQyH-YVsNYG9mKPhVYF1
  117. Max-Forwards: 70
  118. From: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  119. To: <sip:07927800653@85.92.195.146>;tag=as65335e5f
  120. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  121. CSeq: 12392 ACK
  122. Content-Length: 0
  123.  
  124. <------------->
  125. --- (8 headers 0 lines) ---
  126.  
  127. <--- SIP read from UDP:82.147.28.150:61123 --->
  128. INVITE sip:07927800653@85.92.195.146:5160 SIP/2.0
  129. Via: SIP/2.0/UDP 172.16.103.99:61123;rport;branch=z9hG4bKPjMFKJ6Nk7oc.d1ofPc.XsH8snKv.Rdwlk
  130. Max-Forwards: 70
  131. From: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  132. To: <sip:07927800653@85.92.195.146>
  133. Contact: "Henry" <sip:3006@172.16.103.99:61123;ob>
  134. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  135. CSeq: 12393 INVITE
  136. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  137. Supported: replaces, 100rel, norefersub
  138. User-Agent: Telephone 1.2.6
  139. Authorization: Digest username="3006", realm="asterisk", nonce="4ad8e6e2", uri="sip:07927800653@85.92.195.146:5160", response="41158cf5836d421943adc9fb776743ea", algorithm=MD5
  140. Content-Type: application/sdp
  141. Content-Length: 544
  142.  
  143. v=0
  144. o=- 3708949954 3708949954 IN IP4 172.16.103.99
  145. s=pjmedia
  146. b=AS:117
  147. t=0 0
  148. a=X-nat:0
  149. m=audio 4028 RTP/AVP 103 102 104 125 109 3 0 8 9 101
  150. c=IN IP4 172.16.103.99
  151. b=TIAS:96000
  152. a=rtcp:4029 IN IP4 172.16.103.99
  153. a=sendrecv
  154. a=rtpmap:103 speex/16000
  155. a=rtpmap:102 speex/8000
  156. a=rtpmap:104 speex/32000
  157. a=rtpmap:125 opus/48000/2
  158. a=fmtp:125 useinbandfec=1
  159. a=rtpmap:109 iLBC/8000
  160. a=fmtp:109 mode=30
  161. a=rtpmap:3 GSM/8000
  162. a=rtpmap:0 PCMU/8000
  163. a=rtpmap:8 PCMA/8000
  164. a=rtpmap:9 G722/8000
  165. a=rtpmap:101 telephone-event/8000
  166. a=fmtp:101 0-16
  167. <------------->
  168. --- (14 headers 24 lines) ---
  169. Sending to 82.147.28.150:61123 (no NAT)
  170. Using INVITE request as basis request - bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  171. Found peer '3006' for '3006' from 82.147.28.150:61123
  172.   == Using SIP RTP TOS bits 184
  173.   == Using SIP RTP CoS mark 5
  174. Found RTP audio format 103
  175. Found RTP audio format 102
  176. Found RTP audio format 104
  177. Found RTP audio format 125
  178. Found RTP audio format 109
  179. Found RTP audio format 3
  180. Found RTP audio format 0
  181. Found RTP audio format 8
  182. Found RTP audio format 9
  183. Found RTP audio format 101
  184. Found audio description format speex for ID 103
  185. Found audio description format speex for ID 102
  186. Found audio description format speex for ID 104
  187. Found audio description format opus for ID 125
  188. Found audio description format iLBC for ID 109
  189. Found audio description format GSM for ID 3
  190. Found audio description format PCMU for ID 0
  191. Found audio description format PCMA for ID 8
  192. Found audio description format G722 for ID 9
  193. Found audio description format telephone-event for ID 101
  194. 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)
  195. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  196. Peer audio RTP is at port 172.16.103.99:4028
  197. Looking for 07927800653 in from-internal (domain 85.92.195.146)
  198. sip_route_dump: route/path hop: <sip:3006@172.16.103.99:61123;ob>
  199.  
  200. <--- Transmitting (no NAT) to 82.147.28.150:61123 --->
  201. SIP/2.0 100 Trying
  202. Via: SIP/2.0/UDP 172.16.103.99:61123;branch=z9hG4bKPjMFKJ6Nk7oc.d1ofPc.XsH8snKv.Rdwlk;received=82.147.28.150;rport=61123
  203. From: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  204. To: <sip:07927800653@85.92.195.146>
  205. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  206. CSeq: 12393 INVITE
  207. Server: FPBX-14.0.1.1(13.16.0)
  208. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  209. Supported: replaces, timer
  210. Contact: <sip:07927800653@85.92.195.146:5160>
  211. Content-Length: 0
  212.  
  213.  
  214. <------------>
  215.     -- Executing [07927800653@from-internal:1] Macro("SIP/3006-00000008", "user-callerid,LIMIT,EXTERNAL,") in new stack
  216.     -- Executing [s@macro-user-callerid:1] Set("SIP/3006-00000008", "TOUCH_MONITOR=1499961154.14") in new stack
  217.     -- Executing [s@macro-user-callerid:2] Set("SIP/3006-00000008", "AMPUSER=3006") in new stack
  218.     -- Executing [s@macro-user-callerid:3] GotoIf("SIP/3006-00000008", "0?report") in new stack
  219.     -- Executing [s@macro-user-callerid:4] ExecIf("SIP/3006-00000008", "1?Set(__REALCALLERIDNUM=3006)") in new stack
  220.     -- Executing [s@macro-user-callerid:5] Set("SIP/3006-00000008", "AMPUSER=3006") in new stack
  221.     -- Executing [s@macro-user-callerid:6] GotoIf("SIP/3006-00000008", "0?limit") in new stack
  222.     -- Executing [s@macro-user-callerid:7] Set("SIP/3006-00000008", "AMPUSERCIDNAME=Southcoast Payments") in new stack
  223.     -- Executing [s@macro-user-callerid:8] GotoIf("SIP/3006-00000008", "0?report") in new stack
  224.     -- Executing [s@macro-user-callerid:9] Set("SIP/3006-00000008", "AMPUSERCID=3006") in new stack
  225.     -- Executing [s@macro-user-callerid:10] Set("SIP/3006-00000008", "__DIAL_OPTIONS=Ttr") in new stack
  226.     -- Executing [s@macro-user-callerid:11] Set("SIP/3006-00000008", "CALLERID(all)="Southcoast Payments" <3006>") in new stack
  227.     -- Executing [s@macro-user-callerid:12] GotoIf("SIP/3006-00000008", "0?limit") in new stack
  228.     -- Executing [s@macro-user-callerid:13] ExecIf("SIP/3006-00000008", "1?Set(GROUP(concurrency_limit)=3006)") in new stack
  229.     -- Executing [s@macro-user-callerid:14] ExecIf("SIP/3006-00000008", "0?Set(CHANNEL(language)=)") in new stack
  230.     -- Executing [s@macro-user-callerid:15] GotoIf("SIP/3006-00000008", "1?continue") in new stack
  231.     -- Goto (macro-user-callerid,s,29)
  232.     -- Executing [s@macro-user-callerid:29] Set("SIP/3006-00000008", "CALLERID(number)=3006") in new stack
  233.     -- Executing [s@macro-user-callerid:30] Set("SIP/3006-00000008", "CALLERID(name)=Southcoast Payments") in new stack
  234.     -- Executing [s@macro-user-callerid:31] GotoIf("SIP/3006-00000008", "0?cnum") in new stack
  235.     -- Executing [s@macro-user-callerid:32] Set("SIP/3006-00000008", "CDR(cnam)=Southcoast Payments") in new stack
  236.     -- Executing [s@macro-user-callerid:33] Set("SIP/3006-00000008", "CDR(cnum)=3006") in new stack
  237.     -- Executing [s@macro-user-callerid:34] Set("SIP/3006-00000008", "CHANNEL(language)=en") in new stack
  238.     -- Executing [07927800653@from-internal:2] Gosub("SIP/3006-00000008", "sub-record-check,s,1(out,07927800653,dontcare)") in new stack
  239.     -- Executing [s@sub-record-check:1] GotoIf("SIP/3006-00000008", "0?initialized") in new stack
  240.     -- Executing [s@sub-record-check:2] Set("SIP/3006-00000008", "__REC_STATUS=INITIALIZED") in new stack
  241.     -- Executing [s@sub-record-check:3] Set("SIP/3006-00000008", "NOW=1499961154") in new stack
  242.     -- Executing [s@sub-record-check:4] Set("SIP/3006-00000008", "__DAY=13") in new stack
  243.     -- Executing [s@sub-record-check:5] Set("SIP/3006-00000008", "__MONTH=07") in new stack
  244.     -- Executing [s@sub-record-check:6] Set("SIP/3006-00000008", "__YEAR=2017") in new stack
  245.     -- Executing [s@sub-record-check:7] Set("SIP/3006-00000008", "__TIMESTR=20170713-155234") in new stack
  246.     -- Executing [s@sub-record-check:8] Set("SIP/3006-00000008", "__FROMEXTEN=3006") in new stack
  247.     -- Executing [s@sub-record-check:9] Set("SIP/3006-00000008", "__MON_FMT=wav") in new stack
  248.     -- Executing [s@sub-record-check:10] NoOp("SIP/3006-00000008", "Recordings initialized") in new stack
  249.     -- Executing [s@sub-record-check:11] ExecIf("SIP/3006-00000008", "0?Set(ARG3=dontcare)") in new stack
  250.     -- Executing [s@sub-record-check:12] Set("SIP/3006-00000008", "REC_POLICY_MODE_SAVE=") in new stack
  251.     -- Executing [s@sub-record-check:13] ExecIf("SIP/3006-00000008", "0?Set(REC_STATUS=NO)") in new stack
  252.     -- Executing [s@sub-record-check:14] GotoIf("SIP/3006-00000008", "3?checkaction") in new stack
  253.     -- Goto (sub-record-check,s,17)
  254.     -- Executing [s@sub-record-check:17] GotoIf("SIP/3006-00000008", "1?sub-record-check,out,1") in new stack
  255.     -- Goto (sub-record-check,out,1)
  256.     -- Executing [out@sub-record-check:1] NoOp("SIP/3006-00000008", "Outbound Recording Check from 3006 to 07927800653") in new stack
  257.     -- Executing [out@sub-record-check:2] Set("SIP/3006-00000008", "RECMODE=dontcare") in new stack
  258.     -- Executing [out@sub-record-check:3] ExecIf("SIP/3006-00000008", "1?Goto(routewins)") in new stack
  259.     -- Goto (sub-record-check,out,7)
  260.     -- Executing [out@sub-record-check:7] Gosub("SIP/3006-00000008", "recordcheck,1(dontcare,out,07927800653)") in new stack
  261.     -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/3006-00000008", "Starting recording check against dontcare") in new stack
  262.     -- Executing [recordcheck@sub-record-check:2] Goto("SIP/3006-00000008", "dontcare") in new stack
  263.     -- Goto (sub-record-check,recordcheck,3)
  264.     -- Executing [recordcheck@sub-record-check:3] Return("SIP/3006-00000008", "") in new stack
  265.     -- Executing [out@sub-record-check:8] Return("SIP/3006-00000008", "") in new stack
  266.     -- Executing [07927800653@from-internal:3] ExecIf("SIP/3006-00000008", "0 ?Set(CDR(accountcode)=)") in new stack
  267.     -- Executing [07927800653@from-internal:4] Set("SIP/3006-00000008", "MOHCLASS=default") in new stack
  268.     -- Executing [07927800653@from-internal:5] Set("SIP/3006-00000008", "_NODEST=") in new stack
  269.     -- Executing [07927800653@from-internal:6] Macro("SIP/3006-00000008", "dialout-trunk,1,07927800653,,off") in new stack
  270.     -- Executing [s@macro-dialout-trunk:1] Set("SIP/3006-00000008", "DIAL_TRUNK=1") in new stack
  271.     -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/3006-00000008", "0?sub-pincheck,s,1()") in new stack
  272.     -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/3006-00000008", "0?disabletrunk,1") in new stack
  273.     -- Executing [s@macro-dialout-trunk:4] Set("SIP/3006-00000008", "DIAL_NUMBER=07927800653") in new stack
  274.     -- Executing [s@macro-dialout-trunk:5] Set("SIP/3006-00000008", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
  275.     -- Executing [s@macro-dialout-trunk:6] Set("SIP/3006-00000008", "OUTBOUND_GROUP=OUT_1") in new stack
  276.     -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/3006-00000008", "1?nomax") in new stack
  277.     -- Goto (macro-dialout-trunk,s,9)
  278.     -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/3006-00000008", "0?skipoutcid") in new stack
  279.     -- Executing [s@macro-dialout-trunk:10] Set("SIP/3006-00000008", "DIAL_TRUNK_OPTIONS=T") in new stack
  280.     -- Executing [s@macro-dialout-trunk:11] Macro("SIP/3006-00000008", "outbound-callerid,1") in new stack
  281.     -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/3006-00000008", "0?Set(CALLERPRES(name-pres)=)") in new stack
  282.     -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/3006-00000008", "0?Set(CALLERPRES(num-pres)=)") in new stack
  283.     -- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/3006-00000008", "0?Set(REALCALLERIDNUM=3006)") in new stack
  284.     -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/3006-00000008", "1?normcid") in new stack
  285.     -- Goto (macro-outbound-callerid,s,7)
  286.     -- Executing [s@macro-outbound-callerid:7] Set("SIP/3006-00000008", "USEROUTCID=") in new stack
  287.     -- Executing [s@macro-outbound-callerid:8] Set("SIP/3006-00000008", "EMERGENCYCID=") in new stack
  288.     -- Executing [s@macro-outbound-callerid:9] Set("SIP/3006-00000008", "TRUNKOUTCID=hidden") in new stack
  289.     -- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/3006-00000008", "1?trunkcid") in new stack
  290.     -- Goto (macro-outbound-callerid,s,15)
  291.     -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/3006-00000008", "1?Set(CALLERID(all)=hidden)") in new stack
  292.     -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/3006-00000008", "0?Set(CALLERID(all)=)") in new stack
  293.     -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/3006-00000008", "0?Set(CALLERID(all)=)") in new stack
  294.     -- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/3006-00000008", "1?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
  295.     -- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/3006-00000008", "1?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
  296.     -- Executing [s@macro-outbound-callerid:20] Set("SIP/3006-00000008", "CDR(outbound_cnum)=") in new stack
  297.     -- Executing [s@macro-outbound-callerid:21] Set("SIP/3006-00000008", "CDR(outbound_cnam)=hidden") in new stack
  298.     -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/3006-00000008", "0?sub-flp-1,s,1()") in new stack
  299.     -- Executing [s@macro-dialout-trunk:13] Set("SIP/3006-00000008", "OUTNUM=07927800653") in new stack
  300.     -- Executing [s@macro-dialout-trunk:14] Set("SIP/3006-00000008", "custom=SIP/voip") in new stack
  301.     -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/3006-00000008", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
  302.     -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/3006-00000008", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
  303.     -- Executing [s@macro-dialout-trunk:17] Macro("SIP/3006-00000008", "dialout-trunk-predial-hook,") in new stack
  304.     -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/3006-00000008", "") in new stack
  305.     -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/3006-00000008", "0?skipcrm") in new stack
  306.     -- Executing [s@macro-dialout-trunk:19] Set("SIP/3006-00000008", "__CRM_DIRECTION=OUTBOUND") in new stack
  307.     -- Executing [s@macro-dialout-trunk:20] Set("SIP/3006-00000008", "__CRM_DESTINATION=07927800653") in new stack
  308.     -- Executing [s@macro-dialout-trunk:21] Set("SIP/3006-00000008", "__CRM_SOURCE=3006") in new stack
  309.     -- Executing [s@macro-dialout-trunk:22] AGI("SIP/3006-00000008", "sangomacrm.agi") in new stack
  310.     -- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
  311.     -- <SIP/3006-00000008>AGI Script sangomacrm.agi completed, returning 0
  312.     -- Executing [s@macro-dialout-trunk:23] Set("SIP/3006-00000008", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
  313.     -- Executing [s@macro-dialout-trunk:24] NoOp("SIP/3006-00000008", "CRM Finished") in new stack
  314.     -- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/3006-00000008", "0?bypass,1") in new stack
  315.     -- Executing [s@macro-dialout-trunk:26] ExecIf("SIP/3006-00000008", "1?Set(CONNECTEDLINE(num,i)=07927800653)") in new stack
  316.     -- Executing [s@macro-dialout-trunk:27] ExecIf("SIP/3006-00000008", "0?Set(CONNECTEDLINE(name,i)=CID:)") in new stack
  317.     -- Executing [s@macro-dialout-trunk:28] ExecIf("SIP/3006-00000008", "1?Set(CONNECTEDLINE(name,i)=CID:(Hidden))") in new stack
  318.     -- Executing [s@macro-dialout-trunk:29] GotoIf("SIP/3006-00000008", "0?customtrunk") in new stack
  319.     -- Executing [s@macro-dialout-trunk:30] Dial("SIP/3006-00000008", "SIP/voip/07927800653,300,T") in new stack
  320.   == Using SIP RTP TOS bits 184
  321.   == Using SIP RTP CoS mark 5
  322. Audio is at 15250
  323. Adding codec alaw to SDP
  324. Adding non-codec 0x1 (telephone-event) to SDP
  325. Reliably Transmitting (no NAT) to 46.31.231.185:5060:
  326. INVITE sip:07927800653@sip.voipfone.net SIP/2.0
  327. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK28bcb06c
  328. Max-Forwards: 70
  329. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6181670c
  330. To: <sip:07927800653@sip.voipfone.net>
  331. Contact: <sip:30178956@85.92.195.146:5160>
  332. Call-ID: 66879c724b60740f4fd097e8233e2d23@85.92.195.146:5160
  333. CSeq: 102 INVITE
  334. User-Agent: FPBX-14.0.1.1(13.16.0)
  335. Date: Thu, 13 Jul 2017 15:52:34 GMT
  336. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  337. Supported: replaces, timer
  338. Content-Type: application/sdp
  339. Content-Length: 254
  340.  
  341. v=0
  342. o=root 1444822018 1444822018 IN IP4 85.92.195.146
  343. s=Asterisk PBX 13.16.0
  344. c=IN IP4 85.92.195.146
  345. t=0 0
  346. m=audio 15250 RTP/AVP 8 101
  347. a=rtpmap:8 PCMA/8000
  348. a=rtpmap:101 telephone-event/8000
  349. a=fmtp:101 0-16
  350. a=ptime:20
  351. a=maxptime:150
  352. a=sendrecv
  353.  
  354. ---
  355.     -- Called SIP/voip/07927800653
  356.  
  357. <--- SIP read from UDP:46.31.231.185:5060 --->
  358. SIP/2.0 407 Proxy Authentication Required
  359. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK28bcb06c;received=85.92.195.146;rport=5160
  360. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6181670c
  361. To: <sip:07927800653@sip.voipfone.net>;tag=VF8b4ffd7523a056a4ffda5b18c0ff
  362. Call-ID: 66879c724b60740f4fd097e8233e2d23@85.92.195.146:5160
  363. CSeq: 102 INVITE
  364. User-Agent: Voipfone Sip Network
  365. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  366. Contact: <sip:07927800653@46.31.231.185>
  367. Proxy-Authenticate: Digest realm="asterisk", nonce="59a0bed2"
  368. Content-Length: 0
  369.  
  370. <------------->
  371. --- (11 headers 0 lines) ---
  372. Transmitting (no NAT) to 46.31.231.185:5060:
  373. ACK sip:07927800653@sip.voipfone.net SIP/2.0
  374. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK28bcb06c
  375. Max-Forwards: 70
  376. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6181670c
  377. To: <sip:07927800653@sip.voipfone.net>;tag=VF8b4ffd7523a056a4ffda5b18c0ff
  378. Contact: <sip:30178956@85.92.195.146:5160>
  379. Call-ID: 66879c724b60740f4fd097e8233e2d23@85.92.195.146:5160
  380. CSeq: 102 ACK
  381. User-Agent: FPBX-14.0.1.1(13.16.0)
  382. Content-Length: 0
  383.  
  384.  
  385. ---
  386. Audio is at 15250
  387. Adding codec alaw to SDP
  388. Adding non-codec 0x1 (telephone-event) to SDP
  389. Reliably Transmitting (no NAT) to 46.31.231.185:5060:
  390. INVITE sip:07927800653@sip.voipfone.net SIP/2.0
  391. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK36dde228
  392. Max-Forwards: 70
  393. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6181670c
  394. To: <sip:07927800653@sip.voipfone.net>
  395. Contact: <sip:30178956@85.92.195.146:5160>
  396. Call-ID: 66879c724b60740f4fd097e8233e2d23@85.92.195.146:5160
  397. CSeq: 103 INVITE
  398. User-Agent: FPBX-14.0.1.1(13.16.0)
  399. Proxy-Authorization: Digest username="30178956", realm="asterisk", algorithm=MD5, uri="sip:07927800653@sip.voipfone.net", nonce="59a0bed2", response="d10d116be3386e468c9f1778d66d8211"
  400. Date: Thu, 13 Jul 2017 15:52:34 GMT
  401. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  402. Supported: replaces, timer
  403. Content-Type: application/sdp
  404. Content-Length: 254
  405.  
  406. v=0
  407. o=root 1444822018 1444822019 IN IP4 85.92.195.146
  408. s=Asterisk PBX 13.16.0
  409. c=IN IP4 85.92.195.146
  410. t=0 0
  411. m=audio 15250 RTP/AVP 8 101
  412. a=rtpmap:8 PCMA/8000
  413. a=rtpmap:101 telephone-event/8000
  414. a=fmtp:101 0-16
  415. a=ptime:20
  416. a=maxptime:150
  417. a=sendrecv
  418.  
  419. ---
  420.  
  421. <--- SIP read from UDP:46.31.231.185:5060 --->
  422. SIP/2.0 100 Trying
  423. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK36dde228;received=85.92.195.146;rport=5160
  424. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6181670c
  425. To: <sip:07927800653@sip.voipfone.net>;tag=VF4132a7f4e9d1abf0bc1416bb688f
  426. Call-ID: 66879c724b60740f4fd097e8233e2d23@85.92.195.146:5160
  427. CSeq: 103 INVITE
  428. User-Agent: Voipfone Sip Network
  429. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  430. Contact: <sip:07927800653@46.31.231.185>
  431. Content-Length: 0
  432.  
  433. <------------->
  434. --- (10 headers 0 lines) ---
  435.  
  436. <--- SIP read from UDP:46.31.231.185:5060 --->
  437. SIP/2.0 183 Session Progress
  438. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK36dde228;received=85.92.195.146;rport=5160
  439. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6181670c
  440. To: <sip:07927800653@sip.voipfone.net>;tag=VF4132a7f4e9d1abf0bc1416bb688f
  441. Call-ID: 66879c724b60740f4fd097e8233e2d23@85.92.195.146:5160
  442. CSeq: 103 INVITE
  443. User-Agent: Voipfone Sip Network
  444. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  445. Contact: <sip:07927800653@46.31.231.185>
  446. Content-Type: application/sdp
  447. Content-Length: 333
  448.  
  449. v=0
  450. o=root 2431 2431 IN IP4 46.31.231.185
  451. s=session
  452. c=IN IP4 46.31.231.185
  453. t=0 0
  454. m=audio 48100 RTP/AVP 8 2 97 3 110 101
  455. a=sendrecv
  456. a=rtpmap:8 PCMA/8000
  457. a=rtpmap:2 G726-32/8000
  458. a=rtpmap:97 iLBC/8000
  459. a=rtpmap:3 GSM/8000
  460. a=rtpmap:110 speex/8000
  461. a=rtpmap:101 telephone-event/8000
  462. a=fmtp:101 0-16
  463. a=silenceSupp:off - - - -
  464. <------------->
  465. --- (11 headers 15 lines) ---
  466. sip_route_dump: route/path hop: <sip:07927800653@46.31.231.185>
  467. Found RTP audio format 8
  468. Found RTP audio format 2
  469. Found RTP audio format 97
  470. Found RTP audio format 3
  471. Found RTP audio format 110
  472. Found RTP audio format 101
  473. Found audio description format PCMA for ID 8
  474. Found audio description format G726-32 for ID 2
  475. Found audio description format iLBC for ID 97
  476. Found audio description format GSM for ID 3
  477. Found audio description format speex for ID 110
  478. Found audio description format telephone-event for ID 101
  479. Capabilities: us - (alaw), peer - audio=(g726|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (alaw)
  480. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  481. Peer audio RTP is at port 46.31.231.185:48100
  482.     -- SIP/voip-00000009 is making progress passing it to SIP/3006-00000008
  483. Audio is at 14998
  484. Adding codec ulaw to SDP
  485. Adding codec alaw to SDP
  486. Adding codec gsm to SDP
  487. Adding codec g722 to SDP
  488. Adding codec speex to SDP
  489. Adding non-codec 0x1 (telephone-event) to SDP
  490.  
  491. <--- Transmitting (no NAT) to 82.147.28.150:61123 --->
  492. SIP/2.0 183 Session Progress
  493. Via: SIP/2.0/UDP 172.16.103.99:61123;branch=z9hG4bKPjMFKJ6Nk7oc.d1ofPc.XsH8snKv.Rdwlk;received=82.147.28.150;rport=61123
  494. From: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  495. To: <sip:07927800653@85.92.195.146>;tag=as5d0fbd50
  496. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  497. CSeq: 12393 INVITE
  498. Server: FPBX-14.0.1.1(13.16.0)
  499. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  500. Supported: replaces, timer
  501. Contact: <sip:07927800653@85.92.195.146:5160>
  502. Content-Type: application/sdp
  503. Content-Length: 353
  504.  
  505. v=0
  506. o=root 1434435323 1434435323 IN IP4 85.92.195.146
  507. s=Asterisk PBX 13.16.0
  508. c=IN IP4 85.92.195.146
  509. t=0 0
  510. m=audio 14998 RTP/AVP 0 8 3 9 102 101
  511. a=rtpmap:0 PCMU/8000
  512. a=rtpmap:8 PCMA/8000
  513. a=rtpmap:3 GSM/8000
  514. a=rtpmap:9 G722/8000
  515. a=rtpmap:102 speex/8000
  516. a=rtpmap:101 telephone-event/8000
  517. a=fmtp:101 0-16
  518. a=ptime:20
  519. a=maxptime:60
  520. a=sendrecv
  521.  
  522. <------------>
  523.        > 0x7fbf4879c200 -- Probation passed - setting RTP source address to 46.31.231.185:48100
  524.        > 0x7fbf44073260 -- Probation passed - setting RTP source address to 82.147.28.150:4028
  525.  
  526. <--- SIP read from UDP:46.31.231.185:5060 --->
  527. SIP/2.0 200 OK
  528. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK36dde228;received=85.92.195.146;rport=5160
  529. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6181670c
  530. To: <sip:07927800653@sip.voipfone.net>;tag=VF4132a7f4e9d1abf0bc1416bb688f
  531. Call-ID: 66879c724b60740f4fd097e8233e2d23@85.92.195.146:5160
  532. CSeq: 103 INVITE
  533. User-Agent: Voipfone Sip Network
  534. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  535. Contact: <sip:07927800653@46.31.231.185>
  536. Content-Type: application/sdp
  537. Content-Length: 333
  538.  
  539. v=0
  540. o=root 2431 2432 IN IP4 46.31.231.185
  541. s=session
  542. c=IN IP4 46.31.231.185
  543. t=0 0
  544. m=audio 48100 RTP/AVP 8 2 97 3 110 101
  545. a=sendrecv
  546. a=rtpmap:8 PCMA/8000
  547. a=rtpmap:2 G726-32/8000
  548. a=rtpmap:97 iLBC/8000
  549. a=rtpmap:3 GSM/8000
  550. a=rtpmap:110 speex/8000
  551. a=rtpmap:101 telephone-event/8000
  552. a=fmtp:101 0-16
  553. a=silenceSupp:off - - - -
  554. <------------->
  555. --- (11 headers 15 lines) ---
  556. Found RTP audio format 8
  557. Found RTP audio format 2
  558. Found RTP audio format 97
  559. Found RTP audio format 3
  560. Found RTP audio format 110
  561. Found RTP audio format 101
  562. Found audio description format PCMA for ID 8
  563. Found audio description format G726-32 for ID 2
  564. Found audio description format iLBC for ID 97
  565. Found audio description format GSM for ID 3
  566. Found audio description format speex for ID 110
  567. Found audio description format telephone-event for ID 101
  568. Capabilities: us - (alaw), peer - audio=(g726|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (alaw)
  569. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  570. Peer audio RTP is at port 46.31.231.185:48100
  571. sip_route_dump: route/path hop: <sip:07927800653@46.31.231.185>
  572. set_destination: Parsing <sip:07927800653@46.31.231.185> for address/port to send to
  573. set_destination: set destination to 46.31.231.185:5060
  574. Transmitting (no NAT) to 46.31.231.185:5060:
  575. ACK sip:07927800653@46.31.231.185 SIP/2.0
  576. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK353d1d93
  577. Max-Forwards: 70
  578. From: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6181670c
  579. To: <sip:07927800653@sip.voipfone.net>;tag=VF4132a7f4e9d1abf0bc1416bb688f
  580. Contact: <sip:30178956@85.92.195.146:5160>
  581. Call-ID: 66879c724b60740f4fd097e8233e2d23@85.92.195.146:5160
  582. CSeq: 103 ACK
  583. User-Agent: FPBX-14.0.1.1(13.16.0)
  584. Content-Length: 0
  585.  
  586.  
  587. ---
  588.     -- SIP/voip-00000009 answered SIP/3006-00000008
  589. Audio is at 14998
  590. Adding codec ulaw to SDP
  591. Adding codec alaw to SDP
  592. Adding codec gsm to SDP
  593. Adding codec g722 to SDP
  594. Adding codec speex to SDP
  595. Adding non-codec 0x1 (telephone-event) to SDP
  596.  
  597. <--- Reliably Transmitting (no NAT) to 82.147.28.150:61123 --->
  598. SIP/2.0 200 OK
  599. Via: SIP/2.0/UDP 172.16.103.99:61123;branch=z9hG4bKPjMFKJ6Nk7oc.d1ofPc.XsH8snKv.Rdwlk;received=82.147.28.150;rport=61123
  600. From: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  601. To: <sip:07927800653@85.92.195.146>;tag=as5d0fbd50
  602. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  603. CSeq: 12393 INVITE
  604. Server: FPBX-14.0.1.1(13.16.0)
  605. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  606. Supported: replaces, timer
  607. Contact: <sip:07927800653@85.92.195.146:5160>
  608. P-Asserted-Identity: "CID:(Hidden)" <sip:07927800653@85.92.195.146>
  609. Content-Type: application/sdp
  610. Content-Length: 353
  611.  
  612. v=0
  613. o=root 1434435323 1434435323 IN IP4 85.92.195.146
  614. s=Asterisk PBX 13.16.0
  615. c=IN IP4 85.92.195.146
  616. t=0 0
  617. m=audio 14998 RTP/AVP 0 8 3 9 102 101
  618. a=rtpmap:0 PCMU/8000
  619. a=rtpmap:8 PCMA/8000
  620. a=rtpmap:3 GSM/8000
  621. a=rtpmap:9 G722/8000
  622. a=rtpmap:102 speex/8000
  623. a=rtpmap:101 telephone-event/8000
  624. a=fmtp:101 0-16
  625. a=ptime:20
  626. a=maxptime:60
  627. a=sendrecv
  628.  
  629. <------------>
  630.  
  631. <--- SIP read from UDP:82.147.28.150:61123 --->
  632. ACK sip:07927800653@85.92.195.146:5160 SIP/2.0
  633. Via: SIP/2.0/UDP 172.16.103.99:61123;rport;branch=z9hG4bKPjrr5pJYDXDvd4ravfnX7nxyoI8A6jQP-d
  634. Max-Forwards: 70
  635. From: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  636. To: <sip:07927800653@85.92.195.146>;tag=as5d0fbd50
  637. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  638. CSeq: 12393 ACK
  639. Content-Length: 0
  640.  
  641. <------------->
  642. --- (8 headers 0 lines) ---
  643.  
  644. <--- SIP read from UDP:82.147.28.150:61123 --->
  645. INVITE sip:07927800653@85.92.195.146:5160 SIP/2.0
  646. Via: SIP/2.0/UDP 172.16.103.99:61123;rport;branch=z9hG4bKPjMxy4wc9xiFKpq63yzsU0u6vqhU2DV.wh
  647. Max-Forwards: 70
  648. From: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  649. To: <sip:07927800653@85.92.195.146>;tag=as5d0fbd50
  650. Contact: "Henry" <sip:3006@172.16.103.99:61123;ob>
  651. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  652. CSeq: 12394 INVITE
  653. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  654. Supported: replaces, 100rel, norefersub
  655. Content-Type: application/sdp
  656. Content-Length: 278
  657.  
  658. v=0
  659. o=- 3708949954 3708949955 IN IP4 172.16.103.99
  660. s=pjmedia
  661. b=AS:117
  662. t=0 0
  663. a=X-nat:0
  664. m=audio 4028 RTP/AVP 0 101
  665. c=IN IP4 172.16.103.99
  666. b=TIAS:96000
  667. a=rtcp:4029 IN IP4 172.16.103.99
  668. a=rtpmap:0 PCMU/8000
  669. a=rtpmap:101 telephone-event/8000
  670. a=fmtp:101 0-16
  671. a=sendrecv
  672. <------------->
  673. --- (12 headers 14 lines) ---
  674. Sending to 82.147.28.150:61123 (no NAT)
  675. Found RTP audio format 0
  676. Found RTP audio format 101
  677. Found audio description format PCMU for ID 0
  678. Found audio description format telephone-event for ID 101
  679. Capabilities: us - (ulaw|alaw|gsm|g726|g722|g723|speex|siren7|adpcm|g719|g729|slin), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
  680. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  681. Peer audio RTP is at port 172.16.103.99:4028
  682.  
  683. <--- Transmitting (no NAT) to 82.147.28.150:61123 --->
  684. SIP/2.0 100 Trying
  685. Via: SIP/2.0/UDP 172.16.103.99:61123;branch=z9hG4bKPjMxy4wc9xiFKpq63yzsU0u6vqhU2DV.wh;received=82.147.28.150;rport=61123
  686. From: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  687. To: <sip:07927800653@85.92.195.146>;tag=as5d0fbd50
  688. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  689. CSeq: 12394 INVITE
  690. Server: FPBX-14.0.1.1(13.16.0)
  691. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  692. Supported: replaces, timer
  693. Contact: <sip:07927800653@85.92.195.146:5160>
  694. Content-Length: 0
  695.  
  696.  
  697. <------------>
  698. Audio is at 14998
  699. Adding codec ulaw to SDP
  700. Adding non-codec 0x1 (telephone-event) to SDP
  701.  
  702. <--- Reliably Transmitting (no NAT) to 82.147.28.150:61123 --->
  703. SIP/2.0 200 OK
  704. Via: SIP/2.0/UDP 172.16.103.99:61123;branch=z9hG4bKPjMxy4wc9xiFKpq63yzsU0u6vqhU2DV.wh;received=82.147.28.150;rport=61123
  705. From: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  706. To: <sip:07927800653@85.92.195.146>;tag=as5d0fbd50
  707. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  708. CSeq: 12394 INVITE
  709. Server: FPBX-14.0.1.1(13.16.0)
  710. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  711. Supported: replaces, timer
  712. Contact: <sip:07927800653@85.92.195.146:5160>
  713. Content-Type: application/sdp
  714. Content-Length: 254
  715.  
  716. v=0
  717. o=root 1434435323 1434435324 IN IP4 85.92.195.146
  718. s=Asterisk PBX 13.16.0
  719. c=IN IP4 85.92.195.146
  720. t=0 0
  721. m=audio 14998 RTP/AVP 0 101
  722. a=rtpmap:0 PCMU/8000
  723. a=rtpmap:101 telephone-event/8000
  724. a=fmtp:101 0-16
  725. a=ptime:20
  726. a=maxptime:150
  727. a=sendrecv
  728.  
  729. <------------>
  730.     -- Channel SIP/voip-00000009 joined 'simple_bridge' basic-bridge <24eb68d1-16dd-4973-9618-0dcd0497374d>
  731.        > 0x7fbf4879c200 -- Probation passed - setting RTP source address to 46.31.231.185:48100
  732.     -- Channel SIP/3006-00000008 joined 'simple_bridge' basic-bridge <24eb68d1-16dd-4973-9618-0dcd0497374d>
  733.  
  734. <--- SIP read from UDP:82.147.28.150:61123 --->
  735. ACK sip:07927800653@85.92.195.146:5160 SIP/2.0
  736. Via: SIP/2.0/UDP 172.16.103.99:61123;rport;branch=z9hG4bKPjWIJhEx3dz3vkg8fURa.DPyutAZ22Vk91
  737. Max-Forwards: 70
  738. From: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  739. To: <sip:07927800653@85.92.195.146>;tag=as5d0fbd50
  740. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  741. CSeq: 12394 ACK
  742. Content-Length: 0
  743.  
  744. <------------->
  745. --- (8 headers 0 lines) ---
  746.        > 0x7fbf44073260 -- Probation passed - setting RTP source address to 82.147.28.150:4028
  747.  
  748. <--- SIP read from UDP:46.31.231.185:5060 --->
  749. BYE sip:30178956@85.92.195.146:5160 SIP/2.0
  750. Via: SIP/2.0/UDP 46.31.231.185:5060;branch=z9hG4bK377c6fc6;rport
  751. From: <sip:07927800653@sip.voipfone.net>;tag=VF4132a7f4e9d1abf0bc1416bb688f
  752. To: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6181670c
  753. Contact: <sip:07927800653@46.31.231.185>
  754. Call-ID: 66879c724b60740f4fd097e8233e2d23@85.92.195.146:5160
  755. CSeq: 102 BYE
  756. User-Agent: Voipfone Sip Network
  757. Max-Forwards: 70
  758. Content-Length: 0
  759.  
  760. <------------->
  761. --- (10 headers 0 lines) ---
  762. Sending to 46.31.231.185:5060 (no NAT)
  763. Scheduling destruction of SIP dialog '66879c724b60740f4fd097e8233e2d23@85.92.195.146:5160' in 6400 ms (Method: BYE)
  764.  
  765. <--- Transmitting (no NAT) to 46.31.231.185:5060 --->
  766. SIP/2.0 200 OK
  767. Via: SIP/2.0/UDP 46.31.231.185:5060;branch=z9hG4bK377c6fc6;received=46.31.231.185;rport=5060
  768. From: <sip:07927800653@sip.voipfone.net>;tag=VF4132a7f4e9d1abf0bc1416bb688f
  769. To: "Anonymous" <sip:30178956@anonymous.invalid:5160>;tag=as6181670c
  770. Call-ID: 66879c724b60740f4fd097e8233e2d23@85.92.195.146:5160
  771. CSeq: 102 BYE
  772. Server: FPBX-14.0.1.1(13.16.0)
  773. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  774. Supported: replaces, timer
  775. Content-Length: 0
  776.  
  777.  
  778. <------------>
  779.     -- Channel SIP/voip-00000009 left 'simple_bridge' basic-bridge <24eb68d1-16dd-4973-9618-0dcd0497374d>
  780.     -- Channel SIP/3006-00000008 left 'simple_bridge' basic-bridge <24eb68d1-16dd-4973-9618-0dcd0497374d>
  781.   == Spawn extension (macro-dialout-trunk, s, 30) exited non-zero on 'SIP/3006-00000008' in macro 'dialout-trunk'
  782.   == Spawn extension (from-internal, 07927800653, 6) exited non-zero on 'SIP/3006-00000008'
  783.     -- Executing [h@from-internal:1] Macro("SIP/3006-00000008", "hangupcall") in new stack
  784.     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/3006-00000008", "1?theend") in new stack
  785.     -- Goto (macro-hangupcall,s,3)
  786.     -- Executing [s@macro-hangupcall:3] ExecIf("SIP/3006-00000008", "0?Set(CDR(recordingfile)=)") in new stack
  787.     -- Executing [s@macro-hangupcall:4] Hangup("SIP/3006-00000008", "") in new stack
  788.   == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/3006-00000008' in macro 'hangupcall'
  789.   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3006-00000008'
  790.     -- SIP/3006-00000008 Internal Gosub(crm-hangup,s,1) start
  791.     -- Executing [s@crm-hangup:1] NoOp("SIP/3006-00000008", "Sending Hangup to CRM") in new stack
  792.     -- Executing [s@crm-hangup:2] NoOp("SIP/3006-00000008", "HANGUP CAUSE: 16") in new stack
  793.     -- Executing [s@crm-hangup:3] ExecIf("SIP/3006-00000008", "0?Set(__CRM_VOICEMAIL=)") in new stack
  794.     -- Executing [s@crm-hangup:4] NoOp("SIP/3006-00000008", "MASTER CHANNEL: 1499961154.14 = 1499961154.14") in new stack
  795.     -- Executing [s@crm-hangup:5] GotoIf("SIP/3006-00000008", "0?return") in new stack
  796.     -- Executing [s@crm-hangup:6] Set("SIP/3006-00000008", "__CRM_HANGUP=1") in new stack
  797.     -- Executing [s@crm-hangup:7] AGI("SIP/3006-00000008", "sangomacrm.agi") in new stack
  798.     -- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
  799.     -- <SIP/3006-00000008>AGI Script sangomacrm.agi completed, returning 0
  800.     -- Executing [s@crm-hangup:8] Return("SIP/3006-00000008", "") in new stack
  801.   == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3006-00000008'
  802.     -- SIP/3006-00000008 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
  803. Scheduling destruction of SIP dialog 'bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO' in 6400 ms (Method: ACK)
  804. set_destination: Parsing <sip:3006@172.16.103.99:61123;ob> for address/port to send to
  805. set_destination: set destination to 172.16.103.99:61123
  806. Reliably Transmitting (no NAT) to 172.16.103.99:61123:
  807. BYE sip:3006@172.16.103.99:61123;ob SIP/2.0
  808. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK68c51198;rport
  809. Max-Forwards: 70
  810. From: <sip:07927800653@85.92.195.146>;tag=as5d0fbd50
  811. To: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  812. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  813. CSeq: 102 BYE
  814. User-Agent: FPBX-14.0.1.1(13.16.0)
  815. Proxy-Authorization: Digest username="3006", realm="asterisk", algorithm=MD5, uri="sip:85.92.195.146", nonce="4ad8e6e2", response="a4de7974f51682863ba3f228d3968a80"
  816. X-Asterisk-HangupCause: Normal Clearing
  817. X-Asterisk-HangupCauseCode: 16
  818. Content-Length: 0
  819.  
  820.  
  821. ---
  822.  
  823. <--- SIP read from UDP:82.147.28.150:61123 --->
  824. SIP/2.0 200 OK
  825. Via: SIP/2.0/UDP 85.92.195.146:5160;rport=5160;received=85.92.195.146;branch=z9hG4bK68c51198
  826. Call-ID: bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO
  827. From: <sip:07927800653@85.92.195.146>;tag=as5d0fbd50
  828. To: "Henry" <sip:3006@85.92.195.146>;tag=KeK4b-r2NDGlcH4XvBgRRNQJUoC9KRmj
  829. CSeq: 102 BYE
  830. Content-Length: 0
  831.  
  832. <------------->
  833. --- (7 headers 0 lines) ---
  834. SIP Response message for INCOMING dialog BYE arrived
  835. Really destroying SIP dialog 'bYPOZHhn4OLOcKPI4ptmIc45SnssrFfO' Method: ACK
  836.  
  837. <--- SIP read from UDP:82.147.28.150:61123 --->
  838.  
  839. <------------->
  840. Really destroying SIP dialog '66879c724b60740f4fd097e8233e2d23@85.92.195.146:5160' Method: BYE
  841.  
  842. <--- SIP read from UDP:82.147.28.150:61123 --->
  843.  
  844. <------------->
  845. Reliably Transmitting (no NAT) to 46.31.231.185:5060:
  846. OPTIONS sip:sip.voipfone.net SIP/2.0
  847. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK019f7fd5
  848. Max-Forwards: 70
  849. From: "Unknown" <sip:30178956@85.92.195.146:5160>;tag=as363c0203
  850. To: <sip:sip.voipfone.net>
  851. Contact: <sip:30178956@85.92.195.146:5160>
  852. Call-ID: 428a79340af7ad8c5e34b19d627d80c2@85.92.195.146:5160
  853. CSeq: 102 OPTIONS
  854. User-Agent: FPBX-14.0.1.1(13.16.0)
  855. Date: Thu, 13 Jul 2017 15:53:02 GMT
  856. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  857. Supported: replaces, timer
  858. Content-Length: 0
  859.  
  860.  
  861. ---
  862.  
  863. <--- SIP read from UDP:46.31.231.185:5060 --->
  864. SIP/2.0 200 OK
  865. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK019f7fd5;received=85.92.195.146;rport=5160
  866. From: "Unknown" <sip:30178956@85.92.195.146:5160>;tag=as363c0203
  867. To: <sip:sip.voipfone.net>;tag=VFb015619f12ff7fe7645b56452e1e
  868. Call-ID: 428a79340af7ad8c5e34b19d627d80c2@85.92.195.146:5160
  869. CSeq: 102 OPTIONS
  870. User-Agent: Voipfone Sip Network
  871. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
  872. Contact: <sip:46.31.231.185>
  873. Accept: application/sdp
  874. Content-Length: 0
  875.  
  876. <------------->
  877. --- (11 headers 0 lines) ---
  878. Really destroying SIP dialog '428a79340af7ad8c5e34b19d627d80c2@85.92.195.146:5160' Method: OPTIONS
  879. Reliably Transmitting (no NAT) to 82.147.28.150:61123:
  880. OPTIONS sip:3006@172.16.103.99:61123;ob SIP/2.0
  881. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK146e0f4e
  882. Max-Forwards: 70
  883. From: "Unknown" <sip:Unknown@85.92.195.146:5160>;tag=as22adccf8
  884. To: <sip:3006@172.16.103.99:61123;ob>
  885. Contact: <sip:Unknown@85.92.195.146:5160>
  886. Call-ID: 3d53b3626ad3b4df48d0d64000124110@85.92.195.146:5160
  887. CSeq: 102 OPTIONS
  888. User-Agent: FPBX-14.0.1.1(13.16.0)
  889. Date: Thu, 13 Jul 2017 15:53:02 GMT
  890. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  891. Supported: replaces, timer
  892. Content-Length: 0
  893.  
  894.  
  895. ---
  896.  
  897. <--- SIP read from UDP:82.147.28.150:61123 --->
  898. SIP/2.0 200 OK
  899. Via: SIP/2.0/UDP 85.92.195.146:5160;received=85.92.195.146;branch=z9hG4bK146e0f4e
  900. Call-ID: 3d53b3626ad3b4df48d0d64000124110@85.92.195.146:5160
  901. From: "Unknown" <sip:Unknown@85.92.195.146>;tag=as22adccf8
  902. To: <sip:3006@172.16.103.99;ob>;tag=z9hG4bK146e0f4e
  903. CSeq: 102 OPTIONS
  904. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  905. Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
  906. Supported: replaces, 100rel, timer, norefersub
  907. Allow-Events: presence, message-summary, refer
  908. User-Agent: Telephone 1.2.6
  909. Content-Length: 0
  910.  
  911. <------------->
  912. --- (12 headers 0 lines) ---
  913. Really destroying SIP dialog '3d53b3626ad3b4df48d0d64000124110@85.92.195.146:5160' Method: OPTIONS
  914.  
  915. <--- SIP read from UDP:82.147.28.150:61123 --->
  916.  
  917. <------------->
  918. [2017-07-13 15:53:15] NOTICE[6536]: res_pjsip/pjsip_distributor.c:526 log_failed_request: Request 'INVITE' from '"1001" <sip:1001@85.92.195.146>' failed for '155.94.65.77:5070' (callid: 5d2c7a4f58c2bba2e15184cef22f8e32) - No matching endpoint found
  919. [2017-07-13 15:53:17] NOTICE[2906]: chan_sip.c:15722 sip_reregister:    -- Re-registration for  30178956@sip.voipfone.net
  920. REGISTER 12 headers, 0 lines
  921. Reliably Transmitting (NAT) to 46.31.231.185:5060:
  922. REGISTER sip:sip.voipfone.net SIP/2.0
  923. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK50af7044;rport
  924. Max-Forwards: 70
  925. From: <sip:30178956@sip.voipfone.net>;tag=as6f8ec4a6
  926. To: <sip:30178956@sip.voipfone.net>
  927. Call-ID: 7b23c7c724761e89583cbc41071062a4@127.0.0.1
  928. CSeq: 106 REGISTER
  929. Supported: replaces, timer
  930. User-Agent: FPBX-14.0.1.1(13.16.0)
  931. Authorization: Digest username="30178956", realm="asterisk", algorithm=MD5, uri="sip:sip.voipfone.net", nonce="frjjriea", response="fd6f94bfeb127dd6a610408727ae3367"
  932. Expires: 120
  933. Contact: <sip:30178956@85.92.195.146:5160>
  934. Content-Length: 0
  935.  
  936.  
  937. ---
  938.  
  939. <--- SIP read from UDP:46.31.231.185:5060 --->
  940. SIP/2.0 200 OK
  941. Via: SIP/2.0/UDP 85.92.195.146:5160;branch=z9hG4bK50af7044;rport
  942. From: <sip:30178956@sip.voipfone.net>;tag=as6f8ec4a6
  943. To: <sip:30178956@sip.voipfone.net>
  944. Call-ID: 7b23c7c724761e89583cbc41071062a4@127.0.0.1
  945. CSeq: 106 REGISTER
  946. Contact: <sip:30178956@85.92.195.146:5160>;expires=60
  947. Expires: 60
  948. Date: Thu, 13 Jul 2017 15:53:17 GMT
  949. Min-Expires: 60
  950. User-Agent: Voipfone
  951. Content-Length: 0
  952.  
  953. <------------->
  954. --- (12 headers 0 lines) ---
  955. [2017-07-13 15:53:17] NOTICE[2906]: chan_sip.c:24538 handle_response_register: Outbound Registration: Expiry for sip.voipfone.net is 60 sec (Scheduling reregistration in 45 s)
  956. Really destroying SIP dialog '7b23c7c724761e89583cbc41071062a4@127.0.0.1' Method: REGISTER
  957.  
  958. <--- SIP read from UDP:82.147.28.150:61123 --->
  959.  
  960. <------------->
  961. freepbx*CLI>
  962.  

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