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  1. <------------->
  2. --- (13 headers 0 lines) ---
  3. Transmitting (NAT) to 50.56.59.168:5060:
  4. ACK sip:8173195128@trunk2.phonebooth.net SIP/2.0
  5. Via: SIP/2.0/UDP 69.15.69.149:5060;branch=z9hG4bK76116c55;rport
  6. Max-Forwards: 70
  7. From: "8179533349" <sip:8179533349@69.15.69.149>;tag=as0548e321
  8. To: <sip:8173195128@trunk2.phonebooth.net>;tag=Bc6QH2e2Xv5gB
  9. Contact: <sip:8179533349@69.15.69.149:5060>
  10. Call-ID: 5d9372a741e8cced1288e1773b1d70f8@69.15.69.149:5060
  11. CSeq: 103 ACK
  12. User-Agent: FPBX-2.10.0(1.8.11.0)
  13. Content-Length: 0
  14.  
  15.  
  16. ---
  17. Scheduling destruction of SIP dialog '5d9372a741e8cced1288e1773b1d70f8@69.15.69.149:5060' in 6400 ms (Method: INVITE)
  18. == Everyone is busy/congested at this time (1:0/1/0)
  19. -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/150-00000020", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack
  20. -- Executing [s@macro-dialout-trunk:24] Goto("SIP/150-00000020", "s-CONGESTION,1") in new stack
  21. -- Goto (macro-dialout-trunk,s-CONGESTION,1)
  22. -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/150-00000020", "RC=34") in new stack
  23. -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/150-00000020", "34,1") in new stack
  24. -- Goto (macro-dialout-trunk,34,1)
  25. -- Executing [34@macro-dialout-trunk:1] Goto("SIP/150-00000020", "continue,1") in new stack
  26. -- Goto (macro-dialout-trunk,continue,1)
  27. -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/150-00000020", "1?noreport") in new stack
  28. -- Goto (macro-dialout-trunk,continue,3)
  29. -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/150-00000020", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks") in new stack
  30. -- Executing [continue@macro-dialout-trunk:4] Set("SIP/150-00000020", "CALLERID(number)=150") in new stack
  31. -- Executing [8173195128@from-internal:7] Macro("SIP/150-00000020", "outisbusy,") in new stack
  32. -- Executing [s@macro-outisbusy:1] Progress("SIP/150-00000020", "") in new stack
  33. Audio is at 15424
  34. Adding codec 0x4 (ulaw) to SDP
  35. Adding codec 0x8 (alaw) to SDP
  36. Adding non-codec 0x1 (telephone-event) to SDP
  37.  
  38. <--- Transmitting (NAT) to 192.168.1.140:5060 --->
  39. SIP/2.0 183 Session Progress
  40. Via: SIP/2.0/UDP 192.168.1.140:5060;branch=z9hG4bK-acad216f;received=192.168.1.140;rport=5060
  41. From: 150 <sip:150@192.168.1.10>;tag=3e63125ab8a1157o0
  42. To: <sip:8173195128@192.168.1.10>;tag=as4479dbd2
  43. Call-ID: fa56acd6-c8b8d90b@192.168.1.140
  44. CSeq: 102 INVITE
  45. Server: FPBX-2.10.0(1.8.11.0)
  46. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  47. Supported: replaces, timer
  48. Contact: <sip:8173195128@192.168.1.10:5060>
  49. Content-Type: application/sdp
  50. Content-Length: 259
  51.  
  52. v=0
  53. o=root 572245093 572245093 IN IP4 192.168.1.10
  54. s=Asterisk PBX 1.8.11.0
  55. c=IN IP4 192.168.1.10
  56. t=0 0
  57. m=audio 15424 RTP/AVP 0 8 101
  58. a=rtpmap:0 PCMU/8000
  59. a=rtpmap:8 PCMA/8000
  60. a=rtpmap:101 telephone-event/8000
  61. a=fmtp:101 0-16
  62. a=ptime:20
  63. a=sendrecv
  64.  
  65. <------------>
  66. -- Executing [s@macro-outisbusy:2] GotoIf("SIP/150-00000020", "0?emergency,1") in new stack
  67. -- Executing [s@macro-outisbusy:3] GotoIf("SIP/150-00000020", "0?intracompany,1") in new stack
  68. -- Executing [s@macro-outisbusy:4] Playback("SIP/150-00000020", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
  69. -- <SIP/150-00000020> Playing 'all-circuits-busy-now.ulaw' (language 'en')
  70.  
  71. <--- SIP read from UDP:50.56.59.168:5060 --->
  72. SIP/2.0 404 Not Found
  73. Via: SIP/2.0/UDP 69.15.69.149:5060;received=69.15.69.149;branch=z9hG4bK76116c55;rport=48658
  74. From: "8179533349" <sip:8179533349@69.15.69.149>;tag=as0548e321
  75. To: <sip:8173195128@trunk2.phonebooth.net>;tag=Bc6QH2e2Xv5gB
  76. Call-ID: 5d9372a741e8cced1288e1773b1d70f8@69.15.69.149:5060
  77. CSeq: 103 INVITE
  78. User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
  79. Accept: application/sdp
  80. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
  81. Supported: precondition, path, replaces
  82. Allow-Events: talk, refer
  83. Reason: Q.850;cause=3;text="NO_ROUTE_DESTINATION"
  84. Content-Length: 0
  85.  
  86. <------------->
  87. --- (13 headers 0 lines) ---
  88. Transmitting (NAT) to 50.56.59.168:5060:
  89. ACK sip:8173195128@trunk2.phonebooth.net SIP/2.0
  90. Via: SIP/2.0/UDP 69.15.69.149:5060;branch=z9hG4bK76116c55;rport
  91. Max-Forwards: 70
  92. From: "8179533349" <sip:8179533349@69.15.69.149>;tag=as0548e321
  93. To: <sip:8173195128@trunk2.phonebooth.net>;tag=Bc6QH2e2Xv5gB
  94. Contact: <sip:8179533349@69.15.69.149:5060>
  95. Call-ID: 5d9372a741e8cced1288e1773b1d70f8@69.15.69.149:5060
  96. CSeq: 103 ACK
  97. User-Agent: FPBX-2.10.0(1.8.11.0)
  98. Content-Length: 0
  99.  
  100.  
  101. ---
  102. -- <SIP/150-00000020> Playing 'pls-try-call-later.ulaw' (language 'en')
  103. [2012-04-20 12:17:09] NOTICE[-1]: chan_sip.c:13059 sip_reregister: -- Re-registration for 9b540a31@trunk2.phonebooth.net
  104. REGISTER 11 headers, 0 lines
  105. Reliably Transmitting (NAT) to 50.56.59.168:5060:
  106. REGISTER sip:trunk2.phonebooth.net SIP/2.0
  107. Via: SIP/2.0/UDP 69.15.69.149:5060;branch=z9hG4bK78a40e4d;rport
  108. Max-Forwards: 70
  109. From: <sip:9b540a31@trunk2.phonebooth.net>;tag=as0c5889f5
  110. To: <sip:9b540a31@trunk2.phonebooth.net>
  111. Call-ID: 2de41ebd04e7f39601ea90615e0cf00e@127.0.0.1
  112. CSeq: 120 REGISTER
  113. User-Agent: FPBX-2.10.0(1.8.11.0)
  114. Authorization: Digest username="9b540a31", realm="trunk.phonebooth.net", algorithm=MD5, uri="sip:trunk2.phonebooth.net", nonce="37e05f16-e89a-42a2-854b-b5fa46c61808", response="82d1d0f3037bd2be420358e12e265d22", qop=auth, cnonce="40be33c6", nc=00000012
  115. Expires: 120
  116. Contact: <sip:s@69.15.69.149:5060>
  117. Content-Length: 0
  118.  
  119.  
  120. ---
  121.  
  122. <--- SIP read from UDP:50.56.59.168:5060 --->
  123. SIP/2.0 200 OK
  124. Via: SIP/2.0/UDP 69.15.69.149:5060;received=69.15.69.149;branch=z9hG4bK78a40e4d;rport=48658
  125. From: <sip:9b540a31@trunk2.phonebooth.net>;tag=as0c5889f5
  126. To: <sip:9b540a31@trunk2.phonebooth.net>;tag=N7B9Ujjp4jD2H
  127. Call-ID: 2de41ebd04e7f39601ea90615e0cf00e@127.0.0.1
  128. CSeq: 120 REGISTER
  129. Contact: <sip:s@184.106.130.224:5060>;expires=3600
  130. Date: Fri, 20 Apr 2012 17:18:04 GMT
  131. User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
  132. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
  133. Supported: precondition, path, replaces
  134. Path: <sip:pb2proxy-pro-rsp03.phonebooth.net;lr>
  135. Content-Length: 0
  136.  
  137. <------------->
  138. --- (13 headers 0 lines) ---
  139. Scheduling destruction of SIP dialog '2de41ebd04e7f39601ea90615e0cf00e@127.0.0.1' in 32000 ms (Method: REGISTER)
  140. [2012-04-20 12:17:09] NOTICE[-1]: chan_sip.c:20738 handle_response_register: Outbound Registration: Expiry for trunk2.phonebooth.net is 120 sec (Scheduling reregistration in 105 s)
  141. [2012-04-20 12:17:10] NOTICE[-1]: chan_sip.c:13059 sip_reregister: -- Re-registration for 9b540a31@trunk1.phonebooth.net
  142. REGISTER 11 headers, 0 lines
  143. Reliably Transmitting (NAT) to 184.72.227.214:5060:
  144. REGISTER sip:trunk1.phonebooth.net SIP/2.0
  145. Via: SIP/2.0/UDP 69.15.69.149:5060;branch=z9hG4bK6995ee44;rport
  146. Max-Forwards: 70
  147. From: <sip:9b540a31@trunk1.phonebooth.net>;tag=as798acdc9
  148. To: <sip:9b540a31@trunk1.phonebooth.net>
  149. Call-ID: 08b942ea211ba1db2772cb38269fd3f9@127.0.0.1
  150. CSeq: 120 REGISTER
  151. User-Agent: FPBX-2.10.0(1.8.11.0)
  152. Authorization: Digest username="9b540a31", realm="trunk.phonebooth.net", algorithm=MD5, uri="sip:trunk1.phonebooth.net", nonce="6e5a61d3-7386-4eee-b985-16d51f4202d3", response="a149da31e1fd52717a50d3dc7819c9c6", qop=auth, cnonce="0dc09265", nc=00000012
  153. Expires: 120
  154. Contact: <sip:s@69.15.69.149:5060>
  155. Content-Length: 0
  156.  
  157.  
  158. ---
  159.  
  160. <--- SIP read from UDP:184.72.227.214:5060 --->
  161. SIP/2.0 200 OK
  162. Via: SIP/2.0/UDP 69.15.69.149:5060;received=69.15.69.149;branch=z9hG4bK6995ee44;rport=32614
  163. From: <sip:9b540a31@trunk1.phonebooth.net>;tag=as798acdc9
  164. To: <sip:9b540a31@trunk1.phonebooth.net>;tag=a5yH1am07D1aN
  165. Call-ID: 08b942ea211ba1db2772cb38269fd3f9@127.0.0.1
  166. CSeq: 120 REGISTER
  167. Contact: <sip:s@184.106.130.224:5060>;expires=3600
  168. Date: Fri, 20 Apr 2012 17:18:06 GMT
  169. User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
  170. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
  171. Supported: precondition, path, replaces
  172. Path: <sip:pb2proxy-pro-aws03.phonebooth.net;lr>
  173. Content-Length: 0
  174.  
  175. <------------->
  176. --- (13 headers 0 lines) ---
  177. Scheduling destruction of SIP dialog '08b942ea211ba1db2772cb38269fd3f9@127.0.0.1' in 32000 ms (Method: REGISTER)
  178. [2012-04-20 12:17:10] NOTICE[-1]: chan_sip.c:20738 handle_response_register: Outbound Registration: Expiry for trunk1.phonebooth.net is 120 sec (Scheduling reregistration in 105 s)
  179. -- Executing [s@macro-outisbusy:5] Congestion("SIP/150-00000020", "20") in new stack
  180.  
  181. <--- Reliably Transmitting (NAT) to 192.168.1.140:5060 --->
  182. SIP/2.0 503 Service Unavailable
  183. Via: SIP/2.0/UDP 192.168.1.140:5060;branch=z9hG4bK-acad216f;received=192.168.1.140;rport=5060
  184. From: 150 <sip:150@192.168.1.10>;tag=3e63125ab8a1157o0
  185. To: <sip:8173195128@192.168.1.10>;tag=as4479dbd2
  186. Call-ID: fa56acd6-c8b8d90b@192.168.1.140
  187. CSeq: 102 INVITE
  188. Server: FPBX-2.10.0(1.8.11.0)
  189. llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  190. Supported: replaces, timer
  191. X-Asterisk-HangupCause: Circuit/channel congestion
  192. X-Asterisk-HangupCauseCode: 34
  193. Content-Length: 0
  194.  
  195.  
  196. <------------>
  197. [2012-04-20 12:17:11] WARNING[-1]: channel.c:4650 ast_prod: Prodding channel 'SIP/150-00000020' failed
  198. == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/150-00000020' in macro 'outisbusy'
  199. == Spawn extension (from-internal, 8173195128, 7) exited non-zero on 'SIP/150-00000020'
  200. -- Executing [h@from-internal:1] Hangup("SIP/150-00000020", "") in new stack
  201. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/150-00000020'
  202.  
  203. <--- SIP read from UDP:192.168.1.140:5060 --->
  204. ACK sip:8173195128@192.168.1.10 SIP/2.0
  205. Via: SIP/2.0/UDP 192.168.1.140:5060;branch=z9hG4bK-acad216f
  206. From: 150 <sip:150@192.168.1.10>;tag=3e63125ab8a1157o0
  207. To: <sip:8173195128@192.168.1.10>;tag=as4479dbd2
  208. Call-ID: fa56acd6-c8b8d90b@192.168.1.140
  209. CSeq: 102 ACK
  210. Max-Forwards: 70
  211. Authorization: Digest username="150",realm="asterisk",nonce="2e9a1b2f",uri="sip:8173195128@192.168.1.10",algorithm=MD5,response="23a115fa9ea1cae1c354245ddc79903c"
  212. Contact: 150 <sip:150@192.168.1.140:5060>
  213. User-Agent: Linksys/SPA2102-5.2.12
  214. Content-Length: 0
  215.  
  216. <------------->
  217. --- (11 headers 0 lines) ---
  218. Really destroying SIP dialog 'fa56acd6-c8b8d90b@192.168.1.140' Method: ACK
  219. Really destroying SIP dialog '5d9372a741e8cced1288e1773b1d70f8@69.15.69.149:5060' Method: INVITE
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