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  1. [2018-09-04 21:06:25] Asterisk GIT-master-b300c563e8 built by root @ dznet-pbx on a x86_64 running Linux on 2018-09-04 17:11:54 UTC
  2. [2018-09-04 21:06:25] VERBOSE[21958] logger.c: Asterisk Queue Logger restarted
  3. [2018-09-04 21:06:25] VERBOSE[21958] asterisk.c: Remote UNIX connection disconnected
  4. [2018-09-04 21:06:32] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP request (3540 bytes) from UDP:192.168.128.12:5060 --->
  5. INVITE sip:<my 10 digit cell>@mydomain.com:5060 SIP/2.0
  6. Via: SIP/2.0/UDP 192.168.128.12:5060;branch=z9hG4bK360af3f538c5b
  7. From: "My Name" <sip:<my 10dig google voice #>@mydomain.com>;tag=431703~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693947
  8. To: <sip:<my 10 digit cell>@mydomain.com>
  9. Date: Wed, 05 Sep 2018 02:06:32 GMT
  10. Call-ID: 4db9c980-b8f13a28-359b3-c80a8c0@192.168.128.12
  11. Supported: 100rel,timer,resource-priority,replaces
  12. Min-SE: 1800
  13. User-Agent: Cisco-CP-DX650/10.2.5
  14. Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
  15. CSeq: 101 INVITE
  16. Expires: 180
  17. Allow-Events: presence
  18. Supported: X-cisco-srtp-fallback,X-cisco-original-called
  19. Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP;x-cisco-qos-tcl=true
  20. Session-ID: 1997adf200105000a0005017ff96e069;remote=00000000000000000000000000000000
  21. Cisco-Guid: 1304021376-0000065536-0000000413-0209758400
  22. P-Charging-Vector: icid-value="4DB9C980000100000000019C0C80A8C0";icid-generated-at=dznet-ucm;orig-ioi="IMS Inter Operator Identification"
  23. Session-Expires: 1800
  24. P-Asserted-Identity: "My Name" <sip:<my 10dig google voice #>@mydomain.com>
  25. Remote-Party-ID: "My Name" <sip:<my 10dig google voice #>@mydomain.com>;party=calling;screen=yes;privacy=off
  26. Contact: <sip:<my 10dig google voice #>@192.168.128.12:5060>;video;audio;+u.sip!devicename.ccm.cisco.com="SEP5017FF96E069";bfcp
  27. Max-Forwards: 69
  28. Content-Type: application/sdp
  29. Content-Length: 2097
  30.  
  31. v=0
  32. o=CiscoSystemsCCM-SIP 431703 1 IN IP4 192.168.128.12
  33. s=SIP Call
  34. c=IN IP4 192.168.128.134
  35. b=TIAS:384000
  36. b=AS:384
  37. t=0 0
  38. m=audio 19646 RTP/AVP 108 0 18 101
  39. b=TIAS:64000
  40. a=rtpmap:108 MP4A-LATM/90000
  41. a=fmtp:108 bitrate=64000;profile-level-id=24;object=23
  42. a=rtpmap:0 PCMU/8000
  43. a=rtpmap:18 G729/8000
  44. a=rtpmap:101 telephone-event/8000
  45. a=fmtp:101 0-15
  46. a=trafficclass:conversational.audio.avconf.aq:admitted
  47. m=video 19136 RTP/AVP 100 126 97
  48. b=TIAS:384000
  49. a=label:11
  50. a=rtpmap:100 H264/90000
  51. a=fmtp:100 profile-level-id=640016;packetization-mode=1;max-mbps=267300;max-fs=8910;max-rcmd-nalu-size=256000;level-asymmetry-allowed=1;max-fps=6000
  52. a=rtpmap:126 H264/90000
  53. a=fmtp:126 profile-level-id=428016;packetization-mode=1;max-mbps=267300;max-fs=8910;max-rcmd-nalu-size=256000;level-asymmetry-allowed=1;max-fps=6000
  54. a=rtpmap:97 H264/90000
  55. a=fmtp:97 profile-level-id=428016;packetization-mode=0;max-mbps=267300;max-fs=8910;max-rcmd-nalu-size=256000;level-asymmetry-allowed=1;max-fps=6000
  56. a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
  57. a=content:main
  58. a=rtcp-fb:* nack pli
  59. a=rtcp-fb:* ccm fir
  60. a=rtcp-fb:* ccm tmmbr
  61. a=trafficclass:conversational.video.avconf.aq:admitted
  62. m=video 19620 RTP/AVP 100 126 97
  63. b=TIAS:384000
  64. a=label:12
  65. a=rtpmap:100 H264/90000
  66. a=fmtp:100 profile-level-id=640016;packetization-mode=1;max-mbps=267300;max-fs=8910;max-rcmd-nalu-size=256000;level-asymmetry-allowed=1;max-fps=6000
  67. a=rtpmap:126 H264/90000
  68. a=fmtp:126 profile-level-id=428016;packetization-mode=1;max-mbps=267300;max-fs=8910;max-rcmd-nalu-size=256000;level-asymmetry-allowed=1;max-fps=6000
  69. a=rtpmap:97 H264/90000
  70. a=fmtp:97 profile-level-id=428016;packetization-mode=0;max-mbps=267300;max-fs=8910;max-rcmd-nalu-size=256000;level-asymmetry-allowed=1;max-fps=6000
  71. a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
  72. a=content:slides
  73. a=rtcp-fb:* nack pli
  74. a=rtcp-fb:* ccm fir
  75. a=rtcp-fb:* ccm tmmbr
  76. a=trafficclass:conversational.video.avconf.aq:admitted
  77. m=application 19780 UDP/BFCP *
  78. a=floorctrl:s-only c-only
  79. a=floorid:3 mstrm:12
  80. a=confid:1
  81. a=userid:5
  82.  
  83. [2018-09-04 21:06:32] VERBOSE[21916] pbx_variables.c: Setting global variable 'SIPDOMAIN' to 'mydomain.com'
  84. [2018-09-04 21:06:32] VERBOSE[21916] res_pjsip_logger.c: <--- Transmitting SIP response (416 bytes) to UDP:192.168.128.12:5060 --->
  85. SIP/2.0 100 Trying
  86. Via: SIP/2.0/UDP 192.168.128.12:5060;rport=5060;received=192.168.128.12;branch=z9hG4bK360af3f538c5b
  87. Call-ID: 4db9c980-b8f13a28-359b3-c80a8c0@192.168.128.12
  88. From: "My Name" <sip:<my 10dig google voice #>@mydomain.com>;tag=431703~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693947
  89. To: <sip:<my 10 digit cell>@mydomain.com>
  90. CSeq: 101 INVITE
  91. Server: Asterisk PBX GIT-master-b300c563e8
  92. Content-Length: 0
  93.  
  94.  
  95. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] pbx.c: Executing [<my 10 digit cell>@home:1] GotoIf("PJSIP/cucm-0000001d", "1?numeric") in new stack
  96. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] pbx_builtins.c: Goto (home,<my 10 digit cell>,4)
  97. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] pbx.c: Executing [<my 10 digit cell>@home:4] Gosub("PJSIP/cucm-0000001d", "dialprovider,s,1(<my 10 digit cell>)") in new stack
  98. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] pbx.c: Executing [s@dialprovider:1] NoOp("PJSIP/cucm-0000001d", " printing full callerid -- "My Name" <<my 10dig google voice #>>") in new stack
  99. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] pbx.c: Executing [s@dialprovider:2] NoOp("PJSIP/cucm-0000001d", " printing the sip domain -- mydomain.com") in new stack
  100. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] pbx.c: Executing [s@dialprovider:3] Set("PJSIP/cucm-0000001d", "CALLERID(all)=<<my e164 google voice #>>") in new stack
  101. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] pbx.c: Executing [s@dialprovider:4] NoOp("PJSIP/cucm-0000001d", " printing the extension -- <my 10 digit cell>") in new stack
  102. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] pbx.c: Executing [s@dialprovider:5] Dial("PJSIP/cucm-0000001d", "PJSIP/<my e164 cell>@sipbroker-out") in new stack
  103. [2018-09-04 21:06:32] DEBUG[21916] res_pjsip_outbound_registration.c: Outgoing request not associated with a registration. No mangling necessary.
  104. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] app_dial.c: Called PJSIP/<my e164 cell>@sipbroker-out
  105. [2018-09-04 21:06:32] DEBUG[21916] res_pjsip_outbound_registration.c: Outgoing request not associated with a registration. No mangling necessary.
  106. [2018-09-04 21:06:32] VERBOSE[21916] res_pjsip_logger.c: <--- Transmitting SIP request (1195 bytes) to UDP:204.11.194.25:5060 --->
  107. INVITE sip:<my e164 cell>@sipbroker.com:5060 SIP/2.0
  108. Via: SIP/2.0/UDP myexternalip:5060;rport;branch=z9hG4bKPj5e0de5f3-e886-4485-bbd5-53be48f667d4
  109. From: <sip:driz@mydomain.com>;tag=8431a6a2-2432-4f5d-8c4f-c8124e354462
  110. To: <sip:<my e164 cell>@sipbroker.com>
  111. Contact: <sip:driz@myexternalip:5060>
  112. Call-ID: 6950eb67-0e38-478d-acd3-c95b6fb1c051
  113. CSeq: 28671 INVITE
  114. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
  115. Supported: 100rel, timer, replaces, norefersub
  116. Session-Expires: 1800
  117. Min-SE: 90
  118. Remote-Party-ID: <sip:<my e164 google voice #>@mydomain.com>;privacy=off;screen=no
  119. Max-Forwards: 70
  120. User-Agent: Asterisk PBX GIT-master-b300c563e8
  121. Content-Type: application/sdp
  122. Content-Length: 428
  123.  
  124. v=0
  125. o=- 1482607768 1482607768 IN IP4 myexternalip
  126. s=Asterisk
  127. c=IN IP4 myexternalip
  128. t=0 0
  129. m=audio 19358 RTP/AVP 0 101
  130. a=rtpmap:0 PCMU/8000
  131. a=rtpmap:101 telephone-event/8000
  132. a=fmtp:101 0-16
  133. a=ptime:20
  134. a=maxptime:150
  135. a=sendrecv
  136. m=video 19834 RTP/AVP 99
  137. a=rtpmap:99 H264/90000
  138. a=fmtp:99 max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000;packetization-mode=1;level-asymmetry-allowed=1
  139. a=sendrecv
  140.  
  141. [2018-09-04 21:06:32] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP response (581 bytes) from UDP:204.11.194.25:5060 --->
  142. SIP/2.0 100 Trying
  143. Via: SIP/2.0/UDP myexternalip:5060;rport=1024;branch=z9hG4bKPj5e0de5f3-e886-4485-bbd5-53be48f667d4
  144. From: <sip:driz@mydomain.com>;tag=8431a6a2-2432-4f5d-8c4f-c8124e354462
  145. To: <sip:<my e164 cell>@sipbroker.com>
  146. Call-ID: 6950eb67-0e38-478d-acd3-c95b6fb1c051
  147. CSeq: 28671 INVITE
  148. Server: OpenSer (1.1.0-notls (x86_64/linux))
  149. Content-Length: 0
  150. Warning: 392 204.11.194.25:5060 "Noisy feedback tells: pid=3478 req_src_ip=myexternalip req_src_port=1024 in_uri=sip:<my e164 cell>@sipbroker.com:5060 out_uri=sip:<my e164 cell>@sipbroker.com:5060 via_cnt==1"
  151.  
  152.  
  153. [2018-09-04 21:06:32] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP response (669 bytes) from UDP:204.11.194.25:5060 --->
  154. SIP/2.0 300 Redirect
  155. Via: SIP/2.0/UDP myexternalip:5060;rport=1024;branch=z9hG4bKPj5e0de5f3-e886-4485-bbd5-53be48f667d4
  156. From: <sip:driz@mydomain.com>;tag=8431a6a2-2432-4f5d-8c4f-c8124e354462
  157. To: <sip:<my e164 cell>@sipbroker.com>;tag=2b8506bb96abbbb8b95a41b9af69a614.bf8a
  158. Call-ID: 6950eb67-0e38-478d-acd3-c95b6fb1c051
  159. CSeq: 28671 INVITE
  160. Contact: sip:<my 11 digit cell>@mydomain.com
  161. Server: OpenSer (1.1.0-notls (x86_64/linux))
  162. Content-Length: 0
  163. Warning: 392 204.11.194.25:5060 "Noisy feedback tells: pid=3478 req_src_ip=myexternalip req_src_port=1024 in_uri=sip:<my e164 cell>@sipbroker.com:5060 out_uri=sip:<my 11 digit cell>@mydomain.com via_cnt==1"
  164.  
  165.  
  166. [2018-09-04 21:06:32] VERBOSE[21916] res_pjsip_logger.c: <--- Transmitting SIP request (456 bytes) to UDP:204.11.194.25:5060 --->
  167. ACK sip:<my e164 cell>@sipbroker.com:5060 SIP/2.0
  168. Via: SIP/2.0/UDP myexternalip:5060;rport;branch=z9hG4bKPj5e0de5f3-e886-4485-bbd5-53be48f667d4
  169. From: <sip:driz@mydomain.com>;tag=8431a6a2-2432-4f5d-8c4f-c8124e354462
  170. To: <sip:<my e164 cell>@sipbroker.com>;tag=2b8506bb96abbbb8b95a41b9af69a614.bf8a
  171. Call-ID: 6950eb67-0e38-478d-acd3-c95b6fb1c051
  172. CSeq: 28671 ACK
  173. Max-Forwards: 70
  174. User-Agent: Asterisk PBX GIT-master-b300c563e8
  175. Content-Length: 0
  176.  
  177.  
  178. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] app_dial.c: Now forwarding PJSIP/cucm-0000001d to 'Local/<my 11 digit cell>@unauthenticated' (thanks to PJSIP/sipbroker-out-0000001e)
  179. [2018-09-04 21:06:32] NOTICE[21967][C-0000000b] app_dial.c: Not accepting call completion offers from call-forward recipient Local/<my 11 digit cell>@unauthenticated-00000009;1
  180. [2018-09-04 21:06:32] VERBOSE[21916] res_pjsip_logger.c: <--- Transmitting SIP response (687 bytes) to UDP:192.168.128.12:5060 --->
  181. SIP/2.0 181 Call Is Being Forwarded
  182. Via: SIP/2.0/UDP 192.168.128.12:5060;rport=5060;received=192.168.128.12;branch=z9hG4bK360af3f538c5b
  183. Call-ID: 4db9c980-b8f13a28-359b3-c80a8c0@192.168.128.12
  184. From: "My Name" <sip:<my 10dig google voice #>@mydomain.com>;tag=431703~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693947
  185. To: <sip:<my 10 digit cell>@mydomain.com>;tag=4585f087-cac6-4eac-aee5-060e54692981
  186. CSeq: 101 INVITE
  187. Server: Asterisk PBX GIT-master-b300c563e8
  188. Contact: <sip:192.168.128.7:5060>
  189. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
  190. Remote-Party-ID: <sip:s@mydomain.com>;privacy=off;screen=no
  191. Content-Length: 0
  192.  
  193.  
  194. [2018-09-04 21:06:32] NOTICE[21967][C-0000000b] core_local.c: No such extension/context <my 11 digit cell>@unauthenticated while calling Local channel
  195. [2018-09-04 21:06:32] NOTICE[21967][C-0000000b] app_dial.c: Forwarding failed to dial 'Local/<my 11 digit cell>@unauthenticated'
  196. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
  197. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] pbx.c: Executing [s@dialprovider:6] NoOp("PJSIP/cucm-0000001d", " Dial Status: CHANUNAVAIL") in new stack
  198. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] pbx.c: Executing [s@dialprovider:7] Goto("PJSIP/cucm-0000001d", "s-CHANUNAVAIL,1") in new stack
  199. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] pbx_builtins.c: Goto (dialprovider,s-CHANUNAVAIL,1)
  200. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] pbx.c: Executing [s-CHANUNAVAIL@dialprovider:1] Dial("PJSIP/cucm-0000001d", "PJSIP/<my 10 digit cell>@<my 10dig google voice #>,,r") in new stack
  201. [2018-09-04 21:06:32] DEBUG[21916] res_pjsip_outbound_registration.c: Setting transport to 0x7f3a9c410ca8
  202. [2018-09-04 21:06:32] DEBUG[21916] res_pjsip.c: Overriding endpoint transport to use 0x7f3a9c410ca8
  203. [2018-09-04 21:06:32] VERBOSE[21967][C-0000000b] app_dial.c: Called PJSIP/<my 10 digit cell>@<my 10dig google voice #>
  204. [2018-09-04 21:06:32] VERBOSE[21917] res_pjsip_logger.c: <--- Transmitting SIP response (671 bytes) to UDP:192.168.128.12:5060 --->
  205. SIP/2.0 180 Ringing
  206. Via: SIP/2.0/UDP 192.168.128.12:5060;rport=5060;received=192.168.128.12;branch=z9hG4bK360af3f538c5b
  207. Call-ID: 4db9c980-b8f13a28-359b3-c80a8c0@192.168.128.12
  208. From: "My Name" <sip:<my 10dig google voice #>@mydomain.com>;tag=431703~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693947
  209. To: <sip:<my 10 digit cell>@mydomain.com>;tag=4585f087-cac6-4eac-aee5-060e54692981
  210. CSeq: 101 INVITE
  211. Server: Asterisk PBX GIT-master-b300c563e8
  212. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
  213. Contact: <sip:192.168.128.7:5060>
  214. Remote-Party-ID: <sip:s@mydomain.com>;privacy=off;screen=no
  215. Content-Length: 0
  216.  
  217.  
  218. [2018-09-04 21:06:32] DEBUG[21916] res_pjsip_outbound_registration.c: Found matching outbound registration state
  219. [2018-09-04 21:06:32] DEBUG[21916] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:ADW267E74XS3XJ3WDRKAY4FF3O6WO64GCYGTTJCRYIPBDIRQBHUATDWPLB2DR2N:5060;uri-econt=NWXKOLOPHSGWGGPZCFGXSI625RZ2FFKYCK67ANYBG5AUJPTPQYUG4EWMWCYXBLH5TISBPOGA5FW4D5O6R2UVNU4LK5AXX6OMUHO7CMOMGT5XZHS7LEEZ7SHM2NJ3TMZQ62JFUG;lr>
  220. [2018-09-04 21:06:32] DEBUG[21916] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:ADAOKMOFVTHS2X43M4NFJRRDMAJJBCH74MP5HNFPLWSDVHE6FTVV6DVTGNNQMOR:5060;transport=udp;lr;uri-econt=X3QFGCCJY>
  221. [2018-09-04 21:06:32] VERBOSE[21916] res_pjsip_logger.c: <--- Transmitting SIP request (2040 bytes) to TLS:64.9.242.108:5061 --->
  222. INVITE sip:<my 10 digit cell>@obihai.sip.google.com SIP/2.0
  223. Via: SIP/2.0/TLS 192.168.128.7:5061;rport;branch=z9hG4bKPj1bedcf16-39c1-43e1-9b60-bcd3e8dfc955;alias
  224. From: <sip:<my e164 google voice #>@192.168.128.7>;tag=7665ad1a-1d0d-417c-b8bc-c954f0ab3fd8
  225. To: <sip:<my 10 digit cell>@obihai.sip.google.com>
  226. Contact: <sip:asterisk@192.168.128.7:5061;transport=TLS>
  227. Call-ID: 135a18b4-e4ed-4e43-8ebb-a6ff065897bc
  228. CSeq: 1091 INVITE
  229. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
  230. Supported: 100rel, timer, replaces, norefersub, path, outbound
  231. Session-Expires: 1800
  232. Min-SE: 90
  233. Route: <sip:ADW267E74XS3XJ3WDRKAY4FF3O6WO64GCYGTTJCRYIPBDIRQBHUATDWPLB2DR2N:5060;uri-econt=NWXKOLOPHSGWGGPZCFGXSI625RZ2FFKYCK67ANYBG5AUJPTPQYUG4EWMWCYXBLH5TISBPOGA5FW4D5O6R2UVNU4LK5AXX6OMUHO7CMOMGT5XZHS7LEEZ7SHM2NJ3TMZQ62JFUG;lr>
  234. Route: <sip:ADAOKMOFVTHS2X43M4NFJRRDMAJJBCH74MP5HNFPLWSDVHE6FTVV6DVTGNNQMOR:5060;transport=udp;lr;uri-econt=X3QFGCCJY>
  235. P-Preferred-Identity: <sip:BIEWYY3PMZTDGMZVHEJBIMBXG4ZDCOJZGMZTSNZUHAYDSMBYGUZTG===@obihai.sip.google.com>
  236. Max-Forwards: 70
  237. User-Agent: Asterisk PBX GIT-master-b300c563e8
  238. Content-Type: application/sdp
  239. Content-Length: 845
  240.  
  241. v=0
  242. o=- 2012332113 2012332113 IN IP4 192.168.128.7
  243. s=Asterisk
  244. c=IN IP4 192.168.128.7
  245. t=0 0
  246. m=audio 19582 RTP/AVP 0 101
  247. a=ice-ufrag:28cf03962a48fc7c080973944ce06e9d
  248. a=ice-pwd:5780161778b07cda693e867b25dc2926
  249. a=candidate:Ha6e76162 1 UDP 2130706431 fe80::20c:29ff:fe43:c08d 19582 typ host
  250. a=candidate:Hc0a88007 1 UDP 2130706431 192.168.128.7 19582 typ host
  251. a=candidate:S45829cd3 1 UDP 1694498815 myexternalip 19582 typ srflx raddr 192.168.128.7 rport 19582
  252. a=candidate:Ha6e76162 2 UDP 2130706430 fe80::20c:29ff:fe43:c08d 19583 typ host
  253. a=candidate:Hc0a88007 2 UDP 2130706430 192.168.128.7 19583 typ host
  254. a=candidate:S45829cd3 2 UDP 1694498814 myexternalip 19583 typ srflx raddr 192.168.128.7 rport 19583
  255. a=rtpmap:0 PCMU/8000
  256. a=rtpmap:101 telephone-event/8000
  257. a=fmtp:101 0-16
  258. a=ptime:20
  259. a=maxptime:150
  260. a=sendrecv
  261. a=rtcp-mux
  262.  
  263. [2018-09-04 21:06:32] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP response (547 bytes) from TLS:64.9.242.108:5061 --->
  264. SIP/2.0 100 Trying
  265. Via: SIP/2.0/TLS 192.168.128.7:5061;rport=37425;branch=z9hG4bKPj1bedcf16-39c1-43e1-9b60-bcd3e8dfc955;received=myexternalip;alias
  266. Record-Route: <sip:ADAOKMOFJK5DQ6ARMJKX7RJUG334INXXYWDUBWJQGGXKWUDGH3MU4HSKI4YMFNO:5060;lr;transport=udp;uri-econt=GLAGONTWP>
  267. Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
  268. To: <sip:<my 10 digit cell>@obihai.sip.google.com>
  269. From: <sip:<my e164 google voice #>@192.168.128.7>;tag=7665ad1a-1d0d-417c-b8bc-c954f0ab3fd8
  270. Call-ID: 135a18b4-e4ed-4e43-8ebb-a6ff065897bc
  271. CSeq: 1091 INVITE
  272. Content-Length: 0
  273.  
  274.  
  275. [2018-09-04 21:06:33] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP response (1363 bytes) from TLS:64.9.242.108:5061 --->
  276. SIP/2.0 183 Session Progress
  277. Via: SIP/2.0/TLS 192.168.128.7:5061;rport=37425;branch=z9hG4bKPj1bedcf16-39c1-43e1-9b60-bcd3e8dfc955;received=myexternalip;alias
  278. Record-Route: <sip:ADAOKMOFJK5DQ6ARMJKX7RJUG334INXXYWDUBWJQGGXKWUDGH3MU4HSKI4YMFNO:5060;lr;transport=udp;uri-econt=GLAGONTWP>
  279. Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
  280. Contact: <sip:<my e164 google voice #>@AAZZHPMXVMMAVC256NPANQV6ZQPR4CR2FXSWFEEVEEBSEQZEYOBOH7HE3U6GCOO:5060;transport=udp;uri-econt=WA6DUBRQRBEUIFLJRVANF2OIXPZRA>
  281. To: <sip:<my 10 digit cell>@obihai.sip.google.com>;tag=102626655
  282. From: <sip:<my e164 google voice #>@192.168.128.7>;tag=7665ad1a-1d0d-417c-b8bc-c954f0ab3fd8
  283. Call-ID: 135a18b4-e4ed-4e43-8ebb-a6ff065897bc
  284. CSeq: 1091 INVITE
  285. Allow: ACK, BYE, CANCEL, INVITE, UPDATE
  286. Content-Type: application/sdp
  287. Content-Length: 566
  288.  
  289. v=0
  290. o=- 1618086373 1536113193541 IN IP4 74.125.39.28
  291. s=SIP Call
  292. c=IN IP4 74.125.39.28
  293. t=0 0
  294. a=ice-lite
  295. a=ice-pwd:QZb3iaNuqGHexAMrJ2vAwh4j
  296. a=ice-ufrag:NTFmPKiPJkMfpz4r
  297. a=group:BUNDLE audio
  298. a=fingerprint:sha-256 43:EE:1C:08:FA:CD:F9:8C:BB:29:99:AE:9E:A1:63:FC:4C:8D:89:2B:87:9E:9F:A5:52:9A:60:49:A2:BF:BD:90
  299. a=setup:passive
  300. m=audio 19305 RTP/AVP 0 101
  301. a=mid:audio
  302. a=rtpmap:0 PCMU/8000
  303. a=rtpmap:101 telephone-event/8000
  304. a=rtcp-mux
  305. a=candidate:1 1 UDP 1 74.125.39.28 19305 typ host
  306. a=candidate:2 1 UDP 2 2001:4860:4864:2::28 19305 typ host
  307. a=sendrecv
  308.  
  309. [2018-09-04 21:06:33] VERBOSE[21916] res_rtp_asterisk.c: 0x7f3aa0037870 -- Strict RTP learning after remote address set to: 74.125.39.28:19305
  310. [2018-09-04 21:06:33] ERROR[21916] pjproject: icess0x7f3aa005e398 ......Error sending STUN request: Network is unreachable
  311. [2018-09-04 21:06:33] VERBOSE[21967][C-0000000b] app_dial.c: PJSIP/<my 10dig google voice #>-0000001f is making progress passing it to PJSIP/cucm-0000001d
  312. [2018-09-04 21:06:33] VERBOSE[21967][C-0000000b] app_dial.c: PJSIP/<my 10dig google voice #>-0000001f is making progress passing it to PJSIP/cucm-0000001d
  313. [2018-09-04 21:06:33] VERBOSE[30742] res_rtp_asterisk.c: 0x7f3aa0037870 -- Strict RTP learning after ICE completion
  314. [2018-09-04 21:06:34] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP response (755 bytes) from TLS:64.9.242.108:5061 --->
  315. SIP/2.0 180 Ringing
  316. Via: SIP/2.0/TLS 192.168.128.7:5061;rport=37425;branch=z9hG4bKPj1bedcf16-39c1-43e1-9b60-bcd3e8dfc955;received=myexternalip;alias
  317. Record-Route: <sip:ADAOKMOFJK5DQ6ARMJKX7RJUG334INXXYWDUBWJQGGXKWUDGH3MU4HSKI4YMFNO:5060;lr;transport=udp;uri-econt=GLAGONTWP>
  318. Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
  319. Contact: <sip:<my e164 google voice #>@AAZZHPMXVMMAVC256NPANQV6ZQPR4CR2FXSWFEEVEEBSEQZEYOBOH7HE3U6GCOO:5060;transport=udp;uri-econt=WA6DUBRQRBEUIFLJRVANF2OIXPZRA>
  320. To: <sip:<my 10 digit cell>@obihai.sip.google.com>;tag=102626655
  321. From: <sip:<my e164 google voice #>@192.168.128.7>;tag=7665ad1a-1d0d-417c-b8bc-c954f0ab3fd8
  322. Call-ID: 135a18b4-e4ed-4e43-8ebb-a6ff065897bc
  323. CSeq: 1091 INVITE
  324. Allow: ACK, BYE, CANCEL, INVITE, UPDATE
  325. Content-Length: 0
  326.  
  327.  
  328. [2018-09-04 21:06:34] VERBOSE[21967][C-0000000b] app_dial.c: PJSIP/<my 10dig google voice #>-0000001f is ringing
  329. [2018-09-04 21:06:34] VERBOSE[21916] res_pjsip_logger.c: <--- Transmitting SIP response (683 bytes) to UDP:192.168.128.12:5060 --->
  330. SIP/2.0 180 Ringing
  331. Via: SIP/2.0/UDP 192.168.128.12:5060;rport=5060;received=192.168.128.12;branch=z9hG4bK360af3f538c5b
  332. Call-ID: 4db9c980-b8f13a28-359b3-c80a8c0@192.168.128.12
  333. From: "My Name" <sip:<my 10dig google voice #>@mydomain.com>;tag=431703~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693947
  334. To: <sip:<my 10 digit cell>@mydomain.com>;tag=4585f087-cac6-4eac-aee5-060e54692981
  335. CSeq: 101 INVITE
  336. Server: Asterisk PBX GIT-master-b300c563e8
  337. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
  338. Contact: <sip:192.168.128.7:5060>
  339. Remote-Party-ID: <sip:s-CHANUNAVAIL@mydomain.com>;privacy=off;screen=no
  340. Content-Length: 0
  341.  
  342.  
  343. [2018-09-04 21:06:34] VERBOSE[21967][C-0000000b] app_dial.c: PJSIP/<my 10dig google voice #>-0000001f is ringing
  344. [2018-09-04 21:06:36] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP request (398 bytes) from UDP:192.168.128.12:5060 --->
  345. OPTIONS sip:mydomain.com:5060 SIP/2.0
  346. Via: SIP/2.0/UDP 192.168.128.12:5060;branch=z9hG4bK360b129ea6619
  347. From: <sip:192.168.128.12>;tag=469427816
  348. To: <sip:mydomain.com>
  349. Date: Wed, 05 Sep 2018 02:06:36 GMT
  350. Call-ID: 501c2380-b8f13a2c-359b4-c80a8c0@192.168.128.12
  351. User-Agent: Cisco-CUCM11.5
  352. CSeq: 101 OPTIONS
  353. Contact: <sip:192.168.128.12:5060>
  354. Max-Forwards: 0
  355. Content-Length: 0
  356.  
  357.  
  358. [2018-09-04 21:06:36] VERBOSE[21916] res_pjsip_logger.c: <--- Transmitting SIP response (843 bytes) to UDP:192.168.128.12:5060 --->
  359. SIP/2.0 200 OK
  360. Via: SIP/2.0/UDP 192.168.128.12:5060;rport=5060;received=192.168.128.12;branch=z9hG4bK360b129ea6619
  361. Call-ID: 501c2380-b8f13a2c-359b4-c80a8c0@192.168.128.12
  362. From: <sip:192.168.128.12>;tag=469427816
  363. To: <sip:mydomain.com>;tag=z9hG4bK360b129ea6619
  364. CSeq: 101 OPTIONS
  365. Accept: application/pidf+xml, application/simple-message-summary, application/dialog-info+xml, application/xpidf+xml, application/cpim-pidf+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/sdp, message/sipfrag;version=2.0
  366. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
  367. Supported: 100rel, timer, replaces, norefersub
  368. Accept-Encoding: text/plain
  369. Accept-Language: en
  370. Server: Asterisk PBX GIT-master-b300c563e8
  371. Content-Length: 0
  372.  
  373.  
  374. [2018-09-04 21:06:38] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP response (1349 bytes) from TLS:64.9.242.108:5061 --->
  375. SIP/2.0 200 OK
  376. Via: SIP/2.0/TLS 192.168.128.7:5061;rport=37425;branch=z9hG4bKPj1bedcf16-39c1-43e1-9b60-bcd3e8dfc955;received=myexternalip;alias
  377. Record-Route: <sip:ADAOKMOFJK5DQ6ARMJKX7RJUG334INXXYWDUBWJQGGXKWUDGH3MU4HSKI4YMFNO:5060;lr;transport=udp;uri-econt=GLAGONTWP>
  378. Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
  379. Contact: <sip:<my e164 google voice #>@AAZZHPMXVMMAVC256NPANQV6ZQPR4CR2FXSWFEEVEEBSEQZEYOBOH7HE3U6GCOO:5060;transport=udp;uri-econt=WA6DUBRQRBEUIFLJRVANF2OIXPZRA>
  380. To: <sip:<my 10 digit cell>@obihai.sip.google.com>;tag=102626655
  381. From: <sip:<my e164 google voice #>@192.168.128.7>;tag=7665ad1a-1d0d-417c-b8bc-c954f0ab3fd8
  382. Call-ID: 135a18b4-e4ed-4e43-8ebb-a6ff065897bc
  383. CSeq: 1091 INVITE
  384. Allow: ACK, BYE, CANCEL, INVITE, UPDATE
  385. Content-Type: application/sdp
  386. Content-Length: 566
  387.  
  388. v=0
  389. o=- 1618086373 1536113193541 IN IP4 74.125.39.28
  390. s=SIP Call
  391. c=IN IP4 74.125.39.28
  392. t=0 0
  393. a=ice-lite
  394. a=ice-pwd:QZb3iaNuqGHexAMrJ2vAwh4j
  395. a=ice-ufrag:NTFmPKiPJkMfpz4r
  396. a=group:BUNDLE audio
  397. a=fingerprint:sha-256 43:EE:1C:08:FA:CD:F9:8C:BB:29:99:AE:9E:A1:63:FC:4C:8D:89:2B:87:9E:9F:A5:52:9A:60:49:A2:BF:BD:90
  398. a=setup:passive
  399. m=audio 19305 RTP/AVP 0 101
  400. a=mid:audio
  401. a=rtpmap:0 PCMU/8000
  402. a=rtpmap:101 telephone-event/8000
  403. a=rtcp-mux
  404. a=candidate:1 1 UDP 1 74.125.39.28 19305 typ host
  405. a=candidate:2 1 UDP 2 2001:4860:4864:2::28 19305 typ host
  406. a=sendrecv
  407.  
  408. [2018-09-04 21:06:38] VERBOSE[21916] res_pjsip_logger.c: <--- Transmitting SIP request (714 bytes) to TLS:64.9.242.108:5061 --->
  409. ACK sip:<my e164 google voice #>@AAZZHPMXVMMAVC256NPANQV6ZQPR4CR2FXSWFEEVEEBSEQZEYOBOH7HE3U6GCOO:5060;transport=udp;uri-econt=WA6DUBRQRBEUIFLJRVANF2OIXPZRA SIP/2.0
  410. Via: SIP/2.0/TLS 192.168.128.7:5061;rport;branch=z9hG4bKPjbcf1e6a5-10c0-44d1-8853-1231c839fb74;alias
  411. From: <sip:<my e164 google voice #>@192.168.128.7>;tag=7665ad1a-1d0d-417c-b8bc-c954f0ab3fd8
  412. To: <sip:<my 10 digit cell>@obihai.sip.google.com>;tag=102626655
  413. Call-ID: 135a18b4-e4ed-4e43-8ebb-a6ff065897bc
  414. CSeq: 1091 ACK
  415. Route: <sip:64.9.242.108:5061;transport=tls;lr>
  416. Route: <sip:ADAOKMOFJK5DQ6ARMJKX7RJUG334INXXYWDUBWJQGGXKWUDGH3MU4HSKI4YMFNO:5060;transport=udp;lr;uri-econt=GLAGONTWP>
  417. Max-Forwards: 70
  418. User-Agent: Asterisk PBX GIT-master-b300c563e8
  419. Content-Length: 0
  420.  
  421.  
  422. [2018-09-04 21:06:38] VERBOSE[21967][C-0000000b] app_dial.c: PJSIP/<my 10dig google voice #>-0000001f answered PJSIP/cucm-0000001d
  423. [2018-09-04 21:06:38] VERBOSE[21916] res_rtp_asterisk.c: 0x7f3aa00223e0 -- Strict RTP learning after remote address set to: 192.168.128.134:19646
  424. [2018-09-04 21:06:38] VERBOSE[21916] res_rtp_asterisk.c: 0x7f3aa00259d0 -- Strict RTP learning after remote address set to: 192.168.128.134:19136
  425. [2018-09-04 21:06:38] VERBOSE[21916] res_pjsip_logger.c: <--- Transmitting SIP response (1287 bytes) to UDP:192.168.128.12:5060 --->
  426. SIP/2.0 200 OK
  427. Via: SIP/2.0/UDP 192.168.128.12:5060;rport=5060;received=192.168.128.12;branch=z9hG4bK360af3f538c5b
  428. Call-ID: 4db9c980-b8f13a28-359b3-c80a8c0@192.168.128.12
  429. From: "My Name" <sip:<my 10dig google voice #>@mydomain.com>;tag=431703~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693947
  430. To: <sip:<my 10 digit cell>@mydomain.com>;tag=4585f087-cac6-4eac-aee5-060e54692981
  431. CSeq: 101 INVITE
  432. Server: Asterisk PBX GIT-master-b300c563e8
  433. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
  434. Contact: <sip:192.168.128.7:5060>
  435. Supported: 100rel, timer, replaces, norefersub
  436. Session-Expires: 1800;refresher=uac
  437. Require: timer
  438. Remote-Party-ID: <sip:s-CHANUNAVAIL@mydomain.com>;privacy=off;screen=no
  439. Content-Type: application/sdp
  440. Content-Length: 474
  441.  
  442. v=0
  443. o=- 431703 3 IN IP4 192.168.128.7
  444. s=Asterisk
  445. c=IN IP4 192.168.128.7
  446. t=0 0
  447. m=audio 19718 RTP/AVP 0 101
  448. a=rtpmap:0 PCMU/8000
  449. a=rtpmap:101 telephone-event/8000
  450. a=fmtp:101 0-16
  451. a=ptime:20
  452. a=maxptime:150
  453. a=sendrecv
  454. m=video 19144 RTP/AVP 100
  455. a=rtpmap:100 H264/90000
  456. a=fmtp:100 max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000;packetization-mode=1;level-asymmetry-allowed=1
  457. a=sendrecv
  458. m=video 0 RTP/AVP 100 126 97
  459. m=application 0 UDP/BFCP *
  460.  
  461. [2018-09-04 21:06:38] VERBOSE[21977][C-0000000b] bridge_channel.c: Channel PJSIP/<my 10dig google voice #>-0000001f joined 'simple_bridge' basic-bridge <d4123f6e-5330-4d3c-8b2c-b5b959826098>
  462. [2018-09-04 21:06:38] VERBOSE[21967][C-0000000b] bridge_channel.c: Channel PJSIP/cucm-0000001d joined 'simple_bridge' basic-bridge <d4123f6e-5330-4d3c-8b2c-b5b959826098>
  463. [2018-09-04 21:06:38] DEBUG[21916] res_pjsip_outbound_registration.c: Found matching outbound registration state
  464. [2018-09-04 21:06:38] DEBUG[21916] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:ADW267E74XS3XJ3WDRKAY4FF3O6WO64GCYGTTJCRYIPBDIRQBHUATDWPLB2DR2N:5060;uri-econt=NWXKOLOPHSGWGGPZCFGXSI625RZ2FFKYCK67ANYBG5AUJPTPQYUG4EWMWCYXBLH5TISBPOGA5FW4D5O6R2UVNU4LK5AXX6OMUHO7CMOMGT5XZHS7LEEZ7SHM2NJ3TMZQ62JFUG;lr>
  465. [2018-09-04 21:06:38] DEBUG[21916] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:ADAOKMOFVTHS2X43M4NFJRRDMAJJBCH74MP5HNFPLWSDVHE6FTVV6DVTGNNQMOR:5060;transport=udp;lr;uri-econt=X3QFGCCJY>
  466. [2018-09-04 21:06:38] VERBOSE[21916] res_pjsip_logger.c: <--- Transmitting SIP request (2348 bytes) to TLS:64.9.242.108:5061 --->
  467. INVITE sip:<my e164 google voice #>@AAZZHPMXVMMAVC256NPANQV6ZQPR4CR2FXSWFEEVEEBSEQZEYOBOH7HE3U6GCOO:5060;transport=udp;uri-econt=WA6DUBRQRBEUIFLJRVANF2OIXPZRA SIP/2.0
  468. Via: SIP/2.0/TLS 192.168.128.7:5061;rport;branch=z9hG4bKPj420bb8ed-e540-4e6d-b7ca-55537e5a1c05;alias
  469. From: <sip:<my e164 google voice #>@192.168.128.7>;tag=7665ad1a-1d0d-417c-b8bc-c954f0ab3fd8
  470. To: <sip:<my 10 digit cell>@obihai.sip.google.com>;tag=102626655
  471. Contact: <sip:asterisk@192.168.128.7:5061;transport=TLS>
  472. Call-ID: 135a18b4-e4ed-4e43-8ebb-a6ff065897bc
  473. CSeq: 1092 INVITE
  474. Route: <sip:64.9.242.108:5061;transport=tls;lr>
  475. Route: <sip:ADAOKMOFJK5DQ6ARMJKX7RJUG334INXXYWDUBWJQGGXKWUDGH3MU4HSKI4YMFNO:5060;transport=udp;lr;uri-econt=GLAGONTWP>
  476. Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
  477. Supported: 100rel, timer, replaces, norefersub, path, outbound
  478. Session-Expires: 1800
  479. Min-SE: 90
  480. Route: <sip:ADW267E74XS3XJ3WDRKAY4FF3O6WO64GCYGTTJCRYIPBDIRQBHUATDWPLB2DR2N:5060;uri-econt=NWXKOLOPHSGWGGPZCFGXSI625RZ2FFKYCK67ANYBG5AUJPTPQYUG4EWMWCYXBLH5TISBPOGA5FW4D5O6R2UVNU4LK5AXX6OMUHO7CMOMGT5XZHS7LEEZ7SHM2NJ3TMZQ62JFUG;lr>
  481. Route: <sip:ADAOKMOFVTHS2X43M4NFJRRDMAJJBCH74MP5HNFPLWSDVHE6FTVV6DVTGNNQMOR:5060;transport=udp;lr;uri-econt=X3QFGCCJY>
  482. P-Preferred-Identity: <sip:BIEWYY3PMZTDGMZVHEJBIMBXG4ZDCOJZGMZTSNZUHAYDSMBYGUZTG===@obihai.sip.google.com>
  483. Max-Forwards: 70
  484. User-Agent: Asterisk PBX GIT-master-b300c563e8
  485. Content-Type: application/sdp
  486. Content-Length: 867
  487.  
  488. v=0
  489. o=- 2012332113 2012332114 IN IP4 192.168.128.7
  490. s=Asterisk
  491. c=IN IP4 192.168.128.7
  492. t=0 0
  493. m=audio 19582 RTP/AVP 0 101
  494. a=ice-ufrag:28cf03962a48fc7c080973944ce06e9d
  495. a=ice-pwd:5780161778b07cda693e867b25dc2926
  496. a=candidate:Ha6e76162 1 UDP 2130706431 fe80::20c:29ff:fe43:c08d 19582 typ host
  497. a=candidate:Hc0a88007 1 UDP 2130706431 192.168.128.7 19582 typ host
  498. a=candidate:S45829cd3 1 UDP 1694498815 myexternalip 19582 typ srflx raddr 192.168.128.7 rport 19582
  499. a=candidate:Ha6e76162 2 UDP 2130706430 fe80::20c:29ff:fe43:c08d 19583 typ host
  500. a=candidate:Hc0a88007 2 UDP 2130706430 192.168.128.7 19583 typ host
  501. a=candidate:S45829cd3 2 UDP 1694498814 myexternalip 19583 typ srflx raddr 192.168.128.7 rport 19583
  502. a=rtpmap:0 PCMU/8000
  503. a=rtpmap:101 telephone-event/8000
  504. a=fmtp:101 0-16
  505. a=ptime:20
  506. a=maxptime:150
  507. a=sendrecv
  508. a=rtcp-mux
  509. m=video 0 RTP/AVP 32
  510.  
  511. [2018-09-04 21:06:38] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP request (503 bytes) from UDP:192.168.128.12:5060 --->
  512. ACK sip:192.168.128.7:5060 SIP/2.0
  513. Via: SIP/2.0/UDP 192.168.128.12:5060;branch=z9hG4bK360b249efc0e
  514. From: "My Name" <sip:<my 10dig google voice #>@mydomain.com>;tag=431703~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693947
  515. To: <sip:<my 10 digit cell>@mydomain.com>;tag=4585f087-cac6-4eac-aee5-060e54692981
  516. Date: Wed, 05 Sep 2018 02:06:32 GMT
  517. Call-ID: 4db9c980-b8f13a28-359b3-c80a8c0@192.168.128.12
  518. User-Agent: Cisco-CP-DX650/10.2.5
  519. Max-Forwards: 70
  520. CSeq: 101 ACK
  521. Allow-Events: presence
  522. Content-Length: 0
  523.  
  524.  
  525. [2018-09-04 21:06:38] VERBOSE[21967][C-0000000b] res_rtp_asterisk.c: 0x7f3aa00223e0 -- Strict RTP switching to RTP target address 192.168.128.134:19646 as source
  526. [2018-09-04 21:06:38] VERBOSE[21977][C-0000000b] res_rtp_asterisk.c: 0x7f3aa0037870 -- Strict RTP switching to RTP target address 74.125.39.28:19305 as source
  527. [2018-09-04 21:06:38] VERBOSE[21977][C-0000000b] res_rtp_asterisk.c: 0x7f3aa0037870 -- Strict RTP learning complete - Locking on source address 74.125.39.28:19305
  528. [2018-09-04 21:06:39] VERBOSE[21967][C-0000000b] res_rtp_asterisk.c: 0x7f3aa00259d0 -- Strict RTP switching to RTP target address 192.168.128.134:19136 as source
  529. [2018-09-04 21:06:43] VERBOSE[21967][C-0000000b] res_rtp_asterisk.c: 0x7f3aa00223e0 -- Strict RTP learning complete - Locking on source address 192.168.128.134:19646
  530. [2018-09-04 21:06:43] VERBOSE[21967][C-0000000b] res_rtp_asterisk.c: 0x7f3aa00259d0 -- Strict RTP learning complete - Locking on source address 192.168.128.134:19136
  531. [2018-09-04 21:07:07] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP request (551 bytes) from UDP:192.168.128.20:49795 --->
  532. REGISTER sip:dznet-pbx.mydomain.com SIP/2.0
  533. Via: SIP/2.0/UDP 192.168.128.20:49795;rport;branch=z9hG4bKPjd3a788c4fb4d41af8232d28b0515b324
  534. Max-Forwards: 70
  535. From: <sip:dznet1@mydomain.com>;tag=dc944d2e84344f0b9c685bb0f76b293a
  536. To: <sip:dznet1@mydomain.com>
  537. Call-ID: 330d06c3b2a342bea7d42c4c539ac7dd
  538. CSeq: 30637 REGISTER
  539. User-Agent: MicroSIP/3.19.7
  540. Contact: <sip:dznet1@192.168.128.20:49795;ob>
  541. Expires: 300
  542. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  543. Content-Length: 0
  544.  
  545.  
  546. [2018-09-04 21:07:07] VERBOSE[21916] res_pjsip_logger.c: <--- Transmitting SIP response (587 bytes) to UDP:192.168.128.20:49795 --->
  547. SIP/2.0 401 Unauthorized
  548. Via: SIP/2.0/UDP 192.168.128.20:49795;rport=49795;received=192.168.128.20;branch=z9hG4bKPjd3a788c4fb4d41af8232d28b0515b324
  549. Call-ID: 330d06c3b2a342bea7d42c4c539ac7dd
  550. From: <sip:dznet1@mydomain.com>;tag=dc944d2e84344f0b9c685bb0f76b293a
  551. To: <sip:dznet1@mydomain.com>;tag=z9hG4bKPjd3a788c4fb4d41af8232d28b0515b324
  552. CSeq: 30637 REGISTER
  553. WWW-Authenticate: Digest realm="asterisk",nonce="1536113227/b735336ab9d01a7414dadbc546c2c1ff",opaque="3f7ef1636c3b0624",algorithm=md5,qop="auth"
  554. Server: Asterisk PBX GIT-master-b300c563e8
  555. Content-Length: 0
  556.  
  557.  
  558. [2018-09-04 21:07:07] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP request (856 bytes) from UDP:192.168.128.20:49795 --->
  559. REGISTER sip:dznet-pbx.mydomain.com SIP/2.0
  560. Via: SIP/2.0/UDP 192.168.128.20:49795;rport;branch=z9hG4bKPj8992b249b354406e90b97ad5467a95d8
  561. Max-Forwards: 70
  562. From: <sip:dznet1@mydomain.com>;tag=dc944d2e84344f0b9c685bb0f76b293a
  563. To: <sip:dznet1@mydomain.com>
  564. Call-ID: 330d06c3b2a342bea7d42c4c539ac7dd
  565. CSeq: 30638 REGISTER
  566. User-Agent: MicroSIP/3.19.7
  567. Contact: <sip:dznet1@192.168.128.20:49795;ob>
  568. Expires: 300
  569. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
  570. Authorization: Digest username="dznet1", realm="asterisk", nonce="1536113227/b735336ab9d01a7414dadbc546c2c1ff", uri="sip:dznet-pbx.mydomain.com", response="19b244fbe9a0e8312b42a10584b27139", algorithm=md5, cnonce="3d78fd67b1d94220884695371008b11e", opaque="3f7ef1636c3b0624", qop=auth, nc=00000001
  571. Content-Length: 0
  572.  
  573.  
  574. [2018-09-04 21:07:07] VERBOSE[21916] res_pjsip_registrar.c: Added contact 'sip:dznet1@192.168.128.20:49795;ob' to AOR 'dznet1' with expiration of 300 seconds
  575. [2018-09-04 21:07:07] VERBOSE[21916] res_pjsip_logger.c: <--- Transmitting SIP response (540 bytes) to UDP:192.168.128.20:49795 --->
  576. SIP/2.0 200 OK
  577. Via: SIP/2.0/UDP 192.168.128.20:49795;rport=49795;received=192.168.128.20;branch=z9hG4bKPj8992b249b354406e90b97ad5467a95d8
  578. Call-ID: 330d06c3b2a342bea7d42c4c539ac7dd
  579. From: <sip:dznet1@mydomain.com>;tag=dc944d2e84344f0b9c685bb0f76b293a
  580. To: <sip:dznet1@mydomain.com>;tag=z9hG4bKPj8992b249b354406e90b97ad5467a95d8
  581. CSeq: 30638 REGISTER
  582. Date: Wed, 05 Sep 2018 02:07:07 GMT
  583. Contact: <sip:dznet1@192.168.128.20:49795;ob>;expires=299
  584. Expires: 300
  585. Server: Asterisk PBX GIT-master-b300c563e8
  586. Content-Length: 0
  587.  
  588.  
  589. [2018-09-04 21:07:07] VERBOSE[21917] res_pjsip/pjsip_configuration.c: Endpoint dznet1 is now Reachable
  590. [2018-09-04 21:07:10] VERBOSE[30730] res_pjsip_logger.c: <--- Transmitting SIP request (714 bytes) to TLS:64.9.242.108:5061 --->
  591. BYE sip:<my e164 google voice #>@AAZZHPMXVMMAVC256NPANQV6ZQPR4CR2FXSWFEEVEEBSEQZEYOBOH7HE3U6GCOO:5060;transport=udp;uri-econt=WA6DUBRQRBEUIFLJRVANF2OIXPZRA SIP/2.0
  592. Via: SIP/2.0/TLS 192.168.128.7:5061;rport;branch=z9hG4bKPj44d4f26a-eb58-46ee-953d-e53ce39d2a70;alias
  593. From: <sip:<my e164 google voice #>@192.168.128.7>;tag=7665ad1a-1d0d-417c-b8bc-c954f0ab3fd8
  594. To: <sip:<my 10 digit cell>@obihai.sip.google.com>;tag=102626655
  595. Call-ID: 135a18b4-e4ed-4e43-8ebb-a6ff065897bc
  596. CSeq: 1093 BYE
  597. Route: <sip:64.9.242.108:5061;transport=tls;lr>
  598. Route: <sip:ADAOKMOFJK5DQ6ARMJKX7RJUG334INXXYWDUBWJQGGXKWUDGH3MU4HSKI4YMFNO:5060;transport=udp;lr;uri-econt=GLAGONTWP>
  599. Max-Forwards: 70
  600. User-Agent: Asterisk PBX GIT-master-b300c563e8
  601. Content-Length: 0
  602.  
  603.  
  604. [2018-09-04 21:07:10] VERBOSE[21977][C-0000000b] bridge_channel.c: Channel PJSIP/<my 10dig google voice #>-0000001f left 'simple_bridge' basic-bridge <d4123f6e-5330-4d3c-8b2c-b5b959826098>
  605. [2018-09-04 21:07:10] VERBOSE[21967][C-0000000b] bridge_channel.c: Channel PJSIP/cucm-0000001d left 'simple_bridge' basic-bridge <d4123f6e-5330-4d3c-8b2c-b5b959826098>
  606. [2018-09-04 21:07:10] VERBOSE[21967][C-0000000b] pbx.c: Spawn extension (dialprovider, s-CHANUNAVAIL, 1) exited non-zero on 'PJSIP/cucm-0000001d'
  607. [2018-09-04 21:07:10] VERBOSE[21916] res_pjsip_logger.c: <--- Transmitting SIP request (525 bytes) to UDP:192.168.128.12:5060 --->
  608. BYE sip:<my 10dig google voice #>@192.168.128.12:5060 SIP/2.0
  609. Via: SIP/2.0/UDP 192.168.128.7:5060;rport;branch=z9hG4bKPj0f6e47d8-ef7f-4122-b7d3-e8b27307afc0
  610. From: <sip:<my 10 digit cell>@mydomain.com>;tag=4585f087-cac6-4eac-aee5-060e54692981
  611. To: "My Name" <sip:<my 10dig google voice #>@mydomain.com>;tag=431703~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693947
  612. Call-ID: 4db9c980-b8f13a28-359b3-c80a8c0@192.168.128.12
  613. CSeq: 24830 BYE
  614. Reason: Q.850;cause=16
  615. Max-Forwards: 70
  616. User-Agent: Asterisk PBX GIT-master-b300c563e8
  617. Content-Length: 0
  618.  
  619.  
  620. [2018-09-04 21:07:10] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP response (470 bytes) from UDP:192.168.128.12:5060 --->
  621. SIP/2.0 200 OK
  622. Via: SIP/2.0/UDP 192.168.128.7:5060;rport;branch=z9hG4bKPj0f6e47d8-ef7f-4122-b7d3-e8b27307afc0
  623. From: <sip:<my 10 digit cell>@mydomain.com>;tag=4585f087-cac6-4eac-aee5-060e54692981
  624. To: "My Name" <sip:<my 10dig google voice #>@mydomain.com>;tag=431703~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693947
  625. Date: Wed, 05 Sep 2018 02:07:10 GMT
  626. Call-ID: 4db9c980-b8f13a28-359b3-c80a8c0@192.168.128.12
  627. Server: Cisco-CP-DX650/10.2.5
  628. CSeq: 24830 BYE
  629. Content-Length: 0
  630.  
  631.  
  632. [2018-09-04 21:07:10] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP response (595 bytes) from TLS:64.9.242.108:5061 --->
  633. SIP/2.0 200 OK
  634. Via: SIP/2.0/TLS 192.168.128.7:5061;rport=37425;branch=z9hG4bKPj44d4f26a-eb58-46ee-953d-e53ce39d2a70;received=myexternalip;alias
  635. Record-Route: <sip:ADAOKMOFJK5DQ6ARMJKX7RJUG334INXXYWDUBWJQGGXKWUDGH3MU4HSKI4YMFNO:5060;lr;transport=udp;uri-econt=GLAGONTWP>
  636. Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
  637. To: <sip:<my 10 digit cell>@obihai.sip.google.com>;tag=102626655
  638. From: <sip:<my e164 google voice #>@192.168.128.7>;tag=7665ad1a-1d0d-417c-b8bc-c954f0ab3fd8
  639. Call-ID: 135a18b4-e4ed-4e43-8ebb-a6ff065897bc
  640. CSeq: 1093 BYE
  641. Allow: ACK, BYE, CANCEL, INVITE, UPDATE
  642. Content-Length: 0
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