trace2


[2018-09-07 18:53:18] VERBOSE[21226] asterisk.c: Remote UNIX connection
[2018-09-07 18:53:18] Asterisk GIT-master-b300c563e8 built by root @ dznet-pbx on a x86_64 running Linux on 2018-09-04 17:11:54 UTC
[2018-09-07 18:53:18] VERBOSE[23487] logger.c: Asterisk Queue Logger restarted
[2018-09-07 18:53:18] VERBOSE[23487] asterisk.c: Remote UNIX connection disconnected
[2018-09-07 18:53:25] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP request (3540 bytes) from UDP:192.168.128.12:5060 --->
INVITE sip:<my 10D cell>@mydomain.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.12:5060;branch=z9hG4bK380ff14f08507
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445172~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693970
To: <sip:<my 10D cell>@mydomain.com>
Date: Fri, 07 Sep 2018 23:53:25 GMT
Call-ID: 34578e00-b9310f75-379eb-c80a8c0@192.168.128.12
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CP-DX650/10.2.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP;x-cisco-qos-tcl=true
Session-ID: 2ca4f8f400105000a0005017ff96e069;remote=00000000000000000000000000000000
Cisco-Guid: 0878153216-0000065536-0000000434-0209758400
P-Charging-Vector: icid-value="34578E0000010000000001B10C80A8C0";icid-generated-at=dznet-ucm;orig-ioi="IMS Inter Operator Identification"
Session-Expires:  1800
P-Asserted-Identity: "My Name" <sip:<my 10d gvoice>@mydomain.com>
Remote-Party-ID: "My Name" <sip:<my 10d gvoice>@mydomain.com>;party=calling;screen=yes;privacy=off
Contact: <sip:<my 10d gvoice>@192.168.128.12:5060>;video;audio;+u.sip!devicename.ccm.cisco.com="SEP5017FF96E069";bfcp
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 2097

v=0
o=CiscoSystemsCCM-SIP 445172 1 IN IP4 192.168.128.12
s=SIP Call
c=IN IP4 192.168.128.134
b=TIAS:384000
b=AS:384
t=0 0
m=audio 19486 RTP/AVP 108 0 18 101
b=TIAS:64000
a=rtpmap:108 MP4A-LATM/90000
a=fmtp:108 bitrate=64000;profile-level-id=24;object=23
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=trafficclass:conversational.audio.avconf.aq:admitted
m=video 19634 RTP/AVP 100 126 97
b=TIAS:384000
a=label:11
a=rtpmap:100 H264/90000
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
a=rtpmap:126 H264/90000
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
a=rtpmap:97 H264/90000
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
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=content:main
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=trafficclass:conversational.video.avconf.aq:admitted
m=video 19196 RTP/AVP 100 126 97
b=TIAS:384000
a=label:12
a=rtpmap:100 H264/90000
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
a=rtpmap:126 H264/90000
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
a=rtpmap:97 H264/90000
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
a=imageattr:* recv [x=1024,y=600,q=0.60] [x=1280,y=720,q=0.50]
a=content:slides
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=trafficclass:conversational.video.avconf.aq:admitted
m=application 19736 UDP/BFCP *
a=floorctrl:s-only c-only
a=floorid:3 mstrm:12
a=confid:1
a=userid:2

[2018-09-07 18:53:25] VERBOSE[23313] pbx_variables.c: Setting global variable 'SIPDOMAIN' to 'mydomain.com'
[2018-09-07 18:53:25] VERBOSE[23313] res_pjsip_logger.c: <--- Transmitting SIP response (416 bytes) to UDP:192.168.128.12:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.128.12:5060;rport=5060;received=192.168.128.12;branch=z9hG4bK380ff14f08507
Call-ID: 34578e00-b9310f75-379eb-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445172~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693970
To: <sip:<my 10D cell>@mydomain.com>
CSeq: 101 INVITE
Server: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] pbx.c: Executing [<my 10D cell>@home:1] GotoIf("PJSIP/cucm-00000006", "1?numeric") in new stack
[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] pbx_builtins.c: Goto (home,<my 10D cell>,4)
[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] pbx.c: Executing [<my 10D cell>@home:4] Gosub("PJSIP/cucm-00000006", "dialprovider,s,1(<my 10D cell>)") in new stack
[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] pbx.c: Executing [s@dialprovider:1] NoOp("PJSIP/cucm-00000006", " printing full callerid -- "My Name" <<my 10d gvoice>>") in new stack
[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] pbx.c: Executing [s@dialprovider:2] NoOp("PJSIP/cucm-00000006", " printing the sip domain -- mydomain.com") in new stack
[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] pbx.c: Executing [s@dialprovider:3] Set("PJSIP/cucm-00000006", "CALLERID(all)=<<my 164 gvoice>>") in new stack
[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] pbx.c: Executing [s@dialprovider:4] NoOp("PJSIP/cucm-00000006", " printing the extension -- <my 10D cell>") in new stack
[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] pbx.c: Executing [s@dialprovider:5] Dial("PJSIP/cucm-00000006", "PJSIP/<my e164 cell>@sipbroker-out") in new stack
[2018-09-07 18:53:25] DEBUG[23313] res_pjsip_outbound_registration.c: Outgoing request not associated with a registration. No mangling necessary.
[2018-09-07 18:53:25] DEBUG[23313] res_pjsip_outbound_registration.c: Outgoing request not associated with a registration. No mangling necessary.
[2018-09-07 18:53:25] VERBOSE[23313] res_pjsip_logger.c: <--- Transmitting SIP request (865 bytes) to UDP:204.11.194.25:5060 --->
INVITE sip:<my e164 cell>@sipbroker.com:5060 SIP/2.0
Via: SIP/2.0/UDP <my public ip>:5060;rport;branch=z9hG4bKPj1bf4afff-708b-46b8-bf3b-d632e8e2f4a6
From: <sip:driz@mydomain.com>;tag=4a4510ed-e40b-4edb-ad12-91a4dacef60d
To: <sip:<my e164 cell>@sipbroker.com>
Contact: <sip:driz@<my public ip>:5060>
Call-ID: 27dd18a8-085d-49c8-ba6b-50c7e4bd47c5
CSeq: 30056 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Remote-Party-ID: <sip:<my 164 gvoice>@mydomain.com>;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Type: application/sdp
Content-Length:    98

v=0
o=- 2064535687 2064535687 IN IP4 <my public ip>
s=Asterisk
c=IN IP4 <my public ip>
t=0 0

[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] app_dial.c: Called PJSIP/<my e164 cell>@sipbroker-out
[2018-09-07 18:53:25] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP response (581 bytes) from UDP:204.11.194.25:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <my public ip>:5060;rport=1024;branch=z9hG4bKPj1bf4afff-708b-46b8-bf3b-d632e8e2f4a6
From: <sip:driz@mydomain.com>;tag=4a4510ed-e40b-4edb-ad12-91a4dacef60d
To: <sip:<my e164 cell>@sipbroker.com>
Call-ID: 27dd18a8-085d-49c8-ba6b-50c7e4bd47c5
CSeq: 30056 INVITE
Server: OpenSer (1.1.0-notls (x86_64/linux))
Content-Length: 0
Warning: 392 204.11.194.25:5060 "Noisy feedback tells:  pid=3449 req_src_ip=<my public ip> 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"


[2018-09-07 18:53:25] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP response (669 bytes) from UDP:204.11.194.25:5060 --->
SIP/2.0 300 Redirect
Via: SIP/2.0/UDP <my public ip>:5060;rport=1024;branch=z9hG4bKPj1bf4afff-708b-46b8-bf3b-d632e8e2f4a6
From: <sip:driz@mydomain.com>;tag=4a4510ed-e40b-4edb-ad12-91a4dacef60d
To: <sip:<my e164 cell>@sipbroker.com>;tag=2b8506bb96abbbb8b95a41b9af69a614.d170
Call-ID: 27dd18a8-085d-49c8-ba6b-50c7e4bd47c5
CSeq: 30056 INVITE
Contact: sip:<my 11D cell>@mydomain.com
Server: OpenSer (1.1.0-notls (x86_64/linux))
Content-Length: 0
Warning: 392 204.11.194.25:5060 "Noisy feedback tells:  pid=3449 req_src_ip=<my public ip> req_src_port=1024 in_uri=sip:<my e164 cell>@sipbroker.com:5060 out_uri=sip:<my 11D cell>@mydomain.com via_cnt==1"


[2018-09-07 18:53:25] VERBOSE[23313] res_pjsip_logger.c: <--- Transmitting SIP request (456 bytes) to UDP:204.11.194.25:5060 --->
ACK sip:<my e164 cell>@sipbroker.com:5060 SIP/2.0
Via: SIP/2.0/UDP <my public ip>:5060;rport;branch=z9hG4bKPj1bf4afff-708b-46b8-bf3b-d632e8e2f4a6
From: <sip:driz@mydomain.com>;tag=4a4510ed-e40b-4edb-ad12-91a4dacef60d
To: <sip:<my e164 cell>@sipbroker.com>;tag=2b8506bb96abbbb8b95a41b9af69a614.d170
Call-ID: 27dd18a8-085d-49c8-ba6b-50c7e4bd47c5
CSeq: 30056 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] app_dial.c: Now forwarding PJSIP/cucm-00000006 to 'Local/<my 11D cell>@unauthenticated' (thanks to PJSIP/sipbroker-out-00000007)
[2018-09-07 18:53:25] NOTICE[23489][C-00000005] app_dial.c: Not accepting call completion offers from call-forward recipient Local/<my 11D cell>@unauthenticated-00000001;1
[2018-09-07 18:53:25] VERBOSE[23313] res_pjsip_logger.c: <--- Transmitting SIP response (687 bytes) to UDP:192.168.128.12:5060 --->
SIP/2.0 181 Call Is Being Forwarded
Via: SIP/2.0/UDP 192.168.128.12:5060;rport=5060;received=192.168.128.12;branch=z9hG4bK380ff14f08507
Call-ID: 34578e00-b9310f75-379eb-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445172~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693970
To: <sip:<my 10D cell>@mydomain.com>;tag=77bc5c6a-8e33-4c9a-9f12-87685c5d8e46
CSeq: 101 INVITE
Server: Asterisk PBX GIT-master-b300c563e8
Contact: <sip:192.168.128.7:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Remote-Party-ID: <sip:s@mydomain.com>;privacy=off;screen=no
Content-Length:  0


[2018-09-07 18:53:25] NOTICE[23489][C-00000005] core_local.c: No such extension/context <my 11D cell>@unauthenticated while calling Local channel
[2018-09-07 18:53:25] NOTICE[23489][C-00000005] app_dial.c: Forwarding failed to dial 'Local/<my 11D cell>@unauthenticated'
[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] pbx.c: Executing [s@dialprovider:6] NoOp("PJSIP/cucm-00000006", " Dial Status: CHANUNAVAIL") in new stack
[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] pbx.c: Executing [s@dialprovider:7] Goto("PJSIP/cucm-00000006", "s-CHANUNAVAIL,1") in new stack
[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] pbx_builtins.c: Goto (dialprovider,s-CHANUNAVAIL,1)
[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] pbx.c: Executing [s-CHANUNAVAIL@dialprovider:1] Dial("PJSIP/cucm-00000006", "PJSIP/<my 10D cell>@<my 10d gvoice>,,r") in new stack
[2018-09-07 18:53:25] DEBUG[23313] res_pjsip_outbound_registration.c: Setting transport to 0x7f821c1141e8
[2018-09-07 18:53:25] DEBUG[23313] res_pjsip.c: Overriding endpoint transport to use 0x7f821c1141e8
[2018-09-07 18:53:25] VERBOSE[23489][C-00000005] app_dial.c: Called PJSIP/<my 10D cell>@<my 10d gvoice>
[2018-09-07 18:53:25] VERBOSE[23453] res_pjsip_logger.c: <--- Transmitting SIP response (671 bytes) to UDP:192.168.128.12:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.128.12:5060;rport=5060;received=192.168.128.12;branch=z9hG4bK380ff14f08507
Call-ID: 34578e00-b9310f75-379eb-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445172~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693970
To: <sip:<my 10D cell>@mydomain.com>;tag=77bc5c6a-8e33-4c9a-9f12-87685c5d8e46
CSeq: 101 INVITE
Server: Asterisk PBX GIT-master-b300c563e8
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Contact: <sip:192.168.128.7:5060>
Remote-Party-ID: <sip:s@mydomain.com>;privacy=off;screen=no
Content-Length:  0


[2018-09-07 18:53:25] DEBUG[23313] res_pjsip_outbound_registration.c: Found matching outbound registration state
[2018-09-07 18:53:25] DEBUG[23313] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:ADW267E7WKCZWWOSTAMVZM3OX5DURRTYGHZKY3CDLRQBSI5EGDYSO4QTWKU2HP5:5060;uri-econt=6RNT45K7F4X56ZVTLUCLQW5FJ54H3CE5UVSZ7CGBPXDEKWMJRQPREFW6YR25EXS3324EFNQZI5M5CPVKJGFRMP7U5ION76ZHT3DNVE4MMYQLLWVQ2N4A7OIAAYDICQPNJU4QRK;lr>
[2018-09-07 18:53:25] DEBUG[23313] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:ADAOKMOFMOSQQUSE7DVJ4EDSS3US3XEVQCPZZDEBC4FIHGXOSMWD6TPLHLJVRX4:5060;transport=udp;lr;uri-econt=YQVLFKPPJ>
[2018-09-07 18:53:25] VERBOSE[23313] res_pjsip_logger.c: <--- Transmitting SIP request (2041 bytes) to TLS:64.9.242.108:5061 --->
INVITE sip:<my 10D cell>@obihai.sip.google.com SIP/2.0
Via: SIP/2.0/TLS 192.168.128.7:5061;rport;branch=z9hG4bKPj7eca58f8-5756-469b-b22d-537bcb8b5cbf;alias
From: <sip:<my 164 gvoice>@192.168.128.7>;tag=c3a34301-f273-499a-af8f-af5c03e7dda8
To: <sip:<my 10D cell>@obihai.sip.google.com>
Contact: <sip:asterisk@192.168.128.7:5061;transport=TLS>
Call-ID: 84ec934c-d7db-4468-8f36-ea59e778bfd7
CSeq: 28907 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, path, outbound
Session-Expires: 1800
Min-SE: 90
Route: <sip:ADW267E7WKCZWWOSTAMVZM3OX5DURRTYGHZKY3CDLRQBSI5EGDYSO4QTWKU2HP5:5060;uri-econt=6RNT45K7F4X56ZVTLUCLQW5FJ54H3CE5UVSZ7CGBPXDEKWMJRQPREFW6YR25EXS3324EFNQZI5M5CPVKJGFRMP7U5ION76ZHT3DNVE4MMYQLLWVQ2N4A7OIAAYDICQPNJU4QRK;lr>
Route: <sip:ADAOKMOFMOSQQUSE7DVJ4EDSS3US3XEVQCPZZDEBC4FIHGXOSMWD6TPLHLJVRX4:5060;transport=udp;lr;uri-econt=YQVLFKPPJ>
P-Preferred-Identity: <sip:BIEWYY3PMZTDGMZVHEJBIMBXG4ZDCOJZGMZTSNZUHAYDSMBYGUZTG===@obihai.sip.google.com>
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Type: application/sdp
Content-Length:   845

v=0
o=- 1273799627 1273799627 IN IP4 192.168.128.7
s=Asterisk
c=IN IP4 192.168.128.7
t=0 0
m=audio 19066 RTP/AVP 0 101
a=ice-ufrag:4e594efa6b97e4f020210cbe733f91e6
a=ice-pwd:6dbe9326032b458b3b04a93e18127ce3
a=candidate:Ha6e76162 1 UDP 2130706431 fe80::20c:29ff:fe43:c08d 19066 typ host
a=candidate:Hc0a88007 1 UDP 2130706431 192.168.128.7 19066 typ host
a=candidate:S45829cd3 1 UDP 1694498815 <my public ip> 19066 typ srflx raddr 192.168.128.7 rport 19066
a=candidate:Ha6e76162 2 UDP 2130706430 fe80::20c:29ff:fe43:c08d 19067 typ host
a=candidate:Hc0a88007 2 UDP 2130706430 192.168.128.7 19067 typ host
a=candidate:S45829cd3 2 UDP 1694498814 <my public ip> 19067 typ srflx raddr 192.168.128.7 rport 19067
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux

[2018-09-07 18:53:25] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP response (548 bytes) from TLS:64.9.242.108:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.128.7:5061;rport=45855;branch=z9hG4bKPj7eca58f8-5756-469b-b22d-537bcb8b5cbf;received=<my public ip>;alias
Record-Route: <sip:ADAOKMOF66AMLN3RPG2VFJUO67UOJRZMKGS6JY2EUJ4X4RMM3ZRNPRQBVQC6PEU:5060;lr;transport=udp;uri-econt=HMMMRJT6S>
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
To: <sip:<my 10D cell>@obihai.sip.google.com>
From: <sip:<my 164 gvoice>@192.168.128.7>;tag=c3a34301-f273-499a-af8f-af5c03e7dda8
Call-ID: 84ec934c-d7db-4468-8f36-ea59e778bfd7
CSeq: 28907 INVITE
Content-Length: 0


[2018-09-07 18:53:26] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP response (1364 bytes) from TLS:64.9.242.108:5061 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 192.168.128.7:5061;rport=45855;branch=z9hG4bKPj7eca58f8-5756-469b-b22d-537bcb8b5cbf;received=<my public ip>;alias
Record-Route: <sip:ADAOKMOF66AMLN3RPG2VFJUO67UOJRZMKGS6JY2EUJ4X4RMM3ZRNPRQBVQC6PEU:5060;lr;transport=udp;uri-econt=HMMMRJT6S>
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
Contact: <sip:<my 164 gvoice>@AAZZHPMXKUEGJMHQJCEHV4JQEMSR24M3TNJP2T77KXEM7JBOXRX3AZD3IWK3OQT:5060;transport=udp;uri-econt=DOHVD5CVUQE52B2K44GFBWB3TGORQ>
To: <sip:<my 10D cell>@obihai.sip.google.com>;tag=235679531
From: <sip:<my 164 gvoice>@192.168.128.7>;tag=c3a34301-f273-499a-af8f-af5c03e7dda8
Call-ID: 84ec934c-d7db-4468-8f36-ea59e778bfd7
CSeq: 28907 INVITE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Type: application/sdp
Content-Length: 566

v=0
o=- 1735903471 1536364406094 IN IP4 74.125.39.26
s=SIP Call
c=IN IP4 74.125.39.26
t=0 0
a=ice-lite
a=ice-pwd:z9tM6jTWkZGIIbr4XvTaMUS0
a=ice-ufrag:GfKZR+RqZ9ajB3s6
a=group:BUNDLE audio
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
a=setup:passive
m=audio 19305 RTP/AVP 0 101
a=mid:audio
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=rtcp-mux
a=candidate:1 1 UDP 1 74.125.39.26 19305 typ host
a=candidate:2 1 UDP 2 2001:4860:4864:2::26 19305 typ host
a=sendrecv

[2018-09-07 18:53:26] VERBOSE[23313] res_rtp_asterisk.c: 0x7f82300ddd20 -- Strict RTP learning after remote address set to: 74.125.39.26:19305
[2018-09-07 18:53:26] ERROR[23313] pjproject: 	   icess0x7f82300ee1a8 ......Error sending STUN request: Network is unreachable
[2018-09-07 18:53:26] VERBOSE[23489][C-00000005] app_dial.c: PJSIP/<my 10d gvoice>-00000008 is making progress passing it to PJSIP/cucm-00000006
[2018-09-07 18:53:26] VERBOSE[23489][C-00000005] app_dial.c: PJSIP/<my 10d gvoice>-00000008 is making progress passing it to PJSIP/cucm-00000006
[2018-09-07 18:53:26] VERBOSE[21258] res_rtp_asterisk.c: 0x7f82300ddd20 -- Strict RTP learning after ICE completion
[2018-09-07 18:53:26] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP response (756 bytes) from TLS:64.9.242.108:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 192.168.128.7:5061;rport=45855;branch=z9hG4bKPj7eca58f8-5756-469b-b22d-537bcb8b5cbf;received=<my public ip>;alias
Record-Route: <sip:ADAOKMOF66AMLN3RPG2VFJUO67UOJRZMKGS6JY2EUJ4X4RMM3ZRNPRQBVQC6PEU:5060;lr;transport=udp;uri-econt=HMMMRJT6S>
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
Contact: <sip:<my 164 gvoice>@AAZZHPMXKUEGJMHQJCEHV4JQEMSR24M3TNJP2T77KXEM7JBOXRX3AZD3IWK3OQT:5060;transport=udp;uri-econt=DOHVD5CVUQE52B2K44GFBWB3TGORQ>
To: <sip:<my 10D cell>@obihai.sip.google.com>;tag=235679531
From: <sip:<my 164 gvoice>@192.168.128.7>;tag=c3a34301-f273-499a-af8f-af5c03e7dda8
Call-ID: 84ec934c-d7db-4468-8f36-ea59e778bfd7
CSeq: 28907 INVITE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Length: 0


[2018-09-07 18:53:26] VERBOSE[23489][C-00000005] app_dial.c: PJSIP/<my 10d gvoice>-00000008 is ringing
[2018-09-07 18:53:26] VERBOSE[23489][C-00000005] app_dial.c: PJSIP/<my 10d gvoice>-00000008 is ringing
[2018-09-07 18:53:26] VERBOSE[23313] res_pjsip_logger.c: <--- Transmitting SIP response (683 bytes) to UDP:192.168.128.12:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.128.12:5060;rport=5060;received=192.168.128.12;branch=z9hG4bK380ff14f08507
Call-ID: 34578e00-b9310f75-379eb-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445172~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693970
To: <sip:<my 10D cell>@mydomain.com>;tag=77bc5c6a-8e33-4c9a-9f12-87685c5d8e46
CSeq: 101 INVITE
Server: Asterisk PBX GIT-master-b300c563e8
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Contact: <sip:192.168.128.7:5060>
Remote-Party-ID: <sip:s-CHANUNAVAIL@mydomain.com>;privacy=off;screen=no
Content-Length:  0


[2018-09-07 18:53:30] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP response (1350 bytes) from TLS:64.9.242.108:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.128.7:5061;rport=45855;branch=z9hG4bKPj7eca58f8-5756-469b-b22d-537bcb8b5cbf;received=<my public ip>;alias
Record-Route: <sip:ADAOKMOF66AMLN3RPG2VFJUO67UOJRZMKGS6JY2EUJ4X4RMM3ZRNPRQBVQC6PEU:5060;lr;transport=udp;uri-econt=HMMMRJT6S>
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
Contact: <sip:<my 164 gvoice>@AAZZHPMXKUEGJMHQJCEHV4JQEMSR24M3TNJP2T77KXEM7JBOXRX3AZD3IWK3OQT:5060;transport=udp;uri-econt=DOHVD5CVUQE52B2K44GFBWB3TGORQ>
To: <sip:<my 10D cell>@obihai.sip.google.com>;tag=235679531
From: <sip:<my 164 gvoice>@192.168.128.7>;tag=c3a34301-f273-499a-af8f-af5c03e7dda8
Call-ID: 84ec934c-d7db-4468-8f36-ea59e778bfd7
CSeq: 28907 INVITE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Type: application/sdp
Content-Length: 566

v=0
o=- 1735903471 1536364406094 IN IP4 74.125.39.26
s=SIP Call
c=IN IP4 74.125.39.26
t=0 0
a=ice-lite
a=ice-pwd:z9tM6jTWkZGIIbr4XvTaMUS0
a=ice-ufrag:GfKZR+RqZ9ajB3s6
a=group:BUNDLE audio
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
a=setup:passive
m=audio 19305 RTP/AVP 0 101
a=mid:audio
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=rtcp-mux
a=candidate:1 1 UDP 1 74.125.39.26 19305 typ host
a=candidate:2 1 UDP 2 2001:4860:4864:2::26 19305 typ host
a=sendrecv

[2018-09-07 18:53:30] VERBOSE[23313] res_pjsip_logger.c: <--- Transmitting SIP request (715 bytes) to TLS:64.9.242.108:5061 --->
ACK sip:<my 164 gvoice>@AAZZHPMXKUEGJMHQJCEHV4JQEMSR24M3TNJP2T77KXEM7JBOXRX3AZD3IWK3OQT:5060;transport=udp;uri-econt=DOHVD5CVUQE52B2K44GFBWB3TGORQ SIP/2.0
Via: SIP/2.0/TLS 192.168.128.7:5061;rport;branch=z9hG4bKPj73e7a20d-e1cb-400e-912a-ca422cc66fd4;alias
From: <sip:<my 164 gvoice>@192.168.128.7>;tag=c3a34301-f273-499a-af8f-af5c03e7dda8
To: <sip:<my 10D cell>@obihai.sip.google.com>;tag=235679531
Call-ID: 84ec934c-d7db-4468-8f36-ea59e778bfd7
CSeq: 28907 ACK
Route: <sip:64.9.242.108:5061;transport=tls;lr>
Route: <sip:ADAOKMOF66AMLN3RPG2VFJUO67UOJRZMKGS6JY2EUJ4X4RMM3ZRNPRQBVQC6PEU:5060;transport=udp;lr;uri-econt=HMMMRJT6S>
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:53:30] VERBOSE[23489][C-00000005] app_dial.c: PJSIP/<my 10d gvoice>-00000008 answered PJSIP/cucm-00000006
[2018-09-07 18:53:30] VERBOSE[23313] res_rtp_asterisk.c: 0x7f82300d78a0 -- Strict RTP learning after remote address set to: 192.168.128.134:19486
[2018-09-07 18:53:30] VERBOSE[23313] res_rtp_asterisk.c: 0x7f823010c470 -- Strict RTP learning after remote address set to: 192.168.128.134:19634
[2018-09-07 18:53:30] VERBOSE[23313] res_pjsip_logger.c: <--- Transmitting SIP response (1287 bytes) to UDP:192.168.128.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.128.12:5060;rport=5060;received=192.168.128.12;branch=z9hG4bK380ff14f08507
Call-ID: 34578e00-b9310f75-379eb-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445172~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693970
To: <sip:<my 10D cell>@mydomain.com>;tag=77bc5c6a-8e33-4c9a-9f12-87685c5d8e46
CSeq: 101 INVITE
Server: Asterisk PBX GIT-master-b300c563e8
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Contact: <sip:192.168.128.7:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Remote-Party-ID: <sip:s-CHANUNAVAIL@mydomain.com>;privacy=off;screen=no
Content-Type: application/sdp
Content-Length:   474

v=0
o=- 445172 3 IN IP4 192.168.128.7
s=Asterisk
c=IN IP4 192.168.128.7
t=0 0
m=audio 19356 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 19334 RTP/AVP 100
a=rtpmap:100 H264/90000
a=fmtp:100 max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000;packetization-mode=1;level-asymmetry-allowed=1
a=sendrecv
m=video 0 RTP/AVP 100 126 97
m=application 0 UDP/BFCP *

[2018-09-07 18:53:30] VERBOSE[23496][C-00000005] bridge_channel.c: Channel PJSIP/<my 10d gvoice>-00000008 joined 'simple_bridge' basic-bridge <33668dfb-e3e2-45a1-aff3-3bcba979bd22>
[2018-09-07 18:53:30] VERBOSE[23489][C-00000005] bridge_channel.c: Channel PJSIP/cucm-00000006 joined 'simple_bridge' basic-bridge <33668dfb-e3e2-45a1-aff3-3bcba979bd22>
[2018-09-07 18:53:30] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP request (504 bytes) from UDP:192.168.128.12:5060 --->
ACK sip:192.168.128.7:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.12:5060;branch=z9hG4bK3810135da8786
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445172~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693970
To: <sip:<my 10D cell>@mydomain.com>;tag=77bc5c6a-8e33-4c9a-9f12-87685c5d8e46
Date: Fri, 07 Sep 2018 23:53:25 GMT
Call-ID: 34578e00-b9310f75-379eb-c80a8c0@192.168.128.12
User-Agent: Cisco-CP-DX650/10.2.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0


[2018-09-07 18:53:30] VERBOSE[23489][C-00000005] res_rtp_asterisk.c: 0x7f82300d78a0 -- Strict RTP switching to RTP target address 192.168.128.134:19486 as source
[2018-09-07 18:53:30] VERBOSE[23496][C-00000005] res_rtp_asterisk.c: 0x7f82300ddd20 -- Strict RTP switching to RTP target address 74.125.39.26:19305 as source
[2018-09-07 18:53:30] VERBOSE[23489][C-00000005] res_rtp_asterisk.c: 0x7f823010c470 -- Strict RTP switching to RTP target address 192.168.128.134:19634 as source
[2018-09-07 18:53:31] VERBOSE[23496][C-00000005] res_rtp_asterisk.c: 0x7f82300ddd20 -- Strict RTP learning complete - Locking on source address 74.125.39.26:19305
[2018-09-07 18:53:35] VERBOSE[23489][C-00000005] res_rtp_asterisk.c: 0x7f82300d78a0 -- Strict RTP learning complete - Locking on source address 192.168.128.134:19486
[2018-09-07 18:53:35] VERBOSE[23489][C-00000005] res_rtp_asterisk.c: 0x7f823010c470 -- Strict RTP learning complete - Locking on source address 192.168.128.134:19634
[2018-09-07 18:53:42] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP request (415 bytes) from UDP:5.189.226.100:5701 --->
OPTIONS sip:100@<my public ip> SIP/2.0
Via: SIP/2.0/UDP 5.189.226.100:5701;branch=z9hG4bK-3092043321;rport
Content-Length: 0
From: "sipvicious"<sip:100@1.1.1.1>;tag=3435383239636433313363340131393230343037363133
Accept: application/sdp
User-Agent: friendly-scanner
To: "sipvicious"<sip:100@1.1.1.1>
Contact: sip:100@5.189.226.100:5701
CSeq: 1 OPTIONS
Call-ID: 192640479108589336732620
Max-Forwards: 70


[2018-09-07 18:53:42] VERBOSE[23313] res_pjsip_logger.c: <--- Transmitting SIP response (873 bytes) to UDP:5.189.226.100:5701 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 5.189.226.100:5701;rport=5701;received=5.189.226.100;branch=z9hG4bK-3092043321
Call-ID: 192640479108589336732620
From: "sipvicious" <sip:100@1.1.1.1>;tag=3435383239636433313363340131393230343037363133
To: "sipvicious" <sip:100@1.1.1.1>;tag=z9hG4bK-3092043321
CSeq: 1 OPTIONS
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
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: text/plain
Accept-Language: en
Server: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:53:59] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP request (397 bytes) from UDP:192.168.128.12:5060 --->
OPTIONS sip:mydomain.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.12:5060;branch=z9hG4bK381021b9634f
From: <sip:192.168.128.12>;tag=156038798
To: <sip:mydomain.com>
Date: Fri, 07 Sep 2018 23:53:59 GMT
Call-ID: 489b8b00-b9310f97-379ec-c80a8c0@192.168.128.12
User-Agent: Cisco-CUCM11.5
CSeq: 101 OPTIONS
Contact: <sip:192.168.128.12:5060>
Max-Forwards: 0
Content-Length: 0


[2018-09-07 18:53:59] VERBOSE[23313] res_pjsip_logger.c: <--- Transmitting SIP response (841 bytes) to UDP:192.168.128.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.128.12:5060;rport=5060;received=192.168.128.12;branch=z9hG4bK381021b9634f
Call-ID: 489b8b00-b9310f97-379ec-c80a8c0@192.168.128.12
From: <sip:192.168.128.12>;tag=156038798
To: <sip:mydomain.com>;tag=z9hG4bK381021b9634f
CSeq: 101 OPTIONS
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
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: text/plain
Accept-Language: en
Server: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:54:14] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP request (973 bytes) from TLS:64.9.242.108:5061 --->
BYE sip:asterisk@192.168.128.7:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 64.9.242.108:5061;branch=z9hG4bK-524287-1---6c80dbbff2d8f0863e261ebd9a45ab73;rport
Via: SIP/2.0/UDP ADAOKMOFVFRAUO3M4WMLJ4Z464C7IL2XWHOM62EGLHYEVNZ3Q4XYZBUANUL2IS7:5060;branch=z9hG4bK-524287-1---c6a581a8452b906df4cf365b482de994;econt=VLZKSPAVBI47DYWEM4U
Via: SIP/2.0/UDP AAZZHPMXEUSYWQBK6ICDQYMWRWPYSO5G7MJAXWZS2KQKR4Q7QBVW646J3RRHQR3:5060;branch=z9hG4bK72751773;econt=3NUPW5FK5HXJDHFHY4LTRYISE77YYLGQQF7OWCQZPJP7F37EF6UDVFRMS
Max-Forwards: 68
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
Record-Route: <sip:ADAOKMOF66AMLN3RPG2VFJUO67UOJRZMKGS6JY2EUJ4X4RMM3ZRNPRQBVQC6PEU:5060;lr;transport=udp;uri-econt=HMMMRJT6S>
To: <sip:<my 164 gvoice>@192.168.128.7>;tag=c3a34301-f273-499a-af8f-af5c03e7dda8
From: <sip:<my 10D cell>@obihai.sip.google.com>;tag=235679531
Call-ID: 84ec934c-d7db-4468-8f36-ea59e778bfd7
CSeq: 393318 BYE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Length: 0


[2018-09-07 18:54:14] VERBOSE[23313] res_pjsip_logger.c: <--- Transmitting SIP response (943 bytes) to TLS:64.9.242.108:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 64.9.242.108:5061;rport=5061;received=64.9.242.108;branch=z9hG4bK-524287-1---6c80dbbff2d8f0863e261ebd9a45ab73
Via: SIP/2.0/UDP ADAOKMOFVFRAUO3M4WMLJ4Z464C7IL2XWHOM62EGLHYEVNZ3Q4XYZBUANUL2IS7:5060;branch=z9hG4bK-524287-1---c6a581a8452b906df4cf365b482de994;econt=VLZKSPAVBI47DYWEM4U
Via: SIP/2.0/UDP AAZZHPMXEUSYWQBK6ICDQYMWRWPYSO5G7MJAXWZS2KQKR4Q7QBVW646J3RRHQR3:5060;branch=z9hG4bK72751773;econt=3NUPW5FK5HXJDHFHY4LTRYISE77YYLGQQF7OWCQZPJP7F37EF6UDVFRMS
Record-Route: <sip:64.9.242.108:5061;transport=tls;lr>
Record-Route: <sip:ADAOKMOF66AMLN3RPG2VFJUO67UOJRZMKGS6JY2EUJ4X4RMM3ZRNPRQBVQC6PEU:5060;transport=udp;lr;uri-econt=HMMMRJT6S>
Call-ID: 84ec934c-d7db-4468-8f36-ea59e778bfd7
From: <sip:<my 10D cell>@obihai.sip.google.com>;tag=235679531
To: <sip:<my 164 gvoice>@192.168.128.7>;tag=c3a34301-f273-499a-af8f-af5c03e7dda8
CSeq: 393318 BYE
Server: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:54:14] VERBOSE[23496][C-00000005] bridge_channel.c: Channel PJSIP/<my 10d gvoice>-00000008 left 'simple_bridge' basic-bridge <33668dfb-e3e2-45a1-aff3-3bcba979bd22>
[2018-09-07 18:54:14] VERBOSE[23489][C-00000005] bridge_channel.c: Channel PJSIP/cucm-00000006 left 'simple_bridge' basic-bridge <33668dfb-e3e2-45a1-aff3-3bcba979bd22>
[2018-09-07 18:54:14] VERBOSE[23489][C-00000005] pbx.c: Spawn extension (dialprovider, s-CHANUNAVAIL, 1) exited non-zero on 'PJSIP/cucm-00000006'
[2018-09-07 18:54:14] VERBOSE[23313] res_pjsip_logger.c: <--- Transmitting SIP request (524 bytes) to UDP:192.168.128.12:5060 --->
BYE sip:<my 10d gvoice>@192.168.128.12:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.7:5060;rport;branch=z9hG4bKPjc8b81ddf-23a8-48bc-aa54-1d64fde38b72
From: <sip:<my 10D cell>@mydomain.com>;tag=77bc5c6a-8e33-4c9a-9f12-87685c5d8e46
To: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445172~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693970
Call-ID: 34578e00-b9310f75-379eb-c80a8c0@192.168.128.12
CSeq: 4517 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:54:14] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP response (469 bytes) from UDP:192.168.128.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.128.7:5060;rport;branch=z9hG4bKPjc8b81ddf-23a8-48bc-aa54-1d64fde38b72
From: <sip:<my 10D cell>@mydomain.com>;tag=77bc5c6a-8e33-4c9a-9f12-87685c5d8e46
To: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445172~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693970
Date: Fri, 07 Sep 2018 23:54:14 GMT
Call-ID: 34578e00-b9310f75-379eb-c80a8c0@192.168.128.12
Server: Cisco-CP-DX650/10.2.5
CSeq: 4517 BYE
Content-Length: 0