[Sep  4 12:32:12] Asterisk GIT-master-b300c563e8 built by root @ dznet-pbx on a x86_64 running Linux on 2018-09-04 17:11:54 UTC
[Sep  4 12:32:12] VERBOSE[30765] logger.c: Asterisk Queue Logger restarted
[Sep  4 12:32:22] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP request (3681 bytes) from UDP:192.168.128.12:5060 --->
INVITE sip:<my 10 digit cellphone>@mydomain.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.12:5060;branch=z9hG4bK35c8c30fa5bce
From: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;tag=429787~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693932
To: <sip:<my 10 digit cellphone>@mydomain.com>
Date: Tue, 04 Sep 2018 17:32:22 GMT
Call-ID: 79b15c80-b8e1c1a6-355a7-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: 107dd6d000105000a0005017ff96e069;remote=00000000000000000000000000000000
Cisco-Guid: 2041666688-0000065536-0000000400-0209758400
P-Charging-Vector: icid-value="79B15C80000100000000018F0C80A8C0";icid-generated-at=dznet-ucm;orig-ioi="IMS Inter Operator Identification"
Session-Expires:  1800
P-Asserted-Identity: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>
Remote-Party-ID: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;party=calling;screen=yes;privacy=off
Contact: <sip:<my 10 digit CALLING #>@192.168.128.12:5060>;video;audio;+u.sip!devicename.ccm.cisco.com="SEP5017FF96E069";bfcp
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 2238

v=0
o=CiscoSystemsCCM-SIP 429787 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 19646 RTP/AVP 108 9 124 0 8 116 18 101
b=TIAS:64000
a=rtpmap:108 MP4A-LATM/90000
a=fmtp:108 bitrate=64000;profile-level-id=24;object=23
a=rtpmap:9 G722/8000
a=rtpmap:124 iSAC/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=maxptime:20
a=fmtp:116 mode=20
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 19414 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 19466 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 19976 UDP/BFCP *
a=floorctrl:s-only c-only
a=floorid:3 mstrm:12
a=confid:1
a=userid:11

[Sep  4 12:32:22] VERBOSE[30731] pbx_variables.c: Setting global variable 'SIPDOMAIN' to 'mydomain.com'
[Sep  4 12:32:22] VERBOSE[30731] 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=z9hG4bK35c8c30fa5bce
Call-ID: 79b15c80-b8e1c1a6-355a7-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;tag=429787~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693932
To: <sip:<my 10 digit cellphone>@mydomain.com>
CSeq: 101 INVITE
Server: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[Sep  4 12:32:22] VERBOSE[30951][C-00000002] pbx.c: Executing [<my 10 digit cellphone>@home:1] GotoIf("PJSIP/cucm-00000003", "1?numeric") in new stack
[Sep  4 12:32:22] VERBOSE[30951][C-00000002] pbx_builtins.c: Goto (home,<my 10 digit cellphone>,4)
[Sep  4 12:32:22] VERBOSE[30951][C-00000002] pbx.c: Executing [<my 10 digit cellphone>@home:4] Gosub("PJSIP/cucm-00000003", "dialprovider,s,1(<my 10 digit cellphone>)") in new stack
[Sep  4 12:32:22] VERBOSE[30951][C-00000002] pbx.c: Executing [s@dialprovider:1] NoOp("PJSIP/cucm-00000003", " printing full callerid -- "My Name" <<my 10 digit CALLING #>>") in new stack
[Sep  4 12:32:22] VERBOSE[30951][C-00000002] pbx.c: Executing [s@dialprovider:2] NoOp("PJSIP/cucm-00000003", " printing the sip domain -- mydomain.com") in new stack
[Sep  4 12:32:22] VERBOSE[30951][C-00000002] pbx.c: Executing [s@dialprovider:3] Set("PJSIP/cucm-00000003", "CALLERID(all)=<<my e164 CALLING #>>") in new stack
[Sep  4 12:32:22] VERBOSE[30951][C-00000002] pbx.c: Executing [s@dialprovider:4] NoOp("PJSIP/cucm-00000003", " printing the extension -- <my 10 digit cellphone>") in new stack
[Sep  4 12:32:22] VERBOSE[30951][C-00000002] pbx.c: Executing [s@dialprovider:5] Dial("PJSIP/cucm-00000003", "PJSIP/<my e164 cellphone>@sipbroker-out") in new stack
[Sep  4 12:32:22] DEBUG[30731] res_pjsip_outbound_registration.c: Outgoing request not associated with a registration. No mangling necessary.
[Sep  4 12:32:22] VERBOSE[30951][C-00000002] app_dial.c: Called PJSIP/<my e164 cellphone>@sipbroker-out
[Sep  4 12:32:22] DEBUG[30731] res_pjsip_outbound_registration.c: Outgoing request not associated with a registration. No mangling necessary.
[Sep  4 12:32:22] VERBOSE[30731] res_pjsip_logger.c: <--- Transmitting SIP request (1192 bytes) to UDP:204.11.194.25:5060 --->
INVITE sip:<my e164 cellphone>@sipbroker.com:5060 SIP/2.0
Via: SIP/2.0/UDP <my public IP>:5060;rport;branch=z9hG4bKPjd5b0d34c-8b11-417c-839f-e1ee41488964
From: <sip:driz@mydomain.com>;tag=b6a23e40-9fb2-407e-9db7-057379c1123f
To: <sip:<my e164 cellphone>@sipbroker.com>
Contact: <sip:driz@<my public IP>:5060>
Call-ID: f1f4014f-8aa7-4510-8101-9f1573e9659b
CSeq: 7098 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 e164 CALLING #>@mydomain.com>;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Type: application/sdp
Content-Length:   426

v=0
o=- 154115894 154115894 IN IP4 <my public IP>
s=Asterisk
c=IN IP4 <my public IP>
t=0 0
m=audio 19626 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 19596 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 max-mbps=267300;max-fs=8910;max-fps=6000;max-rcmd-nalu-size=256000;packetization-mode=1;level-asymmetry-allowed=1
a=sendrecv

[Sep  4 12:32:22] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP response (580 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=z9hG4bKPjd5b0d34c-8b11-417c-839f-e1ee41488964
From: <sip:driz@mydomain.com>;tag=b6a23e40-9fb2-407e-9db7-057379c1123f
To: <sip:<my e164 cellphone>@sipbroker.com>
Call-ID: f1f4014f-8aa7-4510-8101-9f1573e9659b
CSeq: 7098 INVITE
Server: OpenSer (1.1.0-notls (x86_64/linux))
Content-Length: 0
Warning: 392 204.11.194.25:5060 "Noisy feedback tells:  pid=3452 req_src_ip=<my public IP> req_src_port=1024 in_uri=sip:<my e164 cellphone>@sipbroker.com:5060 out_uri=sip:<my e164 cellphone>@sipbroker.com:5060 via_cnt==1"


[Sep  4 12:32:23] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP response (668 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=z9hG4bKPjd5b0d34c-8b11-417c-839f-e1ee41488964
From: <sip:driz@mydomain.com>;tag=b6a23e40-9fb2-407e-9db7-057379c1123f
To: <sip:<my e164 cellphone>@sipbroker.com>;tag=2b8506bb96abbbb8b95a41b9af69a614.0527
Call-ID: f1f4014f-8aa7-4510-8101-9f1573e9659b
CSeq: 7098 INVITE
Contact: sip:<my 11 digit cellphone>@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=3452 req_src_ip=<my public IP> req_src_port=1024 in_uri=sip:<my e164 cellphone>@sipbroker.com:5060 out_uri=sip:<my 11 digit cellphone>@mydomain.com via_cnt==1"


[Sep  4 12:32:23] VERBOSE[30731] res_pjsip_logger.c: <--- Transmitting SIP request (455 bytes) to UDP:204.11.194.25:5060 --->
ACK sip:<my e164 cellphone>@sipbroker.com:5060 SIP/2.0
Via: SIP/2.0/UDP <my public IP>:5060;rport;branch=z9hG4bKPjd5b0d34c-8b11-417c-839f-e1ee41488964
From: <sip:driz@mydomain.com>;tag=b6a23e40-9fb2-407e-9db7-057379c1123f
To: <sip:<my e164 cellphone>@sipbroker.com>;tag=2b8506bb96abbbb8b95a41b9af69a614.0527
Call-ID: f1f4014f-8aa7-4510-8101-9f1573e9659b
CSeq: 7098 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[Sep  4 12:32:23] VERBOSE[30951][C-00000002] app_dial.c: Now forwarding PJSIP/cucm-00000003 to 'Local/<my 11 digit cellphone>@unauthenticated' (thanks to PJSIP/sipbroker-out-00000004)
[Sep  4 12:32:23] NOTICE[30951][C-00000002] app_dial.c: Not accepting call completion offers from call-forward recipient Local/<my 11 digit cellphone>@unauthenticated-00000001;1
[Sep  4 12:32:23] NOTICE[30951][C-00000002] core_local.c: No such extension/context <my 11 digit cellphone>@unauthenticated while calling Local channel
[Sep  4 12:32:23] NOTICE[30951][C-00000002] app_dial.c: Forwarding failed to dial 'Local/<my 11 digit cellphone>@unauthenticated'
[Sep  4 12:32:23] VERBOSE[30951][C-00000002] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[Sep  4 12:32:23] VERBOSE[30951][C-00000002] pbx.c: Executing [s@dialprovider:6] NoOp("PJSIP/cucm-00000003", " Dial Status: CHANUNAVAIL") in new stack
[Sep  4 12:32:23] VERBOSE[30951][C-00000002] pbx.c: Executing [s@dialprovider:7] Goto("PJSIP/cucm-00000003", "s-CHANUNAVAIL,1") in new stack
[Sep  4 12:32:23] VERBOSE[30951][C-00000002] pbx_builtins.c: Goto (dialprovider,s-CHANUNAVAIL,1)
[Sep  4 12:32:23] VERBOSE[30951][C-00000002] pbx.c: Executing [s-CHANUNAVAIL@dialprovider:1] Dial("PJSIP/cucm-00000003", "PJSIP/<my 10 digit cellphone>@<my 10 digit CALLING #>,,r") in new stack
[Sep  4 12:32:23] VERBOSE[30954] 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=z9hG4bK35c8c30fa5bce
Call-ID: 79b15c80-b8e1c1a6-355a7-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;tag=429787~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693932
To: <sip:<my 10 digit cellphone>@mydomain.com>;tag=20b77b06-a4c6-4983-8851-a8db829e1847
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


[Sep  4 12:32:23] DEBUG[30954] res_pjsip_outbound_registration.c: Setting transport to 0x7f3a9c410ca8
[Sep  4 12:32:23] DEBUG[30954] res_pjsip.c: Overriding endpoint transport to use 0x7f3a9c410ca8
[Sep  4 12:32:23] VERBOSE[30951][C-00000002] app_dial.c: Called PJSIP/<my 10 digit cellphone>@<my 10 digit CALLING #>
[Sep  4 12:32:23] VERBOSE[30954] 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=z9hG4bK35c8c30fa5bce
Call-ID: 79b15c80-b8e1c1a6-355a7-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;tag=429787~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693932
To: <sip:<my 10 digit cellphone>@mydomain.com>;tag=20b77b06-a4c6-4983-8851-a8db829e1847
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


[Sep  4 12:32:23] DEBUG[30731] res_pjsip_outbound_registration.c: Found matching outbound registration state
[Sep  4 12:32:23] DEBUG[30731] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:ADW267E7QDYVJULDIFALPWCPJCVK75WYZLLUAC5O2A2BV226R6PVGIYD4RD22MY:5060;uri-econt=HCRAAPX65ESRY3USRTVGMCBBV7O2LVZB433NF2XKTWGWOK7OK2OAROWCL2Q4IZBXL6BOTDV47DXQ64YLFHGEFGE4AJ4MB5PUN6AXPXIDOUGUA7VQTLUZDYIEO5RKD2CJ7RY2DM;lr>
[Sep  4 12:32:23] DEBUG[30731] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:ADAOKMOFVTHS2X43M4NFJRRDMAJJBCH74MP5HNFPLWSDVHE6FTVV6DVTGNNQMOR:5060;transport=udp;lr;uri-econt=X3QFGCCJY>
[Sep  4 12:32:23] VERBOSE[30954] res_pjsip_logger.c: <--- Transmitting SIP request (2039 bytes) to TLS:64.9.242.108:5061 --->
INVITE sip:<my 10 digit cellphone>@obihai.sip.google.com SIP/2.0
Via: SIP/2.0/TLS 192.168.128.7:5061;rport;branch=z9hG4bKPj54a1e01a-85a3-484a-bf3f-9ac3fce29907;alias
From: <sip:<my e164 CALLING #>@192.168.128.7>;tag=df13a723-ffba-4539-bbf1-1fb58f8c0f0d
To: <sip:<my 10 digit cellphone>@obihai.sip.google.com>
Contact: <sip:asterisk@192.168.128.7:5061;transport=TLS>
Call-ID: 38759f4b-023f-41f4-9201-13d65c681bf1
CSeq: 30481 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:ADW267E7QDYVJULDIFALPWCPJCVK75WYZLLUAC5O2A2BV226R6PVGIYD4RD22MY:5060;uri-econt=HCRAAPX65ESRY3USRTVGMCBBV7O2LVZB433NF2XKTWGWOK7OK2OAROWCL2Q4IZBXL6BOTDV47DXQ64YLFHGEFGE4AJ4MB5PUN6AXPXIDOUGUA7VQTLUZDYIEO5RKD2CJ7RY2DM;lr>
Route: <sip:ADAOKMOFVTHS2X43M4NFJRRDMAJJBCH74MP5HNFPLWSDVHE6FTVV6DVTGNNQMOR:5060;transport=udp;lr;uri-econt=X3QFGCCJY>
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:   843

v=0
o=- 656189483 656189483 IN IP4 192.168.128.7
s=Asterisk
c=IN IP4 192.168.128.7
t=0 0
m=audio 19274 RTP/AVP 0 101
a=ice-ufrag:6c23a59d18283a176df8399d31a35ed8
a=ice-pwd:2f83db0360693c8f346d670210785d81
a=candidate:Ha6e76162 1 UDP 2130706431 fe80::20c:29ff:fe43:c08d 19274 typ host
a=candidate:Hc0a88007 1 UDP 2130706431 192.168.128.7 19274 typ host
a=candidate:S45829cd3 1 UDP 1694498815 <my public IP> 19274 typ srflx raddr 192.168.128.7 rport 19274
a=candidate:Ha6e76162 2 UDP 2130706430 fe80::20c:29ff:fe43:c08d 19275 typ host
a=candidate:Hc0a88007 2 UDP 2130706430 192.168.128.7 19275 typ host
a=candidate:S45829cd3 2 UDP 1694498814 <my public IP> 19275 typ srflx raddr 192.168.128.7 rport 19275
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

[Sep  4 12:32:23] VERBOSE[30730] 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=37425;branch=z9hG4bKPj54a1e01a-85a3-484a-bf3f-9ac3fce29907;received=<my public IP>;alias
Record-Route: <sip:ADAOKMOFOOFRCQ2I33VKRLIQIYKFPIMPYJF7A2KHNOC5IMN4ETZFWQGI376LMP3:5060;lr;transport=udp;uri-econt=BJOHZWFQE>
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
To: <sip:<my 10 digit cellphone>@obihai.sip.google.com>
From: <sip:<my e164 CALLING #>@192.168.128.7>;tag=df13a723-ffba-4539-bbf1-1fb58f8c0f0d
Call-ID: 38759f4b-023f-41f4-9201-13d65c681bf1
CSeq: 30481 INVITE
Content-Length: 0


[Sep  4 12:32:23] VERBOSE[30730] 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=37425;branch=z9hG4bKPj54a1e01a-85a3-484a-bf3f-9ac3fce29907;received=<my public IP>;alias
Record-Route: <sip:ADAOKMOFOOFRCQ2I33VKRLIQIYKFPIMPYJF7A2KHNOC5IMN4ETZFWQGI376LMP3:5060;lr;transport=udp;uri-econt=BJOHZWFQE>
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
Contact: <sip:<my e164 CALLING #>@AAZZHPMXVTLHDSWAVIJZAGIHIWNBLLYZLSRXDZK4FSLA7QTXOPWDZLP72GEUDQI:5060;transport=udp;uri-econt=4DRGJR3JLSNO2Z2QEIH5345R6PDSQ>
To: <sip:<my 10 digit cellphone>@obihai.sip.google.com>;tag=1722332733
From: <sip:<my e164 CALLING #>@192.168.128.7>;tag=df13a723-ffba-4539-bbf1-1fb58f8c0f0d
Call-ID: 38759f4b-023f-41f4-9201-13d65c681bf1
CSeq: 30481 INVITE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Type: application/sdp
Content-Length: 565

v=0
o=- 922692888 1536082343721 IN IP4 74.125.39.60
s=SIP Call
c=IN IP4 74.125.39.60
t=0 0
a=ice-lite
a=ice-pwd:6dh91hQ3khRtOY7kaWEyVEsE
a=ice-ufrag:98RifgViJ+31wknQ
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 74.125.39.60 19305 typ host
a=candidate:2 1 UDP 2 2001:4860:4864:2::60 19305 typ host
a=sendrecv

[Sep  4 12:32:23] VERBOSE[30951][C-00000002] app_dial.c: PJSIP/<my 10 digit CALLING #>-00000005 is making progress passing it to PJSIP/cucm-00000003
[Sep  4 12:32:23] VERBOSE[30731] res_rtp_asterisk.c: 0x7f3a9c45ee70 -- Strict RTP learning after remote address set to: 74.125.39.60:19305
[Sep  4 12:32:23] ERROR[30731] pjproject: 	   icess0x7f3a9c485348 ......Error sending STUN request: Network is unreachable
[Sep  4 12:32:23] VERBOSE[30951][C-00000002] app_dial.c: PJSIP/<my 10 digit CALLING #>-00000005 is making progress passing it to PJSIP/cucm-00000003
[Sep  4 12:32:23] VERBOSE[30742] res_rtp_asterisk.c: 0x7f3a9c45ee70 -- Strict RTP learning after ICE completion
[Sep  4 12:32:24] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP response (757 bytes) from TLS:64.9.242.108:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 192.168.128.7:5061;rport=37425;branch=z9hG4bKPj54a1e01a-85a3-484a-bf3f-9ac3fce29907;received=<my public IP>;alias
Record-Route: <sip:ADAOKMOFOOFRCQ2I33VKRLIQIYKFPIMPYJF7A2KHNOC5IMN4ETZFWQGI376LMP3:5060;lr;transport=udp;uri-econt=BJOHZWFQE>
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
Contact: <sip:<my e164 CALLING #>@AAZZHPMXVTLHDSWAVIJZAGIHIWNBLLYZLSRXDZK4FSLA7QTXOPWDZLP72GEUDQI:5060;transport=udp;uri-econt=4DRGJR3JLSNO2Z2QEIH5345R6PDSQ>
To: <sip:<my 10 digit cellphone>@obihai.sip.google.com>;tag=1722332733
From: <sip:<my e164 CALLING #>@192.168.128.7>;tag=df13a723-ffba-4539-bbf1-1fb58f8c0f0d
Call-ID: 38759f4b-023f-41f4-9201-13d65c681bf1
CSeq: 30481 INVITE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Length: 0


[Sep  4 12:32:24] VERBOSE[30951][C-00000002] app_dial.c: PJSIP/<my 10 digit CALLING #>-00000005 is ringing
[Sep  4 12:32:24] VERBOSE[30951][C-00000002] app_dial.c: PJSIP/<my 10 digit CALLING #>-00000005 is ringing
[Sep  4 12:32:24] VERBOSE[30731] 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=z9hG4bK35c8c30fa5bce
Call-ID: 79b15c80-b8e1c1a6-355a7-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;tag=429787~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693932
To: <sip:<my 10 digit cellphone>@mydomain.com>;tag=20b77b06-a4c6-4983-8851-a8db829e1847
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


[Sep  4 12:32:28] VERBOSE[30730] 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=37425;branch=z9hG4bKPj54a1e01a-85a3-484a-bf3f-9ac3fce29907;received=<my public IP>;alias
Record-Route: <sip:ADAOKMOFOOFRCQ2I33VKRLIQIYKFPIMPYJF7A2KHNOC5IMN4ETZFWQGI376LMP3:5060;lr;transport=udp;uri-econt=BJOHZWFQE>
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
Contact: <sip:<my e164 CALLING #>@AAZZHPMXVTLHDSWAVIJZAGIHIWNBLLYZLSRXDZK4FSLA7QTXOPWDZLP72GEUDQI:5060;transport=udp;uri-econt=4DRGJR3JLSNO2Z2QEIH5345R6PDSQ>
To: <sip:<my 10 digit cellphone>@obihai.sip.google.com>;tag=1722332733
From: <sip:<my e164 CALLING #>@192.168.128.7>;tag=df13a723-ffba-4539-bbf1-1fb58f8c0f0d
Call-ID: 38759f4b-023f-41f4-9201-13d65c681bf1
CSeq: 30481 INVITE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Type: application/sdp
Content-Length: 565

v=0
o=- 922692888 1536082343721 IN IP4 74.125.39.60
s=SIP Call
c=IN IP4 74.125.39.60
t=0 0
a=ice-lite
a=ice-pwd:6dh91hQ3khRtOY7kaWEyVEsE
a=ice-ufrag:98RifgViJ+31wknQ
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 74.125.39.60 19305 typ host
a=candidate:2 1 UDP 2 2001:4860:4864:2::60 19305 typ host
a=sendrecv

[Sep  4 12:32:28] VERBOSE[30731] res_pjsip_logger.c: <--- Transmitting SIP request (716 bytes) to TLS:64.9.242.108:5061 --->
ACK sip:<my e164 CALLING #>@AAZZHPMXVTLHDSWAVIJZAGIHIWNBLLYZLSRXDZK4FSLA7QTXOPWDZLP72GEUDQI:5060;transport=udp;uri-econt=4DRGJR3JLSNO2Z2QEIH5345R6PDSQ SIP/2.0
Via: SIP/2.0/TLS 192.168.128.7:5061;rport;branch=z9hG4bKPj78e5908b-5994-4120-9dbc-5b12b300c7e8;alias
From: <sip:<my e164 CALLING #>@192.168.128.7>;tag=df13a723-ffba-4539-bbf1-1fb58f8c0f0d
To: <sip:<my 10 digit cellphone>@obihai.sip.google.com>;tag=1722332733
Call-ID: 38759f4b-023f-41f4-9201-13d65c681bf1
CSeq: 30481 ACK
Route: <sip:64.9.242.108:5061;transport=tls;lr>
Route: <sip:ADAOKMOFOOFRCQ2I33VKRLIQIYKFPIMPYJF7A2KHNOC5IMN4ETZFWQGI376LMP3:5060;transport=udp;lr;uri-econt=BJOHZWFQE>
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[Sep  4 12:32:28] VERBOSE[30951][C-00000002] app_dial.c: PJSIP/<my 10 digit CALLING #>-00000005 answered PJSIP/cucm-00000003
[Sep  4 12:32:28] VERBOSE[30731] res_rtp_asterisk.c: 0x7f3a9c475b10 -- Strict RTP learning after remote address set to: 192.168.128.134:19646
[Sep  4 12:32:28] VERBOSE[30731] res_rtp_asterisk.c: 0x7f3a9c47a6d0 -- Strict RTP learning after remote address set to: 192.168.128.134:19414
[Sep  4 12:32:28] VERBOSE[30731] res_pjsip_logger.c: <--- Transmitting SIP response (1311 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=z9hG4bK35c8c30fa5bce
Call-ID: 79b15c80-b8e1c1a6-355a7-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;tag=429787~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693932
To: <sip:<my 10 digit cellphone>@mydomain.com>;tag=20b77b06-a4c6-4983-8851-a8db829e1847
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:   498

v=0
o=- 429787 3 IN IP4 192.168.128.7
s=Asterisk
c=IN IP4 192.168.128.7
t=0 0
m=audio 19100 RTP/AVP 9 0 101
a=rtpmap:9 G722/8000
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 19966 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 *

[Sep  4 12:32:28] VERBOSE[30966][C-00000002] bridge_channel.c: Channel PJSIP/<my 10 digit CALLING #>-00000005 joined 'simple_bridge' basic-bridge <86303b96-ce45-4278-8146-b6c3935ebada>
[Sep  4 12:32:28] VERBOSE[30951][C-00000002] bridge_channel.c: Channel PJSIP/cucm-00000003 joined 'simple_bridge' basic-bridge <86303b96-ce45-4278-8146-b6c3935ebada>
[Sep  4 12:32:28] DEBUG[30731] res_pjsip_outbound_registration.c: Found matching outbound registration state
[Sep  4 12:32:28] DEBUG[30731] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:ADW267E7QDYVJULDIFALPWCPJCVK75WYZLLUAC5O2A2BV226R6PVGIYD4RD22MY:5060;uri-econt=HCRAAPX65ESRY3USRTVGMCBBV7O2LVZB433NF2XKTWGWOK7OK2OAROWCL2Q4IZBXL6BOTDV47DXQ64YLFHGEFGE4AJ4MB5PUN6AXPXIDOUGUA7VQTLUZDYIEO5RKD2CJ7RY2DM;lr>
[Sep  4 12:32:28] DEBUG[30731] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:ADAOKMOFVTHS2X43M4NFJRRDMAJJBCH74MP5HNFPLWSDVHE6FTVV6DVTGNNQMOR:5060;transport=udp;lr;uri-econt=X3QFGCCJY>
[Sep  4 12:32:28] VERBOSE[30731] res_pjsip_logger.c: <--- Transmitting SIP request (2348 bytes) to TLS:64.9.242.108:5061 --->
INVITE sip:<my e164 CALLING #>@AAZZHPMXVTLHDSWAVIJZAGIHIWNBLLYZLSRXDZK4FSLA7QTXOPWDZLP72GEUDQI:5060;transport=udp;uri-econt=4DRGJR3JLSNO2Z2QEIH5345R6PDSQ SIP/2.0
Via: SIP/2.0/TLS 192.168.128.7:5061;rport;branch=z9hG4bKPj8029d37b-8205-4cb9-97d6-55c689e85987;alias
From: <sip:<my e164 CALLING #>@192.168.128.7>;tag=df13a723-ffba-4539-bbf1-1fb58f8c0f0d
To: <sip:<my 10 digit cellphone>@obihai.sip.google.com>;tag=1722332733
Contact: <sip:asterisk@192.168.128.7:5061;transport=TLS>
Call-ID: 38759f4b-023f-41f4-9201-13d65c681bf1
CSeq: 30482 INVITE
Route: <sip:64.9.242.108:5061;transport=tls;lr>
Route: <sip:ADAOKMOFOOFRCQ2I33VKRLIQIYKFPIMPYJF7A2KHNOC5IMN4ETZFWQGI376LMP3:5060;transport=udp;lr;uri-econt=BJOHZWFQE>
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:ADW267E7QDYVJULDIFALPWCPJCVK75WYZLLUAC5O2A2BV226R6PVGIYD4RD22MY:5060;uri-econt=HCRAAPX65ESRY3USRTVGMCBBV7O2LVZB433NF2XKTWGWOK7OK2OAROWCL2Q4IZBXL6BOTDV47DXQ64YLFHGEFGE4AJ4MB5PUN6AXPXIDOUGUA7VQTLUZDYIEO5RKD2CJ7RY2DM;lr>
Route: <sip:ADAOKMOFVTHS2X43M4NFJRRDMAJJBCH74MP5HNFPLWSDVHE6FTVV6DVTGNNQMOR:5060;transport=udp;lr;uri-econt=X3QFGCCJY>
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:   865

v=0
o=- 656189483 656189484 IN IP4 192.168.128.7
s=Asterisk
c=IN IP4 192.168.128.7
t=0 0
m=audio 19274 RTP/AVP 0 101
a=ice-ufrag:6c23a59d18283a176df8399d31a35ed8
a=ice-pwd:2f83db0360693c8f346d670210785d81
a=candidate:Ha6e76162 1 UDP 2130706431 fe80::20c:29ff:fe43:c08d 19274 typ host
a=candidate:Hc0a88007 1 UDP 2130706431 192.168.128.7 19274 typ host
a=candidate:S45829cd3 1 UDP 1694498815 <my public IP> 19274 typ srflx raddr 192.168.128.7 rport 19274
a=candidate:Ha6e76162 2 UDP 2130706430 fe80::20c:29ff:fe43:c08d 19275 typ host
a=candidate:Hc0a88007 2 UDP 2130706430 192.168.128.7 19275 typ host
a=candidate:S45829cd3 2 UDP 1694498814 <my public IP> 19275 typ srflx raddr 192.168.128.7 rport 19275
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
m=video 0 RTP/AVP 32

[Sep  4 12:32:28] VERBOSE[30730] 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=z9hG4bK35c8e39046cf7
From: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;tag=429787~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693932
To: <sip:<my 10 digit cellphone>@mydomain.com>;tag=20b77b06-a4c6-4983-8851-a8db829e1847
Date: Tue, 04 Sep 2018 17:32:22 GMT
Call-ID: 79b15c80-b8e1c1a6-355a7-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


[Sep  4 12:32:28] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP request (2743 bytes) from UDP:192.168.128.12:5060 --->
INVITE sip:192.168.128.7:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.12:5060;branch=z9hG4bK35c8f365345d0
From: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;tag=429787~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693932
To: <sip:<my 10 digit cellphone>@mydomain.com>;tag=20b77b06-a4c6-4983-8851-a8db829e1847
Date: Tue, 04 Sep 2018 17:32:28 GMT
Call-ID: 79b15c80-b8e1c1a6-355a7-c80a8c0@192.168.128.12
Supported: 100rel,timer,resource-priority,replaces
User-Agent: Cisco-CP-DX650/10.2.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Session-ID: 107dd6d000105000a0005017ff96e069;remote=d31b200c4e2a70c8e1fae054ab429787
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP;x-cisco-qos-tcl=true
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires:  1800;refresher=uac
Min-SE:  1800
P-Asserted-Identity: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>
Remote-Party-ID: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;party=calling;screen=yes;privacy=off
Contact: <sip:dd2b9c1a-cbb4-48eb-9c01-2920967e52ac@192.168.128.12:5060>;video;audio;+u.sip!devicename.ccm.cisco.com="SEP5017FF96E069";bfcp
Content-Type: application/sdp
Content-Length: 1431

v=0
o=CiscoSystemsCCM-SIP 429787 2 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 19646 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=trafficclass:conversational.audio.avconf.aq:admitted
m=video 19414 RTP/AVP 100
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=content:main
a=trafficclass:conversational.video.avconf.aq:admitted
m=video 0 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=inactive
a=rtcp-fb:* nack pli
a=rtcp-fb:* ccm fir
a=rtcp-fb:* ccm tmmbr
a=trafficclass:conversational.video.avconf.aq:admitted
m=application 0 UDP/BFCP *
c=IN IP4 0.0.0.0

[Sep  4 12:32:28] VERBOSE[30731] res_rtp_asterisk.c: 0x7f3a9c475b10 -- Strict RTP learning after remote address set to: 192.168.128.134:19646
[Sep  4 12:32:28] VERBOSE[30731] res_rtp_asterisk.c: 0x7f3a9c47a6d0 -- Strict RTP learning after remote address set to: 192.168.128.134:19414
[Sep  4 12:32:28] VERBOSE[30731] res_pjsip_logger.c: <--- Transmitting SIP response (1247 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=z9hG4bK35c8f365345d0
Call-ID: 79b15c80-b8e1c1a6-355a7-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;tag=429787~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693932
To: <sip:<my 10 digit cellphone>@mydomain.com>;tag=20b77b06-a4c6-4983-8851-a8db829e1847
CSeq: 102 INVITE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:192.168.128.7:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX GIT-master-b300c563e8
Content-Type: application/sdp
Content-Length:   513

v=0
o=- 429787 4 IN IP4 192.168.128.7
s=Asterisk
c=IN IP4 192.168.128.7
t=0 0
m=audio 19100 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 19966 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
b=TIAS:384000
m=application 0 UDP/BFCP *
c=IN IP4 192.168.128.7

[Sep  4 12:32:28] VERBOSE[30730] 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=z9hG4bK35c90246413a3
From: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;tag=429787~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693932
To: <sip:<my 10 digit cellphone>@mydomain.com>;tag=20b77b06-a4c6-4983-8851-a8db829e1847
Date: Tue, 04 Sep 2018 17:32:28 GMT
Call-ID: 79b15c80-b8e1c1a6-355a7-c80a8c0@192.168.128.12
User-Agent: Cisco-CP-DX650/10.2.5
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0


[Sep  4 12:32:28] VERBOSE[30951][C-00000002] res_rtp_asterisk.c: 0x7f3a9c475b10 -- Strict RTP switching to RTP target address 192.168.128.134:19646 as source
[Sep  4 12:32:28] VERBOSE[30966][C-00000002] res_rtp_asterisk.c: 0x7f3a9c45ee70 -- Strict RTP switching to RTP target address 74.125.39.60:19305 as source
[Sep  4 12:32:28] VERBOSE[30966][C-00000002] res_rtp_asterisk.c: 0x7f3a9c45ee70 -- Strict RTP learning complete - Locking on source address 74.125.39.60:19305
[Sep  4 12:32:28] VERBOSE[30951][C-00000002] res_rtp_asterisk.c: 0x7f3a9c47a6d0 -- Strict RTP switching to RTP target address 192.168.128.134:19414 as source
[Sep  4 12:32:33] VERBOSE[30951][C-00000002] res_rtp_asterisk.c: 0x7f3a9c475b10 -- Strict RTP learning complete - Locking on source address 192.168.128.134:19646
[Sep  4 12:32:33] VERBOSE[30951][C-00000002] res_rtp_asterisk.c: 0x7f3a9c47a6d0 -- Strict RTP learning complete - Locking on source address 192.168.128.134:19414
[Sep  4 12:32:59] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP request (398 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=z9hG4bK35c93585b43cc
From: <sip:192.168.128.12>;tag=910318971
To: <sip:mydomain.com>
Date: Tue, 04 Sep 2018 17:32:59 GMT
Call-ID: 8fbf1d00-b8e1c1cb-355a9-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


[Sep  4 12:32:59] VERBOSE[30731] res_pjsip_logger.c: <--- Transmitting SIP response (843 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=z9hG4bK35c93585b43cc
Call-ID: 8fbf1d00-b8e1c1cb-355a9-c80a8c0@192.168.128.12
From: <sip:192.168.128.12>;tag=910318971
To: <sip:mydomain.com>;tag=z9hG4bK35c93585b43cc
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


[Sep  4 12:33:00] VERBOSE[30966][C-00000002] bridge_channel.c: Channel PJSIP/<my 10 digit CALLING #>-00000005 left 'simple_bridge' basic-bridge <86303b96-ce45-4278-8146-b6c3935ebada>
[Sep  4 12:33:00] VERBOSE[30730] res_pjsip_logger.c: <--- Transmitting SIP request (716 bytes) to TLS:64.9.242.108:5061 --->
BYE sip:<my e164 CALLING #>@AAZZHPMXVTLHDSWAVIJZAGIHIWNBLLYZLSRXDZK4FSLA7QTXOPWDZLP72GEUDQI:5060;transport=udp;uri-econt=4DRGJR3JLSNO2Z2QEIH5345R6PDSQ SIP/2.0
Via: SIP/2.0/TLS 192.168.128.7:5061;rport;branch=z9hG4bKPj976e8ac4-d7f0-4dc2-b0b0-aac7d894cca0;alias
From: <sip:<my e164 CALLING #>@192.168.128.7>;tag=df13a723-ffba-4539-bbf1-1fb58f8c0f0d
To: <sip:<my 10 digit cellphone>@obihai.sip.google.com>;tag=1722332733
Call-ID: 38759f4b-023f-41f4-9201-13d65c681bf1
CSeq: 30483 BYE
Route: <sip:64.9.242.108:5061;transport=tls;lr>
Route: <sip:ADAOKMOFOOFRCQ2I33VKRLIQIYKFPIMPYJF7A2KHNOC5IMN4ETZFWQGI376LMP3:5060;transport=udp;lr;uri-econt=BJOHZWFQE>
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[Sep  4 12:33:00] VERBOSE[30951][C-00000002] bridge_channel.c: Channel PJSIP/cucm-00000003 left 'simple_bridge' basic-bridge <86303b96-ce45-4278-8146-b6c3935ebada>
[Sep  4 12:33:00] VERBOSE[30951][C-00000002] pbx.c: Spawn extension (dialprovider, s-CHANUNAVAIL, 1) exited non-zero on 'PJSIP/cucm-00000003'
[Sep  4 12:33:00] VERBOSE[30731] res_pjsip_logger.c: <--- Transmitting SIP request (551 bytes) to UDP:192.168.128.12:5060 --->
BYE sip:dd2b9c1a-cbb4-48eb-9c01-2920967e52ac@192.168.128.12:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.128.7:5060;rport;branch=z9hG4bKPj04dce981-2042-430c-a9ff-9652032b6e2d
From: <sip:<my 10 digit cellphone>@mydomain.com>;tag=20b77b06-a4c6-4983-8851-a8db829e1847
To: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;tag=429787~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693932
Call-ID: 79b15c80-b8e1c1a6-355a7-c80a8c0@192.168.128.12
CSeq: 11766 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[Sep  4 12:33:00] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP response (470 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=z9hG4bKPj04dce981-2042-430c-a9ff-9652032b6e2d
From: <sip:<my 10 digit cellphone>@mydomain.com>;tag=20b77b06-a4c6-4983-8851-a8db829e1847
To: "My Name" <sip:<my 10 digit CALLING #>@mydomain.com>;tag=429787~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693932
Date: Tue, 04 Sep 2018 17:33:00 GMT
Call-ID: 79b15c80-b8e1c1a6-355a7-c80a8c0@192.168.128.12
Server: Cisco-CP-DX650/10.2.5
CSeq: 11766 BYE
Content-Length: 0


[Sep  4 12:33:00] VERBOSE[30730] res_pjsip_logger.c: <--- Received SIP response (597 bytes) from TLS:64.9.242.108:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.128.7:5061;rport=37425;branch=z9hG4bKPj976e8ac4-d7f0-4dc2-b0b0-aac7d894cca0;received=<my public IP>;alias
Record-Route: <sip:ADAOKMOFOOFRCQ2I33VKRLIQIYKFPIMPYJF7A2KHNOC5IMN4ETZFWQGI376LMP3:5060;lr;transport=udp;uri-econt=BJOHZWFQE>
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
To: <sip:<my 10 digit cellphone>@obihai.sip.google.com>;tag=1722332733
From: <sip:<my e164 CALLING #>@192.168.128.7>;tag=df13a723-ffba-4539-bbf1-1fb58f8c0f0d
Call-ID: 38759f4b-023f-41f4-9201-13d65c681bf1
CSeq: 30483 BYE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Length: 0


[Sep  4 12:33:19] VERBOSE[30731] res_pjsip_registrar.c: Added contact 'sip:<my daughters phone>@172.56.21.79:34633;ob' to AOR '<my daughters phone>' with expiration of 900 seconds
[Sep  4 12:33:19] VERBOSE[30731] res_pjsip_registrar.c: Removed contact 'sip:<my daughters phone>@172.56.21.79:38973;ob' from AOR '<my daughters phone>' due to request
[Sep  4 12:33:19] VERBOSE[30955] res_pjsip/pjsip_options.c: Contact <my daughters phone>/sip:<my daughters phone>@172.56.21.79:38973;ob has been deleted
[Sep  4 12:33:19] VERBOSE[30731] res_pjsip_registrar.c: Attempted to remove non-existent contact 'sip:<my daughters phone>@172.50.139.231:39568;ob' from AOR '<my daughters phone>' by request
[Sep  4 12:33:19] VERBOSE[30731] res_pjsip_registrar.c: Attempted to remove non-existent contact 'sip:<my daughters phone>@172.56.21.79:38973;ob' from AOR '<my daughters phone>' by request
[Sep  4 12:33:29] WARNING[30731] res_pjsip_registrar.c: Endpoint 'anonymous' has no configured AORs