[2018-09-07 18:46:56] 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 18:46:56] VERBOSE[21283] logger.c: Asterisk Queue Logger restarted
[2018-09-07 18:47:41] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP request (647 bytes) from UDP:141.101.157.105:53112 --->
INVITE sip:0000000011972592277524@<my public IP> SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:53112;branch=z9hG4bK537365819
Max-Forwards: 70
From: <sip:000000001169130156211@<my public IP>>;tag=2112017032
To: <sip:0000000011972592277524@<my public IP>>
Call-ID: 1009269589-317066280-1001414125
CSeq: 1 INVITE
Contact: <sip:000000001169130156211@212.129.10.158:53112>
User-Agent: pplsip
Content-Type: application/sdp
Content-Length: 210

v=0
o=000000001169130156211 16264 18299 IN IP4 0.0.0.0
s=pplsip
c=IN IP4 0.0.0.0
t=0 0
m=audio 25282 RTP/AVP 100 6 0 8 3 18 5 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

[2018-09-07 18:47:41] ERROR[23094] pjproject: 	             sip_inv.c .Error parsing/validating SDP body: Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)
[2018-09-07 18:47:41] VERBOSE[23094] res_pjsip_logger.c: <--- Transmitting SIP response (470 bytes) to UDP:141.101.157.105:53112 --->
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 0.0.0.0:53112;rport=53112;received=141.101.157.105;branch=z9hG4bK537365819
Call-ID: 1009269589-317066280-1001414125
From: <sip:000000001169130156211@<my public IP>>;tag=2112017032
To: <sip:0000000011972592277524@<my public IP>>;tag=z9hG4bK537365819
CSeq: 1 INVITE
Warning: 399 SIP "Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)"
Server: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:47:44] 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=z9hG4bK380ef4123cc21
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445146~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693968
To: <sip:<my 10d cell>@mydomain.com>
Date: Fri, 07 Sep 2018 23:47:44 GMT
Call-ID: 69171580-b9310e20-379de-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: 1309bbdb00105000a0005017ff96e069;remote=00000000000000000000000000000000
Cisco-Guid: 1763120512-0000065536-0000000432-0209758400
P-Charging-Vector: icid-value="6917158000010000000001AF0C80A8C0";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 445146 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 19882 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 19210 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 19860 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 19412 UDP/BFCP *
a=floorctrl:s-only c-only
a=floorid:3 mstrm:12
a=confid:1
a=userid:1

[2018-09-07 18:47:44] VERBOSE[23094] pbx_variables.c: Setting global variable 'SIPDOMAIN' to 'mydomain.com'
[2018-09-07 18:47:44] VERBOSE[23094] 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=z9hG4bK380ef4123cc21
Call-ID: 69171580-b9310e20-379de-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445146~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693968
To: <sip:<my 10d cell>@mydomain.com>
CSeq: 101 INVITE
Server: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] pbx.c: Executing [<my 10d cell>@home:1] GotoIf("PJSIP/cucm-00000003", "1?numeric") in new stack
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] pbx_builtins.c: Goto (home,<my 10d cell>,4)
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] pbx.c: Executing [<my 10d cell>@home:4] Gosub("PJSIP/cucm-00000003", "dialprovider,s,1(<my 10d cell>)") in new stack
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] pbx.c: Executing [s@dialprovider:1] NoOp("PJSIP/cucm-00000003", " printing full callerid -- "My Name" <<my 10d gvoice>>") in new stack
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] pbx.c: Executing [s@dialprovider:2] NoOp("PJSIP/cucm-00000003", " printing the sip domain -- mydomain.com") in new stack
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] pbx.c: Executing [s@dialprovider:3] Set("PJSIP/cucm-00000003", "CALLERID(all)=<<my e164 gvoice>>") in new stack
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] pbx.c: Executing [s@dialprovider:4] NoOp("PJSIP/cucm-00000003", " printing the extension -- <my 10d cell>") in new stack
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] pbx.c: Executing [s@dialprovider:5] Dial("PJSIP/cucm-00000003", "PJSIP/<my e164 cell>@sipbroker-out") in new stack
[2018-09-07 18:47:44] DEBUG[23094] res_pjsip_outbound_registration.c: Outgoing request not associated with a registration. No mangling necessary.
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] app_dial.c: Called PJSIP/<my e164 cell>@sipbroker-out
[2018-09-07 18:47:44] DEBUG[23094] res_pjsip_outbound_registration.c: Outgoing request not associated with a registration. No mangling necessary.
[2018-09-07 18:47:44] VERBOSE[23094] res_pjsip_logger.c: <--- Transmitting SIP request (1195 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=z9hG4bKPjae5eb2b7-7e9a-4e46-a92f-745ef1117830
From: <sip:driz@mydomain.com>;tag=6be0e08c-d06d-4884-b373-2c779d9848c9
To: <sip:<my e164 cell>@sipbroker.com>
Contact: <sip:driz@<my public IP>:5060>
Call-ID: 4d739cc5-e287-48f8-a0b8-7d6ae7591a3b
CSeq: 10426 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 gvoice>@mydomain.com>;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Type: application/sdp
Content-Length:   428

v=0
o=- 1167749074 1167749074 IN IP4 <my public IP>
s=Asterisk
c=IN IP4 <my public IP>
t=0 0
m=audio 19334 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 19758 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

[2018-09-07 18:47:44] 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=z9hG4bKPjae5eb2b7-7e9a-4e46-a92f-745ef1117830
From: <sip:driz@mydomain.com>;tag=6be0e08c-d06d-4884-b373-2c779d9848c9
To: <sip:<my e164 cell>@sipbroker.com>
Call-ID: 4d739cc5-e287-48f8-a0b8-7d6ae7591a3b
CSeq: 10426 INVITE
Server: OpenSer (1.1.0-notls (x86_64/linux))
Content-Length: 0
Warning: 392 204.11.194.25:5060 "Noisy feedback tells:  pid=3471 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 18:47:44] 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=z9hG4bKPjae5eb2b7-7e9a-4e46-a92f-745ef1117830
From: <sip:driz@mydomain.com>;tag=6be0e08c-d06d-4884-b373-2c779d9848c9
To: <sip:<my e164 cell>@sipbroker.com>;tag=2b8506bb96abbbb8b95a41b9af69a614.8b80
Call-ID: 4d739cc5-e287-48f8-a0b8-7d6ae7591a3b
CSeq: 10426 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=3471 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 18:47:44] VERBOSE[23094] 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=z9hG4bKPjae5eb2b7-7e9a-4e46-a92f-745ef1117830
From: <sip:driz@mydomain.com>;tag=6be0e08c-d06d-4884-b373-2c779d9848c9
To: <sip:<my e164 cell>@sipbroker.com>;tag=2b8506bb96abbbb8b95a41b9af69a614.8b80
Call-ID: 4d739cc5-e287-48f8-a0b8-7d6ae7591a3b
CSeq: 10426 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] app_dial.c: Now forwarding PJSIP/cucm-00000003 to 'Local/<my 11d cell>@unauthenticated' (thanks to PJSIP/sipbroker-out-00000004)
[2018-09-07 18:47:44] NOTICE[23200][C-00000004] app_dial.c: Not accepting call completion offers from call-forward recipient Local/<my 11d cell>@unauthenticated-00000000;1
[2018-09-07 18:47:44] VERBOSE[23094] 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=z9hG4bK380ef4123cc21
Call-ID: 69171580-b9310e20-379de-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445146~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693968
To: <sip:<my 10d cell>@mydomain.com>;tag=e58da897-b459-46f3-b258-788f9ace1aa0
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 18:47:44] NOTICE[23200][C-00000004] core_local.c: No such extension/context <my 11d cell>@unauthenticated while calling Local channel
[2018-09-07 18:47:44] NOTICE[23200][C-00000004] app_dial.c: Forwarding failed to dial 'Local/<my 11d cell>@unauthenticated'
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] pbx.c: Executing [s@dialprovider:6] NoOp("PJSIP/cucm-00000003", " Dial Status: CHANUNAVAIL") in new stack
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] pbx.c: Executing [s@dialprovider:7] Goto("PJSIP/cucm-00000003", "s-CHANUNAVAIL,1") in new stack
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] pbx_builtins.c: Goto (dialprovider,s-CHANUNAVAIL,1)
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] pbx.c: Executing [s-CHANUNAVAIL@dialprovider:1] Dial("PJSIP/cucm-00000003", "PJSIP/<my 10d cell>@<my 10d gvoice>,,r") in new stack
[2018-09-07 18:47:44] DEBUG[23094] res_pjsip_outbound_registration.c: Setting transport to 0x7f821c1141e8
[2018-09-07 18:47:44] DEBUG[23094] res_pjsip.c: Overriding endpoint transport to use 0x7f821c1141e8
[2018-09-07 18:47:44] VERBOSE[23200][C-00000004] app_dial.c: Called PJSIP/<my 10d cell>@<my 10d gvoice>
[2018-09-07 18:47:44] VERBOSE[23203] 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=z9hG4bK380ef4123cc21
Call-ID: 69171580-b9310e20-379de-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445146~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693968
To: <sip:<my 10d cell>@mydomain.com>;tag=e58da897-b459-46f3-b258-788f9ace1aa0
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 18:47:44] DEBUG[23094] res_pjsip_outbound_registration.c: Found matching outbound registration state
[2018-09-07 18:47:44] DEBUG[23094] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:ADW267E7WKCZWWOSTAMVZM3OX5DURRTYGHZKY3CDLRQBSI5EGDYSO4QTWKU2HP5:5060;uri-econt=6RNT45K7F4X56ZVTLUCLQW5FJ54H3CE5UVSZ7CGBPXDEKWMJRQPREFW6YR25EXS3324EFNQZI5M5CPVKJGFRMP7U5ION76ZHT3DNVE4MMYQLLWVQ2N4A7OIAAYDICQPNJU4QRK;lr>
[2018-09-07 18:47:44] DEBUG[23094] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:ADAOKMOFMOSQQUSE7DVJ4EDSS3US3XEVQCPZZDEBC4FIHGXOSMWD6TPLHLJVRX4:5060;transport=udp;lr;uri-econt=YQVLFKPPJ>
[2018-09-07 18:47:44] VERBOSE[23094] res_pjsip_logger.c: <--- Transmitting SIP request (2040 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=z9hG4bKPjf86a678a-ba90-4a8f-925f-096309a3b415;alias
From: <sip:<my e164 gvoice>@192.168.128.7>;tag=3882d807-0338-4223-9835-c3310f054eef
To: <sip:<my 10d cell>@obihai.sip.google.com>
Contact: <sip:asterisk@192.168.128.7:5061;transport=TLS>
Call-ID: b455cbb2-9d00-4c07-9212-ec2ddd0e32ce
CSeq: 5793 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=- 2028413573 2028413573 IN IP4 192.168.128.7
s=Asterisk
c=IN IP4 192.168.128.7
t=0 0
m=audio 19796 RTP/AVP 0 101
a=ice-ufrag:3c7bf915333bf881290752a14921fcf0
a=ice-pwd:55f2cb5a4eca5105127990bb29296d59
a=candidate:Ha6e76162 1 UDP 2130706431 fe80::20c:29ff:fe43:c08d 19796 typ host
a=candidate:Hc0a88007 1 UDP 2130706431 192.168.128.7 19796 typ host
a=candidate:S45829cd3 1 UDP 1694498815 <my public IP> 19796 typ srflx raddr 192.168.128.7 rport 19796
a=candidate:Ha6e76162 2 UDP 2130706430 fe80::20c:29ff:fe43:c08d 19797 typ host
a=candidate:Hc0a88007 2 UDP 2130706430 192.168.128.7 19797 typ host
a=candidate:S45829cd3 2 UDP 1694498814 <my public IP> 19797 typ srflx raddr 192.168.128.7 rport 19797
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 18:47:44] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP response (547 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=z9hG4bKPjf86a678a-ba90-4a8f-925f-096309a3b415;received=<my public IP>;alias
Record-Route: <sip:ADAOKMOFAYVR22PFLCCITF6SG4XI3D75NWOTYTHZO6DPP7Y2CDMIVF4BW2SPKE2:5060;lr;transport=udp;uri-econt=QFC2HLX2Q>
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
To: <sip:<my 10d cell>@obihai.sip.google.com>
From: <sip:<my e164 gvoice>@192.168.128.7>;tag=3882d807-0338-4223-9835-c3310f054eef
Call-ID: b455cbb2-9d00-4c07-9212-ec2ddd0e32ce
CSeq: 5793 INVITE
Content-Length: 0


[2018-09-07 18:47:45] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP response (1363 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=z9hG4bKPjf86a678a-ba90-4a8f-925f-096309a3b415;received=<my public IP>;alias
Record-Route: <sip:ADAOKMOFAYVR22PFLCCITF6SG4XI3D75NWOTYTHZO6DPP7Y2CDMIVF4BW2SPKE2:5060;lr;transport=udp;uri-econt=QFC2HLX2Q>
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
Contact: <sip:<my e164 gvoice>@AAZZHPMX45LTPUT7NG5WJ6OFRYEUDC7ERX77YV5R6XVTGARROA53RUJSK6C2745:5060;transport=udp;uri-econt=FEF4D6DA4DD7GGIUDYUK4I52IPO3Q>
To: <sip:<my 10d cell>@obihai.sip.google.com>;tag=842616855
From: <sip:<my e164 gvoice>@192.168.128.7>;tag=3882d807-0338-4223-9835-c3310f054eef
Call-ID: b455cbb2-9d00-4c07-9212-ec2ddd0e32ce
CSeq: 5793 INVITE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Type: application/sdp
Content-Length: 566

v=0
o=- 1106899807 1536364065336 IN IP4 74.125.39.21
s=SIP Call
c=IN IP4 74.125.39.21
t=0 0
a=ice-lite
a=ice-pwd:Y1o6k1y2OPxXxu2Syrr7qJ0K
a=ice-ufrag:7lmKHeVdFZawRPvD
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 74.125.39.21 19305 typ host
a=candidate:2 1 UDP 2 2001:4860:4864:2::21 19305 typ host
a=sendrecv

[2018-09-07 18:47:45] VERBOSE[23094] res_rtp_asterisk.c: 0x7f8228019a90 -- Strict RTP learning after remote address set to: 74.125.39.21:19305
[2018-09-07 18:47:45] ERROR[23094] pjproject: 	   icess0x7f8228042a08 ......Error sending STUN request: Network is unreachable
[2018-09-07 18:47:45] VERBOSE[23200][C-00000004] app_dial.c: PJSIP/<my 10d gvoice>-00000005 is making progress passing it to PJSIP/cucm-00000003
[2018-09-07 18:47:45] VERBOSE[23200][C-00000004] app_dial.c: PJSIP/<my 10d gvoice>-00000005 is making progress passing it to PJSIP/cucm-00000003
[2018-09-07 18:47:45] VERBOSE[21258] res_rtp_asterisk.c: 0x7f8228019a90 -- Strict RTP learning after ICE completion
[2018-09-07 18:47:46] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP response (755 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=z9hG4bKPjf86a678a-ba90-4a8f-925f-096309a3b415;received=<my public IP>;alias
Record-Route: <sip:ADAOKMOFAYVR22PFLCCITF6SG4XI3D75NWOTYTHZO6DPP7Y2CDMIVF4BW2SPKE2:5060;lr;transport=udp;uri-econt=QFC2HLX2Q>
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
Contact: <sip:<my e164 gvoice>@AAZZHPMX45LTPUT7NG5WJ6OFRYEUDC7ERX77YV5R6XVTGARROA53RUJSK6C2745:5060;transport=udp;uri-econt=FEF4D6DA4DD7GGIUDYUK4I52IPO3Q>
To: <sip:<my 10d cell>@obihai.sip.google.com>;tag=842616855
From: <sip:<my e164 gvoice>@192.168.128.7>;tag=3882d807-0338-4223-9835-c3310f054eef
Call-ID: b455cbb2-9d00-4c07-9212-ec2ddd0e32ce
CSeq: 5793 INVITE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Length: 0


[2018-09-07 18:47:46] VERBOSE[23200][C-00000004] app_dial.c: PJSIP/<my 10d gvoice>-00000005 is ringing
[2018-09-07 18:47:46] VERBOSE[23200][C-00000004] app_dial.c: PJSIP/<my 10d gvoice>-00000005 is ringing
[2018-09-07 18:47:46] VERBOSE[23094] 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=z9hG4bK380ef4123cc21
Call-ID: 69171580-b9310e20-379de-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445146~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693968
To: <sip:<my 10d cell>@mydomain.com>;tag=e58da897-b459-46f3-b258-788f9ace1aa0
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 18:47:50] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP response (1349 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=z9hG4bKPjf86a678a-ba90-4a8f-925f-096309a3b415;received=<my public IP>;alias
Record-Route: <sip:ADAOKMOFAYVR22PFLCCITF6SG4XI3D75NWOTYTHZO6DPP7Y2CDMIVF4BW2SPKE2:5060;lr;transport=udp;uri-econt=QFC2HLX2Q>
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
Contact: <sip:<my e164 gvoice>@AAZZHPMX45LTPUT7NG5WJ6OFRYEUDC7ERX77YV5R6XVTGARROA53RUJSK6C2745:5060;transport=udp;uri-econt=FEF4D6DA4DD7GGIUDYUK4I52IPO3Q>
To: <sip:<my 10d cell>@obihai.sip.google.com>;tag=842616855
From: <sip:<my e164 gvoice>@192.168.128.7>;tag=3882d807-0338-4223-9835-c3310f054eef
Call-ID: b455cbb2-9d00-4c07-9212-ec2ddd0e32ce
CSeq: 5793 INVITE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Type: application/sdp
Content-Length: 566

v=0
o=- 1106899807 1536364065336 IN IP4 74.125.39.21
s=SIP Call
c=IN IP4 74.125.39.21
t=0 0
a=ice-lite
a=ice-pwd:Y1o6k1y2OPxXxu2Syrr7qJ0K
a=ice-ufrag:7lmKHeVdFZawRPvD
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 74.125.39.21 19305 typ host
a=candidate:2 1 UDP 2 2001:4860:4864:2::21 19305 typ host
a=sendrecv

[2018-09-07 18:47:50] VERBOSE[23094] res_pjsip_logger.c: <--- Transmitting SIP request (714 bytes) to TLS:64.9.242.108:5061 --->
ACK sip:<my e164 gvoice>@AAZZHPMX45LTPUT7NG5WJ6OFRYEUDC7ERX77YV5R6XVTGARROA53RUJSK6C2745:5060;transport=udp;uri-econt=FEF4D6DA4DD7GGIUDYUK4I52IPO3Q SIP/2.0
Via: SIP/2.0/TLS 192.168.128.7:5061;rport;branch=z9hG4bKPj48aee165-52f1-449d-b268-04b5af4e10fb;alias
From: <sip:<my e164 gvoice>@192.168.128.7>;tag=3882d807-0338-4223-9835-c3310f054eef
To: <sip:<my 10d cell>@obihai.sip.google.com>;tag=842616855
Call-ID: b455cbb2-9d00-4c07-9212-ec2ddd0e32ce
CSeq: 5793 ACK
Route: <sip:64.9.242.108:5061;transport=tls;lr>
Route: <sip:ADAOKMOFAYVR22PFLCCITF6SG4XI3D75NWOTYTHZO6DPP7Y2CDMIVF4BW2SPKE2:5060;transport=udp;lr;uri-econt=QFC2HLX2Q>
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:47:50] VERBOSE[23200][C-00000004] app_dial.c: PJSIP/<my 10d gvoice>-00000005 answered PJSIP/cucm-00000003
[2018-09-07 18:47:50] VERBOSE[23094] res_rtp_asterisk.c: 0x7f8228030860 -- Strict RTP learning after remote address set to: 192.168.128.134:19882
[2018-09-07 18:47:50] VERBOSE[23094] res_rtp_asterisk.c: 0x7f822814bcb0 -- Strict RTP learning after remote address set to: 192.168.128.134:19210
[2018-09-07 18:47:50] VERBOSE[23094] 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=z9hG4bK380ef4123cc21
Call-ID: 69171580-b9310e20-379de-c80a8c0@192.168.128.12
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445146~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693968
To: <sip:<my 10d cell>@mydomain.com>;tag=e58da897-b459-46f3-b258-788f9ace1aa0
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=- 445146 3 IN IP4 192.168.128.7
s=Asterisk
c=IN IP4 192.168.128.7
t=0 0
m=audio 19324 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 19314 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 18:47:50] VERBOSE[23214][C-00000004] bridge_channel.c: Channel PJSIP/<my 10d gvoice>-00000005 joined 'simple_bridge' basic-bridge <ad2e1573-1420-440a-a2d8-9cb812c98c1d>
[2018-09-07 18:47:50] VERBOSE[23200][C-00000004] bridge_channel.c: Channel PJSIP/cucm-00000003 joined 'simple_bridge' basic-bridge <ad2e1573-1420-440a-a2d8-9cb812c98c1d>
[2018-09-07 18:47:50] 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=z9hG4bK380f173af932a
From: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445146~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693968
To: <sip:<my 10d cell>@mydomain.com>;tag=e58da897-b459-46f3-b258-788f9ace1aa0
Date: Fri, 07 Sep 2018 23:47:44 GMT
Call-ID: 69171580-b9310e20-379de-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 18:47:50] VERBOSE[23200][C-00000004] res_rtp_asterisk.c: 0x7f8228030860 -- Strict RTP switching to RTP target address 192.168.128.134:19882 as source
[2018-09-07 18:47:50] VERBOSE[23214][C-00000004] res_rtp_asterisk.c: 0x7f8228019a90 -- Strict RTP switching to RTP target address 74.125.39.21:19305 as source
[2018-09-07 18:47:50] VERBOSE[23214][C-00000004] res_rtp_asterisk.c: 0x7f8228019a90 -- Strict RTP learning complete - Locking on source address 74.125.39.21:19305
[2018-09-07 18:47:51] VERBOSE[23200][C-00000004] res_rtp_asterisk.c: 0x7f822814bcb0 -- Strict RTP switching to RTP target address 192.168.128.134:19210 as source
[2018-09-07 18:47:51] VERBOSE[21246] 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=z9hG4bK380f21795a96e
From: <sip:192.168.128.12>;tag=933663006
To: <sip:mydomain.com>
Date: Fri, 07 Sep 2018 23:47:51 GMT
Call-ID: 6d433300-b9310e27-379df-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 18:47:51] VERBOSE[23094] 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=z9hG4bK380f21795a96e
Call-ID: 6d433300-b9310e27-379df-c80a8c0@192.168.128.12
From: <sip:192.168.128.12>;tag=933663006
To: <sip:mydomain.com>;tag=z9hG4bK380f21795a96e
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 18:47:55] VERBOSE[23200][C-00000004] res_rtp_asterisk.c: 0x7f822814bcb0 -- Strict RTP learning complete - Locking on source address 192.168.128.134:19210
[2018-09-07 18:47:55] VERBOSE[23200][C-00000004] res_rtp_asterisk.c: 0x7f8228030860 -- Strict RTP learning complete - Locking on source address 192.168.128.134:19882
[2018-09-07 18:48:14] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP request (669 bytes) from UDP:68.46.145.125:44507 --->
SUBSCRIBE sip:<my public IP>:5060 SIP/2.0
Via: SIP/2.0/UDP 68.46.145.125:44507;branch=z9hG4bK1783236069;rport
From: <sip:<my parents>@dznet-pbx.mydomain.com>;tag=107759677
To: <sip:<my parents>@dznet-pbx.mydomain.com>;tag=accdac79-58ef-48d6-8288-d90fdc218f9d
Call-ID: 848737224-44507-7@BA.A.A.CG
CSeq: 20525 SUBSCRIBE
Contact: <sip:<my parents>@68.46.145.125:44507>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3140 1.0.7.80
Expires: 900
Event: message-summary
Accept: application/simple-message-summary
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


[2018-09-07 18:48:14] VERBOSE[23094] res_pjsip_logger.c: <--- Transmitting SIP response (603 bytes) to UDP:68.46.145.125:44507 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.46.145.125:44507;rport=44507;received=68.46.145.125;branch=z9hG4bK1783236069
Call-ID: 848737224-44507-7@BA.A.A.CG
From: <sip:<my parents>@dznet-pbx.mydomain.com>;tag=107759677
To: <sip:<my parents>@dznet-pbx.mydomain.com>;tag=accdac79-58ef-48d6-8288-d90fdc218f9d
CSeq: 20525 SUBSCRIBE
Expires: 900
Contact: <sip:<my public IP>: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-Length:  0


[2018-09-07 18:48:14] VERBOSE[23094] res_pjsip_logger.c: <--- Transmitting SIP request (702 bytes) to UDP:68.46.145.125:44507 --->
NOTIFY sip:<my parents>@68.46.145.125:44507 SIP/2.0
Via: SIP/2.0/UDP <my public IP>:5060;rport;branch=z9hG4bKPjd74c1cd4-a248-422c-a97a-a557b96dc961
From: <sip:<my parents>@dznet-pbx.mydomain.com>;tag=accdac79-58ef-48d6-8288-d90fdc218f9d
To: <sip:<my parents>@dznet-pbx.mydomain.com>;tag=107759677
Contact: <sip:<my public IP>:5060>
Call-ID: 848737224-44507-7@BA.A.A.CG
CSeq: 20501 NOTIFY
Event: message-summary
Subscription-State: active;expires=900
Allow-Events: message-summary, presence, dialog, refer
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Type: application/simple-message-summary
Content-Length:    49

Messages-Waiting: yes
Voice-Message: 1/0 (0/0)

[2018-09-07 18:48:14] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP response (631 bytes) from UDP:68.46.145.125:44507 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <my public IP>:5060;rport=5060;branch=z9hG4bKPjd74c1cd4-a248-422c-a97a-a557b96dc961
From: <sip:<my parents>@dznet-pbx.mydomain.com>;tag=accdac79-58ef-48d6-8288-d90fdc218f9d
To: <sip:<my parents>@dznet-pbx.mydomain.com>;tag=107759677
Call-ID: 848737224-44507-7@BA.A.A.CG
CSeq: 20501 NOTIFY
Contact: <sip:<my parents>@68.46.145.125:44507>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3140 1.0.7.80
Warning: 399 10.0.0.26 "Detected NAT type is UDP Blocked"
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


[2018-09-07 18:48:25] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP request (616 bytes) from UDP:141.101.157.105:65526 --->
INVITE sip:8011972567088721@<my public IP> SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:65526;branch=z9hG4bK1762655200
Max-Forwards: 70
From: <sip:801169130156211@<my public IP>>;tag=751370776
To: <sip:8011972567088721@<my public IP>>
Call-ID: 416977163-1714714042-508182786
CSeq: 1 INVITE
Contact: <sip:801169130156211@212.129.10.158:65526>
User-Agent: pplsip
Content-Type: application/sdp
Content-Length: 204

v=0
o=801169130156211 16264 18299 IN IP4 0.0.0.0
s=pplsip
c=IN IP4 0.0.0.0
t=0 0
m=audio 25282 RTP/AVP 100 6 0 8 3 18 5 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

[2018-09-07 18:48:25] ERROR[23094] pjproject: 	             sip_inv.c .Error parsing/validating SDP body: Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)
[2018-09-07 18:48:25] VERBOSE[23094] res_pjsip_logger.c: <--- Transmitting SIP response (458 bytes) to UDP:141.101.157.105:65526 --->
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 0.0.0.0:65526;rport=65526;received=141.101.157.105;branch=z9hG4bK1762655200
Call-ID: 416977163-1714714042-508182786
From: <sip:801169130156211@<my public IP>>;tag=751370776
To: <sip:8011972567088721@<my public IP>>;tag=z9hG4bK1762655200
CSeq: 1 INVITE
Warning: 399 SIP "Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)"
Server: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:48:40] VERBOSE[21246] res_pjsip_logger.c: <--- Received SIP request (974 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---fc25ce3b6e0551031dab8a13f916247a;rport
Via: SIP/2.0/UDP ADAOKMOFOOEBAOUQJFYJSWESKDM5YQ7NKMN4TCVH632NMMMEVMPP3GBFB2XVA6L:5060;branch=z9hG4bK-524287-1---6fcbdd4f2a7eb6212005163409b1ca7e;econt=UNQVQ7BBZMU4O7M5WNY
Via: SIP/2.0/UDP AAZZHPMXCACNO66R63JHPXE7FXF54FLV7WFQXAFTNCJXOALWMWZKTWC7UIP7BYS:5060;branch=z9hG4bK611058854;econt=7I7CXUAGLFB3CZKPCDKKTAMMTT6BGMLFQLFWUXLO2H5T4H5N7J2I7CSNF
Max-Forwards: 68
Record-Route: <sip:64.9.242.108:5061;lr;transport=tls>
Record-Route: <sip:ADAOKMOFAYVR22PFLCCITF6SG4XI3D75NWOTYTHZO6DPP7Y2CDMIVF4BW2SPKE2:5060;lr;transport=udp;uri-econt=QFC2HLX2Q>
To: <sip:<my e164 gvoice>@192.168.128.7>;tag=3882d807-0338-4223-9835-c3310f054eef
From: <sip:<my 10d cell>@obihai.sip.google.com>;tag=842616855
Call-ID: b455cbb2-9d00-4c07-9212-ec2ddd0e32ce
CSeq: 264553 BYE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Length: 0


[2018-09-07 18:48:40] VERBOSE[23094] res_pjsip_logger.c: <--- Transmitting SIP response (944 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---fc25ce3b6e0551031dab8a13f916247a
Via: SIP/2.0/UDP ADAOKMOFOOEBAOUQJFYJSWESKDM5YQ7NKMN4TCVH632NMMMEVMPP3GBFB2XVA6L:5060;branch=z9hG4bK-524287-1---6fcbdd4f2a7eb6212005163409b1ca7e;econt=UNQVQ7BBZMU4O7M5WNY
Via: SIP/2.0/UDP AAZZHPMXCACNO66R63JHPXE7FXF54FLV7WFQXAFTNCJXOALWMWZKTWC7UIP7BYS:5060;branch=z9hG4bK611058854;econt=7I7CXUAGLFB3CZKPCDKKTAMMTT6BGMLFQLFWUXLO2H5T4H5N7J2I7CSNF
Record-Route: <sip:64.9.242.108:5061;transport=tls;lr>
Record-Route: <sip:ADAOKMOFAYVR22PFLCCITF6SG4XI3D75NWOTYTHZO6DPP7Y2CDMIVF4BW2SPKE2:5060;transport=udp;lr;uri-econt=QFC2HLX2Q>
Call-ID: b455cbb2-9d00-4c07-9212-ec2ddd0e32ce
From: <sip:<my 10d cell>@obihai.sip.google.com>;tag=842616855
To: <sip:<my e164 gvoice>@192.168.128.7>;tag=3882d807-0338-4223-9835-c3310f054eef
CSeq: 264553 BYE
Server: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:48:40] VERBOSE[23214][C-00000004] bridge_channel.c: Channel PJSIP/<my 10d gvoice>-00000005 left 'simple_bridge' basic-bridge <ad2e1573-1420-440a-a2d8-9cb812c98c1d>
[2018-09-07 18:48:40] VERBOSE[23200][C-00000004] bridge_channel.c: Channel PJSIP/cucm-00000003 left 'simple_bridge' basic-bridge <ad2e1573-1420-440a-a2d8-9cb812c98c1d>
[2018-09-07 18:48:40] VERBOSE[23200][C-00000004] pbx.c: Spawn extension (dialprovider, s-CHANUNAVAIL, 1) exited non-zero on 'PJSIP/cucm-00000003'
[2018-09-07 18:48:40] VERBOSE[23094] res_pjsip_logger.c: <--- Transmitting SIP request (525 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=z9hG4bKPj41eab8dc-a42d-482b-9d92-8ee7e8922592
From: <sip:<my 10d cell>@mydomain.com>;tag=e58da897-b459-46f3-b258-788f9ace1aa0
To: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445146~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693968
Call-ID: 69171580-b9310e20-379de-c80a8c0@192.168.128.12
CSeq: 25582 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-b300c563e8
Content-Length:  0


[2018-09-07 18:48:40] VERBOSE[21246] 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=z9hG4bKPj41eab8dc-a42d-482b-9d92-8ee7e8922592
From: <sip:<my 10d cell>@mydomain.com>;tag=e58da897-b459-46f3-b258-788f9ace1aa0
To: "My Name" <sip:<my 10d gvoice>@mydomain.com>;tag=445146~58340c4c-2bc2-48dc-96bc-4e4d906b83a8-26693968
Date: Fri, 07 Sep 2018 23:48:40 GMT
Call-ID: 69171580-b9310e20-379de-c80a8c0@192.168.128.12
Server: Cisco-CP-DX650/10.2.5
CSeq: 25582 BYE
Content-Length: 0