Error in terminating session timer: Unsupported transport (PJSIP_EUNSUPTRANSPORT), and closing channels immediately

We are using 2 browsers as user agents with JSSIP. Everything is working fine. However when a disconnect appears on one of the 2 user agents the channels of both stay open. After a minute or so asterisk is closing the channels responding with the error message.

Asterisk version
Asterisk 18.13.0

Question
How can we tell asterisk to immediately close the channels instead of letting it wait for a 1 minute( It has probably something to do with a grace time so an user agent can reconnect?)

The channels

janus*CLI> pjsip show channels

  Channel:  <ChannelId........................................>  <State.....>  <Time.....>
      Exten: <DialedExten.............>  CLCID: <ConnectedLineCID.......>
==========================================================================================

  Channel: PJSIP/14663CC9-7C1F-447F-AAF1-2BD2F451E6AD-00000063/D Up            00:00:22   
      Exten: 1                           CLCID: "" <1>

  Channel: PJSIP/D0F460EC-E02D-47EF-B39F-88BC8DE7DFB0-00000065/A Up            00:00:20   
      Exten:                             CLCID: "" <+xxxx>


Objects found: 2

The error

[Nov  4 17:16:44] WARNING[1394777]: pjproject: <?>: 	     dlg0x7f9ee400a0c0 Error in terminating session timer: Unsupported transport (PJSIP_EUNSUPTRANSPORT)

The log

<--- Received SIP request (2563 bytes) from WSS:xxx.xxx.xxx:57445 --->
INVITE sip:xxx.xxx.xxx@xxx.xxx.xxx:8443 SIP/2.0
Via: SIP/2.0/WSS revbqlg629ue.invalid;branch=z9hG4bK9916778
Max-Forwards: 69
To: <sip:xxx.xxx.xxx@xxx.xxx.xxx:8443>
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=v42h90lrg2
Call-ID: tfd43f2sbnv4n4i9s07o
CSeq: 8560 INVITE
ctbEndPointPin: 234731
ctbEndPointRef: 001676
ctbIsEndPoint: 1
ctbEndPointEndpoint: D0F460EC-E02D-47EF-B39F-88BC8DE7DFB0
Contact: <sip:8dl59l7f@revbqlg629ue.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.9.1
Content-Length: 1858

v=0
o=- 1251817391069597861 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS bb960539-69c8-40e7-8803-d28ffb10c046
m=audio 51683 UDP/TLS/RTP/SAVPF 111 63 103 9 0 8 105 13 110 113 126
c=IN IP4 10.8.0.3
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:4091210427 1 udp 2122260223 10.8.0.3 51683 typ host generation 0 network-id 1 network-cost 50
a=candidate:3176732235 1 tcp 1518280447 10.8.0.3 9 typ host tcptype active generation 0 network-id 1 network-cost 50
a=ice-ufrag:GnU2
a=ice-pwd:JcJvTY3uLqODpI+5jVDXUGr9
a=ice-options:trickle
a=fingerprint:sha-256 86:E8:F1:75:CF:B1:3C:79:13:F9:1E:E7:6F:C6:0A:61:52:CA:74:F0:C3:15:D6:06:3B:3D:EC:5F:BA:4D:B1:A3
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:bb960539-69c8-40e7-8803-d28ffb10c046 f55aeb97-2e9f-439d-ac20-4ebfe4214044
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:103 ISAC/16000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:3001809347 cname:KgZyvMyOk/thwV4S
a=ssrc:3001809347 msid:bb960539-69c8-40e7-8803-d28ffb10c046 f55aeb97-2e9f-439d-ac20-4ebfe4214044
a=ssrc:3001809347 mslabel:bb960539-69c8-40e7-8803-d28ffb10c046
a=ssrc:3001809347 label:f55aeb97-2e9f-439d-ac20-4ebfe4214044

<--- Transmitting SIP response (506 bytes) to WSS:xxx.xxx.xxx:57445 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS revbqlg629ue.invalid;rport=57445;received=xxx.xxx.xxx;branch=z9hG4bK9916778
Call-ID: tfd43f2sbnv4n4i9s07o
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=v42h90lrg2
To: <sip:xxx.xxx.xxx@xxx.xxx.xxx>;tag=z9hG4bK9916778
CSeq: 8560 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1667578170/634fbce1a69f577df530e65fa06d2e23",opaque="40a630c47ec861ec",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.13.0
Content-Length:  0


<--- Received SIP request (452 bytes) from WSS:xxx.xxx.xxx:57445 --->
ACK sip:xxx.xxx.xxx@xxx.xxx.xxx:8443 SIP/2.0
Via: SIP/2.0/WSS revbqlg629ue.invalid;branch=z9hG4bK9916778
Max-Forwards: 69
To: <sip:xxx.xxx.xxx@xxx.xxx.xxx>;tag=z9hG4bK9916778
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=v42h90lrg2
Call-ID: tfd43f2sbnv4n4i9s07o
CSeq: 8560 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.9.1
Content-Length: 0


<--- Received SIP request (2881 bytes) from WSS:xxx.xxx.xxx:57445 --->
INVITE sip:xxx.xxx.xxx@xxx.xxx.xxx:8443 SIP/2.0
Via: SIP/2.0/WSS revbqlg629ue.invalid;branch=z9hG4bK158201
Max-Forwards: 69
To: <sip:xxx.xxx.xxx@xxx.xxx.xxx:8443>
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=v42h90lrg2
Call-ID: tfd43f2sbnv4n4i9s07o
CSeq: 8561 INVITE
Authorization: Digest algorithm=MD5, username="C45841A4-A2F6-49D4-8CA2-9D0C554D9919", realm="asterisk", nonce="1667578170/634fbce1a69f577df530e65fa06d2e23", uri="sip:xxx.xxx.xxx@xxx.xxx.xxx:8443", response="5df5c9c7e1b314c0038251559f3b4751", opaque="40a630c47ec861ec", qop=auth, cnonce="r76u9qqu7nac", nc=00000001
ctbEndPointPin: 234731
ctbEndPointRef: 001676
ctbIsEndPoint: 1
ctbEndPointEndpoint: D0F460EC-E02D-47EF-B39F-88BC8DE7DFB0
Contact: <sip:8dl59l7f@revbqlg629ue.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.9.1
Content-Length: 1858

v=0
o=- 1251817391069597861 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS bb960539-69c8-40e7-8803-d28ffb10c046
m=audio 51683 UDP/TLS/RTP/SAVPF 111 63 103 9 0 8 105 13 110 113 126
c=IN IP4 10.8.0.3
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:4091210427 1 udp 2122260223 10.8.0.3 51683 typ host generation 0 network-id 1 network-cost 50
a=candidate:3176732235 1 tcp 1518280447 10.8.0.3 9 typ host tcptype active generation 0 network-id 1 network-cost 50
a=ice-ufrag:GnU2
a=ice-pwd:JcJvTY3uLqODpI+5jVDXUGr9
a=ice-options:trickle
a=fingerprint:sha-256 86:E8:F1:75:CF:B1:3C:79:13:F9:1E:E7:6F:C6:0A:61:52:CA:74:F0:C3:15:D6:06:3B:3D:EC:5F:BA:4D:B1:A3
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:bb960539-69c8-40e7-8803-d28ffb10c046 f55aeb97-2e9f-439d-ac20-4ebfe4214044
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:103 ISAC/16000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:3001809347 cname:KgZyvMyOk/thwV4S
a=ssrc:3001809347 msid:bb960539-69c8-40e7-8803-d28ffb10c046 f55aeb97-2e9f-439d-ac20-4ebfe4214044
a=ssrc:3001809347 mslabel:bb960539-69c8-40e7-8803-d28ffb10c046
a=ssrc:3001809347 label:f55aeb97-2e9f-439d-ac20-4ebfe4214044

<--- Transmitting SIP response (334 bytes) to WSS:xxx.xxx.xxx:57445 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS revbqlg629ue.invalid;rport=57445;received=xxx.xxx.xxx;branch=z9hG4bK158201
Call-ID: tfd43f2sbnv4n4i9s07o
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=v42h90lrg2
To: <sip:xxx.xxx.xxx@xxx.xxx.xxx>
CSeq: 8561 INVITE
Server: Asterisk PBX 18.13.0
Content-Length:  0


<--- Transmitting SIP response (1703 bytes) to WSS:xxx.xxx.xxx:57445 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS revbqlg629ue.invalid;rport=57445;received=xxx.xxx.xxx;branch=z9hG4bK158201
Call-ID: tfd43f2sbnv4n4i9s07o
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=v42h90lrg2
To: <sip:xxx.xxx.xxx@xxx.xxx.xxx>;tag=bc8d9db8-553f-446d-ba24-e673a4b33abf
CSeq: 8561 INVITE
Server: Asterisk PBX 18.13.0
Contact: <sip:172.26.5.114:8443;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 90;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:  1041

v=0
o=- 3864793253 4 IN IP4 172.26.5.114
s=Asterisk
c=IN IP4 172.26.5.114
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 15326 UDP/TLS/RTP/SAVPF 8 0 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 A1:11:12:2D:A4:75:B6:B9:6C:A8:F7:29:A0:7C:55:CB:F1:97:DD:71:3F:0B:C2:F5:C3:75:F3:0D:C3:07:AB:F6
a=ice-ufrag:0440097f7ceb8f002cbf91c704b5feef
a=ice-pwd:2ae755491f8f9db32a97602a6ab18764
a=candidate:Hac1a0572 1 UDP 2130706431 172.26.5.114 15326 typ host
a=candidate:H1a9c7137 1 UDP 2130706431 fe80::885:d1ff:fe6e:9799 15326 typ host
a=candidate:S3f212553 1 UDP 1694498815 xxx.xxx.xxx 15326 typ srflx raddr 172.26.5.114 rport 15326
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1932454137 cname:713b2d08-22df-4164-891d-db209a327ab7
a=msid:b7104718-022f-41b2-aaae-fb5e9d59d564 a87913cc-abd5-459c-ad0f-5b19dba6b7b4
a=rtcp-fb:* transport-cc
a=mid:0

<--- Transmitting SIP response (1703 bytes) to WSS:xxx.xxx.xxx:57445 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS revbqlg629ue.invalid;rport=57445;received=xxx.xxx.xxx;branch=z9hG4bK158201
Call-ID: tfd43f2sbnv4n4i9s07o
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=v42h90lrg2
To: <sip:xxx.xxx.xxx@xxx.xxx.xxx>;tag=bc8d9db8-553f-446d-ba24-e673a4b33abf
CSeq: 8561 INVITE
Server: Asterisk PBX 18.13.0
Contact: <sip:172.26.5.114:8443;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 90;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:  1041

v=0
o=- 3864793253 4 IN IP4 172.26.5.114
s=Asterisk
c=IN IP4 172.26.5.114
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 15326 UDP/TLS/RTP/SAVPF 8 0 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 A1:11:12:2D:A4:75:B6:B9:6C:A8:F7:29:A0:7C:55:CB:F1:97:DD:71:3F:0B:C2:F5:C3:75:F3:0D:C3:07:AB:F6
a=ice-ufrag:0440097f7ceb8f002cbf91c704b5feef
a=ice-pwd:2ae755491f8f9db32a97602a6ab18764
a=candidate:Hac1a0572 1 UDP 2130706431 172.26.5.114 15326 typ host
a=candidate:H1a9c7137 1 UDP 2130706431 fe80::885:d1ff:fe6e:9799 15326 typ host
a=candidate:S3f212553 1 UDP 1694498815 xxx.xxx.xxx 15326 typ srflx raddr 172.26.5.114 rport 15326
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1932454137 cname:713b2d08-22df-4164-891d-db209a327ab7
a=msid:b7104718-022f-41b2-aaae-fb5e9d59d564 a87913cc-abd5-459c-ad0f-5b19dba6b7b4
a=rtcp-fb:* transport-cc
a=mid:0

channeluniqueid 1667578170.165
initialise chattabai DIALPLAN
SessionInit
callerId xxx.xxx.xxx
Do we need pin validation cause its a variable call?
Pin validation
Variable EndPoint pin validation
Variable EndPoint pin success
Creating variable inbound call
Variable ref validation
Variable EndPoint ref validation
<--- Transmitting SIP response (1703 bytes) to WSS:xxx.xxx.xxx:57445 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS revbqlg629ue.invalid;rport=57445;received=xxx.xxx.xxx;branch=z9hG4bK158201
Call-ID: tfd43f2sbnv4n4i9s07o
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=v42h90lrg2
To: <sip:xxx.xxx.xxx@xxx.xxx.xxx>;tag=bc8d9db8-553f-446d-ba24-e673a4b33abf
CSeq: 8561 INVITE
Server: Asterisk PBX 18.13.0
Contact: <sip:172.26.5.114:8443;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 90;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:  1041

v=0
o=- 3864793253 4 IN IP4 172.26.5.114
s=Asterisk
c=IN IP4 172.26.5.114
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 15326 UDP/TLS/RTP/SAVPF 8 0 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 A1:11:12:2D:A4:75:B6:B9:6C:A8:F7:29:A0:7C:55:CB:F1:97:DD:71:3F:0B:C2:F5:C3:75:F3:0D:C3:07:AB:F6
a=ice-ufrag:0440097f7ceb8f002cbf91c704b5feef
a=ice-pwd:2ae755491f8f9db32a97602a6ab18764
a=candidate:Hac1a0572 1 UDP 2130706431 172.26.5.114 15326 typ host
a=candidate:H1a9c7137 1 UDP 2130706431 fe80::885:d1ff:fe6e:9799 15326 typ host
a=candidate:S3f212553 1 UDP 1694498815 xxx.xxx.xxx 15326 typ srflx raddr 172.26.5.114 rport 15326
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1932454137 cname:713b2d08-22df-4164-891d-db209a327ab7
a=msid:b7104718-022f-41b2-aaae-fb5e9d59d564 a87913cc-abd5-459c-ad0f-5b19dba6b7b4
a=rtcp-fb:* transport-cc
a=mid:0

Variable EndPoint ref success
Ending variable inbound call
Variable inbound call succesfully ended
Creating variable outbound call
callerid: 14663CC9-7C1F-447F-AAF1-2BD2F451E6AD
dialednumber: xxx.xxx.xxx
ctbEmployeeOutboundNumber: 
ctbVisitorWalletId: 1670
Variable outbound call start success
Callrecording: 1
<--- Received SIP request (472 bytes) from WSS:xxx.xxx.xxx:57445 --->
ACK sip:172.26.5.114:8443;transport=ws SIP/2.0
Via: SIP/2.0/WSS revbqlg629ue.invalid;branch=z9hG4bK2551852
Max-Forwards: 69
To: <sip:xxx.xxx.xxx@xxx.xxx.xxx>;tag=bc8d9db8-553f-446d-ba24-e673a4b33abf
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=v42h90lrg2
Call-ID: tfd43f2sbnv4n4i9s07o
CSeq: 8561 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.9.1
Content-Length: 0


<--- Received SIP request (472 bytes) from WSS:xxx.xxx.xxx:57445 --->
ACK sip:172.26.5.114:8443;transport=ws SIP/2.0
Via: SIP/2.0/WSS revbqlg629ue.invalid;branch=z9hG4bK3877204
Max-Forwards: 69
To: <sip:xxx.xxx.xxx@xxx.xxx.xxx>;tag=bc8d9db8-553f-446d-ba24-e673a4b33abf
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=v42h90lrg2
Call-ID: tfd43f2sbnv4n4i9s07o
CSeq: 8561 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.9.1
Content-Length: 0


<--- Received SIP request (472 bytes) from WSS:xxx.xxx.xxx:57445 --->
ACK sip:172.26.5.114:8443;transport=ws SIP/2.0
Via: SIP/2.0/WSS revbqlg629ue.invalid;branch=z9hG4bK9597998
Max-Forwards: 69
To: <sip:xxx.xxx.xxx@xxx.xxx.xxx>;tag=bc8d9db8-553f-446d-ba24-e673a4b33abf
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=v42h90lrg2
Call-ID: tfd43f2sbnv4n4i9s07o
CSeq: 8561 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.9.1
Content-Length: 0


[Nov  4 17:09:32] NOTICE[1459680][C-00000025]: app_stack.c:1079 gosub_run: PJSIP/twilio-00000061 Abnormal 'Gosub(ctbOutboundCallStartHeaders,addheader,1)' exit.  Popping routine return locations.
[Nov  4 17:09:32] NOTICE[1459680][C-00000025]: app_stack.c:1079 gosub_run: PJSIP/D0F460EC-E02D-47EF-B39F-88BC8DE7DFB0-00000062 Abnormal 'Gosub(ctbOutboundCallStartHeaders,addheader,1)' exit.  Popping routine return locations.
<--- Transmitting SIP request (1800 bytes) to WSS:xxx.xxx.xxx:57441 --->
INVITE sip:8uca8051@xxx.xxx.xxx:57441;transport=ws SIP/2.0
Via: SIP/2.0/WSS 172.26.5.114:8443;rport;branch=z9hG4bKPjdf9fabe2-a5e6-4c52-b7eb-4a6ab35791c6;alias
From: <sip:+xxx.xxx.xxx@xxx.xxx.xxx>;tag=9a869087-2368-4160-b927-71755396847a
To: <sip:8uca8051@xxx.xxx.xxx>
Contact: <sip:asterisk@xxx.xxx.xxx:5060;transport=ws>
Call-ID: 69405790-5926-4239-b2ff-e2395640f136
CSeq: 29648 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
CtbVisitorUserId: 1675
Max-Forwards: 70
User-Agent: Asterisk PBX 18.13.0
Content-Type: application/sdp
Content-Length:  1063

v=0
o=- 1345366189 1345366189 IN IP4 172.26.5.114
s=Asterisk
c=IN IP4 172.26.5.114
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 11184 UDP/TLS/RTP/SAVPF 8 0 107 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 C4:36:31:05:48:5A:FE:08:5F:E1:A4:0C:A7:64:24:C7:B5:BA:6A:B8:EE:F2:92:4D:D7:40:91:7E:9B:06:28:F0
a=ice-ufrag:6d00066c4e7b35f62f0547c0340b9c8c
a=ice-pwd:551579364371639d694396467b0c31f3
a=candidate:Hac1a0572 1 UDP 2130706431 172.26.5.114 11184 typ host
a=candidate:H1a9c7137 1 UDP 2130706431 fe80::885:d1ff:fe6e:9799 11184 typ host
a=candidate:S3f212553 1 UDP 1694498815 xxx.xxx.xxx 11184 typ srflx raddr 172.26.5.114 rport 11184
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:2005242914 cname:18ac60a4-243b-4eb7-a442-4d086d0da86c
a=msid:72316601-6bf1-4243-b998-00dabd207a55 2fde1f69-0ca3-4bfe-a5a7-e33eaae56faa
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Transmitting SIP request (1147 bytes) to TLS:54.171.127.193:5061 --->
INVITE sip:+@janus.pstn.dublin.twilio.com SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.xxx:5061;rport;branch=z9hG4bKPjbf72bdbc-4948-4d95-a69d-a1ff9c856426;alias
From: <sip:+xxx.xxx.xxx@172.26.5.114>;tag=df657bcf-11ee-480a-918a-f4dce5b72f46
To: <sip:+@janus.pstn.dublin.twilio.com>
Contact: <sip:asterisk@xxx.xxx.xxx:5061;transport=TLS>
Call-ID: 8f0990b0-d5e5-47fc-9012-0d46c805de0f
CSeq: 4429 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
CtbVisitorUserId: 1675
Max-Forwards: 70
User-Agent: Asterisk PBX 18.13.0
Content-Type: application/sdp
Content-Length:   415

v=0
o=- 1424902494 1424902494 IN IP4 xxx.xxx.xxx
s=Asterisk
c=IN IP4 xxx.xxx.xxx
t=0 0
m=audio 18634 RTP/SAVP 8 0 18 9 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:mk2sCFmlMBwvAUEhiqCFv4x0dtgua2FQm3lN453s
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (413 bytes) from TLS:54.171.127.193:5061 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/TLS xxx.xxx.xxx:5061;rport=56007;branch=z9hG4bKPjbf72bdbc-4948-4d95-a69d-a1ff9c856426;alias;received=xxx.xxx.xxx
From: <sip:+xxx.xxx.xxx@172.26.5.114>;tag=df657bcf-11ee-480a-918a-f4dce5b72f46
To: <sip:+@janus.pstn.dublin.twilio.com>
Call-ID: 8f0990b0-d5e5-47fc-9012-0d46c805de0f
CSeq: 4429 INVITE
Server: Twilio Gateway
Content-Length: 0


<--- Received SIP response (364 bytes) from WSS:xxx.xxx.xxx:57441 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 172.26.5.114:8443;rport;branch=z9hG4bKPjdf9fabe2-a5e6-4c52-b7eb-4a6ab35791c6;alias
To: <sip:8uca8051@xxx.xxx.xxx>
From: <sip:+xxx.xxx.xxx@xxx.xxx.xxx>;tag=9a869087-2368-4160-b927-71755396847a
Call-ID: 69405790-5926-4239-b2ff-e2395640f136
CSeq: 29648 INVITE
Supported: timer,ice,replaces,outbound
Content-Length: 0


<--- Received SIP response (439 bytes) from WSS:xxx.xxx.xxx:57441 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 172.26.5.114:8443;rport;branch=z9hG4bKPjdf9fabe2-a5e6-4c52-b7eb-4a6ab35791c6;alias
To: <sip:8uca8051@xxx.xxx.xxx>;tag=8545pnh7av
From: <sip:+xxx.xxx.xxx@xxx.xxx.xxx>;tag=9a869087-2368-4160-b927-71755396847a
Call-ID: 69405790-5926-4239-b2ff-e2395640f136
CSeq: 29648 INVITE
Contact: <sip:8uca8051@hgvuvmvd4har.invalid;transport=ws>
Supported: timer,ice,replaces,outbound
Content-Length: 0


<--- Received SIP response (682 bytes) from TLS:54.171.127.193:5061 --->
SIP/2.0 407 Proxy Authentication required
CSeq: 4429 INVITE
Call-ID: 8f0990b0-d5e5-47fc-9012-0d46c805de0f
From: <sip:+xxx.xxx.xxx@172.26.5.114>;tag=df657bcf-11ee-480a-918a-f4dce5b72f46
To: <sip:+@janus.pstn.dublin.twilio.com>;tag=95830338_c3356d0b_a5b29cf8-3b61-48ec-886b-c5d2188abed1
Via: SIP/2.0/TLS xxx.xxx.xxx:5061;received=xxx.xxx.xxx;rport=56007;branch=z9hG4bKPjbf72bdbc-4948-4d95-a69d-a1ff9c856426;alias
Server: Twilio
Contact: <sip:172.18.197.135:5060>
Proxy-Authenticate: Digest realm="sip.twilio.com",qop="auth",nonce="651k4brTUPL75Aq75tddVd5cT_t16bTH9IbFiV77p_1nA6GZ",opaque="95a4620a38c909c2038ad88bfbf92dc1"
X-Twilio-TlsPolicy: TLSv1.2+
Content-Length: 0


<--- Transmitting SIP request (467 bytes) to TLS:54.171.127.193:5061 --->
ACK sip:+@janus.pstn.dublin.twilio.com SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.xxx:5061;rport;branch=z9hG4bKPjbf72bdbc-4948-4d95-a69d-a1ff9c856426;alias
From: <sip:+xxx.xxx.xxx@172.26.5.114>;tag=df657bcf-11ee-480a-918a-f4dce5b72f46
To: <sip:+@janus.pstn.dublin.twilio.com>;tag=95830338_c3356d0b_a5b29cf8-3b61-48ec-886b-c5d2188abed1
Call-ID: 8f0990b0-d5e5-47fc-9012-0d46c805de0f
CSeq: 4429 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.13.0
Content-Length:  0


<--- Transmitting SIP request (1480 bytes) to TLS:54.171.127.193:5061 --->
INVITE sip:+@janus.pstn.dublin.twilio.com SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.xxx:5061;rport;branch=z9hG4bKPj12081800-ee06-40c0-a045-68b4dd05a9a9;alias
From: <sip:+xxx.xxx.xxx@172.26.5.114>;tag=df657bcf-11ee-480a-918a-f4dce5b72f46
To: <sip:+@janus.pstn.dublin.twilio.com>
Contact: <sip:asterisk@xxx.xxx.xxx:5061;transport=TLS>
Call-ID: 8f0990b0-d5e5-47fc-9012-0d46c805de0f
CSeq: 4430 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
CtbVisitorUserId: 1675
Max-Forwards: 70
User-Agent: Asterisk PBX 18.13.0
Proxy-Authorization: Digest username="xxx.xxx.xxx", realm="sip.twilio.com", nonce="651k4brTUPL75Aq75tddVd5cT_t16bTH9IbFiV77p_1nA6GZ", uri="sip:+@janus.pstn.dublin.twilio.com", response="b26ca3118aea4c06d57123f57bc0cb10", cnonce="cac6dd4b32384c10bbc021a6f8c0674c", opaque="95a4620a38c909c2038ad88bfbf92dc1", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   415

v=0
o=- 1424902494 1424902494 IN IP4 xxx.xxx.xxx
s=Asterisk
c=IN IP4 xxx.xxx.xxx
t=0 0
m=audio 18634 RTP/SAVP 8 0 18 9 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:mk2sCFmlMBwvAUEhiqCFv4x0dtgua2FQm3lN453s
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (413 bytes) from TLS:54.171.127.193:5061 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/TLS xxx.xxx.xxx:5061;rport=56007;branch=z9hG4bKPj12081800-ee06-40c0-a045-68b4dd05a9a9;alias;received=xxx.xxx.xxx
From: <sip:+xxx.xxx.xxx@172.26.5.114>;tag=df657bcf-11ee-480a-918a-f4dce5b72f46
To: <sip:+@janus.pstn.dublin.twilio.com>
Call-ID: 8f0990b0-d5e5-47fc-9012-0d46c805de0f
CSeq: 4430 INVITE
Server: Twilio Gateway
Content-Length: 0

<--- Received SIP response (581 bytes) from TLS:54.171.127.193:5061 --->
SIP/2.0 400 Invalid phone number
CSeq: 4430 INVITE
Call-ID: 8f0990b0-d5e5-47fc-9012-0d46c805de0f
From: <sip:+xxx.xxx.xxx@172.26.5.114>;tag=df657bcf-11ee-480a-918a-f4dce5b72f46
To: <sip:+@janus.pstn.dublin.twilio.com>;tag=20365519_c3356d0b_2c626b87-9cd9-4cb9-a427-b68da037d1c4
Via: SIP/2.0/TLS xxx.xxx.xxx:5061;received=xxx.xxx.xxx;rport=56007;branch=z9hG4bKPj12081800-ee06-40c0-a045-68b4dd05a9a9;alias
Server: Twilio
Contact: <sip:172.18.212.166:5060>
X-Twilio-Error: 32101 The called number is not correctly formatted.
X-Twilio-TlsPolicy: TLSv1.2+
Content-Length: 0


<--- Transmitting SIP request (467 bytes) to TLS:54.171.127.193:5061 --->
ACK sip:+@janus.pstn.dublin.twilio.com SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.xxx:5061;rport;branch=z9hG4bKPj12081800-ee06-40c0-a045-68b4dd05a9a9;alias
From: <sip:+xxx.xxx.xxx@172.26.5.114>;tag=df657bcf-11ee-480a-918a-f4dce5b72f46
To: <sip:+@janus.pstn.dublin.twilio.com>;tag=20365519_c3356d0b_2c626b87-9cd9-4cb9-a427-b68da037d1c4
Call-ID: 8f0990b0-d5e5-47fc-9012-0d46c805de0f
CSeq: 4430 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.13.0
Content-Length:  0


<--- Received SIP response (1511 bytes) from WSS:xxx.xxx.xxx:57441 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 172.26.5.114:8443;rport;branch=z9hG4bKPjdf9fabe2-a5e6-4c52-b7eb-4a6ab35791c6;alias
To: <sip:8uca8051@xxx.xxx.xxx>;tag=8545pnh7av
From: <sip:+xxx.xxx.xxx@xxx.xxx.xxx>;tag=9a869087-2368-4160-b927-71755396847a
Call-ID: 69405790-5926-4239-b2ff-e2395640f136
CSeq: 29648 INVITE
Contact: <sip:8uca8051@hgvuvmvd4har.invalid;transport=ws>
Session-Expires: 1800;refresher=uas
Supported: timer,ice,replaces,outbound
Content-Type: application/sdp
Content-Length: 1006

v=0
o=- 5449372610522312373 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio-0
a=msid-semantic: WMS d1f1d529-652f-4f13-b532-88efb019a8e8
m=audio 58780 UDP/TLS/RTP/SAVPF 8 0 107 101
c=IN IP4 10.8.0.3
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:4091210427 1 udp 2122260223 10.8.0.3 58780 typ host generation 0 network-id 1 network-cost 50
a=candidate:3176732235 1 tcp 1518280447 10.8.0.3 9 typ host tcptype active generation 0 network-id 1 network-cost 50
a=ice-ufrag:a8fZ
a=ice-pwd:4YiYhVkU1Wf7qM/sYN63MDhy
a=ice-options:trickle
a=fingerprint:sha-256 60:97:D8:05:D5:17:EE:7E:94:10:9D:6A:2B:A4:BD:09:A7:A9:83:C1:B1:B0:98:F6:DB:83:39:42:28:14:9E:8F
a=setup:active
a=mid:audio-0
a=sendrecv
a=msid:d1f1d529-652f-4f13-b532-88efb019a8e8 cdaefa36-dfd0-4ef1-b503-0860369a1376
a=rtcp-mux
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=rtcp-fb:107 transport-cc
a=fmtp:107 minptime=10;useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=ssrc:2328851244 cname:ghYsQClPAKObRSTz

<--- Transmitting SIP request (426 bytes) to WSS:xxx.xxx.xxx:57441 --->
ACK sip:8uca8051@xxx.xxx.xxx:57441;transport=ws SIP/2.0
Via: SIP/2.0/WSS 172.26.5.114:8443;rport;branch=z9hG4bKPje85110bf-07fe-4fa3-86f3-441df6701f8b;alias
From: <sip:+xxx.xxx.xxx@xxx.xxx.xxx>;tag=9a869087-2368-4160-b927-71755396847a
To: <sip:8uca8051@xxx.xxx.xxx>;tag=8545pnh7av
Call-ID: 69405790-5926-4239-b2ff-e2395640f136
CSeq: 29648 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.13.0
Content-Length:  0


<--- Received SIP request (625 bytes) from WSS:xxx.xxx.xxx:57586 --->
REGISTER sip:xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/WSS oeee0q6kumpu.invalid;branch=z9hG4bK8818310
Max-Forwards: 69
To: <sip:D0F460EC-E02D-47EF-B39F-88BC8DE7DFB0@xxx.xxx.xxx>
From: <sip:D0F460EC-E02D-47EF-B39F-88BC8DE7DFB0@xxx.xxx.xxx>;tag=v0ai2ongki
Call-ID: hmepuje7fu03r2fqeoed6n
CSeq: 1 REGISTER
Contact: <sip:lnpambpr@oeee0q6kumpu.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:0dd202eb-f77d-47ff-a36f-262283e8e8b8>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: JsSIP 3.9.1
Content-Length: 0

<--- Transmitting SIP response (531 bytes) to WSS:xxx.xxx.xxx:57586 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS oeee0q6kumpu.invalid;rport=57586;received=xxx.xxx.xxx;branch=z9hG4bK8818310
Call-ID: hmepuje7fu03r2fqeoed6n
From: <sip:D0F460EC-E02D-47EF-B39F-88BC8DE7DFB0@xxx.xxx.xxx>;tag=v0ai2ongki
To: <sip:D0F460EC-E02D-47EF-B39F-88BC8DE7DFB0@xxx.xxx.xxx>;tag=z9hG4bK8818310
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1667578190/31eaef5c02dc301b76f33913e2a76d5f",opaque="61d37b7c6382fabf",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.13.0
Content-Length:  0

<--- Received SIP request (926 bytes) from WSS:xxx.xxx.xxx:57586 --->
REGISTER sip:xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/WSS oeee0q6kumpu.invalid;branch=z9hG4bK6034078
Max-Forwards: 69
To: <sip:D0F460EC-E02D-47EF-B39F-88BC8DE7DFB0@xxx.xxx.xxx>
From: <sip:D0F460EC-E02D-47EF-B39F-88BC8DE7DFB0@xxx.xxx.xxx>;tag=v0ai2ongki
Call-ID: hmepuje7fu03r2fqeoed6n
CSeq: 2 REGISTER
Authorization: Digest algorithm=MD5, username="9F37BA9D-AA0C-430E-82A5-EE26411EA559", realm="asterisk", nonce="1667578190/31eaef5c02dc301b76f33913e2a76d5f", uri="sip:xxx.xxx.xxx", response="e183071447ff3dc1a7ba80867a888ecc", opaque="61d37b7c6382fabf", qop=auth, cnonce="9fp3r0dqqef7", nc=00000001
Contact: <sip:lnpambpr@oeee0q6kumpu.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:0dd202eb-f77d-47ff-a36f-262283e8e8b8>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: JsSIP 3.9.1
Content-Length: 0

<--- Transmitting SIP response (497 bytes) to WSS:xxx.xxx.xxx:57586 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS oeee0q6kumpu.invalid;rport=57586;received=xxx.xxx.xxx;branch=z9hG4bK6034078
Call-ID: hmepuje7fu03r2fqeoed6n
From: <sip:D0F460EC-E02D-47EF-B39F-88BC8DE7DFB0@xxx.xxx.xxx>;tag=v0ai2ongki
To: <sip:D0F460EC-E02D-47EF-B39F-88BC8DE7DFB0@xxx.xxx.xxx>;tag=z9hG4bK6034078
CSeq: 2 REGISTER
Date: Fri, 04 Nov 2022 16:09:50 GMT
Contact: <sip:lnpambpr@oeee0q6kumpu.invalid;transport=ws>;expires=599
Expires: 600
Server: Asterisk PBX 18.13.0
Content-Length:  0

<--- Received SIP request (625 bytes) from WSS:xxx.xxx.xxx:57595 --->
REGISTER sip:xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/WSS vr07cfrvinpe.invalid;branch=z9hG4bK8011378
Max-Forwards: 69
To: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=ismjim71gk
Call-ID: 4u0kn832tcuopr836s4g3r
CSeq: 1 REGISTER
Contact: <sip:kbptpeop@vr07cfrvinpe.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:14623be6-498e-4a39-9286-c8f6ac935d8e>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: JsSIP 3.9.1
Content-Length: 0


<--- Transmitting SIP response (531 bytes) to WSS:xxx.xxx.xxx:57595 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS vr07cfrvinpe.invalid;rport=57595;received=xxx.xxx.xxx;branch=z9hG4bK8011378
Call-ID: 4u0kn832tcuopr836s4g3r
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=ismjim71gk
To: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=z9hG4bK8011378
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1667578191/3a9d1287d4766f1dc877efae35b3edde",opaque="5a9130c04c5c82d9",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.13.0
Content-Length:  0

<--- Received SIP request (926 bytes) from WSS:xxx.xxx.xxx:57595 --->
REGISTER sip:xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/WSS vr07cfrvinpe.invalid;branch=z9hG4bK3084336
Max-Forwards: 69
To: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=ismjim71gk
Call-ID: 4u0kn832tcuopr836s4g3r
CSeq: 2 REGISTER
Authorization: Digest algorithm=MD5, username="C45841A4-A2F6-49D4-8CA2-9D0C554D9919", realm="asterisk", nonce="1667578191/3a9d1287d4766f1dc877efae35b3edde", uri="sip:xxx.xxx.xxx", response="26e1dee875c27e7a97419b749c475fd4", opaque="5a9130c04c5c82d9", qop=auth, cnonce="5jv9ch7qrlm2", nc=00000001
Contact: <sip:kbptpeop@vr07cfrvinpe.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:14623be6-498e-4a39-9286-c8f6ac935d8e>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: JsSIP 3.9.1
Content-Length: 0


<--- Transmitting SIP response (497 bytes) to WSS:xxx.xxx.xxx:57595 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS vr07cfrvinpe.invalid;rport=57595;received=xxx.xxx.xxx;branch=z9hG4bK3084336
Call-ID: 4u0kn832tcuopr836s4g3r
From: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=ismjim71gk
To: <sip:14663CC9-7C1F-447F-AAF1-2BD2F451E6AD@xxx.xxx.xxx>;tag=z9hG4bK3084336
CSeq: 2 REGISTER
Date: Fri, 04 Nov 2022 16:09:51 GMT
Contact: <sip:kbptpeop@vr07cfrvinpe.invalid;transport=ws>;expires=599
Expires: 600
Server: Asterisk PBX 18.13.0
Content-Length:  0

<--- Transmitting SIP request (460 bytes) to TLS:54.171.127.192:5061 --->
OPTIONS sip:janus.pstn.dublin.twilio.com SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.xxx:5061;rport;branch=z9hG4bKPj239ce0e9-c396-4229-b117-3d50eaf70f4d;alias
From: <sip:twilio@172.26.5.114>;tag=55906e6d-9b5a-4b76-8845-4d9880245182
To: <sip:janus.pstn.dublin.twilio.com>
Contact: <sip:twilio@xxx.xxx.xxx:5061;transport=TLS>
Call-ID: a128fbcc-392d-4730-a044-7288ad6f8c26
CSeq: 58633 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 18.13.0
Content-Length:  0

<--- Received SIP response (416 bytes) from TLS:54.171.127.192:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS xxx.xxx.xxx:5061;rport=53625;branch=z9hG4bKPj239ce0e9-c396-4229-b117-3d50eaf70f4d;alias;received=xxx.xxx.xxx
From: <sip:twilio@172.26.5.114>;tag=55906e6d-9b5a-4b76-8845-4d9880245182
To: <sip:janus.pstn.dublin.twilio.com>;tag=22c56222765cdc5bdea4795da03a6302.8ea6d15e
Call-ID: a128fbcc-392d-4730-a044-7288ad6f8c26
CSeq: 58633 OPTIONS
Server: Twilio Gateway
Content-Length: 0