Can't register Xlite Asterisk 11.3

Im using Asterisk 11.3 and Xlite for Mac (Asterisk running in VMware), using FreePBX 11.
Regardless my config in file or in Freepbx I always get invalid password. I have configured multiple accounts and same issue. (This is the first time I configure Asterisk but cant find issue)

in Xlite for Mac (192.168.1.69)
username 202
password 202
domain 192.168.1.10
Auth name 202
Register with domain and receive calls
Proxy 192.168.1.10

I setup packet capture in Mac and I do get the messages. I have tried:
Disable firewall in Asterisk
TCP or UDP
remove rport
Use NAT, and no NAT
alwaysauthreject=no
I disabled strong secrets and same issue.
Jitsi causes same issue.

any ideas?

vmexten=*97
disallow=all
allow=ulaw
allow=alaw
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=2.11.0beta1(11.3.0)

/etc/asterisk/sip_additional.conf

[202]
deny=0.0.0.0/0.0.0.0
secret=202
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=tcp,udp,tls
encryption=no
callgroup=
pickupgroup=
dial=SIP/202
mailbox=202@device
permit=0.0.0.0/0.0.0.0
callerid=202 <202>
callcounter=yes
faxdetect=no

[Apr 17 22:53:17] NOTICE[19800] chan_sip.c: Registration from ‘"202"sip:202@192.168.1.10’ failed for ‘192.168.1.69:61098’ - Wrong password
[Apr 17 22:53:17] VERBOSE[19800] chan_sip.c: Scheduling destruction of SIP dialog ‘ZTljMDFiMjM0YWRhMmEzMTkyNjljZTU4NWJmYTdkMWM’ in 32000 ms (Method: REGISTER)
[Apr 17 22:53:19] VERBOSE[19800] chan_sip.c:
<— SIP read from UDP:192.168.1.69:2180 —>
REGISTER sip:192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.69:2180;branch=z9hG4bK-d8754z-6a113c13596b2e69-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:202@192.168.1.69:2180;rinstance=737c2a79ebff1ba6
To: "202"sip:202@192.168.1.10
From: "202"sip:202@192.168.1.10;tag=d17fdd0d
Call-ID: OTNjODRiOGQ0ZTM0ZjdmY2M3ZjA4MDA0ZWYxMDdkODY
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 4.5 stamp 69608
Content-Length: 0

<------------->
[Apr 17 22:53:19] VERBOSE[19800] chan_sip.c: — (12 headers 0 lines) —
[Apr 17 22:53:19] VERBOSE[19800] chan_sip.c: Sending to 192.168.1.69:2180 (no NAT)
[Apr 17 22:53:19] VERBOSE[19800] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.69:2180 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.69:2180;branch=z9hG4bK-d8754z-6a113c13596b2e69-1—d8754z-;received=192.168.1.69;rport=2180
From: "202"sip:202@192.168.1.10;tag=d17fdd0d
To: "202"sip:202@192.168.1.10;tag=as0ef8878f
Call-ID: OTNjODRiOGQ0ZTM0ZjdmY2M3ZjA4MDA0ZWYxMDdkODY
CSeq: 1 REGISTER
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="61160794"
Content-Length: 0

<------------>
[Apr 17 22:53:19] VERBOSE[19800] chan_sip.c: Scheduling destruction of SIP dialog ‘OTNjODRiOGQ0ZTM0ZjdmY2M3ZjA4MDA0ZWYxMDdkODY’ in 32000 ms (Method: REGISTER)
[Apr 17 22:53:19] VERBOSE[19800] chan_sip.c:
<— SIP read from UDP:192.168.1.69:2180 —>
REGISTER sip:192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.69:2180;branch=z9hG4bK-d8754z-8531935020d62a12-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:202@192.168.1.69:2180;rinstance=737c2a79ebff1ba6
To: "202"sip:202@192.168.1.10
From: "202"sip:202@192.168.1.10;tag=d17fdd0d
Call-ID: OTNjODRiOGQ0ZTM0ZjdmY2M3ZjA4MDA0ZWYxMDdkODY
CSeq: 2 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 4.5 stamp 69608
Authorization: Digest username=“202”,realm=“asterisk”,nonce=“61160794”,uri=“sip:192.168.1.10”,response=“62c6e544d0c7bec09132491b034ba0c7”,algorithm=MD5
Content-Length: 0

<------------->
[Apr 17 22:53:19] VERBOSE[19800] chan_sip.c: — (13 headers 0 lines) —
[Apr 17 22:53:19] VERBOSE[19800] chan_sip.c: Sending to 192.168.1.69:2180 (no NAT)
[Apr 17 22:53:19] VERBOSE[19800] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.69:2180 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.69:2180;branch=z9hG4bK-d8754z-8531935020d62a12-1—d8754z-;received=192.168.1.69;rport=2180
From: "202"sip:202@192.168.1.10;tag=d17fdd0d
To: "202"sip:202@192.168.1.10;tag=as0ef8878f
Call-ID: OTNjODRiOGQ0ZTM0ZjdmY2M3ZjA4MDA0ZWYxMDdkODY
CSeq: 2 REGISTER
Server: Asterisk PBX 11.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Global Settings:

UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: 0.0.0.0:5060
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promisc. redir: No
Enable call counters: No
SIP domain support: Yes
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX 11.3.0
SDP Session Name: Asterisk PBX 11.3.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:

IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No

Network Settings:

SIP address remapping: Disabled, no localnet list
Externhost:
Externaddr: (null)
Externrefresh: 10

Global Signalling Settings:

Codecs: (gsm|ulaw|alaw|h263|testlaw)
Codec Order: none
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: default
Record on feature: automon
Record off feature: automon
Force rport: Auto (No)
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language:
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk