Manual Outbound not working. Please guide and help

[Salahi]

Hi,

I need your help on what's the correct configuration that I need to use so I can do an outbound call? Here's my current setup including the version of my ViciDial server:

Registration String:
register => register => Salahi:xxxxxxxxxxx@as1.doorvaani.com:5060

Account Entry:
[Salahi]
type=peer
port=5060
nat=yes
insecure=invite
ignoresdpversion=yes
username=Salahi
secret=xxxxxxxxxxxx
host=as1.doorvaani.com
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw
allow=g729

Globals String: TESTSIPTRUNK = SIP/Salahi

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@Salahi,,tTo)
exten => _91XXXXXXXXXX,3,Hangup

VERSION: 2.14-904a
BUILD: 231204-0858
© 2023 ViciDial Group

The carrier successfully registered. However, when I tried to perform an outbound calls, the Asterisk logs always shows below and the calls didn't go thru:

<--- Transmitting (NAT) to 192.168.142.1:51357 --->
[Jan 22 16:22:30] SIP/2.0 401 Unauthorized
[Jan 22 16:22:30] Via: SIP/2.0/UDP 192.168.142.1:51357;branch=z9hG4bK-524287-1---1b15ff4914598574;received=192.168.142.1;rport=51357
[Jan 22 16:22:30] From: "7252401324"<sip:1001@192.168.142.130;transport=UDP>;tag=9d710949
[Jan 22 16:22:30] To: "7252401324"<sip:1001@192.168.142.130;transport=UDP>;tag=as17cd8a0d
[Jan 22 16:22:30] Call-ID: TXWLfAUzSWx-c3ZWanpD6w..
[Jan 22 16:22:30] CSeq: 31 REGISTER
[Jan 22 16:22:30] Server: Asterisk PBX 16.30.0-vici
[Jan 22 16:22:30] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jan 22 16:22:30] Supported: replaces, timer
[Jan 22 16:22:30] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="228de381"
[Jan 22 16:22:30] Content-Length: 0


<--- SIP read from UDP:52.19.173.147:5060 --->
[Jan 22 16:22:53] SIP/2.0 404 Not Found
[Jan 22 16:22:53] Via: SIP/2.0/UDP 223.233.74.123:5060;rport=4948;received=223.233.77.80;branch=z9hG4bK0ed5e88f
[Jan 22 16:22:53] From: "asterisk" <sip:asterisk@223.233.74.123>;tag=as1138555c
[Jan 22 16:22:53] To: <sip:as1.doorvaani.com>;tag=as7af93eba
[Jan 22 16:22:53] Call-ID: 053caf9d6a609e4e5a68342b48ed083c@223.233.74.123:5060
[Jan 22 16:22:53] CSeq: 102 OPTIONS
[Jan 22 16:22:53] Server: Asterisk PBX 13.5.0
[Jan 22 16:22:53] Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
[Jan 22 16:22:53] Supported: replaces,timer
[Jan 22 16:22:53] Accept: application/sdp
[Jan 22 16:22:53] Content-Length: 0
[Jan 22 16:22:53]
[Jan 22 16:22:53] <------------->
[Jan 22 16:22:53] --- (11 headers 0 lines) ---
[Jan 22 16:22:53] Really destroying SIP dialog '053caf9d6a609e4e5a68342b48ed083c@223.233.74.123:5060' Method: OPTIONS

Hope you can help me. Thank you in advance!

[carpenox]

Edit out your personal info and set a caller id then you should be fine

[Salahi]

Caller ID is already set to 7252401324 but then also error is coming

[carpenox]

Edit your personal info bro, your username and password, your IPs, someone gonna use your account

[Salahi]

Done brother. If you can pls guide me on the error part

[carpenox]

your carrier isnt allowing the call from that DID, you need to get with them to see why the call is being rejected