How to get call from anywhere of world

Hello, I want to get call from all numbers but I dont know whether should I change settings in sip.conf or extensions.conf. I have also got forwarding number from sip provider but its without any login details. I am also getting following error when trying to call on.asterisk from outside.

Retransmitting #10 (no NAT) to
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP;branch=z9hG4bK-25c67de1944eacc4dec18d5140554dd8;received=;rport=5070
From: 503<sip:503[@167](>;tag=b555c9d2
To: 0146406820577<sip:0146406820577[@167](>;tag=as25f83ad3
Call-ID: 25c67de1944eacc4dec18d5140554dd8
Server: VitalPBX

401 is not an error. It is a request to send your password. Did you configure it?

I got this number from didx and dont know about any password. Do I need password?

This is my sip.conf file


and this is extensions.conf

exten => 15672446095,1,Goto(cos-all,*789,1)

If you dont use secret on your peer, INVITE request won’t be challenged to Authenticate (SIP 401)

Also you can use insecure=invite

I got confused here. The OP says he was calling out from Asterisk, in which case the peer would generate the 401, and the user agent is VitalPBX, rather than Asterisk. However, it is a retransmission, so it must really be an incoming call to Asterisk.

For outgong calls, with chan_sip, I would suggest remotesecret is cleaner than secret+insecure. However, for a new system, chan_pjsip is better than chan_sip.

If it is up to retransmit number 10, there is something more fundamental than authentication that is broken. The OP should check the Contact header, which he seems to have missed from the protocol capture.

I thought he was talking about inbound calls so, if is inbound asterisk could challenge auth, depending on his configuration, if is outbound it is up to his Carrier to authorize the call

The log does seem to show an inbound call, but the body of the text seems to say otherwise (and the user agent didn’t contain Asterisk, or one of the well known GUIs). But, in any case, they are failing to get an ACK, so there is something more than authentication wrong.

First time I heard about this system VitalPBX ,but they say they re Fastest growing PBX system based on Asterisk

It loooks like a GUI, in which case support needs to be obtained from VitaPBX.

On the other hand, I failed to find the page for configuring a SIP ITSP peer on their demo. I couldn’t work out their design metaphor.

VitalPBX used to be Ombutel, the name which people might be more familiar with.

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