Can't make outbound calls through trunk

[buba]

Hello, I have a trunk that i want to make calls both ways. It’s from orange polska

I was provided with a username and password:
[email protected]
password

When I input this into microsip all is well i can use it both ways.

However in FreePBX I can only receive calls.

In the log FreePBX spits out this error:

[2025-10-20 19:21:47] WARNING[499372]: res_pjsip_outbound_authenticator_digest.c:577 digest_create_request_with_auth: noendpoint:83.0.8.193: No auth objects matching realm/algorithm(s) '' from challenge found.

This is my PJSIP config in the GUI.

Please help and thank you in advance.

[david55]

You are being asked to prove that you know the value of this:

but haven’t configured it. They may call it the password.

[buba]

if i dont hover my mouse over this, it blurs it. With an extension that turns light mode pages into dark mode it appears that there is nothing there but I assure you the password is filled out. If I hover over it, it appears. If i refresh it’s still there, even on another device it’s still there

[buba]

[david55]

I think we need to see what is in the challenge, in their 401 response. Use pjsip set logger on, to enable output of the SIP, or use sngrep, or use tcpdump/wireshark.

[buba]

<--- Received SIP response (597 bytes) from UDP:83.0.8.193:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.3:5060;received=x.x.x.x(my ip);branch=z9hG4bKPja5b7ed60-49cf-4803-b16e-2a998a1b7991;rport=5060
From: <sip:[email protected]>;tag=042b5528-1f71-41b6-b5c3-fefaf126926d
To: <sip:[email protected]>;tag=918976478
Call-ID: eefcbc70-b098-4053-a366-a4939478b15c
CSeq: 11004 REGISTER
WWW-Authenticate: Digest realm="neofon.tp.pl",
   nonce="413f50ed68f698ff1OF8fffe66c33090b8fbcfdb2bb3ce142500c9",
   opaque="ALU:QbkRBthODQ0HFwYVBA5ZW0UFAB1BBAFcAxhPQ0dPSUEtMSg0KCV6GwYMCQ__",
   algorithm=MD5,
   qop="auth"
Content-Length: 0

and I’m also getting this proxy authenticaiton required

<--- Received SIP response (435 bytes) from UDP:83.0.8.193:5060 --->
SIP/2.0 407 Proxy Authentication Required
Call-ID: 902ea984-0592-463d-ad44-bfa18213f1d2
Via: SIP/2.0/UDP 192.168.1.3:5060;received=x.x.x.x(my ip);branch=z9hG4bKPj2ceb8372-1caf-4c44-a58c-61377d248761;rport=5060
To: <sip:[email protected]>;tag=67a62a89-68f6998a1c4f910f
From: <sip:[email protected]>;tag=df76db09-0244-47ce-beb6-160c60f593f0
CSeq: 31834 OPTIONS
Date: Mon, 20 Oct 2025 20:20:26 GMT
Content-Length: 0

[buba]

I should also mention that this happens intermitently, sometimes it’ll say all circuits are busy sometimes it will connnect and sometimes it’ll ring once or twice and say all circuits are busy

[david55]

I think the 407 response is the problem. See RFC 2617: HTTP Authentication: Basic and Digest Access Authentication which I think says that the WWW-Authenticate header would be mandatory, in this case. Certainly Asterisk is not going to be able to honour it without that.

However, only getting this for OPTIONS should not be a problem, as all Asterisk requires is some response.

Also, the presence of the 192.168… address means that you haven’t correctly configured your external address and/or local networks. This should be the public address of your NAT router.

[buba]

In the place of my public ip i put x.x.x.x, so you’re saying i should put my public ip in the network settings in freepbx?

[david55]

If you are inside NAT, and the provider is outside of NAT (the most common case, I would suggest), and noting that the following page is from an older version of FreePBX, in the following, External Address should be your address as seen from outside NAT (typically a public address, although some providers may use shared addresses), Local Networks should cover the FreePBX machine, and all the terminals on your LAN and any VPN.

Sangoma Documentation

[buba]

I did that to no avail.

[karimeusz]

I assume you’re writing from Poland.
If Neofon offers a free trial period for a few days, I could try testing it out.