For 3CX, FreePBX, Asterisk, Yeastar and SIP apps

SIP trunk configuration is four settings and one rule.

Every phone system asks for the same things under different names: a server, a username and a password to register the trunk, and an outbound rule that sends numbers in + format. Pick your system and this page speaks its language.

Which phone system do you run?

3CX

FreePBX

Asterisk

Yeastar

SIP app

One call, start to finishSIP

Someone dials
020 7946 0018, the way they always have.
The outbound rule rewrites it
+44 20 7946 0018: the 0 goes, the country code comes.
The trunk sends it
Registered with your server, username and password.
LinkedSIP connects it
At the published rate for that number, from your credit.

What you need before SIP trunk setup

Three things to have open before you touch the PBX.

IDYour SIP login
Server, username and password. We set each login up by hand and message you when it is in your account, under SIP logins.
$20Some credit
A $20 top-up is enough to test. Calls are held against your balance before they ring, so a test call needs credit. What calls cost.
PBXAdmin access to your system
Any PBX, desk phone or app that registers with a username and password. Grandstream UCM, FusionPBX and Issabel take the same values.

SIP trunk settings

Eight settings. Five names for each. One value.

The labels change from system to system; what you type never does. Your system's column is lit once you pick it above.

SIP trunk settings and what each phone system calls them
Setting	What to type	3CX	FreePBX	Asterisk	Yeastar	SIP app
Server	The server shown with your login, exactly. No http:// and no path.	Registrar/Server/Gateway Hostname or IP	SIP Server, SIP Server Port	server_uri and client_uri (registration), contact (aor)	Hostname/IP, Port, Domain	Domain or SIP server
Username	Your SIP login username.	Authentication ID (aka SIP User ID)	Username	username (auth)	Username	Username or User ID
Auth username	The same username, unless your login shows a different one. Check this first when registration keeps failing with 401.	Same field as the username	Auth username	username (auth)	Authentication Name	Auth username or Login
Password	Shown with your login. Type it if pasting adds a space.	Authentication Password	Secret	password (auth)	Password	Password
Transport and port	As your login shows: UDP, TCP or TLS, and its port.	From the template; the port goes with the server	Transport	transport, and the port in server_uri	Transport, Port	Transport
Outbound proxy	Empty, unless your login shows one.	Outbound Proxy	Outbound Proxy (Advanced tab)	outbound_proxy	Enable Outbound Proxy	Outbound proxy
Codecs	The ones your login lists. G.711 (ulaw, alaw) first if you are unsure.	The trunk's codec list	Codecs tab	allow= on the endpoint	Codec Setting	Audio codecs
Caller ID	A number on your account, in + format. Leave it as your login shows if you have none.	Main Trunk No	Outbound CallerID	CALLERID(num) in the dialplan	Outbound Caller ID	Set with your login

Labels as of October 2026: 3CX V20 manual and provider guides, FreePBX chan_pjsip, Asterisk pjsip.conf, Yeastar P-Series. Versions rename things; if a label differs, the setting is the same.

Outbound dial rule for international calls

Your dial rule, built for the country your phones are in.

LinkedSIP takes numbers in + or 00 format and refuses anything else rather than guess. These rules turn the way your staff already dial into that, in your system's own fields.

Where your phones are

Your phone system

Try a number the way your staff dial it

Sends+447400123456

Rule 3 takes it. LinkedSIP accepts it.

Outbound rules

1Already in + formatCalls to numbers starting with prefix: +Calls to numbers with a length of: emptyStrip Digits: 0Prepend: empty
2Abroad, dialled 00Calls to numbers starting with prefix: 00Calls to numbers with a length of: emptyStrip Digits: 0Prepend: empty
3National, dialled 0Calls to numbers starting with prefix: 0Calls to numbers with a length of: 8,10,11Strip Digits: 1Prepend: +44

Where: Outbound Rules, Add. New rules go last, so select this one and press Move Up.

Built from Google's libphonenumber data for each country: its trunk prefix, its prefix for calling abroad and its number lengths. To keep local calls on your current carrier and send only calls abroad here, use just the first two rules: the second trunk setup.

How to check a SIP trunk

Three checks before anyone else picks up a phone.

It registers
Your system shows a registration status on the trunk or account: registered, online, or a green dot.
If it does not, the code in the log points at the setting. See the table below.
A real call, past 40 seconds
Call your own mobile. Talk both ways, stay on longer than 40 seconds, then call a voicemail menu and press a key.
This catches one-way audio, the 30-second drop and keypad problems in one call.
It shows in your account
Open Calls in your account. The call is there with its talk time, the seconds billed and the charge.
A call missing here never reached LinkedSIP: look at your outbound rule.

401

SIP registration failed and other errors

When it fails, the symptom names the setting.

Most failed setups come down to the same handful of settings. Find what you see, change one thing, test again.

SIP trunk problems, their usual cause and the fix
What you see	Code in the log	Usual cause	What to change
Registration fails again and again	401, 407	Username, auth username or password	Put the SIP username in the auth username field too, and type the password by hand.
Registration refused outright	403	Wrong server for this login, or the login is switched off	Copy the server exactly as your login shows it. Check the login is active in your account.
Nothing answers the registration	408, timeout	Server, port or transport	Match the transport and port to your login. A firewall that blocks outgoing SIP looks the same.
Calls fail at once, registration is fine	404, 484	The number is not in + or 00 format	Add the dial rule below, above your other rules, and dial the number again.
Calls fail when answered or before ringing	488, 606	No shared codec	Offer G.711 (ulaw and alaw) on the trunk, alongside anything else.
Calls connect, but one side hears nothing	no code	NAT: the PBX sends its private address	Turn SIP ALG off on the router. Set the PBX's public address and local networks.
Calls drop after about 30 seconds	no code	The answer never reaches the PBX, so the call is torn down	Usually the same NAT problem. Turn SIP ALG off and check the PBX's public address.
Keypad presses do nothing in phone menus	no code	DTMF mode	Set DTMF to RFC 2833 (RFC 4733) on the trunk and on the phones.

Still stuck? Email [email protected] with your system's name and the code from its log, and we will tell you which setting to change.

FAQ

SIP trunk setup questions

How do I configure a SIP trunk?

Add a trunk with the server, username and password from your SIP login, set the auth username to the same username, offer G.711, then add an outbound rule that sends numbers in + format. Check that it registers, and make a test call.

How do I check that a SIP trunk is working?

Look for a registered status on the trunk, then call your own mobile and talk both ways for more than 40 seconds. On LinkedSIP the call then shows under Calls in your account, with its cost.

How many SIP trunk channels do I need?

As many as the calls you make at once in your busiest hour. Each LinkedSIP login has its own limit on calls at once: it starts at 1 and can be raised to 100 on request, with no charge per channel. Size it from your busiest hour.

How do I configure a SIP phone?

A SIP desk phone takes the same values on its account page: server, username, password. Most offices register phones to their PBX as extensions and give only the PBX the trunk. A single phone can also register to the trunk directly, like an app.

Do I need a static IP address?

No. LinkedSIP logins register with a username and password, so your PBX can sit behind an ordinary office or home router. If your login lists allowed IP addresses, calls work only from those.

Do I need to open ports on my firewall?

Usually not for calls out: a registered trunk is a connection your PBX opens. Turn SIP ALG off on the router (how to disable SIP ALG), and make sure the firewall lets out the audio ports your PBX uses.

Which codecs should I use?

The ones your login lists, with G.711 (ulaw and alaw) first if you are unsure. Almost every SIP system supports it. Calls cost the same whatever the codec.

My phone system is not listed. Will it work?

Yes, if it can register a SIP trunk or account with a username and password. Grandstream UCM, FusionPBX, Issabel and most desk phones use the same settings under their own names.

Can I make emergency calls?

No. Emergency calls are not supported. Keep a normal phone line or a mobile for emergency numbers, and keep your current carrier for them if your PBX has one.

Four settings, one rule. Then the first call.

Open an account with an email. We set up your SIP login, you add $20 of credit, and the settings above go in once.

Get a SIP login What calls cost