Connecting a trunk to freepbx

[Magicaldj]

Hello,

I am trying to connect my newly rented phone number to freepbx

The number is bought from SEWAN Belgium (3starsnet)

The innfos that i have are those:
Domein : voip.3starsnet.com (IP=188.66.8.19)è UDP port 6060
Phone number :04NNNNNNN
Sip server : 188.66.8.19 or voip.3starsnet.com
Sip username :04NNNNNNN
Sip password : PPPPPPPPPPPP

I went to Connectivity -> Trunks -> Add ChanSip Trunk:

Gave it a name, and an Outbound CID <04NNNNNNN>
Here is my SIP Settings tab:

host=188.66.8.19
port=6060 (because the provider told me to use it)
username=04NNNNNNN
secret=PPPPPPPPPPPP
type=peer

Then i went to Connectivity -> Inbound Route -> Add inbound route
And here are the settings of the inbound route:
DID Number: ANY
CID Number: ANY
Set destination: IVR

And finally, i went to Connectivity -> Outbound Route -> Add outbound route
Here are the settings:
Trunk sequence: My SEWAN TRUNK

Dial Patterns: well… i don’t know exacly what to set there because i want to be able to call numbers as:

For example 0471462362 to stay in my country (Belgium)
And be able to call numbers from other countries by adding +“country prefix” and the number like so:

0471462362 would call a cellphone in belgium
+336254236 would call a cellphone in france
+15552563 would call a phone in the US

THOSE NUMBERS ARE RANDOM BTW !

And so far the only thing that i hear while calling from or to the number are error messages…

So if someone knows a deep lost forum somewhere where details on configuring a 3starsnet trunk with freepbx or if someone knows what to modify it would be great

Have a nice day !

[Stewart1 (Stewart)]

I am assuming that you have a new FreePBX 15 system and that calls from one extension to another work properly. If not, please provide details.

I recommend that you delete the chan_sip trunk and create a pjsip trunk. Leave all settings not mentioned below at their default values.

Trunk Name: 3starsnet
Outbound CallerID: 04NNNNNNN
Username: 04NNNNNNN
Secret: PPPPPPPPPPPP
SIP Server: voip.3starsnet.com
SIP Server Port: 6060
Expiration: 120
From Domain: voip.3starsnet.com
From User: 04NNNNNNN

Also, for initial testing, set the destination for your ANY/ANY route to a working extension, as that is easier to debug.

After Apply Config, go to Reports -> Asterisk Info -> Registries and see whether the trunk status is Registered. If not, tell us about your system: On-site or cloud? Virtual or physical? Router/firewall? Trunk status?

If registration is ok, report results of attempted incoming and outgoing calls.

[Magicaldj]

Well, i assume i owe you a drink !

It works great
I can now call from and to my PBX using my number

The only thing left to do for me is to undertand concepts of interactive menus and how to create automations like “after 6PM redirect any inbound calls to the IVR” etc…
But be sure that if i need help again, i will come around here

Again a great thanks to you Stewart <3