The number you have dialed in not in service. Please check the number and try again

[ageorgescu]

I have outbound and extension2extension calls working but I can’t figure out what I’m doing wrong for inbound calls.

When calling there’s a female voice saying: The number you have dialed in not in service. Please check the number and try again.

This is the trunk configuration (replaced characters with “x” for security reasons.
This trunk configuration worked for the old PBX configuration that I am replacing with FreePBX:
host=xxx.xxx.xxx.xxx
username=372xxxxxx
secret=aqiHuxxxxx
type=friend&friend
fromuser=372xxxxxx
insecure=port,invite
qualify=yes
canreinvite=no
dtmfmode=inband
fromdomain=sip.xxx.xxx
disallow=all
allow=alaw

The Inbound Route has the DID number: 372xxxxx and Set Destination to a Specific Extension.

Basically I have configured extensions, trunk, outbound route, inbound route and firewall.

Is there anything else I need to do?

[david55]

You probably haven’t registered with the provider.

[ageorgescu]

Thanks for the answer. Eventually I found out that I needed to select Allow anonymous inbound SIP Calls from Asterisk SIP Settings.

[david55]

That’s a very good reason for moving to chan_pjsip as you can specify whole networks as the possible source address, and therefore don’t need to open up your system to the hackers. Make sure you compensate by setting firewall rules that only allow traffic from the service provider, and, if they allow it change local SIP port number to something unpredictable.

[ageorgescu]

Under Networks I have:

• Internal networks listed as Local (local trusted traffic);
• ISP IP as Internet (Default Firewall).

Interfaces:
eth0 and eth1 - Internet (Default Firewall).

Responsive Firewall and Intrusion Detection are both on.

Do you recommend adding anything else?

[david55]

Restrict port 5060 to just the address ranges used by the ITSP, (If possible change 5060 and apply the restriction to the new port number.)