Inbound call only busy signal

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rogergodefroy

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Hi,

I have tested the hosted solution with cm.com as the sip-trunk, which all worked perfectly. Now I have setup a new VPS using the guide on:

https://computingforgeeks.com/how-to-install-and-configure-3cx-pbx-on-debian/

As my VPS provider doesn't allow me to use the out-of-the-box ISO from 3CX.

It installed all perfectly and added all network interfaces and firewall rules during setup (as far as I could see), but on the new server all inbound calls are getting the busy signal when I want to test if all is working. I ran the firewall-test on the dashboard and all turned out green. Also I checked if port 5060 is available from the outside and I made an inbound rule to go to an extension (which is available) and also tried to forward the call to the digital receptionist. All without success, because I still get the busy signal.

I also changed the IP on cm.com to my new static IP-address.

Am I missing something why I constantly get the busy signal?

(PS. outbound calls are working)
 
Run a wireshark from the PBX, perform a test, check if call is hitting 3CX.

If no, speak to provider and check your firewall again.

Also, provision a 3CX mobile client and perform an audio test dialling *777
 
Run a wireshark from the PBX, perform a test, check if call is hitting 3CX.

If no, speak to provider and check your firewall again.

Also, provision a 3CX mobile client and perform an audio test dialling *777
As this is debian with no GUI installed, it's quite hard to run wireshark on the PBX VPS. I tried with tshark, but don't see any external traffic incoming which I could match with the incoming call.

The audio test is working with a problem.
 
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Did you try the 3CX Activity Log, to see if there is a log (Invite) of an incoming call? If there is not, then , as mentioned earlier, you might want to talk to your provider, as they may be able to quickly provide an explanation of what is happening, rather than trying to guess.
 
I didn't find any activity log, so I was looking into the sip provider. Somehow they have separate routing for inbound number and sip trunk. After changing the URL of the distribution group endpoint of CM.com calls are incoming.
 
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