So I have basic internal phones working; I have signed up for a trial with Sipgate UK … is there any documentation or guidance of the steps I need to take to now configure for use for external outgoing/incoming calls?
I have configured my router firewall to allow sipgate through the firewall; fixing it to their ip address …
Thanks
There are a couple of bits of info that might make your implementation easier.
Inbound calling and outbound calling are largely unrelated. People often conflate them because, well this is a phone system.
Lock port 5060 (inbound) down to people that will be connecting to your server. This list is short - any outside phones you might want to set up (don’t yet - KISS) and SIPGate (since they will be calling you). If you don’t have outside-the-network phones, the opening to SIPGate is enough.
Make sure you have your inbound set up to accept any number from SIPGate. You can set up specific inbound numbers on your inbound routes, but always leave an “Any/Any” route in case your provider changes what they are sending to your system (prefixes, number of digits, etc.) It also gives you a solid indication of what SIPGate is actually sending you for DIDs.
When setting up the outbound route, find out if you need to match your outbound CID to what they are expecting. If they care about “foreign” Caller ID (some do, some don’t) they will silently block your calls if the CID they are expecting is different than the one you send. The easiest way to make sure this is right is to ask them. They should be able to tell you what they expect.
@lgaetz has a list of ports that need to be opened on your border firewall for certain services to run.
Do not abandon the integrated firewall. I understand that it’s like running your server in a condom, but until you are familiar with your partners, it’s always a good idea. Since you don’t have “mobile” phone users, you don’t need the Adaptive Firewall - you can leave that turned off.
You may need to port forward some ports from the border router to your phone system. This is one of the places that FreePBX gets tricky, since setting up NAT for SIP and RTP is sometimes counter-intuitive. Once you understand what’s happening, it’s simple as falling off a log. Until then, it’s like getting hit by one.
Try stuff - you’ll learn more from mistakes than you will from successes. If you get frustrated ask - we cover every major timezone in the world, so you should be able to get some kind of answer in short order.