How to become your own SIP trunk?

We have been building more and more VOIP phone systems for our clients and are looking for a way for us to increase our support. Is there a way for us to use a provider like Anveo Direct for the lines then have it connect to our phone system in our datacenter, then have our clients system connect through SIP to ours?

We’re wanting an easy way for us to take more control of the system and be able to monitor all sip trunk statuses and have more features. All VOIP providers are either focused on retail (single users) or large massive trunks (anveodirect) so we either have to create a separate account for each client and have decent features, or be able to use all in one but not have any features.

Consider VOIP Innovations - they are a wholesaler that allows for rebranding, so this might be a good middle-ground for you, without the expense of becoming (in effect) a telephony carrier.

You can do what you are talking about, but without LOTS of redundancy, you become a single point of failure - and then you incur the wrath of people who can never understand why anything breaks ever :slight_smile:!

Having said that, if you invest in the redundancy you need, you can indeed start routing calls through your own equipment - it’s not hard.

Have you looked at Vitelity? They have tremendous visibility of what all your customers are doing, you can set up your own sub accounts, and all the billing comes to you - then you send out the bills you want.

We like them VERY much!


We have a fully redundant datacenter with very low usage so we should be pretty good on that front. We’re slowly building API’s and other platforms to help manage our clients and automate many tasks and make monitoring easier.

We’re actually starting a project with Cloudco currently so we’ll see how their products work before looking at other vendors.

It sounds like you want to setup your own trunks to your customers equipment rather than connect each server to your upstream carriers? If thats the case you may want to look at at a sip proxy like opensips or kamailio. There is a large learning curve there, but you could start out on a prebuilt turnkey solution/distro that does the same, and uses opensips/kamailio like sipwise sip provider CE.