3CX Trunks

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Ryan Schroller

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Are there some trunks that just will not work with 3CX. I work for a small ISP. We have several FreePBX servers working with our SIP services. We are in the process of looking for another solution for our PBX deployments.

I have gone through setting up the unsupported trunk set up in 3CX, however there is no way to manipulate the outbound register string.

In FreepbX the to and from on the register are of OUR pbx, not the SIP trunk provider
in 3CX, the to and from on the register are of the SIP trunk provider, not the pbx
We just get 404 errors. This works fine in FreePBX. Could someone just provide a simple answer. Are we just out of luck?
 
If you provide the name of the unsupported provider, someone else may have experience with getting them to work on 3CX.
 
It is the ISP I work for. S & T Communications. They use a GenBand switch for SIP trunks.
 
They use a GenBand switch for SIP trunks.

This should be fine, I know of some big ITSP's using the Ribbon C20 ATCA/Softswitch solution without issues - I don't think that the platform matters, more the configuration - what is being sent.

As a rule of thumb 3CX always say that a "SIP Connect" Certified SIP trunk will work with the system I would check this out: https://www.sipforum.org/technology/sipconnect/
 
We have a local CLEC that has a Genband switch and have had virtually no issue getting SIP trunks connected. We use the Generic - Generic SIP Trunk setup. On the Genband we use the IP address validation - to secure the connection. They do establish a username and password - but that method doesn't seem to work. I often set that up, then switch it back to the IP based connection. Then no issues with in/out calls once rules are setup for INbound and outbound. Has been flawless over a year or so. I can probably priovide screen shots and printout on Genband side as well to show you a working configiuration.
 
Hello @Ryan Schroller

Without going in depth into the SIP RFC i do not remember seeing something that explicitly dictates what should be added in the host part of the From and the To SIP fields. Also the RFC in open to interpretation some times. I am also not an expert on this but as we have over 160 supported SIP providers (most of the are authentication based) including some of the biggest providers in the world that work this way then i do think the 3CX phone system is doing this wrong.
As you are looking into 3CX for your PBX deployments i would recommend applying for an interop test so that our dedicated team can test your trunk against the 3CX phone system.
You can start by applying here: https://www.3cx.com/partners/sip-trunks/voip-provider-interop-form/
 
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