3CX to a FreePBX

[RageX]

Hi,

My current VOIP provider uses 3CX, and I'm trying to get my FreePBX setup to connect properly.

I have no issues connecting to my VOIP provider using softphones and deskphones with the usual settings. However, I've not been able to get the FreePBX to successfully connect to it despite being able to connect to many other VOIP providers that are not using 3CX.

I have had a look at the guide here at https://web.archive.org/web/20161213151353/http://www.3cx.com/docs/bridging-asterisk-pbx/ but I'm not sure that its applicable to my use case, since I probably have no way of getting a special bridge set up. I've also tried those settings and I'm not able to register, in additional to many other variations as suggested by others. The current error code that I have is:

WARNING[1898] chan_sip.c: Got 404 Not found on SIP register to service [user]@[domain], giving up

In terms of setting everything up, the quirk here is that apparently I have a different "username" and "authentication name". The username is 5 numbers, while the authentication name is different. Does anyone know the correct settings to input into FreePBX to get everything to work?

In particular the settings for outgoing PEER Details and incoming USER Context/USER Details and Register String.

Thank you!

 

[cobaltit]

I'm hoping you are also asking on the Asterisk/FreePBX forums since your asking for Asterisk/FreePBX settings.

 

[RageX]

Thanks for the response. Their forums are currently down for me but I'll try when they are up again.

From what I've seen though it appears that there are certain quirks in 3CX setups such that the usual FreePBX/Asterisk configuration settings do not work properly. Was hoping that someone here with experience connecting a 3CX setup to FreePBX/Asterisk could help with the settings.

 

[leejor]

Am I to understand that your provider uses 3CX to provide service to you? You actually register to them? Or...am I missing something.

 

[RageX]

The provider's backend uses 3CX. But from my perspective it is just another SIP line, with the usual configuration details. So yes I register to them.

Registration works fine using softphones and VOIP deskphones.

However it appears that because of the 3CX backend, the usual settings with FreePBX/Asterisk do not work and I'm not able to get registered with my PBX.

As background I use two VOIP providers. I was able to get the first VOIP provider connected to my FreePBX/Asterisk with the usual settings. Its the one with the 3CX backend that's giving me problems.

 

[leejor]

So you are attempting to register a FreePBX trunk to a 3CX extension, and not a 3CX Bridge trunk.

 

[RageX]

I'm not certain but it is possible that you may be correct.

In my country these VOIP products are known as "SIP accounts" that come with their own phone numbers. Both the ones I have cost about the same as well, but one is able to connect to FreePBX and the other (3CX one) is not.

The configuration information that I'm provided is the Authentication Name, Authentication Password, Username, Display name, Domain, Proxy Server Address, Registration Server Port, STUN server address, STUN server port.

Since they have also provided STUN settings it appears to me that they are setup to connect directly to PBXs, so I'm not sure whether its considered an "extension" or a "trunk".

 

[leejor]

If what you are saying is, in fact, correct...a provider is offering services using 3CX, then it is the first account that I can recall. One thing that would cause me to wonder, is that the phone number (extension) digit length, is normally limited in 3CX. In other words, 3CX normally cannot offer phone numbers that correspond in length to (mimic) standard PSTN numbers, unless there was some sort of software "modification".

 

[RageX]

I'm not certain on the specifics but the provider is relatively large. In addition to the various configuration and login details that they have provided me (which successfully connects to my softphone and deskphone), I am also able to login to the 3CX dashboard separately to make calls.

Based on this information do you now the possible settings that I may use to connect my FreePBX/Asterisk setup to them? So far no matter what settings and permutations I've tried the error that I get is:

WARNING[1898] chan_sip.c: Got 404 Not found on SIP register to service [user]@[domain], giving up

The same settings on a softphone/deskphone successfully register fine.

 

[leejor]

I would speak to the provider. They should be able to tell you what it is, that you are sending, that they do not like. It could be something to do with the UserAgent you are sending, if all other parameters are correct.

http://forums.asterisk.org/viewtopic.php?f=1&t=68686

 

[TBWD]

I would find a different VOIP provider, 3cx is a PBX, not a SIP Trunk server.