3CX Initial Install

[Truckin]

Hey all,

I have recently installed and set up my 3CX PBX for my small business on a PI 4. I have a SIP Trunk provider and entered all the information as the 3CX help guides stated for SIP Trunking. Once everything was done and configured I called the office number with my cell and the cordless phones (Didn't have the desk phones configured yet) rang just fine on the Grandstream HT802 and once answered you can hear on both ends. However, whenever I tried to call out the call would ring but once answered it would end abruptly and would beep like when you call someone and the line is busy. I went over the outbound rules and pretty much just copied the configuration of the 3CX outbound guide just changing the items to match my info. So I am trying to figure out why the call drops when going outbound? I'm also having trouble with the CID on Inbound calls...if someone calls the office it will say the "SIP TRUNK Providers name" then the CID of the number calling itself.....is there a way to remove the SIP Trunk providers name?? Lastly, is there a way to change the Outbound CID that shows up from my numbers? I want it to state my business name not the number..? Any help would be greatly appreciated!
Thanks,
Truckin

 

[ChrisC_3CX]

However, whenever I tried to call out the call would ring but once answered it would end abruptly and would beep like when you call someone and the line is busy

• Is this happening for all calls?
• Also, how long after it has been answered does it drop?
• If it drops straight away before you can even speak with the external party, how can you tell that the call has been answered?
• Are you using a 3CX Supported SIP Provider? What SIP Trunk are you using?

What I would recommend you do first is to check if you can reproduce the issue by calling one of your mobile phones so that you are in control of both the caller and the callee. If you can, try making the call from a 3CX Webclient to eliminate the possibility of this having anything to do with the Grandstream FXS gateway. Do you still get the same issue?

I'm also having trouble with the CID on Inbound calls...if someone calls the office it will say the "SIP TRUNK Providers name" then the CID of the number calling itself.....is there a way to remove the SIP Trunk providers name?

Whether this is possible and how it can be done ,will depend on a few things. First you'd need to figure out who is actually adding the name to the Caller ID. If the Inbound Rule configured in 3CX has been named after the SIP Trunk Provider's name, then it might actually be 3CX adding it. If this is the case, just go to Contacts >> Options and where it says "Add Group, Queue or DID/DDI Names to Caller ID" just select "Do not Add".

Lastly, is there a way to change the Outbound CID that shows up from my numbers? I want it to state my business name not the number..?

This will mostly depend on what your SIP Provider allows. You could set the Outbound Caller ID in 3CX to be a name, but whether your SIP Provider will accept it and forward it as is we cannot know. You might want to check this with them. If it is a 3CX Supported SIP Provider you might want to check for the "Clip No Screening" capability which if they support, might mean you'll be able to do this.

 

[Truckin]

• Is this happening for all calls?
• Also, how long after it has been answered does it drop?
• If it drops straight away before you can even speak with the external party, how can you tell that the call has been answered?
• Are you using a 3CX Supported SIP Provider? What SIP Trunk are you using?

It happened with every call I made last night yes.
Hmm, maybe 1-3 seconds?? That's a guess I really didn't time it.
When I called my cell I could hear me pickup the line and it connect then drop.
Yes, I am using a 3CX supported SIP Provider. "DIDLOGIC"

What I would recommend you do first is to check if you can reproduce the issue by calling one of your mobile phones so that you are in control of both the caller and the callee. If you can, try making the call from a 3CX Webclient to eliminate the possibility of this having anything to do with the Grandstream FXS gateway. Do you still get the same issue?

I will try that this evening for sure, but I don't have the Webclient installed. I can try to call my cell again using one of the desktop phones we have installed instead so it doesn't go through the FXS gateway.

Whether this is possible and how it can be done ,will depend on a few things. First you'd need to figure out who is actually adding the name to the Caller ID. If the Inbound Rule configured in 3CX has been named after the SIP Trunk Provider's name, then it might actually be 3CX adding it. If this is the case, just go to Contacts >> Options and where it says "Add Group, Queue or DID/DDI Names to Caller ID" just select "Do not Add".

I think I did name it after the SIP Trunk Provider as I thought that's what the instructions stated....I probably misunderstood them. I will verify and if it is DIDLOGIC I will attempt the recommended solution to correct.

This will mostly depend on what your SIP Provider allows. You could set the Outbound Caller ID in 3CX to be a name, but whether your SIP Provider will accept it and forward it as is we cannot know. You might want to check this with them. If it is a 3CX Supported SIP Provider you might want to check for the "Clip No Screening" capability which if they support, might mean you'll be able to do this.

Ok, I will check with DIDLOGIC and see if they will allow this and also check this setting you mentioned.

Truckin

 

[ChrisC_3CX]

When I called my cell I could hear me pickup the line and it connect then drop.
Yes, I am using a 3CX supported SIP Provider. "DIDLOGIC"

In that case, the call must be answered meaning one of the two ends is ending the call. To determine which of the two it is you might need to check the Activity Log and then find who sent the BYE SIP message. You would need to enable verbose logging under Dashboard >> Activity Log >> Settings. Then, reproduce the issue and check the activity Log. You can use the "Filter" button to filter which call you are looking at which will make it easier to trace.

I will try that this evening for sure, but I don't have the Webclient installed. I can try to call my cell again using one of the desktop phones we have installed instead so it doesn't go through the FXS gateway.

That is the beauty of the webclient! It does not need to be installed, all you need to do is launch a web browser such as Google Chrome and go to https://FQDN.3CX.com:5001/webclient. You will need the extension number and it's "Web Authentication" password which can be found in Management Console >> Extensions >> Edit >> General. As you mentioned, you could indeed just use an IP Phone but the webclient will allow you to do it from anywhere. Alternatively you could also use the 3CX Mobile clients (Android/iOS).

Ok, I will check with DIDLOGIC and see if they will allow this and also check this setting you mentioned.

Based on the info we have for our Supported Providers, it seems they do not. I guess it wouldn't hurt asking though:

 

[Truckin]

In that case, the call must be answered meaning one of the two ends is ending the call. To determine which of the two it is you might need to check the Activity Log and then find who sent the BYE SIP message. You would need to enable verbose logging under Dashboard >> Activity Log >> Settings. Then, reproduce the issue and check the activity Log. You can use the "Filter" button to filter which call you are looking at which will make it easier to trace.


That is the beauty of the webclient! It does not need to be installed, all you need to do is launch a web browser such as Google Chrome and go to https://FQDN.3CX.com:5001/webclient. You will need the extension number and it's "Web Authentication" password which can be found in Management Console >> Extensions >> Edit >> General. As you mentioned, you could indeed just use an IP Phone but the webclient will allow you to do it from anywhere. Alternatively you could also use the 3CX Mobile clients (Android/iOS).


Based on the info we have for our Supported Providers, it seems they do not. I guess it wouldn't hurt asking though:
View attachment 22831

Chris,

Thank you for getting back to me. I will try those recommendations for sure when I get home this evening and see if it can be determined who is ending the call. Regarding DIDLOGIC that is disappointing, but we really didn't know what that meant so now that we do it may be time to look at another provider. 3CX's list has some SIP providers that aren't really easy to work with so I think you might want to look at expanding that list as we have tried to go through a couple of them, but it hasn't worked out.

 

[ChrisC_3CX]

I will try those recommendations for sure when I get home this evening and see if it can be determined who is ending the call

Do let us know what happens!

...so I think you might want to look at expanding that list...

I can assure you that we're always working on it!

 

[Truckin]

Thank you! The system seems to be working normally now. My next question would be is it possible to add my Audiocodes HTTPS fax adapter so that I can fax using my 2nd phone number? I have the mac address of the Audiocodes HTTPS fax adapter.....

 

[Truckin]

Bump for response/help......

 

[YiannisH_3CX]

We are not familiar with the Audiocodes HTTPS fax adapter as it is not a supported device so we cannot assist with the configuration but from a short search it seems that it connects directly with the providers fax to email service (if your provider supports this) so this is something that needs to be configured from the providers side and will bypass the PBX completely. You can also use the 3CX fax server and receive fax to email directly from 3CX without the need of a 3rd party device.

 

[Truckin]

Thank you for the response. We really would like to use our faxes the "Traditional" way via faxing with the machine and receiving via the fax machine....that's why we looked into the HTTPS fax adapter. I will check with our SIP provider and see if it's something they can use.
Truckin

 

[SweetAction]

If you want fax via a physical machine I would recommend grabbing a supported fxs device like a Grandstream HT801. It's sub $50 USD and can be added as a fax extension.

More info here:https://www.3cx.com/voip-gateways/grandstream-ht-fxs/

Fax over ATA can be very hit or miss. But it's an option if you want to.

 

[Truckin]

Ok thanks for the suggestion.