Users can call extensions but not the main PSTN phone

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27310

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Just started working with this PBX system. The system is an instance of Elastix (v2.4.0-19)running on a VM on the main server. The office I'm in is about 40 miles away from another branch office. The users in each branch can dial each other by using their extension numbers, but when someone wants to call the main PSTN phone number of the other branch the calls do not go through.

Just started on this issue with this client and still gathering all the information I can about the setup. I am able to use ssh to get into the Elastix server but will be using the web interface.
 
Are calls to the other branch using an outgoing trunk?
If so, are any other outside calls affected?
If other calling other numbers works, then I'd first suspect an outbound rules issue.
If all users are on the same PBX, then it may have originally be set up so that calls to the other PSTN number be kept internal, to avoid provider charges. The main number could be converted to a destination in the outbound rules, that is having issues. I would first check the Activity Log (Verbose) to see how the calls route.
 
Hi @27310 Elastix its EOL since 2016, you will need to migrate that installation to FREE/SMB or a dedicated installation.

For more info please enter here to sign up.

With FREE/SMB you will have to use a Supported VoIP Provider to connect with the PSTN.
 
Are calls to the other branch using an outgoing trunk?
If so, are any other outside calls affected?
If other calling other numbers works, then I'd first suspect an outbound rules issue.
If all users are on the same PBX, then it may have originally be set up so that calls to the other PSTN number be kept internal, to avoid provider charges. The main number could be converted to a destination in the outbound rules, that is having issues. I would first check the Activity Log (Verbose) to see how the calls route.
leejor, thanks for your reply. Let me see if I can reply concisely on this:

Cannot answer your first point about calls to the other branch. Too little information for the moment but I'm working it.
All other calls to the PSTN work fine, in either branch. It's only when someone dials the direct PSTN number to the other branch that the calls drop. Any by drop, at this moment I simply mean nothing happens. No clicks, whistles or bells, only silience.
And I'm still waiting on information on the real setup here. It does appear that they are running Elastix in a VM instance on a local Windows 2022 server that is running in each branch.
It's been a long time since I configured Asterisk conf files but I think I can work through this. Could someone point me in the right direction on how to check the Activity Log. I have to be very careful because this is a live system and a typo could really mess things up.
 
leejor, thanks for your reply. Let me see if I can reply concisely on this:

Cannot answer your first point about calls to the other branch. Too little information for the moment but I'm working it.
All other calls to the PSTN work fine, in either branch. It's only when someone dials the direct PSTN number to the other branch that the calls drop. Any by drop, at this moment I simply mean nothing happens. No clicks, whistles or bells, only silience.
And I'm still waiting on information on the real setup here. It does appear that they are running Elastix in a VM instance on a local Windows 2022 server that is running in each branch.
It's been a long time since I configured Asterisk conf files but I think I can work through this. Could someone point me in the right direction on how to check the Activity Log. I have to be very careful because this is a live system and a typo could really mess things up.
Hi @27310 you are using a FREE Version installation, there is not any chance that you connect this elastix server to the FREE version since it is only supported voip providers.

This old elastix installation must be migrated to a hosted dedicated machine or using FREE/SMB according to their needs.
 
Thanks Alejandro. Sorry to be so clueless for the moment on this. The login I'm using does not have permissions to access the log files. But they are getting that taken care of.

To confirm, you're saying that Elastix v2.4.0-19 is a free version? The gentleman who set this up is no longer with us so it's hard to pin down the exact details. But I'm working on it.
 
To confirm, you're saying that Elastix v2.4.0-19 is a free version?
No, this is an End of Life system, you have to migrate this installation to a dedicated 3CX or using SMB/FREE version according to their needs.

Then in order to connect to the PSTN you will need to use a supported voip provider.

You may try to search a partner in your area that can assist your further in this migration
 
Tried again this morning to dial from my extension to the main PSTN number in the other branch office. The call rang twice then went silent for about 15 seconds then I got a busy signal. Checking the CDR report using the Elastix web interface I see the call.

Src. Channel = SIP/6305-0000000d
Dst. Channel = SIP/ATT1_OUT-0000000e
Status = ANSWERED
Duration = 10s

I'm not sure why the call status is shown as "ANSWERED". I will have to wait until Friday when another tech will be in that office to check on what's really happening there at the other branch.
 
Alejandro,

Thank you again for your replies on this. I need to ask for a little more clarification. And of course the lack of information on this end is abundant. The previous IT team are long gone, did not document anything and the client has little to no understanding of the issue. So that said, could you please explain with a little more detail on what exactly isn't working about this setup. One way or another they are going to replace their PBX system with a cloud based application. But they want to be fully informed of why.

So, without sounding clueless, which I mostly am at this stage, may I ask you the following questions:
  1. The term FREE/SMB is referring to a generic product by anyone that is available at no cost? BTW When I search "FREE/SMB" unless I put something like VOIP in the search string it brings up Super Mario Brothers... LOL
  2. Is there a product you're referring to "FREE/SMB" or is that just a generic term to refer to any such app?
  3. I like what I see with 3cx but we're not ready to make a decision yet.
  4. Providing SIP trunks is not a problem, but we want to replace this outdated Elastix system with a suitable product. 3cx is one of our options.
Sorry again to be such a newbie at this point but I can use all the simple advice I can get.
 
Honestly, you need a 3CX partner to help you here.

3CX has 4 SKUs - Free, SMB, Pro, Dedicated

Free - Hosted by 3CX, limited interface and options, free
SMB - Hosted by 3CX, more features but still limited interface and options, max 20 users, costs $
Pro - Hosted by 3CX or you, full interface and most options, licensed per sim call and not user, costs $
Enterprise - Hosted by 3CX or you, full interface and all options, licensed per sim call and not user, costs $

SIP trunks are restricted in Free and SMB

Quick feature list: https://www.3cx.com/ordering/pricing/
Full feature list: https://www.3cx.com/ordering/pricing/features/
Price List for Pro/Enterprise: https://www.3cx.com/ordering/pricing/pricelist/
 
Thanks. Now I see that FREE/SMB refers to two of the different versions of 3cx.
 
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