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Softphone Integration
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Ring group failover when SIP endpoints unreachable
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How to setup Queues like a ring group?
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Queuemetrics transferred calls
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Want to setup complete PBX solution linked with Website
But where to start ?
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FreePBX to FreePBX SIP URI calling
@BlazeStudios, I realize that all calls leaving the PBX are in a full SIP URI format, but up until now all calls (on PBXs we’ve setup) are required to traverse a PSTN trunk. What I’m trying to achieve is to allow our client’s to call us directly without having to tie up a PSTN trunk.
Let me illustrate the challenge with a bit more detail:
We are a decent sized MSP/IT consulting firm. We have several hundred client organizations that we manage, of which we have about 50 clients that we provide high-level HelpDesk services for - meaning the employees at these organizations will pickup the phone and call us at any time, for any reason. About 30/40 of those HelpDesk clients have FreePBX deployments that we manage. So at any time our PBX could be handling calls to several of these 30/40 HelpDesk clients at once, and occupying space on our PSTN trunks (for which there is a limited capacity and we pay per minute for additional connections above our “limit”). My thinking is that we can minimize our PSTN trunk costs by giving these HelpDesk clients’ PBX’s a “direct connection” to our phone system without having to go out on the PSTN and come back in on the PSTN, effectively saving 2 PSTN trunk channels.
The goal is for our clients to dial “00” for the IT HelpDesk and they would automatically be put into a priority queue on our end that gets them to the department they are looking for fast and efficiently - without tying up 2 PSTN channels.
That’s the end goal. How it get’s done is what I’m trying to figure out. Setting up a trunk in our PBX with a shared secret, etc isn’t a problem. What I’m trying to avoid is setting up a dedicated PBX <–> PBX trunk in our system solely for the use of each of our clients (30/40 dedicated trunks). Right?
Thoughts?
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Extension unable to call other extension
@Stewart1 here I think this is better
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Extensions in queue ringing for 1 second
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Add external link to menu
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Connected line update to ...... prevented
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FreePBX to FreePBX SIP URI calling
PBX A wants to call PBX B and vice versa, they need trunks on each of them for the other. You have 40 PBXes you want to do this to and have all 40 call each other directly, you need to peer them all together.
As pointed out, PJSIP will let you match against multiple IPs/subnets on one trunk. So you need at least one trunk per PBX that has all the IPs/subnets it should accept calls for. Then you can Dial() in full SIP URI format so you don’t have to create 40 outbound endpoints. It just means you either have 40 Dial() lines or you are storing those 40 URIs in someplace like AstDB.
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Want to setup complete PBX solution linked with Website
Hire some software engineers and some SIP engineers that know one of the popular SIP switches (i.e. Asterisk, FreeSwitch, Kamailio, etc.) and then a some accounting/legal folks who know the billing / taxation / regulatory requirements for the jurisdictions you’ll service (if you’ll have customers in the US, this will be especially fun), and some UX designers, marketing people, and get some capital to make it all happen.
Where to start? Maybe a business plan for all the above? Or just start coding and see what you can come up with. I doubt you can build what you described using purely off the shelf stuff.
Good luck!
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FreePBX to FreePBX SIP URI calling
No, that’s my point. They don’t all need to call each other. It’s a one-way hub-and-spoke. They all need to be able to call us ONLY. We don’t even need to call them. If we need to call them we can place an outbound call on the PSTN, that happens RARELY. We receive inbound calls constantly. I just want client PBXs to be able to call into our PBX directly.
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Zulu login prompt when accessing UCP
FreePBX: v14.0.11
Asterisk: v13.22.0
PBX Firmware: 12.7.6-1904-1.sng7
PBX Service Pack: 1.0.0.0
Zulu module: 14.0.56.16 (not licensed)
We had been on a trial of Zulu, but that has ended now. I have tried to turn off Zulu where I can, but when I login to the UCP I am getting this prompt for Zulu credentials:
I have tried setting the Zulu permissions in UCP to “No”. My FreePBX user account has this set to “Inherit” under Admin -> User Management -> my account -> UCP -> Zulu. Then, the groups I am a member of have this Zulu option set to “No”. I also just tried setting UCP -> Zulu to “No” for my account, and I still see this prompt.
The odd thing is… if I’m logged out of the UCP, I do not see the Zulu credential prompt on first login. It’s only when I’m already logged in and I open a new tab directly to the UCP or click the UCP link from the admin console – then I see the Zulu prompt. I’ve tried this in a private Firefox window with the same results.
Does this seem like a configuration error on my part, or is this potentially a bug?
Is it best to just disable this module since we don’t use it? I still like to keep all modules updated to maintain compatibility if we ever decide to license one of the commercial modules.
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FreePBX to FreePBX SIP URI calling
If the help-desk is in one location, you dont need a mesh of trunks , you need one trunk at the help desk location, all the other pbx’s need a trunk to that central location.they could be identically provisioned.
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Zulu login prompt when accessing UCP
I believe stopping the service and disabling the module will not cause the module to stop receiving updates, I think you have to remove it.
I’d try removing Zulu from pm2 which should also stop it:
fwconsole pm2 --delete zulu
You can manually stop it if you want to be sure:
fwconsole stop zulu
And if it persists, disable the module
fwconsole ma disable zulu
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Limit simultaneous calls via group count of incoming and outgoing call totals
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S705 Provision over wifi
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FreePBX to FreePBX SIP URI calling
My thoughts exactly…
But how do we do this? This level of programming is beyond my ability. Is there a tutorial out there that describes how to set this sort of thing up?
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FreePBX to FreePBX SIP URI calling
The proper way to do this is by creating a peer for each IP from which you need to receive an INVITE.
I can think of another way to do this, but I’m reluctant to put it in writing due to the security implications. You could enable “Allow SIP Guests” (but not “Allow Anonymous Inbound SIP Calls”) in Asterisk SIP settings, and set the guest context to custom dialplan you control. Then create the dialplan to look for SIP invites to a random string that is highly unlikely to be guessable like lFuAs9HhgHeY8tq2. Send all other calls to hangup. You would not want to do this using normal SIP ports because spurious invites will no longer trigger any bans, but using a high random SIP port would probably be safe. Then SIP calls to lFuAs9HhgHeY8tq2@ipaddress:highrandomport can be sent to an inbound route.
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