Sync inbound rules between two systems


(SilverHiro) #1

Hi All

What would be the best way to synchronize inbound rules between two systems?

The systems would be configured and running independently, I just want to make sure that inbound rules that are created on the main system get to the secondary system in the recommended way.

I’m happy to make a backup and restore that if that can be done just for inbound rules, but obviously want to make sure I’m following the best practices before this goes into production.

I am in the process of building the two servers as my up stream provider can set up a Primary destination and a secondary destination which would allow me to some additional redundancy should there be a problem with the Primary host.

Many Thanks!


(Itzik) #2

If you are looking for redundancy, why not get the whole system configuration on the secondary?

You can do so with setting up a warm spare: https://wiki.freepbx.org/display/F2/Warm+Spare+Setup

However, if you want ONLY the inbound routes, you’ll need to setup the warm spare backup to only include the inbound routes DB Table.


(SilverHiro) #3

I would definitely be down with trying this out, would I need any of the commercial modules for this or could I do it with what comes with the ISO?


(Itzik) #4

Depends what commercial modules you are using.

For example, if said client uses EPM, You’ll only need EPM on the Primary to configure the phones, and you can also set the secondary SIP server through EPM so as soon as the Primary goes out the phones will automatically connect to the secondary.


(SilverHiro) #5

I don’t use commercial modules so I should be OK then as the Warm Standby functionality is built in?


(Itzik) #6

Yes .


(SilverHiro) #7

Brilliant thanks very much!


(Tom Ray) #8

As per the other post you asked about things in, use an SBC or a SIP Proxy. Asterisk should not be used a switch.


(SilverHiro) #9

Oh yeah? How come?


(Tom Ray) #10

Because Asterisk is not a switch, it’s a B2BUA. They handle things differently. It also doesn’t give you the control or other features like load balancing, failover, etc… I’m not saying don’t use Asterisk because you’ll need it for media but you don’t need to do all the switching functions on it.


(SilverHiro) #11

Are there alternatives that can switch on telephone number based rules that would be a better alternative? Free unfortunately is the only budget I have.


(Tom Ray) #12

Free is never free. Don’t confuse free to use as free of cost because it’s not. There are a few open source SIP proxy options including Kamailio and OpenSIPs but you’ll need to learn them, configure them and test them. There is a cost to that.

There is even a cost to your current plan. So nothing is free. What are you actually doing with all this? It would help understand what options are best for you.


(Lorne Gaetz) #13

I wish you well but you are setting yourself up for failure.


(SilverHiro) #14

Training costs aren’t a problem, just product costs and recurring ones in particular.

I’ll check out OpenSIPs and Kamailio right away.

Basically I’m wanting to put a SIP Switch between myself and my vendor so I can control the destinations of each call.


(SilverHiro) #15

Thats pretty much why I wanted to come here and ask for advice/help before even starting down this path.


(Tom Ray) #16

And this is why you want something like Kamailio or OpenSIPs because they have numerous methods for handling this. They also allow you to build destination sets that you can cycle through based on the replies you get back. They just allow more control over the routing and the actual SIP message as you can modify/add headers at any point and made other adjustments.


(SilverHiro) #17

I had a look at OpenSIPs yesterday and I think I’m definitely going to get the Udemy training for it straight away this morning. Are there any other good Video training resources I should know of other than their own Bootcamp?