Lets talk multi-tenant

[netswork]

I have a few questions for anyone using the multi-tenant setup, mainly how to get dialtone to each tenant.

If you have a SIP trunk provider how are you trunking each tenant to the provider? Are you giving your server multiple IP addresses?

Is there a way to bring in a digital trunk like a PRI into a device and have it shared by multiple PBX's? I would like to host the customers systems in our data center and provide dial tone via PRI's. Just not sure if there is a device out there that will allow for multiple registrations.

 

[leejor]

Is there a way to bring in a digital trunk like a PRI into a device and have it shared by multiple PBX's?

My understanding is that each client, is completely separate right down to the port range used for each.
It depends a lot on what sort of PRI service you purchase from a provider. Did you plan on purchasing a PRI with a range, of say 100 DID, numbers, and then dividing them up amongst multiple customers?
In most cases (I'm assuming), each customer would have their own PRI (single or multiple T1's), their own numbers, going to one ISDN Gateway device, so one per customer.

That way the contract with the phone company/provider would be in their name. If they went out of business, or otherwise defaulted, before the contract expired, then it would be between them and the provider. You would just be the inter-connector, and not on the hook for any amount still owed.

 

[netswork]

My goal was to oversubscribe the PRI since here they only come in 12 or 24 channels.

Is there a device that will terminate multiple PRI's and allow me to connect it to multiple PBX's? It would be expensive to have to buy a separate patton for each customer.

 

[leejor]

I'm sure that there are ISDN gateways that will take multiple T1/E1's. What you want is one that can handle more than one trunk group, multiple D channels (and not just the second, back-up, D channel).

 

[netswork]

There are pattons that will take multiple PRI trunks, But I dont think you can register a single patton to multiple 3cx systems.

 

[new29]

I know this is a little late, but we've done this in a lab recently. We're using a Cisco gateway (theoretically a Patton would work the same way) and installed Kamailio to handle the registrations. This way you can scale a single gateway (or a cluster of them) to support any number of 3cx instances. Works flawlessly.

 

[netswork]

Very cool. I had never heard of that kamailio software. I will check that out.

Do you mind sharing your cisco gateway configuration that you were able to get working with 3cx?

 

[new29]

Yes, I believe if you want to do multi-tenant right, you should use a SIP proxy to pass the calls between the 3cx and the gateway. My Cisco config is still in "lab mode" and is subject to change. I may not need one of those dial-peers as I think an extra one got inserted when playing around and the PRI needs to be cleaned up. Here are the relative snippets of my Cisco config:

controller T1 1/0
framing esf
linecode b8zs
cablelength short 133
pri-group timeslots 1-24
!
interface Serial1/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn guard-timer 3000
isdn supp-service name calling
isdn outgoing ie facility
isdn outgoing ie caller-number
isdn outgoing ie called-number
fair-queue 64 256 0
no cdp enable
!
voice-port 1/0:23
no vad
bearer-cap Speech
!
dial-peer voice 1 pots
description Incoming from PSTN
preference 10
incoming called-number .T
direct-inward-dial
port 1/0:23
!
dial-peer voice 100 voip
description Outgoing To SIP Proxy
huntstop
preference 1
max-conn 23
destination-pattern .T
voice-class codec 10
session protocol sipv2
session target ipv4:x.x.x.x <------Kamailio Proxy IP
session transport udp
dtmf-relay rtp-nte
no vad
!
dial-peer voice 101 voip
description Outgoing To Cisco
huntstop
preference 10
max-conn 23
destination-pattern .T
voice-class codec 10
session protocol sipv2
session target ipv4:x.x.x.x <-----------Kamailio IP
session transport udp
dtmf-relay rtp-nte
no vad
!
dial-peer voice 200 pots
description Outgoing to PSTN
destination-pattern .T
port 1/0:23
!

 

[netswork]

Thank you very much for that.

On the proxy do you direct inbound calls based on what number was dialed and point it to the 3cx server you want that number to ring on?

How do you configure 3cx to interact with the proxy?

 

[new29]

The kamailio server acts as a router for 3cx and passes the call from PSTN gateway to the correct 3CX instance. 3CX is configured to connect to kamailio via SIP trunk authentication.

 

[netswork]

Is there any documentation out there on setting up the trunk inside 3cx to kamailio?

 

[new29]

We set it up using the 'generic SIP trunk' option when setting us the trunking in the 3cx. We then use the settings that correspond with the configuration of Kamailio. There are so many options in Kamailio that you'd need to really set that up first and then configure 3cx to match Kamailio. For example, you can have your DID be the sip trunk username or you can bind it to another username. You'd need to know that prior to setting up 3cx.

 

[netswork]

With kamailio how are you routing the calls? Destination caller id?