Voice Gateway with Subinterfaces

[Juraj Papic]

Hello,

I have my voice gateway with 2 interfaces, one for the customer network and the other will be set in 2 sub interfaces. This 2 sub-interfaces will be to connect sip trunks to the PSTN. I would like to know if there's any special configuration I have to do on the gateway or in the cucm to make this work?

thanks. 

[Ritesh Desai]

@Juraj Papic 

Yes, that is possible but I would recommend to keep seperate interfaces for PSTN SIP trunks as I consider as best practice. There is no chance of redundancy in case the interfaces goes down.

Lets take e.g: Gig 0/0/0 for LAN and Gig 0/0/1 for 2 SIP trunks. In this case, you would need to configure 2 sub interfaces.

interface GigabitEthernet0/0/1.1
encapsulation dot1Q 100

!
interface GigabitEthernet0/0/1.2
encapsulation dot1Q 200

Values 100 and 200 refer to VLAN ID's. This will help you to segregate voice packets at Switch level. Underneath respective interfaces you can configure ip address, media, etc.

For dedicated interface, you need to check compatibility for additional interfaces. for ISR 4K routers you can check here 

Wish this info finds helpful to you.

thanks and regards,

Ritesh Desai

Please rate helpful votes.

[Juraj Papic]

Hello,

Having the 2 sub-interfaces how can I configure my dial plan in the cucm to go via which sub-interface  I need to route the calls.

Thanks. 

[Ritesh Desai]

Hi,

Eg. From CUCM if you want department 1 to use trunk-1 And department-2 to use trunk-2. Create similar route pattern with different prefixes for 2 departments.
Eg: prefix 8 for dept-1 and 9 for dept-2 to dial mobile / national calls.

CUCM will route to Voice Gateway via Sip trunk. On CUBE, you will use translation rule (manipulate digits) to segregate the call to go from trunk-1 gig0/0/1.1 or trunk-2 gig0/0/1.2

Using translation rule you will remove 9 and 8 and apply to outgoing dial-peer.

E.g
dial-peer-1 voip
destination-pattern 9.T
Session-target ipv4:
translation profile remove9
Bind interface control gig0/0/1.1
Bind interface media gig0/0/1.1

dial-peer-2voip
destination-pattern 8.T
Session-target ipv4:
translation profile remove8
Bind interface control gig0/0/1.2
Bind interface media gig0/0/1.2

*** Above dial-peer config is for understanding purpose. Actual config commands will differ.

Hope you understand the logic.

[Ritesh Desai]

Hi @Juraj Papic 

using the example above, you can create different partition and CSS for dept-1 and dept-2 to restrict inter dept calling OR just to segregate the 2 department calls at CUCM.

[Juraj Papic]

Hello,

For both new groups I have to dial 9 to go to the PSTN, how can I configure the dial plans to know via wich interface they have to go out?

thanks!

[Roger Kallberg]

How a user dials is dictated by configuration in CUCM. That doesn’t mean that it has to be identical in the voice gateway. You can modify the called number on egress from CUCM so that when it reaches the voice gateway it can be sent out to different SIP trunks. For example you could add 890 in front of called number on calls from department 1 and 790 in front of called number for calls from department 2. Then you use these as a sort of route prefix or whatever you’d like to call it to send the call via different dial peers. Not currently on a computer, so can’t provide you with any more details around this, but hopefully you get what I mean. If not please get back to me and I’ll provide additional details.