Connecting two CUCM's that are remote to each other with SIP trunk

[Samer R. Saleem]

Hello,

I have a new requirement from my manager to connect two sites of two companies in order to allow calls from Site1 CUCM to come to Site2 CUCM.

however, I did not work on something like this before and I really need your help with this.

A.What do I need in order to accomplish this? for example site to site VPN?

B.Will site to site VPN allow SIP trunk to be formed?

C.what will happen if Site1 and Site2 are using same range of numbers?

any other advise and comments, please share with me!

Thanks in advance

[Nithin Eluvathingal]

You need to have reachability between two sites, both CUCM severs   and phone subnet. It can be over vpn or point to point to point or anything. 

what will happen if Site1 and Site2 are using same range of numbers?

use prefix for calling between sites.

For example site A has number range 1XXX and Number range for site B 1XXX.

when calling from site A To B users call by extra digit *11XXX. And when calling from B to A user can use *21XXX. 
 Extra digits can be stripped before sending or while receiving the call by CUCM.

[Jonas Fraga]

You need to think basically on 2 things to get any Trunk working:

Reachability for signaling,

Reachability for media.

CUCM by default centralize all signaling traffic because it's the call control portion which need reachability from following flow:

PHONE 1 <-> CUCM1 <-> CUCM2 <-> PHONE 2

For Media, by default CUCM lets Devices communicates directly which make you have some options to choice:

1) With no changes media goes directly from and to phones:

Phone 1 <-> Phone 2

2) Enable MTP on Trunk both sides to media goes through CUCM, following same path as signaling:

PHONE 1 <-> CUCM1 <-> CUCM2 <-> PHONE 2

3) Using other device as media node using MTP or Traversal as resource like Voice gateway or Expressway:

PHONE 1 <-> CUCM1 <-> GW1 <-> GW2 <-> CUCM2 <-> PHONE 2

If your'e using NAT you need to take care of media because on SDP negotiation the NAT address should be placed to work. I'd no recommend using NAT on these kind of integration.

For dial plan as already explained you can have same extension range if you use some digits to identify when the calls needs to be traversed to other site.