Help!

nkillgore · June 2013

I'm not entirely sure where to even ask this, but I'm looking for some resources to learn about CUCM/Unity/Presence.

The voice person left the company, and now I need to learn it. More immediately, in about two weeks, we are moving all of our PRIs and ELDs to SIP trunks. We currently have one SIP trunk with 10 new numbers on it delivered from the ISP for testing, but I don't know where to even start (beyond going to Device -> Trunk and adding a new one). Where can I look for more info?

shodown · June 2013

This depends on what version of CUCM you are on. You will have to create dial peers on the gateway pointing towards CUCM and the SIP trunk. Also if they are giving you a SIP trunk on your MPLS could you may have to advertise routes to them or make sure they are being redistrubuted into BGP correctly. You have 2 weeks, but I would engage your cisco partner or external consultants if this is your 1st SIP trunk implementation to ensure you have things setup correctly. They may have given you a cut sheet to use as well. That should help you with the basics.

nkillgore · June 2013

We are on 7.1.5. I am also supposed to upgrade that to 9.1. Soon.

I was really planning to take the CCNP: R&S exams soon, but I may end up having to study for the voice track just so that I can learn it for work.

shodown · June 2013

I would go with what you want to know. The company shouldn't be expecting you to do that level of work and have everything running. There is a reason why you had a separate voice engineer.

sieff · June 2013

There's the Cisco Interoperability guide somewhere on CCO. It pretty much has sample SIP trunk configs for most carriers. SIP deployments can be kind of tricky, which is why it's good to start with a pool of test DID's. So it's good that you have them.

There's no quick way to learn CUCM/Unity/CUPS. I did buy the CUPS book from Cisco Press, which was surprisingly pretty good for getting me through my first install of the application.

Happy Hunting!

wintermute000 · June 2013

You really want a CUBE between your CUCM and carrier. you WILL NEED a CUBE if there is any NAT issue present e.g. carrier can't provision in your RFC1918 address space. (actually in reality, any session border controller will do - acme, sonus, cisco CUBE, but the easiest to learn/interop with CUCM is obviously CUBE). There's also codec fun times if the carrier does not support your native CUCM codec (u or a law), DTMF, have you figured out if you need MTP resources on that SIP trunk, I could go on and on (I was tech lead on our ISDN to SIP migration which was 28 ISDNS --> 3 SIP trunks so I've gone through the pain myself)

Almost nobody SIP trunks directly to the carrier, in addition to the NAT issue you are opening your voice core to external (i really hope its not internet SIP trunks!!!). And if there are any interop issues like DTMF or fax stream or whatever, CUCM is notorious for offering very little in the way of interop options which is why Cisco tells you to use a border gateway i.e. CUBE to do the SIP normalisation.

If you've got two weeks and are a CUCM rookie you are in big trouble. I suggest getting professional services in.

Help!

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