[asterisk-users] Asterisk + Cisco Call Manager

[David Backeberg]

On Sat, Apr 4, 2009 at 11:18 AM, Timothy Smith <timotsmith at gmail.com> wrote:
> We're migrating from Cisco to asterisk because cisco is expensive to
> maintain, besides we can achieve more with asterisk like customised
> IVRs etc.

I don't know what expensive to maintain means. We spend more on our
phone bill than what the gear costs by a significant margin.

> This being a large production environment, we can't just change over
> without testing thoroughly..

makes sense

>Now, i'd like to
> completely get rid of the cisco gateways by routing incoming calls
> through asterisk too (to the call manager, and finally the phones).

I don't have an architecture diagram of your call flow, but you'll
need to be picking off calls with some criteria, perhaps the number
dialed or DNIS if these are PRIs and route accordingly. You should be
able to make asterisk talk directly to your phones by putting some
asterisk hardware onto the same address space as the phones. If these
are Cisco phones most of them are multi-line, and you can go ahead and
push a change that one of the lines will SIP-register with an asterisk
system. At that point you don't need to route _through_ Cisco because
you would be routing around Call Manager.

If you want to keep Call Manager in the loop you need to have the
traffic going to Cisco continue to look like it did before you put
Asterisk in the loop, or you need to change the Call Manager to act
according to the revised traffic it will be receiving from Asterisk. I
recommend enabling call debugging on the Asterisk side and the Cisco
side, and figure out which side(s) you want to change to keep both
sides happy.

> After that, they'll be satisfied and i'll start registering the phones
> to asterisk until everything is asterisk. It has to be a smooth
> transition, just fyi, we're about 200 employees.
> I'll appreciate any advice towards achieving this.

My advice is to test changes in a small lab setup before you cut them
loose, assuming that no downtime is acceptable. No idea what your
budget is, nor your type of call traffic. If this is real telco lines,
you may need a test telco line for your experiments. (which you can
also fake with asterisk and/or Cisco). If this is just SIP you can
fake it with asterisk and cheaper used Cisco gear. Hopefully you
already have a lab environment where you can test things out.