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Craig
29-Jun-2007, 08:22 PM
Does anyone know if you can integrate Asterisk with a Traditional PBX and
have them work together seamlessly? For example, can either switch call the
other switch using only extensions? What hardware, if any, is required?

Thanks!

unsigned@ @digerati.us
29-Jun-2007, 08:54 PM
Look at q.sig compatibility is you have digital trunks. I believe the
basic functionality you are talking about is available on the Asterisk
side, it just depends on the PBX you are trying to talk to. If not
digital lines it can be done with analog ports and a good call routing
setup. In either case you might need hardware.

Craig wrote:
> Does anyone know if you can integrate Asterisk with a Traditional PBX and
> have them work together seamlessly? For example, can either switch call the
> other switch using only extensions? What hardware, if any, is required?
>
> Thanks!
>
>

Adam Johnson
29-Jun-2007, 09:52 PM
Craig wrote:

> Does anyone know if you can integrate Asterisk with a Traditional PBX and
> have them work together seamlessly? For example, can either switch call the
> other switch using only extensions? What hardware, if any, is required?

> Thanks!

Well some type of connection is required, if the "traditional" PBX can do
VoIP trunks, it might be as little as a crossover Ethernet cable. If it
doesn't do VoIP, you'll need some other type of trunk port, which would
require hardware on both systems that match (analog tie line, ISDN BRI?
ISDN PRI/T1). Which type you get will depend on how many circuits you'll
need between the systems, and hardware availability.

We migrated from a couple NEC PBX's and a really old Mitel over to Cisco
CallManagers, and we hooked them together with a Cisco router with a voice
network module and a couple voice V-WIC's set up for PRI use. One went to
the PRI circuit from the local telco, the other went to the NEC PBX which
was the hub of the traditional PBXs. As far as the NEC was concerned, it
still had a PRI circuit from the phone company. The Cisco router had a
long list of "dial peers" so that as incoming phone calls arrived, it new
where to send the call by extension, either to the VoIP phone system over
ethernet, or the traditional PBX over the second PRI circuit.

This did mean we had 3 places to make changes every time we moved a phone.
The old PBX needed to know to send calls to VoIP extensions out the PRI,
the voice router needed to know to send that call to the Cisco CallManger
(Asterisk in your case) and CallManager needed to have the extension
defined in itself.

Since we were also migrating to a new voicemail system, things were not
totally seamless - users on the traditional PBX could not forward
voicemails to users on the IP phones. IP Phone users could forward
voicemails to PBX users, but only as an email with a wave attachment (and
not everyone had sound cards/speakers). It was a bit of a pain, but now
that we're totally switched over, it's nice not to have to re-crosswire
every time someone moves two cubes over :-)