[asterisk-users] Bridging old system (ESI IVX E) with new Asterisk server

Ryan Wagoner rswagoner at gmail.com
Mon May 3 23:27:59 CDT 2010


I was in a similar situation with a Toshiba CIX PBX. I had 150 phones
on the Toshiba and wanted to switch over to SIP phones slowly. The
Toshiba already had PRI cards connecting to the phone company. I
purchased Sangoma PRI cards for the Asterisk server. I connected the
Toshiba PRIs to the Asterisk PRIs and used QSIG signaling so Caller ID
names would be correctly shown for internal calls. I connected the
Asterisk PRIs to the phone company.

To make calls between the two systems seamless I had to program the
extensions in both systems. We currently used 4 digit extensions. I
ended up reprogramming all the Toshiba extensions to 2+ext. Toshiba
has StrataNet which allows me to say extension x, y, z are available
at this remote node. I programmed all the 4 digit extensions in here.
This way when a 4 digit extension is dialed it will go to Asterisk
which then can decide how to handle the call. This was a requirement
as I needed everyone to be able to have a SIP softphone.

On the Asterisk side I setup a trunk and route to the Toshiba. For the
route I setup a pattern so I could dial Toshiba phones with 2+ext. I
then created all the 4 digit extensions and setup follow me for each
with 2+ext.

At this point you should be able to register a SIP phone and dial a 4
digit extension. It will use follow me to ring the Toshiba phone. From
a Toshiba phone dialing a 4 digit extension will route the call
through Asterisk and back with follow me to the Toshiba.

With Asterisk handling all the calls I can easily transition users by
setting them up a SIP hardphone and then removing the follow me.
Eventually as funds allow I can move everyone over to a SIP phone.
Then it is as simply as turning the Toshiba off.

Ryan

On Mon, May 3, 2010 at 10:30 AM, Eddie Mikell <eddie at rimmkaufman.com> wrote:
> All:
>
> My company has an existing ESI IVX E-class system with 45 phones.  I can
> add one more card, to expand it another 6 phones, but it's $8000, and
> then the system will have to be replaced.
>
> I have the Asterisk server up and running, with 2 sip lines from the
> local phone service.  (Thanks to you guys, it is working great!).  I'm
> pretty sure this is the way the company will move, and I've have
> installed 8 phones for different people to test.
>
> My question is this:  Is there some way I can bridge the two systems
> together, even temporarily?  So if Jake is at extension 120 on the ESI
> system, and Regis is at extensions 155 on the asterisk server, Jake can
> call Regis and vice versa.
>
> I've pondered on this over the week-end, but don't see an easy way to
> handle this.
>
> Thanks!
>
> Eddie Mikell
> Senior Systems Engineer
> The Rimm-Kaufman Group
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



More information about the asterisk-users mailing list