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 <ed...@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 > -- _____________________________________________________________________ -- 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