Hi, Did you created your normal Inbound and Outbound routes in freepbx? For use with your zap channels?
You'll problably have to change your routes on your pbx too... Regards, Carlos M Cruz 2011/7/28 michael k <[email protected]> > Hello All, > > I don't even know the relevancy of my question. Please answer me if my > question have some sense. > > I have recently implemented an asterisk server with freepbx. I have created > 100 extentions and i can make successful calls between extensions from > anywhere. But my office have three different land-line numbers and three of > them are terminating into an internal PBX ( normal matrix telephone PBX) > with more than 60 extensions. This internal PBX is the live PBX where we can > call local, STD and ISD from extensions. > > At present i have some practical difficulties to configure telephone lines > at the end of asterisk PBX. So i am trying to connect my asterisk PBX to the > normal telephone PBX. > > I have installed 1 port x100p FXO card in my asterisk PBX and detected by > my freepbx. Then i removed my normal telephone extension cable from phone > and connected to the FXO port of my asterisk PBX. > > Ultimately my intention is that > > 1) if somebody call to my normal telephone extension, that should reach to > my asterisk server, and asterisk server should send this call to my asterisk > extension. > 2) if i am calling from my asterisk extension, call should go to the normal > telephone PBX via FXO card in my asterisk server and ultimately the call > should send outside via the telephone PBX. > > > Is my approach is correct ? If it is wrong please somebody assist me to > connect my asterisk PBX to normal telephone PBX. > > Michael.K > > > -- > _____________________________________________________________________ > -- 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
