Thanks for the reply. I am using an analog phone in normal PBX. I have an extension called 199 in asterisk and an extension 264 in analog PBX. So how do i create an inbound or outbound routes for call between these two extentions ?
On Thu, Jul 28, 2011 at 1:39 PM, Carlos M Cruz <carlosmoc...@gmail.com>wrote: > 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 <mich...@inapp.com> > >> 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 >
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