You will need to setup a SIP trunk between the asterisk server and the CCM server. Then in your asterisk config, you'll need to direct any extensions that are handled by the CCM server to that trunk. You'll also need to configure the CCM server to send calls to the specific extension through the asterisk sip trunk.
-- Thanks, Warren Selby, dCAP On Aug 9, 2012, at 6:05 PM, Eduardo Giacoman <giaco...@gmail.com> wrote: > Danny, thanks for your input... > > Can you tell me if I am wrong with the following or give me a brief guide of > what to look at? > I was planning on using Asterisk + chan_sccp to control the VOIP phone. > Asterisk will NOT replace the current CCM/PBX at work, it will have just one > phone but in a way that I still can call extensions at work from asterisk. > > I can point the phone to another TFTP server with the proper SEM file, etc. > so it will talk to Asterisk. But after that, if I call an internal extension > at work will it find it or I have to do something else? I am a little > confused because I think that since the phone is not pointing anymore to the > CCM at work, it won't find any other internal extensions, just the ones I may > add to the asterisk setup. > > Excuse me I have very basic voip knowledge. > > > On Thu, Aug 9, 2012 at 3:25 PM, Danny Nicholas <da...@debsinc.com> wrote: > This shouldn’t be a problem. Asterisk is basically “flavor-blind” as to what > type and quantity of phones you put on it. > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jorge Díaz > Sent: Thursday, August 09, 2012 4:21 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Asterisk to control just one phone within current > CCM? > > > > > > Hi, > > I have used basic Asterisk as a PBX controlling few extensions. My question > is simple, at work there is an existing Call Manager/PBX and all which > manages all the extensions for SCCP VOIP phones. Can Asterisk be used to > manage just 1 VOIP phone and still can make internal calls to the other > extensions? > > Thanks, > Jorge > > > -- > _____________________________________________________________________ > -- 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
-- _____________________________________________________________________ -- 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
