Basically, you run the phone lines into the asterisk box, then out of the Asterisk system into the PABX.
This works reasonably well, and gives you the option to migrate to a full asterisk setup in the future. PaulH Syed Nasruddin wrote: > Thanks Steve, > > How I can use it "Asterisk" as Man In The Middle. Since we have to keep > our Native PBX intact and functioning but only thing it doesn't handle > is Voice Recording. I thought if I can get some Channel Variable or some > system generated event regarding OFF-HOOK and ON-HOOK condition through > Asterisk I will easily handle this requirement. > > It will be a great help if you can elaborate how I can use asterisk as > man-in-the-middle configuration along with my current PBX. > > Thanks a lot for your prompt response > > Syed Nasruddin (CISSP) > > Assistant Manager - Systems > National Commodity Exchange Limited > 9th Floor, PIC Towers > 32-A Lalazar Drive > M.T. Khan Road > Karachi > Phone: 111623623 ext 217 > Fax: 5611263 > Web: www.ncel.com.pk > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Steve > Totaro > Sent: Thursday, June 12, 2008 7:39 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Using Asterisk Only as Voice > RecordingSolution. > > On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin <[EMAIL PROTECTED]> > wrote: > >> HI, >> >> >> >> I am using TDM800P Digium Card with Asterisk 1.4.* version. I have >> > fair > >> command over Asterisk up till now and have run it in different >> > scenarios > >> such as Call Center Solution, PBX solution. >> >> >> >> There is a requirement to use Asterisk only as Voice Recording >> > solution in > >> following manner: >> >> >> >> Physical POT lines before entering into our native PBX will be >> > splitted and > >> one of each of those lines will also enter into our Asterisk System. >> Once the call is routed by our native PBX and recipient picks up the >> > phone > >> (either SIP phone or Analog Phone) I should be able to start recording >> > the > >> call. >> When the call ends, the recording should stop. >> >> >> >> Problem being faced by me is this that I am able to catch the call in >> > my > >> diaplan and initialize MixMonitor but since my diaplan never detects >> OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up >> > while in > >> actual the call is running through our PBX. >> >> >> >> Is there any channel variable or any other mechanism by which I can >> accomplish this task? Since i will not be using any Dial() or similar >> application I will be needing some kind of OFF-HOOK trigger/Event in >> > my > >> dialplan. >> >> >> >> Your help will be highly appreciated. >> >> >> >> regards >> >> >> >> Syed Nasruddin >> >> > > It may not be possible to do this in parallel the way you are trying > now. In series should be a simple task. > > Just pass the call through Asterisk as the man in the middle, the > dialplan will be very simple. > > Thanks, > Steve T > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
