Step five: Profit ;-) I am not going to write your dialplan for you but here is a clue. http://www.voip-info.org/wiki/view/Asterisk+legacy+integration
Of those various setups, you can extract what you need. Thanks, Steve T On Fri, Jun 13, 2008 at 8:05 AM, Syed Nasruddin <[EMAIL PROTECTED]> wrote: > Dear PaulH, > > I have 5 PSTN Lines going into my legacy PBX. There is an active IVR > present on legacy PBX which the client wants to keep. So what I have to > do is: > > 1. Install 10 Port i.e 5-FXO/5-FXS card on asterisk machine. > 2. Insert All those PSTN directly to my 5-Port FXO. > 3. Take out 5-FXS Port lines and insert them into my legacy PBX. > 4. Since as I mentioned previously that my client wants to keep its IVR > intact on its Legacy system so I will not be handling IVR in my Asterisk > Dialplan. > 5. when the call arrives at asterisk....what should I do?? Should I > simply call Dial(FXS channel) or something else. > > Kindly provide some info regarding Step 5. > > Thanks > > Syed Nasruddin > > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales > Sent: Friday, June 13, 2008 9:38 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Using Asterisk Only as Voice > RecordingSolution. > > > 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 > > _______________________________________________ > -- 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