[asterisk-users] Using Asterisk Only as Voice RecordingSolution.

[Steve Totaro]

You will need exactly two times the number of ports that your legacy
system has.  Asterisk takes the call on _.,1,DAHDI, starts monitor and
dials out the second DAHDI port to your legacy system.

It is about ten lines in extensions.conf.

Thanks,
Steve T

On Thu, Jun 12, 2008 at 12:01 PM, Syed Nasruddin <nasruddin at ncel.com.pk> 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: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] 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 <nasruddin at ncel.com.pk>
> 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
>
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