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

Syed Nasruddin nasruddin at ncel.com.pk
Thu Jun 12 11:01:33 CDT 2008


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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