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

[Paul Hales]

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: 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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