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

[Syed Nasruddin]

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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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: 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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