[asterisk-users] Using asterisk as the recording server

[Miguel Molina]

>>> I imagine this setup will need those two communicating entities to  
>>> be part
>>> of the pabx. But support extension 100 of PABX A (legacy) calls 101  
>>> on the
>>> same platform. I want asterisk connected to PABX A via E1/T1 to  
>>> know about
>>> that call and start recording (tap) without bridging or being part  
>>> of that
>>> conversation
>>>
>>>       
>> Hi,
>>
>> Asterisk won't work as a recording server if the call doesn't go  
>> through
>> it. In the IP world it means that both media (RTP) and signalling must
>> pass through asterisk, and in the E1/T1 digital or analog world it  
>> means
>> that the call must be bridged through asterisk. A simple dialplan  
>> would
>> explain it:
>>
>> exten => s,1,Answer() ;Asterisk receives the call, from the lecagy PBX
>> or from the external link (this should be two different contexts)
>> exten => s,n,MixMonitor(blah....) ; Records the conversation,
>> exten => s,n,Dial(Tech/peer/number...,30,rtTwhatever) ; and sends the
>> call back to the legacy PBX or to an external link
>>
>> If you want to record 100% calls, you would have to route every call
>> through asterisk, even internal PBX calls. Even if you want to tap  
>> your
>> legacy PBX to a non-asterisk recording server like the ones suggested
>> before in this thread, the calls must go through a link to make  
>> tapping
>> possible and you should seek an alternate solution to the internal  
>> calls
>> within your legacy PBX. The beauty of asterisk and open source IP-PBXs
>> relies on the native recording capabilities which makes things really
>> easy. When you see that asterisk works and that can do the recordings
>> and much more, you would start thinking on making asterisk your main  
>> PBX
>> solution and leaving that legacy PBX for minimal uses.
>>
>> Cheers,
>>
>> -- 
>> Ing. Miguel Molina
>> Grupo de Tecnología
>> Millenium Phone Center
>>     
Erik de Wild escribió:
> using mixmonitor might not be such a good idea. afaik the mixing of  
> the recordings of the two channels starts after ending the call  
> causing a high cpu load. if you have recordings going on all the time  
> moving the 2 files that has to be mixed to a dedicated mixing server  
> might be a good idea. after mixing it should be stored in a  
> retrievable way.
>
>   
No, that was the old behavior of Monitor() with the m option that at the 
end of the recording it launched an underneath sox process which did the 
mix, causing a CPU spike on every conversation end and putting asterisk 
on trouble if there were many mixes at the same time. Mixmonitor took 
care of that, and it does the mixing while the conversation is taking 
place, thus generating the single file with no CPU spikes or external 
process calls.  Your idea about the separate mixing server was what our 
company did about three years ago with the old first 1.2 asterisk 
versions, where MixMonitor used to be buggy and we were forced to 
implement that kind of solution. But times are a lot better now!

Cheers,

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center