[asterisk-users] Call Recording Questions

[Antonio Berrios]

  1) I use a bash script I wrote to check if call recordings are being 
written to and if not then move them. I move them to a locally mounted 
NFS share but this will work with any type of locally mounted share 
(Samba for Windows). I run the script every minute with cron. It also 
sorts the recordings in directories based on date. If you just want to 
sync files rather than move, just change the mv commands to cp commands.

Script attached.


On 09/02/2010 12:27 PM, Gareth Blades wrote:
> The DTMF mode can cause problems. The main rule is to make sure
> everything is using the same method. I normally use SIP-Info as the
> method as it allows to rtp stream to be switch directly between the two
> end points but asterisk still sees all the dtmf digits.
>
> Dan Journo wrote:
>>> 1) I want to create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the
>>> recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over.
>>> 2) I tried setting up *1 in features.conf but when I press *1, all that happens is that the caller hears the tones but no recording starts. I've added
>>> wW to the Dial() command, and also Set(DYNAMIC_FEATURES=automon) (don't know if the last one is necessary). The line in features.conf says automon =>
>>> *1 and I restarted asterisk once the changes were made.
>> Sorry, just re-read my email and realised I didn't ask any questions and it sounded quite rude.
>>
>> Basically, I'm trying to allow one of my clients to record calls and download them onto their PC. I'm thinking of creating a web interface for this, which is where my first question comes in.
>>
>> However, I can't seem to get it working. I think it's something to do with inband and rfc2833 but when I change it, the menu systems seem to stop working.
>>
>> Can anyone assist?
>>
>> Thanks
>> Dan
>>
>


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