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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MoveCallRecs.sh
Description: Bourne shell script
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