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 > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
