On 15 Jan 2009, at 07:30, Wolfgang Pichler wrote: > Hi all, > > thanks Tim and Mexuar for releasing this here... > > I have already taken the source - and compiled a little java applet > which is self signed to test the whole thing. >
That was quick :-) > I will put it on my site (and allow users to enter > host/user/pass/Calling Number,Calling Name,Number to dial...) for demo > usage.... > > I would be happy to get some feedback about problems - because i am > interessted to integrate it in my callcenter project > > Tim - can you tell me which audio features it does have - as far as i > can see there is alaw and gsm - is there also an echo canceller - > jitter > buffer ? I don't think the GSM codec is actually in there, from memory it does ULAW/ALaw and Slin There is a jitterbuffer of sorts. I never managed to get the echo canceller to work, although the code for it is in the codebase. > > > I will post it here as soon as i have the page up ... If you plan to do significant work on it, please could you put it on sourceforge so others can chip in ? (That's kinda the point of GPLing it) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
