This is the best linux sip phone I've used so far. Audio quality has been perfect and it seems really stable, so hopefully it will be out of beta soon. I might actually pay for the full version! (not counting console games, that would be the second piece of software I've purchaced since 1987).
Kris Edwards <krisedwards@gmail.com> writes:> This is the best linux sip phone I've used so far. Audio quality has > been perfect and it seems really stable, so hopefully it will be out of > beta soon. > > I might actually pay for the full version! (not counting console games, > that would be the second piece of software I've purchaced since 1987).Sounds rather like you want to sell the full version. Myself, I don't know about recent betas since, frankly, I didn't care anymore after initial experiences being pretty much disappointing. The first beta I got produced no audio at all, and we had a tough time to convince the developer that it wasn't a driver issue. The next releases then had huge latencies, primarily due to the Xlite audio setup. Now, I admit that setting up audio for interactive/'realtime' apps on linux is a mess, but various open source projects have already done much better. So no, in contrast to your plug I'm not as enthusiastic myself, especially since audio quality resp. latency is the one major trouble I had with linux softphones. E.g. iaxcomm would be great and totally satisfying for me if latency were (significantly) less than 1 second. Regards, Bruno.
Kris Edwards wrote:>This is the best linux sip phone I've used so far. Audio quality has >been perfect and it seems really stable, so hopefully it will be out of >beta soon. > >I might actually pay for the full version! (not counting console games, >that would be the second piece of software I've purchaced since 1987). >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >Where can i get that version? Not found any link on xten site... Thanks
the idea is this: I have a servant ASTERISK with two cards clone x100p, one of them I want it to form so that it makes calls by the telefonica line and the other to connect I telephone stops from ahi to be able to make calls. the question is podra to become that? for but information I have a servant gentoo 2004.3 Kernel 2.4.28-gentoo-r8 1Giga Byte RAM. 2 clone X100P thanks