Re: [linux-audio-user] fighting skype ...

Eric Dantan Rzewnicki <rzewnickie(a)rfa.org> writes: SIP: http://www.sipfoundry.org/sipXphone/ (never really played with it) http://www.wirlab.net/kphone/ (use it sometimes, making calls through asterisk) http://www.gnophone.com/ (devel seized) IAX: http://iaxclient.sourceforge.net/iaxcomm/ (use sometimes; calls through asterisk or p2p) H.323 http://www.gnomemeeting.org/ Also take a look at http://voip-info.org/wiki-VOIP+Phones and generally on http://voip-info.org There are many trying to lure you;). Always locate the license and make sure it has p2p calls. I don't know of a list of libre software voip phones and since voip-info wiki is not publicly editable, it's such a hassle to make one. It's maybe preferable to have an account with a SIP provider. Then you would have the number like sip:sip-provider.void/your-username or if dialed from a POTS telephone, maybe 1-800-CALL-ME-TOTO. You would then use your account your-username(a)sip-provider.void to register using your softphone enabling you to make and break calls from and to anywhere in the world. I currently have an asterisk box reachable through my FQDN at sip:esben-stien.name - this translates to sip:esben-stien.name/s as s is the default extension; it reaches the main menu system. You could also call sip:esben-stien.name/777 to directly call the my_house_built_on_stilts hotline. Please call me sometime and we can test your setup. I test a lot at the moment so please try again if you can't reach me. Asterisk, http://asterisk.org is a software PBX, supporting the major SIP (and H.323) plus some other protocols, like IAX. It has an interactive interface allowing you to make and break calls over any protocol that asterisk supports. I've never played with sipXphone, but a little with kphone I use asterisk (SIP/IAX) myself for dialing since there is no proper client, but I have many friends that still use windows. I give them iaxcomm. I've had great deals of problems and still have with some users. I'm currently doing heavy testing to have a stable SIP and IAX solution. What I really need is a client daemon hooked up to jack that I can interface to with a non-interactive interface or an interactive ncurses app (and maybe a graphical gtk+ frontend for some eyecandy). It would also be nice to have a sip softphone on my web page. I took a look at mozphone, but have not played with it yet. Proprietary software is not an option. -- Esben Stien is b0ef(a)esben-stien.name http://www.esben-stien.name irc://irc.esben-stien.name/%23contact [sip|iax]:esben-stien.name