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For all you people touting Yate as a replacement to asterisk, you do realize it doesn't even have transfer and 3 way capability built in don't you?

http://yate.null.ro/pmwiki/index.php/Main/Transfer s

Transfering a call and three way calling is listed as a "feature request", I don't know how you can possibly recommend such obviously alpha telephony software.

I'm not trying to say that Asterisk is the end all be all, its not, and maybe Yate will come up and be a better solution, but right now, without basic PBX features, it is not.

Check out Yate, it's open source, and scalable, and is in use in many callcenters in Europe without problems.

All people talk about is Asterisk. Meanwhile there's the OpenSource solution (even GPL) called Yate; which handles a magnitude larger number of calls than Asterisk on the same hardware, it has the (currently still unique) perfect NAT-proof algorithm for SIP, it has excellent support for H.323, and, last but not least, the company supporting it insists to do paid work only when it results in (new) GPLed code.

Yate handles business-logic integration just fine with predefined hooks (I used a PostgreSQL backend to integrate it with).

Did you have a look at Yate (http://yate.null.ro/)? I have also used Asterisk long enough to learn to hate it, but Yate looks very nice. It doesn't have as many features as Asterisk, but the code base is very clean and it does look rather easy to extend.

VoIP not only needs to catch up but also be open like email, and unlike the divided IM space.
Unfortunately Skype is not the application which connects to an open network.

Only applications like Gizmo http://www.gizmoproject.com/ and many other ones (which I don't use) connect to the International Standard-compliant Protocol known as SIP.

If you want voice chat (VoIP) on Linux then you have a good selection too (I don't know which are SIP compliant and which are not though):

http://www.phonegaim.com/
http://cockatoo.mozdev.org/
http://www.gizmoproject.com/
http://www.linphone.org/
http://www.wirlab.net/kphone/
http://www.minisip.org/
http://www.sflphone.org/
http://www.sipfoundry.org/
http://www.twinklephone.com/
http://www.openwengo.com/
http://yate.null.ro/
http://www.divmod.org/projects/shtoom

I've just took a look at then Sofia-SIP stack. One of the most horribile pieces or code i've saw lately. I mean even oSIP which is the most rubbish SIP stack from the free world look way better than this.
I won't compare it with YASS (Yate SIP stack) which is a piece of art if you compare it with SIP stack.
I can't belive that in this days someone will write code in the way Sofia-SIP is writte. Just compare how complicated it is.

http://voip.null.ro/cgi-bin/cvsweb.cgi/yate/contri b/ysip/ - Yate SIP stack

http://savannah.gnu.org/cgi-bin/viewcvs/osip/osip/ - oSIP

http://cvs.sourceforge.net/viewcvs.py/sofia-sip/so fia-sip/ - Sofia-SIP

I think in the end that what Nokia did was just to throw some rubbish code arround hoping to get some more bug fixes.

In addition to the number of good IM clients for Linux (especially GAIM), if you want voice chat (VoIP) on Linux then you have a good selection too: PhoneGaim : http://www.phonegaim.com/ ( http://cockatoo.mozdev.org/ ( http://www.gizmoproject.com/ ( http://www.linphone.org/ KPhone : http://www.wirlab.net/kphone/ Skype : http://www.skype.com/ ( http://www.minisip.org/ SFLphone : http://www.sflphone.org/ SIPfoundry : http://www.sipfoundry.org/ Twinkle : http://www.twinklephone.com/ openwengo : http://www.openwengo.com/ Yate : http://yate.null.ro/ shtoom : http://www.divmod.org/projects/shtoom Best to get one that connects via 'SIP' and is entirely standard-compliant, then you can connect to anyone on other standard networks (except those in closed networks like Skype(???)).

There is also the tryly well designed and promising Yate:
http://www.yate.null.ro/pmwiki/index.php
There is a C++ core with telephone basic functionality and you use a lot of languages you like to do whatever you want (php, perl, python, ruby...).