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...driver has been enhanced significantly since the 0.9.0 version present in FreeBSD's ports. So, the Asterisk application will not build or run with FreeBSD's zaptel 0.9.0 as it stands. CHANGES THIS WEEK: We widened support for flavors of FreeBSD and OpenBSD. Between Olle Johansson, Rich Neese and I, we're working on: - FreeBSD 4.9, 5.2.1 and -current, as well as - OpenBSD 3.5. Work has begun on merging FreeBSD support into the zaptel driver in Asterisk-current. Maxim Sobolev, the maintainer of FreeBSD's zaptel port, wants to work together on it, and Mark Spencer supports the g...

Whatever happened to the Zapatatelephony project? The last information there is from 2001 and a lot of new cards have been released since. Has GNU hardware development completely stopped?

Hi I have just got Asterisk going with an spa-2000. however when I look through the userpdf every function on the sipura and asterisk seems to require on-hook or flash button , all of the phones i have do net seem to have either, is there a way round this ? does anyone know. Or do i ahve to go out and buy more phones? Anyhelp appreciated Simon

Hi, I have the following problem. I have 5 phones behind the same nat (canreinvite=yes). it works fine to receive calls and to make calls. sound quality is good, so everything works fine. The poblem is that the phone behind nat cant call each other. It works if canreinvite=no. But i want to do this. Does anyone have an idea? Regards, cjk.

Hi Guys, I'm new in VoIP, I heard from a friend that asterisk is good in VoIP service especially on SIP. I'm planning to replace our old PBX system (legacy of Panasonic) to VoIP so that even out of the country we can still communicate cheaper than regular phone. But I have a few questions though before I change our OLD PBX to VoIP. 1. Does asterisk generate CDR? If yes how do I