Martin: Status: I downloaded asterisk, compiled and did the "example" test. I needed to install the follwoing packages: readline-dev kernel sources ncurses-dev Doxgen And I did the follwoing commands: cd /usr/src export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot cvs login cvs checkout -r v1-0_stable asterisk cd asterisk make clean make install make samples make progdocs The cvs login gave me this warinig: cvs login: warning: failed to open /root/.cvspass for reading: No such file or directory the "make install" took a little over 2 mins on my 1000 mhz P3 laptop I executed the test command asterisk -vvvvc (Use "STOP NOW" to exit) which gave screens of stuff I didn't bother to read. I then tried to get 2 SIP clients to talk, but was unsuccessfull.. I'm calling it a night Jerry P.S. Subscribed to asterix-users, as you suggested... On Wed, 2004-02-18 at 13:41, Martin Mielke wrote:
Okay, sorry about that..... need to emember read before write... the AVM Fritz pci card is supported...
:-))
Okay Martin, Asterix looks real good, I'll go for it... what are your/mine/our goals?
OK, this is my shot... The company I work for has some offices in Spain (also in Europe, but I'll start with a "local" solution). I aim to implement a VoIP/PSTN gateway so that, for a start, phone calls among our offices are carried at low cost (as low as zero, which makes our accountants happy ;-)).
In a second step, say some months after de "local" solution has been proven to be stable and blah blah blah, that VoIP/PSTN gateway could interconnect our offices across Europe and/or even worldwide... right now I can't say it would be me who will do the whole job but such a solution could be implemented soon.
I'd like to have an VOIP/PSTN gateway up and running ASAP. Unfortunatetly I fear defineing "As Soon as Posible", is quite a let down on my side...
What is your schedule like?
From now, I have around 3 weeks (or less) to setup a (working) demo... keep your fingers crossed! :)) I'll be posting my efforts...
How do we want to go forward?
Hmmm... good question, next question! :DD I'd recommend you to subscribe to the asterisk-users mailing list, which is a good source for support...
Furthermore, I guess this thread is interesing enough to keep it open for a while until all interested people get their answers (well, they can search the list archives) and, at least, one working solution has been (clearly) explained. If this sounds like a bad idea to the rest not interested in VoIP, I'd suggest to create a separate group (suse-voip, or the like), in SuSE servers or somewhere else, and people interested in this subject could join us.
What do you think?
Regards,
Martin