The nice thing about the Digium TDMXXX series of analog cards is that it allows you to mix and match FXO or FXS as you see fit. And, in FXS's defence, it is a good way (well, the ONLY way) to bridge analog to IP.
A P3 class box should scale to a few dozen extensions no problem. The caveat is if the box is doing other stuff (it shouldn't) and / or if it is transcoding (converting) codecs from one format to another, you might run into performance issues. Best practice is to always use the same codec end-to-end if Asterisk remains in the media stream (which you select using canreinvite=yes or canreinvite=no in sip.conf) and this avoids transcoding on the Asterisk box. Depending on your point of view and your luck in getting a box running, Asterisk is either the coolest thing ever or a gigantic pain in the ass. Patience and problem solving skills will serve you well. -----Original Message----- From: Jim Freeze [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 2:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXS or VOIP On 1/11/06, William Boehlke <[EMAIL PROTECTED]> wrote: > > A single computer will handle hundreds of telephones. Just get a card with > more ports, or use an external gateway. I am sorry, I don't understand. Are you talking about analog FXS phones? All the PCI cards I have seen have a max of 4 FXS lines and the external boxes seem very expensive. -- Jim Freeze _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
