IMHO, If you plan to use analog phones the cheapest is to buy a bunch of sipuras instead of TDM40B. (TDM40B = 4 FXS for $300, $75 each; sipura SPA2000 = 2 FXS for $100, $50 each)
Order one TDM04B (4 FXO) and 1 Sipura 2000 for each 2 analog extensions. You can also mix it up, lets say drop two incoming lines and order a 2 FXS, 2 FXO TDM instead. Then subscribe to a VoIP provider like Voicepulse Connect to dial-out through IAX. Use the FXS on the TDM for fax machines and make these the only to dial-out through the analog outgoing lines. Of course, I would rather buy some IP phones instead of analog ones through FXS. For your remote office, depending on the number of extensions, you can either setup a small asterisk box or just use Sipuras 3000 and 2000 connected to your main office's asterisk server. On Tue, 24 Aug 2004 17:11:36 -0600, Andrew Elchuk <[EMAIL PROTECTED]> wrote: > Hi > > I am interested in setting up an Asterisk PBX in my office with digium > hardware, and I just have a few questions in regards to what I would > need. It is my understanding that an FXO card is used to interface with > an incoming/outgoing phone line, and an FXS card is used for interfacing > with a phone within the system. Currently we have 4 incoming/outgoing > phone lines and would like to have 20 phones in the system. In order to > accomodate this, would you either reccommend having 1 TDM04B (4 FXO > modules on it) for the 4 incoming/outgoing lines, and 5 TDM40B (4 FXS > modules on each) for the 20 phones we would have in the system. Or > would you reccommend 1 TDM04B for the 4 incoming/outgoing lines, and a > T100P connected to a channel bank of some sort to connect to the > internal phones? If you reccommend the T100P and channel bank, where do > you suggest I get an FXS channel bank? Please let me know if I got any > of this mixed up (like if I got the FXO and FXS cards mixed up) and > thank you in advance for your help in us deciding the hardware we need > for a new PBX. > > Andrew Elchuk > > P.S. We are currently using an X-Like software phone with a free world > dialup account for communication with our other office in a different > city. My question is if I can configure the extensions.conf to connect > to a free world dialup number when executing a dial command, or would I > need to edit sip.conf as well or some other configs or is that even > possible? Thank you again. > > -- > Andrew Elchuk > Technical Associate > Cronus Technologies > 248 - 111 Research Drive > Saskatoon, SK S7N 2X8 > Tel: (306) 652-5798 ext. 112 > Fax: (306) 652-5799 > Toll Free: 1-877-655-5798 > http://www.cronustech.com > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
