- The FXO and FXS modules are sold separately and also come bundled with the pci card. The modules snap into sockets on the telephone card. (I have 4-port "Wildcard TDM400P" cards with FXO and FXS modules). The modules are
configured with zaptel and can also be tuned with fxotune. If you encounter echo problems, I'd recommend OSLEC.
Since installing OSLEC I've had no echo problems on my asterisk server.
The FXO modules are used to connect your incoming POTS lines to your pbx server. The FXS modules
are used to connect an ordinary analog telephone to your pbx server (hence these modules are not needed
if you plan on using softphones or ipphones. You need one FXO module for each telephone line (number).
Digium sells a TDM01B pci card with one FXO module for ~$150:
http://www.digiumcards.com/digium_cards_fxo.html
Sangoma cards are cheaper but I have not tried them.
- These cards function as a regular modem. In fact, I use mine with hylafax to both send/receive faxes
I simply have a line in /etc/inittab:
f0
respawn:/usr/sbin/faxgetty /dev/ttyIAX0
where /dev/ttyIAX0 is my FXO port.
- Asterisk and Callweaver are free. FreePBX is a php application which simplifies writing an asterisk dialplan.
I bought a couple FXS modules to use with my old analog phones in an emergency, but honestly the FXS modules
are a probably a waste of money. It would be better to apply that $50 to a good ipphone. Snom phones can be bought on eBay for $50-100. I've bought several used Snom360 for ~$70-90 on eBay (retail $@225+) and I've also bought some
Snom320 phones for ~$50 (new ~$100). There are lots of free SIP and IAX softphones if you don't mind the quality of
your soundcard's speaker/microphone. Some examples are Zoiper, Xtensoftphone, Kiax. But I prefer a phone with a handset and buttons and lights.
- PRICES
- PentiumIII or faster with floating point hardware (Intel/AMD instead of Celeron/Sempron) ~$100
- Telephone card + FXO module ~$150
- softphone/ipphone $0/$75
- router ~$50 (for connecting your computer with your ipphone
VOIP Adapters (Cisco/Linksys SPA3102, Sipura SPA-3000, IAXy)
These adapters allow you to make voip phone calls over the internet. Your phone must register itself with a VOIP provider. When you make a phone call with one of these devices, your call will be converted to SIP format and routed over the internet (instead of your phone company's dedicated backbone). Your VOIP provider will be running some kind of pbx software, e.g. Vonage runs asterisk. These adapters have FXS modules to allow you to use your old analog telephone as an ipphone to place calls over the internet. There is a bypass switch to allow you to place local POTS calls.
You do not need a telephone pci card if you do not plan on having a trunk to your local POTS. If you only plan on making ipphone calls over your LAN or internet, then you do not need to convert digital SIP/IAX protocol to an analog signal for your local POTS line. For each POTS line you wish to connect to asterisk, you need to connect an RJ11 jack to the corresponding FXO port on your pci card. (If you have RJ14 lines, you will need to split the double lines into two single RJ11 lines)
But I thought you wanted an answering machine which would pickup your local POTS and take voicemail messages.
Of course you could buy an answering machine, but an asterisk pbx is much more flexible since you can program your own IVR. And asterisk can distinguish between voice calls and fax calls and handle each appropriately.
In summary, if you wish to make ipphone calls, you need
- pbx (You could run asterisk yourself or purchase commercial voip service, e.g. Vonage)
- ipphone If you only have old analog phones, you need FXS port to allow ADC conversion and connection to your pbx provider. The SPA3102 and SPA-3000 are devices which allow you to salvage your old analog phone instead of trashing it. If you have an ipphone, then you simply configure your phone's VOIP URL, account and password. After plugging your ipphone into your LAN, your phone will register itself with the configured VOIP provider and you can now make voip phone calls.
- Trunk Termination If you wish to make voip calls to your local telco (terminate phone calls), you need to setup an interface (TRUNK) between your voip system and the local POTS. You can pay a VOIP-provider for this service, or you can buy a telephone pci card with FXO modules for each line of your trunk. You can configure asterisk to allow/disallow access to your POTS trunk. You could configure asterisk to only allow registered ipphones on your LAN to use your POTS trunk. Or you could allow any registered ipphone anywhere on the net to use your local POTS trunk. You could even charge for access to your POTS trunk (which is what commercial VOIP providers do)
respawn:/usr/sbin/faxgetty /dev/ttyIAX0