Re: Debian way to essentially call block phone number
On Sat, Jul 17, 2010 at 08:18:25PM +0100, Alan Chandler wrote:
> On 17/07/10 17:37, Mike Viau wrote:
>> Hello list,
>> I was wondering if anyone runs a system that essentially blocks unwanted
>> calls from a list of programmed phone numbers. I have the caller display
>> option on my phones service that returns the phone number of most
>> incoming calls.
>> I came across this idea after reading:
>> If you want to go the freebie route, you can download free software that
>> claims to block unwanted calls. This software works by hooking up your
>> phone to your computer and letting the software screen incoming calls.
>> Some of these downloads require you to have other hardware though. 
>> Is there a package in Debian I should be aware of, something I can run
>> on a personal computer? I was also considering designing embedded system
>> to do the job, but I cringe at the thought of integrating a POTS (Plain
>> Old Telephone System) modem in the embedded world.
Debian includes Asterisk and YATE. That said, both (and FreeSwitch,
mentioned below) are an overkill for a simple home system. That is:
unless you want to add your own logic in at the price of extra
You'll need a telephony adapter. A simple modem will not do, as there's
no standard interface that I'm aware of to use a modem as a voice
Zaptel/DAHDI provides interfaces for (mostly) some specialized hardware.
It is an out-of-tree kernel module, though included in Debian and
(Disclaimer: I work for one company that makes such specialized hardware
and am involved in DAHDI and Asterisk)
Alternatively, if your PSTN is ISDN, you have some other options.
Or you could try switching to a whole VoIP system - replacing your
landline with a SIP "line". Though that adds yet anthoer point of
complexity and faliure.
There's an interesting project around Asterisk on a cheap Blackfin-based
embedded system. See http://www.rowetel.com/ucasterisk/ucasterisk.html
However I tend to avoid the Blackfin, due to the system lacking an MMU
and thus lacking long-term stability. But that's me.
Also, as mentioend by Alan, you can use an external SIP-connected
adapter. In that case I guess you could basically use just about any
> I haven't got it set up right now, but in the past I have run
> freeswitch. Its essentially a better package than asterisk. You can
> download a .deb file from freeswitch.org. You can configure it pretty
> easily to do almost anything you want.
FreeSwitch is a huge package of bundled software. Their tarball includes
everything from arp to whatnot. They seem to avoid system copies on
purpose. Good luck maintaining your system.
> I have some linksys boxes one of which takes standard phones and turns
> them into voip phones and the other of which will connect to a standard
> POTS line and convert the data from it into SIP traffic. PAP2T is the
> one that supports two phones. The other, I don't have the exact name of
> right now, but is a similar type of box to the PAP2T except one of the
> ports connects to the old telephone network.
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