Thanks,
Scott
Asterisk can easily do that, coupled to an ATA with an FXO port such as the
Linksys SPA-3102 to interface the telephone line (the existing phone can be
kept operational by connecting it to the SPA-3102's FXS port).
Enzo
Mypeople.com (www.mypeople.com) has a feature that will allow you to
use their VOIP services from any phone.
http://ydoesitmatter.blogspot.com/2006/04/my-remote-feature-in-mypeople.html
Gopi
The SPA-3102 can do that on its own. Easier if your IP phone & your SPA are
connected to the same VoIPSP
Yes, but authenticating the user to prevent calls to the PSTN by other
unauthorized VoIP callers may prove tricky: the caller ID may be faked, PIN
authentication is inconvenient and not so reliable (especially if the VoIPSP
only support inband DTMF), and HTTP Digest Authentication may also fail
depending on the VoIPSP's proxy.
Enzo
How much more secure would Asterisk be, unless callback was employed? I can
only think of two methods of authentication - by CLID or PIN (Three if you
count both)
You can use SIP's "HTTP Authentication" (also called "WWW authentication":
see Section 22 of http://www.ietf.org/rfc/rfc3261.txt). In practice, with
Asterisk you create for your SIP phone (softphone in this case) a
"type=user" or (if you want it to receive calls as well as placing them)
"type=friend" entry in sip.conf, and associate it to a context in
extensions.conf that is not accessible by any unauthenticated device:
[mysoftphone]
context=candialout
type=friend
host=dynamic
secret=sphsecret
Of course, you'll have to define an entry also for the "Line1" section of
the SPA-3102, which, if you are only interested in placing outgoing calls,
might be:
[spa3102l1]
type=peer
host=dynamic
secret=spa3102secret
Then, you place in extensions.conf:
[candialout]
exten => _X.,1,Dial(SIP/${EXTEN}@spa3102l1)
exten => _X.,2,Hangup
...and also configure:
- the SPA-3102 to register as user: spa3102l1, password: spa3102secret
- the softphone to register as user: mysoftphone, password: sphsecret
Now, any incoming call claiming to come from the user "mysoftphone" (with a
SIP header "From: mysoftphone@...") will be challenged by Asterisk to
authenticate (with HTTP Digest Authentication). Only devices that do it with
the password "sphsecret" will trigger the context [candialout] and the
execution of the Dial() command; any other will be rejected.
Enzo
Thanks,
Thomas
http://www.betterphone.org
Thanks,
Scott
1 line clone cards. Check ebay...