Jiri Kuthan <jiri(a)iptel.org> writes:
We will appreciate your feedback -- that's one of
the quickest
ways for us to learn about things deserving improvement.
I'll have it. :) It mostly includes what I see as lack of useful
authorization vs. authentication support. The short story:
I want to be able to say "user graff has passwor foo, and can receive calls
on and dial out using identities sip:7004@isc.org, sip:graff@isc.org, and
sip:michael_graff@isc.org"
What open
source products are people using for voice mail,
I'm not aware of one I could recommend, a reason why we started
developing our own. I hope a beta version will be out by end of
February (may be to optimistic forecast, though). But it may be
just my ignorance -- the asterisk project may perhaps work.
Want assistance? We're an open source shop here, and I might be able to
spend some time on things if there's something already happening.
Columbia university used to develop a conferencing
system, but I'm
not sure what its status is. I personally use mitel hardphones for
3-party conferencing -- the phone has the mixing capability built
in it.
I tried contacting the people who have an "exclusive license from Columbia"
for the code base, but they don't answer. They also don't list a SIP
phone number on their pages.
The PSTN interworking is orthogonal to whether you run
conference
or normal calls...
Yep, we're using a Cisco, or will shortly. We have a 4-line Cisco here
now as a temporary measure.
--Michael