[Serusers] ser up and running, now on to voice mail / conferencing

[Michael Graff]

Jiri Kuthan <jiri at 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 at isc.org, sip:graff at isc.org, and
sip:michael_graff at 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