[Asterisk-Users] Silly question: Why multiple lines on SIP phones?

[Paul Dugas]

This is probably going to sound really silly and I must be confused about
it.  Maybe someone can set me straight.

I've been tinkering for a while with * and a number of different FXO/FXS
cards, SIP phones, and ATAs trying to get a feel for what works and what
doesn't.  In the SIP phone group, I have a Cisco 7940G, a Polycom 500, and
now an SPA-841.  Each of these allow me to configure at least 2 "lines". 
What I'm wrestling with is what to do with the second lines.  With the
cisco and the polycom, I have only one line configured and can
successfully conference and transfer calls.  When I'm on the phone and a
new call comes in, they display the second call and controls for switching
between them.  (hmmm... what happens when the third comes along?)  The
Sipura appears to be unable to do this without configuring a second
extension.  (anybody got any insight on that?)  My question is basically,
assuming all I want is the same kind of service I get on a traditional
analog line with respect to call waiting and 3-way, do I need a multi-line
phone?

Back to the Sipura... If I configure it to only register the first line,
it seems it cannot conference or transfer calls.  With my current configs,
a second incoming call to the extension dumps straight to "busy" voice
mail.  I guess I could setup my dial plan to roll to the phone's line 2
but that seems like a mess.  It's not needed with the cisco or polycom so
it's possible, right?

Curious,

Paul

--
Paul A. Dugas                    Dugas Enterprises, LLC
paul at dugasenterprises.com        1711 Indian Ridge Drive
p:404-932-1355  f:770-516-4841   Woodstock, GA 30189-6856 USA