[asterisk-users] Getting presence working in 1.6.2

[Jared Smith]

On Fri, 2010-05-07 at 08:25 -0500, Danny Nicholas wrote:
> In which future release of Asterisk are we (since it is open-source, we
> theoretically have "some" control) going to stop renaming and deprecating
> features?

It's obviously more complicated that you make it seem with your comment.
Let me try to explain the history of this particular change.

In earlier versions of Asterisk (1.2, 1.4, 1.6.0 and deprecated but
still working in 1.6.1), you had to set the "call-limit" setting to get
Asterisk to keep track of SIP device state.  The majority of the people
using this call-limit setting set it to an arbitrarily high value (such
as 99) so that it didn't really limit the number of concurrent calls,
but simply turned on SIP device state tracking.  (And, to be honest, it
was a whole lot easier to use the GROUP() and GROUP_COUNT() functions in
the dialplan to enforce arbitrary call limits.)

To make it more clear and less cryptic, we split out the "callcounter"
functionality in sip.conf, so that you could turn on/off the SIP device
state tracking without limiting calls, and encouraged people to use the
GROUP() and GROUP_COUNT() functions in the dialplan to enforce call
limits.

Clear as mud?

--
Jared Smith
Digium, Inc.