- Deprecate "call-limit" in chan_sip. No other channel driver enforces call-limits

  and we now have the groupcount system to implement call-limits in the dialplan. You
  can use the "setvar" option in realtime/sip.conf to set limits per device.

- Implement "callcounter" as a new option to enable the call counting we need to
  report device status to queue, manager and SIP subscriptions.

The call counter setting is now enabled in the code by setting the device call-limit
to 999. When we remove the call limit, we can simply enable this with a boolean
setting.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89554 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 7 insertions, 1 deletions

diff --git a/CHANGES b/CHANGES
index 193f24f80..52d438a81 100644
--- a/CHANGES
+++ b/CHANGES
@@ -89,8 +89,14 @@ SIP changes
   * SIP now adds a header to the CANCEL if the call was answered by another phone
      in the same dial command, or if the new c option in dial() is used.
   * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
-     states it is not needed. For phones, however, that do require it the registertrying option
+     states it is not needed. For phones, however, that do require it the "registertrying" option
      has been added so it can be enabled. 
+  * The "call-limit" option is marked as deprecated. It still works in this version of
+    Asterisk, but will be removed in the following version. Please use the groupcount functions
+    in the dialplan to enforce call limits.
+  * A new option called "callcounter" (global/peer/user level) enables call counters needed
+    for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
+    used to enable this functionality).
 
 IAX2 changes
 ------------