diff options
| author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-11-26 19:24:23 +0000 |
|---|---|---|
| committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-11-26 19:24:23 +0000 |
| commit | d33873fadedf231a92c67bcca8d488d7611bffd6 (patch) | |
| tree | b66533550538f9fcc69059d42ca9ee0921abf701 /CHANGES | |
| parent | 78ee740bbf12bd9ed01ee04c6bfc81bc944f0a47 (diff) | |
- Mark "concise" as deprecated
- Restructure other changes to UPGRADE.txt and CHANGES
We're still looking for scripts that replace
asterisk -rx "show shannels concise"
by using the manager interface, but still produces the same output.
Anyone?
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89606 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'CHANGES')
| -rw-r--r-- | CHANGES | 3 |
1 files changed, 0 insertions, 3 deletions
@@ -92,9 +92,6 @@ SIP changes * 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 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). |