- 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: https://origsvn.digium.com/svn/asterisk/trunk@89606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2007-11-26 19:24:23 +00:00
parent 0619fb1248
commit 130a2051fa
4 changed files with 13 additions and 7 deletions

View File

@@ -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).

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@@ -47,6 +47,9 @@ Core:
'rotatestrategy'. This new option supports a 'rotate' strategy that more
closely mimics the system logger in terms of file rotation.
* The concise versions of various CLI commands are now deprecated. We recommend
using the manager interface (AMI) for application integration with Asterisk.
Voicemail:
* The voicemail configuration values 'maxmessage' and 'minmessage' have
@@ -106,14 +109,18 @@ Formats:
Channel Drivers:
* chan_sip.c: a small upgrade to support the "Record" button on the SNOM360,
* SIP: a small upgrade to support the "Record" button on the SNOM360,
which sends a sip INFO message with a "Record: on" or "Record: off"
header. If asterisk is set up (via features.conf) to accept "One Touch Monitor"
header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor"
requests (by default, via '*1'), then the user-configured dialpad sequence
is generated, and recording can be started and stopped via this button. The
file names and formats are all controlled via the normal mechanisms. If the
user has not configured the automon feature, the normal "415 Unsupported media type"
is returned, and nothing is done.
* SIP: 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. The "limitonpeer" configuration option is
now renamed to "counteronpeer".
* chan_local.c: the comma delimiter inside the channel name has been changed to a
semicolon, in order to make the Local channel driver compatible with the comma

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@@ -963,7 +963,7 @@ static char *meetme_cmd(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a
case CLI_INIT:
e->command = "meetme";
e->usage =
"Usage: meetme concise|(un)lock|(un)mute|kick|list [concise] <confno> <usernumber>\n"
"Usage: meetme (un)lock|(un)mute|kick|list [concise] <confno> <usernumber>\n"
" Executes a command for the conference or on a conferee\n";
return NULL;
case CLI_GENERATE:

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@@ -592,7 +592,9 @@ static char *handle_chanlist(struct ast_cli_entry *e, int cmd, struct ast_cli_ar
" 'concise' is specified, the format is abridged and in a more easily\n"
" machine parsable format. If 'verbose' is specified, the output includes\n"
" more and longer fields. If 'count' is specified only the channel and call\n"
" count is output.\n";
" count is output.\n"
" The 'concise' option is deprecated and will be removed from future versions\n"
" of Asterisk.\n";
return NULL;
case CLI_GENERATE: