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- 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
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3
CHANGES
3
CHANGES
@@ -92,9 +92,6 @@ SIP changes
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* The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
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states it is not needed. For phones, however, that do require it the "registertrying" option
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has been added so it can be enabled.
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* The "call-limit" option is marked as deprecated. It still works in this version of
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Asterisk, but will be removed in the following version. Please use the groupcount functions
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in the dialplan to enforce call limits.
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* A new option called "callcounter" (global/peer/user level) enables call counters needed
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for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
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used to enable this functionality).
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11
UPGRADE.txt
11
UPGRADE.txt
@@ -47,6 +47,9 @@ Core:
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'rotatestrategy'. This new option supports a 'rotate' strategy that more
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closely mimics the system logger in terms of file rotation.
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* The concise versions of various CLI commands are now deprecated. We recommend
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using the manager interface (AMI) for application integration with Asterisk.
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Voicemail:
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* The voicemail configuration values 'maxmessage' and 'minmessage' have
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@@ -106,14 +109,18 @@ Formats:
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Channel Drivers:
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* chan_sip.c: a small upgrade to support the "Record" button on the SNOM360,
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* SIP: a small upgrade to support the "Record" button on the SNOM360,
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which sends a sip INFO message with a "Record: on" or "Record: off"
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header. If asterisk is set up (via features.conf) to accept "One Touch Monitor"
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header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor"
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requests (by default, via '*1'), then the user-configured dialpad sequence
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is generated, and recording can be started and stopped via this button. The
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file names and formats are all controlled via the normal mechanisms. If the
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user has not configured the automon feature, the normal "415 Unsupported media type"
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is returned, and nothing is done.
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* SIP: The "call-limit" option is marked as deprecated. It still works in this version of
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Asterisk, but will be removed in the following version. Please use the groupcount functions
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in the dialplan to enforce call limits. The "limitonpeer" configuration option is
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now renamed to "counteronpeer".
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* chan_local.c: the comma delimiter inside the channel name has been changed to a
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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
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case CLI_INIT:
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e->command = "meetme";
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e->usage =
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"Usage: meetme concise|(un)lock|(un)mute|kick|list [concise] <confno> <usernumber>\n"
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"Usage: meetme (un)lock|(un)mute|kick|list [concise] <confno> <usernumber>\n"
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" Executes a command for the conference or on a conferee\n";
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return NULL;
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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
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" 'concise' is specified, the format is abridged and in a more easily\n"
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" machine parsable format. If 'verbose' is specified, the output includes\n"
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" more and longer fields. If 'count' is specified only the channel and call\n"
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" count is output.\n";
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" count is output.\n"
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" The 'concise' option is deprecated and will be removed from future versions\n"
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" of Asterisk.\n";
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return NULL;
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case CLI_GENERATE:
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