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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2007-11-26 19:24:23 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2007-11-26 19:24:23 +0000
commitd33873fadedf231a92c67bcca8d488d7611bffd6 (patch)
treeb66533550538f9fcc69059d42ca9ee0921abf701 /UPGRADE.txt
parent78ee740bbf12bd9ed01ee04c6bfc81bc944f0a47 (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 'UPGRADE.txt')
-rw-r--r--UPGRADE.txt11
1 files changed, 9 insertions, 2 deletions
diff --git a/UPGRADE.txt b/UPGRADE.txt
index 7d8eb6a42..a8eccb4ea 100644
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -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