diff options
-rw-r--r-- | CHANGES | 24 | ||||
-rw-r--r-- | UPGRADE.txt | 5 |
2 files changed, 12 insertions, 17 deletions
@@ -50,12 +50,12 @@ IAX2 Changes Applications ------------ * Added progress option to the app_dial D() option. When progress DTMF is - present, those values are sent immediatly upon receiving a PROGRESS message + present, those values are sent immediately upon receiving a PROGRESS message regardless if the call has been answered or not. * Added functionality to the app_dial F() option to continue with execution at the current location when no parameters are provided. * Added c() option to app_chanspy. This option allows custom DTMF to be set - to cycle through the next avaliable channel. By default this is still '*'. + to cycle through the next available channel. By default this is still '*'. * Added x() option to app_chanspy. This option allows DTMF to be set to exit the application. * The Voicemail application has been improved to automatically ignore messages @@ -87,7 +87,7 @@ Dialplan Functions The possible values are: - on - normal mode (the echo canceller is actually reinitalized) + on - normal mode (the echo canceller is actually reinitialized) off - disabled fax - FAX/data mode (NLP disabled if possible, otherwise completely disabled) @@ -166,12 +166,6 @@ Asterisk Manager Interface reflect this change. Previous options such as 'sslenable' still work, but options with the 'tls' prefix are preferred. -Logger ------- - * The rarely used 'event_log' and LOG_EVENT channel have been removed; the few - users of this channel in the tree have been converted to LOG_NOTICE or removed - (in cases where the same message was already generated to another channel). - Channel Event Logging --------------------- * A new interface, CEL, is introduced here. CEL logs single events, much like @@ -505,12 +499,8 @@ SIP Changes as well as periodically updating the IP address. These properties allow for better performance as well as recovery in the event of an IP change. * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve - load/reload of large numbers of peers/users by ~40x (for large lists of peers. - Initially, we saw 4x improvement in call setup/destruction, but at the time - of merging, this gain has disappeared; further research will be done to try - and restore this performance improvement. Astobj2 refcounting is now used - for users, peers, and dialogs. Users are encouraged to assist in regression - testing and problem reporting! + load/reload of large numbers of peers/users by ~40x (for large lists of peers). + These changes also provide performance improvements for call setup and tear down. * Added ability to specify registration expiry time on a per registration basis in the register line. * Added support for T140 RED - redundancy in T.140 to prevent text loss due to @@ -1146,11 +1136,11 @@ Logger changes -------------- * Added rotatestrategy option to logger.conf, along with two new options: "timestamp" which will use the time to name the logger files instead of - sequence number; and "rotate", which rotates the names of the logfiles, + sequence number; and "rotate", which rotates the names of the log files, similar to the way syslog rotates files. * Added exec_after_rotate option to logger.conf, which allows a system command to be run after rotation. This is primarily useful with - rotatestrategry=rotate, to allow a limit on the number of logfiles kept + rotatestrategry=rotate, to allow a limit on the number of log files kept and to ensure that the oldest log file gets deleted. * Added realtime support for the queue log diff --git a/UPGRADE.txt b/UPGRADE.txt index 6272b9948..d670e414d 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -20,11 +20,16 @@ From 1.6.2 to 1.6.3: +* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few + users of this channel in the tree have been converted to LOG_NOTICE or removed + (in cases where the same message was already generated to another channel). + * The usage of RTP inside of Asterisk has now become modularized. This means the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk. If you are not using autoload=yes in modules.conf you will need to ensure it is set to load. If not, then any module which uses RTP (such as chan_sip) will not be able to send or receive calls. + * The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still remains. It now exists within app_chanspy.c and retains the exact same functionality as before. |