Asterisk version 1.0.8 släpptes nu ikväll.
Den är mycket efterlängtad och innehåller många buggfixar.
Lite av bugfixarna:
-- chan_zap
-- Asterisk will now also look in the regular context for the fax extension while
executing a macro. Previously, for this to work, the fax extension would have
to be included in the macro definition.
-- On some systems, ALERTING will be sent after PROCEEDING, so code has been
added to account for this case.
-- If no extension is specified on an overlap call, the 's' extension will be used.
-- chan_sip
-- We no longer send a "to" tag on "100 Trying" messages, as it is inappropriate
to do so.
-- We now respond correctly to an invite for T.38 with a "488 Not acceptable here"
-- We now discard saved tags on 401/407 responses in case the provider we're talking
to tries to pull a dirty trick on us and change it.
-- rtptimeout options will now be correctly set on a peer basis rather than only global
-- chan_mgcp
-- Fixed setting of accountcode
-- Fixed where *67 to block callerid only worked for first call
-- chan_agent
-- We now will not pass audio until the agent has acked the call if the configuration
is set up for the agent to do so.
-- chan_alsa
-- Fixed problems with the unloading of this module
-- res_agi
-- A fix has been added to prevent calls from being hung up when more than one
call is executing an AGI script calling the GET DATA command.
-- AGI scripts will now continue to run even if a file was not found with the
GET DATA command.
-- When calling SAY NUMBER with a number like 09, we will now say "nine" instead
of "zero"
-- app_dial
-- There was a problem where text frames would not be forwarded before the channel
has been answered.
-- app_disa
-- Fixed the timeout used when no password is set
-- app_queue
-- Distinctive ring has been fixed to work for queue members
-- rtp
-- Fixed a logic error when setting the "rtpchecksums" option
-- say.c
-- A problem has been fixed with saying the date in Spanish.
-- Makefile
-- A line was missing for the autosupport script that caused "make rpm" to fail
-- format_wav_gsm
-- Fixed a problem with wav formatting that prevented files from being played
in some media players
-- pbx_spool
-- Fixed if the last line of text in a file for the call spool did not contain
a new line, it would not be processed
-- logger
-- Fixed the logger so that color escape sequences wouldn't be sent to the logs
-- format_sln
-- A lot of changes were made to correctly handle signed linear format on
big endian machines
-- asterisk.conf
-- fix 'highpriority' option for asterisk.conf
Mer information på: http://dev.asteriskdocs.org/
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