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path: root/comms/asterisk/patches/patch-af
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2006-08-18Changes 1.2.10:adam1-0/+13
* Number of bug fixes * New option to help to avoid a potential denial of service in IAX2 channel driver * Support for TE407P and TE412P quad T1/E1 interface cards
2006-05-31Changes 1.2.8:adam1-38/+0
* Number of bug fixes, including IAX2 channel driver fixes.
2006-04-13Changes 1.2.7:adam1-6/+7
* Important bug fixes * SIP handling * MixMonitor call recording
2006-01-13Update asterisk to version 1.2.1. Many, many bugfixes, and someriz1-16/+33
new features, including support for DUNDi. (http://www.dundi.com/ for more information) The initial framework and porting of this package upgrade was done by Martin J. Laubach, with lots of feature/PLIST fixes by me. DragonFly support added by Joerg Sonnenberger.
2005-09-02Changes 1.0.9:adam1-8/+9
-- fix bug in callerid matching in the dialplan that was introduced in 1.0.8 Changes 1.0.8: -- 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
2005-05-24Fix the build of asterisk on powerpc platforms. Approved by jmcneill.riz1-0/+19