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doesn't immediately break when the next version is released.
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-- 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
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as the INSTALL and DEINSTALL scripts no longer distinguish between
the two types of files. Drop SUPPORT_FILES{,_PERMS} and modify the
packages in pkgsrc accordingly.
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USE_GNU_TOOLS -> USE_TOOLS
awk -> gawk
m4 -> gm4
make -> gmake
sed -> gsed
yacc -> bison
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And always is defined as share/examples/rc.d
which was the default before.
This rc.d scripts are not automatically added to PLISTs now also.
So add to each corresponding PLIST as required.
This was discussed on tech-pkg in late January and late April.
Todo: remove the RCD_SCRIPTS_EXAMPLEDIR uses in MESSAGES and elsewhere
and remove the RCD_SCRIPTS_EXAMPLEDIR itself.
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stand a chance of working. Fixes build on NetBSD/amd64 - have not
tested functionality. Approved by jmcneill.
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There are still some features not enabled by default, but this is a
solid foundation upon which to build - a fully-functional PBX can be
built, including PSTN gatewaying using the comms/zaptel-netbsd package.
From the DESCR:
Asterisk is a complete PBX in software. It provides
all of the features you would expect from a PBX and more. Asterisk
does voice over IP in three protocols, and can interoperate with
almost all standards-based telephony equipment using relatively
inexpensive hardware.
Asterisk provides Voicemail services with Directory, Call Conferencing,
Interactive Voice Response, Call Queuing. It has support for
three-way calling, caller ID services, ADSI, SIP and H.323 (as both
client and gateway).
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