https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51350 | qwell | 2007-01-20 00:53:49 -0600 (Sat, 20 Jan 2007) | 5 lines
Fix Italian numeral support in say.conf for "_[2-9]00" case.
"2131" would've translated to something along the lines of (pardon my..Italian {or lack thereof})
"duecentocentotrentuno", which makes no sense at all.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51348 | qwell | 2007-01-20 00:16:06 -0600 (Sat, 20 Jan 2007) | 8 lines
Fix German language support in say.conf
Properly support 21, 31, 41, 51, 61, 71, 81, and 91.
einundzwanzig has the same format as zweiundzwanzig (as do all other "_ZX" spoken numerals)
Fix support for numbers in the 10,000,000 to 99,999,999 range.
Add support for numbers in the 100,000,000 to 999,999,999 range.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19 Jan 2007) | 5 lines
Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are
actually the same thing. So, a digit would have been interpreted incorrectly
here. Since the channel driver will always have the begin and end callbacks
called for a digit, only support the button-down and button-up messages.
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
AST_INLINE_API() is a macro that takes a block of code as an argument.
Using preprocessor #directives in the argument is not supported by all
compilers, and it is a bit of an obfuscation anyways, so avoid it.
As a workaround, define a macro that produces either its argument
or nothing, and use that instead of #ifdef/#endif within the
argument to AST_INLINE_API().
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51312 f38db490-d61c-443f-a65b-d21fe96a405b
has been found so that 'cat' is non-NULL. This way, users.conf is only checked
when necessary. (issue #8852, akohlsmith, committed patch a bit different)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51296 f38db490-d61c-443f-a65b-d21fe96a405b
the location of the header files.
On passing, add a cast to insure -Werror clean compilation
on FreeBSD 6.x, where time_t does not match %ld
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51293 f38db490-d61c-443f-a65b-d21fe96a405b
place, rather than repeating the check on every single file.
Changes to the individual files are coming.
The header file name has been suggested by kevin.
Approved by: kpfleming
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do not work on FreeBSD - presumably they depend on some
auto* feature that is not installed by default.
I am not sure on what is a proper fix. In my local copy
i simply comment them out.
The AST_PROG_LD is a long standing isse, there were attempts
to fix it in the past but probably not enough has been copied
to acinclude.m4, and i had forgotten about it because i
commented out this call in configure.ac long ago
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(for linux, this is functionally equivalent to the previous
method; for FreeBSD, it re-adds inspection in $PREFIX/zaptel.h).
Please wait to regenerate the "configure" file as i have
another few pending changes to configure.ac
Not applicable to 1.4 until acinclude.m4 is also updated.
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r51262 | russell | 2007-01-18 15:54:23 -0600 (Thu, 18 Jan 2007) | 5 lines
Ensure that the locations given to the Asterisk configure script for ncurses,
curses, termcap, or tinfo are further passed along to the editline configure
script. This fixes some cross-compilation environments.
(issue #8637, reported by ovi, patch by me)
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