Steve Murphy [Wed, 24 Jan 2007 20:35:54 +0000 (20:35 +0000)]
As per bug 8859 (Add option to revert old ChanIsAvail() with 's' option behavior), this update makes the 't' option available, which calls ast_parse_device_state instead of ast_device_state. This option will not dive into the channel driver to find the status of the device (which could be good if sip devicestate isn't returning full status, for various reasons).
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r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1 line
updated check_expr via 8322 (refactoring of expression checking impl); elfring contributed a nice code reorg, I contributed some time to get it working again, better messages
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Fix changing channel formats when joint capability changes and there are no audio formats... I didn't break it originally! (issue #8535 reported by ivoc)
Russell Bryant [Wed, 24 Jan 2007 01:00:57 +0000 (01:00 +0000)]
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r51848 | russell | 2007-01-23 18:59:58 -0600 (Tue, 23 Jan 2007) | 14 lines
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r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) | 6 lines
Fix an issue related to synchronization of recordings when using Monitor().
The bug is a miscalculation of the amount to seek the stream for writing to
disk when the number of samples coming in and out of a channel do not match up.
(issue #8298, #8887, report and patch by guillecabeza, patch files created and
testing done by whoiswes)
Russell Bryant [Wed, 24 Jan 2007 00:21:32 +0000 (00:21 +0000)]
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r51829 | russell | 2007-01-23 18:19:55 -0600 (Tue, 23 Jan 2007) | 12 lines
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r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) | 4 lines
Don't set a new value for the END_ variable on the channel before using the
old value. If you do, it will lead to accessing a memory address that has
been free()'d. (issue #8895, arkadia)
Russell Bryant [Tue, 23 Jan 2007 22:09:23 +0000 (22:09 +0000)]
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r51781 | russell | 2007-01-23 16:04:01 -0600 (Tue, 23 Jan 2007) | 6 lines
Fix some bugs in process_message(). The manager session lock needs to be held
when sending some sort of response, or calling one of the manager action
callbacks. This resolves an issue where people using the GUI would get random
crashes when they start clicking around a lot.
(issue #8711, reported and debugged by zandbelt)
Russell Bryant [Tue, 23 Jan 2007 21:38:31 +0000 (21:38 +0000)]
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r51750 | russell | 2007-01-23 15:33:15 -0600 (Tue, 23 Jan 2007) | 4 lines
When traversing the list of manager actions, the iterator needs to be
initialized to the list head *after* locking the list. Also, lock the actions
list in one place it is being accessed where it was not being done.
Steve Murphy [Tue, 23 Jan 2007 19:11:53 +0000 (19:11 +0000)]
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r51683 | murf | 2007-01-23 11:58:27 -0700 (Tue, 23 Jan 2007) | 1 line
via 8748 (callerid.c loses name when returning PRIVATE_NUMBER flag), the user suggested this mod, saying it would allow 'WITHHELD' to appear in the name field, which would be useful
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Joshua Colp [Tue, 23 Jan 2007 03:15:04 +0000 (03:15 +0000)]
Add SRV Lookup support on outbound calls to chan_iax2. It's listed in the RFC so we might want to support it and please don't hurt me Marko ... (issue #7812 reported by drorlb)
Move filestream creation to Mixmonitor loop. This will prevent a blank file from being created if no frames ever pass through to be recorded. (issue #7589 reported by steve_mcneil)
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.
Russell Bryant [Fri, 19 Jan 2007 19:09:04 +0000 (19:09 +0000)]
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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.
Russell Bryant [Fri, 19 Jan 2007 18:06:03 +0000 (18:06 +0000)]
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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.
Luigi Rizzo [Fri, 19 Jan 2007 18:00:00 +0000 (18:00 +0000)]
As the comment in the diff says:
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().
Russell Bryant [Fri, 19 Jan 2007 17:02:36 +0000 (17:02 +0000)]
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r51302 | russell | 2007-01-19 10:56:17 -0600 (Fri, 19 Jan 2007) | 12 lines
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r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) | 4 lines
Fix a memory leak on command line tab completion. The container for the
matches was freed, but the individual matches themselves were not.
(issue #8851, arkadia)
Russell Bryant [Fri, 19 Jan 2007 16:30:00 +0000 (16:30 +0000)]
Break out of the config processing loop for manager.conf once the correct user
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)
Luigi Rizzo [Fri, 19 Jan 2007 09:44:55 +0000 (09:44 +0000)]
add comments that AC_USE_SYSTEM_EXTENSIONS and AST_PROG_LD
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
Luigi Rizzo [Fri, 19 Jan 2007 09:32:13 +0000 (09:32 +0000)]
simplify checking for zaptel version and location
(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.
issue 7877: chan_zap module reload does not use default/initialized values on subsequent loads. Reset configuration variables to default values prior to parsing configuration file.
Russell Bryant [Thu, 18 Jan 2007 21:57:55 +0000 (21:57 +0000)]
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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)
Copy MOH settings when calling a peer so that if they put someone on hold or get put on hold themselves they get the right music class. (issue #8840 reported by mdu113)
Build the IMAP remote directory string better and properly. Fix an issue with encoding the GSM voicemail when attaching to the voicemail. (issue #8808 reported by akohlsmith)
Russell Bryant [Wed, 17 Jan 2007 23:35:43 +0000 (23:35 +0000)]
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r51205 | russell | 2007-01-17 17:31:11 -0600 (Wed, 17 Jan 2007) | 5 lines
Fix some instances where when loading func_odbc, a double-free could occur.
Also, remove an unneeded error message. If the failure condition is
actually a memory allocation failure, a log message will already be
generated automatically.
Russell Bryant [Wed, 17 Jan 2007 23:31:40 +0000 (23:31 +0000)]
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r51204 | russell | 2007-01-17 16:09:52 -0600 (Wed, 17 Jan 2007) | 4 lines
Instead of dividing the offset by 2 directly, make it more clear that the
offset is being scaled by the size of the elements in the buffer.
(Inspired by a discussing on the asterisk-dev list about this code)
When ast_strip_quoted was called with a zero-length string, it would treat a
NULL as if it were the quoting character (and would thus return the string
in memory immediately following the passed-in string).
Return the correct result when directly writing out a packet so that the core doesn't then decide to handle it the regular way again. (issue #8833 reported by rcourtna)
projects / thirdparty / asterisk.git / log
