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r117086 | tilghman | 2008-05-19 11:05:05 -0500 (Mon, 19 May 2008) | 2 lines
The addition of usleep(2) within ast_assert requires the inclusion of the unistd.h header
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file, line, and function name twice.
(Closes issue #12650)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) | 4 lines
Add ast_assert(), which can be used to handle fatal errors. It is only compiled
in if dev-mode is enabled, and only aborts if DO_CRASH is defined.
(inspired by issue #12650)
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in text stream
Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116088 | mmichelson | 2008-05-13 18:47:49 -0500 (Tue, 13 May 2008) | 12 lines
A change to the way channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.
After debugging a deadlock, it was noticed that when DEBUG_CHANNEL_LOCKS
is enabled in menuselect, the actual origin of channel locks is obscured
by the fact that all channel locks appear to happen in the function
ast_channel_lock(). This code change redefines ast_channel_lock to be a
macro which maps to __ast_channel_lock(), which then relays the proper
file name, line number, and function name information to the core lock
functions so that this information will be displayed in the case that
there is some sort of locking error or core show locks is issued.
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marked "urgent" are considered to be higher priority than other messages
and so they will be played before any other messages in a user's mailbox.
There are two ways to leave an urgent message.
1. send the 'U' option to VoiceMail().
2. Set review=yes in voicemail.conf. This will give instructions for
a caller to mark a message as urgent after the message has been recorded.
I have tested that this works correctly with file and ODBC storage, and James
Rothenberger (who wrote initial support for this feature) has tested its use
with IMAP storage.
(closes issue #11817)
Reported by: jaroth
Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
Tested by: putnopvut, jaroth
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ENUM* functions. The TXCIDNAME lookup function has also gotten a
new paramater. The new options for ENUM* functions include 'u', 's', 'i', and 'd' which return the full uri, trigger isn specific rewriting, look
for branches into an infrastructure enum tree, or do a direct dns lookup of a number respectively. The new paramater for TXCIDNAME adds a
zone-suffix argument for looking up caller id's in DNS that aren't e164.arpa.
This patch is based on the original code from otmar, modified by snuffy, and tested by jtodd, me, and others.
(closes issue #8089)
Reported by: otmar
Patches:
20080508_bug8089-1.diff
- original code by otmar (license 480),
- revised by snuffy (license 35)
Tested by: oej, otmar, jtodd, Corydon76, snuffy, alexnikolov, bbryant
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115579 | file | 2008-05-09 13:34:08 -0300 (Fri, 09 May 2008) | 2 lines
Improve res_ninit and res_ndestroy autoconf logic on the Darwin platform.
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r115512 | russell | 2008-05-07 11:24:09 -0500 (Wed, 07 May 2008) | 11 lines
Merged revisions 115511 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008) | 3 lines
Remove remnants of dlinkedlists. I didn't actually use them in the final version
of my IAX2 improvements.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115327 | file | 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) | 2 lines
Make sure that either the main speex library contains preprocess functions or that speexdsp does. If both fail then speex stuff can not be built.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115308 | tilghman | 2008-05-05 14:55:55 -0500 (Mon, 05 May 2008) | 2 lines
Err, the documentation on the return value of ast_odbc_backslash_is_escape is exactly backwards.
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r115279 | file | 2008-05-03 22:50:59 -0300 (Sat, 03 May 2008) | 2 lines
For my next trick I will make these work with what our autoconf header file gives us.
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processing thread. Modules reference a taskprocessor, push tasks into the taskprocessor as needed, and unreference the taskprocessor when the taskprocessor is no longer needed.
A task wraps a callback function pointer and a data pointer and is managed internal to the taskprocessor subsystem. The callback function is responsible for releasing task data.
Taskprocessor API
* ast_taskprocessor_get(..) - returns a reference to a taskprocessor
* ast_taskprocessor_unreference(..) - releases reference to a taskprocessor
* ast_taskprocessor_push(..) - push a task into a taskprocessor queue
Check doxygen for more details
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r115196 | mmichelson | 2008-05-02 09:28:19 -0500 (Fri, 02 May 2008) | 6 lines
Clarify a comment that was, well, just wrong. It turns out that
ignoring the way that macros expand. Instead, I have clarified in the
comment why the macro will work even if the scheduler id for the
task to be deleted changes during the execution of the macro.
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deprecated function)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115102 | tilghman | 2008-05-01 18:20:25 -0500 (Thu, 01 May 2008) | 2 lines
Change the comment of deprecated to an actual compiler deprecation
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(closes issue #10540)
Reported by: spendergrass
Patches:
20080417__bug10540.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
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and denoising to a channel, AGC() and DENOISE(). Also included, is a change
to the audiohook API to add a new function (ast_audiohook_remove) that can
remove an audiohook from a channel before it is detached.
This code is based on a contribution from Switchvox.
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the spied-on
party to be spoken instead of the channel name or number.
This was accomplished by adding a new function pointer to point to a function in app_voicemail
which retrieves the name file and plays it. This makes for an easy way that applications may play
a user's name should it be necessary. app_directory, in particular, can be simplified greatly by
this change.
This change comes as a suggestion from Switchvox, which already has this feature. AST-23
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(closes issue #12351)
Reported by: Corydon76
Patches:
20080402__pbx_incomplete__3.diff.txt uploaded by Corydon76 (license 14)
pbx_incomplete_with_timeout.diff uploaded by fabled (license 448)
Tested by: Corydon76, fabled
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(closes issue #11328)
Reported by: rain
Patches:
20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14)
brought up-to-date to trunk by me
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114591 | russell | 2008-04-23 12:55:31 -0500 (Wed, 23 Apr 2008) | 5 lines
Store the manager session ID explicitly as 4 byte ID instead of a ulong. The
mansession_id cookie is coded to be limited to 8 characters of hex, and this
could break logins from 64-bit machines in some cases.
(inspired by AST-20)
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This fixes a bug that was thought to be fixed already.
app_voicemail, if using IMAP_STORAGE, has a problem because
the IMAP header files include syslog.h, which define LOG_WARNING
and LOG_DEBUG to be different than what Asterisk uses for those
same macros. This was "fixed" in the past by including all the
IMAP header files prior to including asterisk.h. This fix worked...
unless you were to try to compile with MALLOC_DEBUG. MALLOC_DEBUG
prepends the inclusion of astmm.h to every file, which means that no
matter what order the includes are in in app_voicemail, the unexpected
values for LOG_WARNING and LOG_DEBUG will be in place.
The action taken for this fix was to define AST_LOG_* macros in addition
to the LOG_* macros already defined. These new macros are used in app_voicemail.c,
logger.h, and astobj.h right now, and their use will be encouraged in the future.
In consideration of those who have written third-party modules which use
the LOG_* macros, these will NOT be removed from the source, however future use
of these macros is discouraged.
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(closes issue #12490)
Reported by: bcnit
Patches:
12490-queuevars-3.diff uploaded by qwell (license 4)
Tested by: qwell
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Reported by: oej
Tested by: jpeeler
This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.
Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.
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r114207 | mmichelson | 2008-04-17 11:28:03 -0500 (Thu, 17 Apr 2008) | 12 lines
It was possible for a reference to a frame which was part of a freed DSP to still be
referenced, leading to memory corruption and eventual crashes. This code change ensures
that the dsp is freed when we are finished with the frame. This change is very similar
to a change Russell made with translators back a month or so ago.
(closes issue #11999)
Reported by: destiny6628
Patches:
11999.patch uploaded by putnopvut (license 60)
Tested by: destiny6628, victoryure
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in one place).
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the mentioned lock has been acquired, this routine will log to the console the normal info about that lock you'd see from the CLI when you do a 'core show locks'. It's solely for debug-- if the lock is NOT acquired, there is no output. I use it to show 'unexpected' locks, to see where/why a lock is pre-locked. This command is to be called from points of interest, like just before a trylock, and helps to spot fleeting, highly temporal locks that normally are not locked...
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eating up lots of cpu cycles. See CHANGES. From the team/murf/bug11210 branch.
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feature, and various documentation updates in astobj2.h, and the addition of standalone utility, refcounter, that will filter the trace output for unbalanced, unfreed objects. This comes from the team/murf/bug11210 branch.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114051 | mmichelson | 2008-04-10 15:59:49 -0500 (Thu, 10 Apr 2008) | 3 lines
Fix 1.4 build when LOW_MEMORY is enabled.
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be used by modules.
(closes issue #12384)
Reported by: fnordian
Patches:
features.patch uploaded by fnordian (license 110)
(patch modified by me, to give FEATURE_RETURN_* an AST_ prefix)
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recording.
(closes issue #11962)
Reported by: garlew
Patches:
recording.patch uploaded by garlew (license 376)
bug-11962.diff uploaded by snuffy (license 35)
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for a given domain, the hostname and port listed in the SRV record will be used. If no SRV record exists or a SRV lookup is not attempted, the DNS lookup on the specified domain will be performed as normal. Chan_sip has been modified to take advantage of the new SRV support.
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r112125 | file | 2008-04-01 13:45:14 -0300 (Tue, 01 Apr 2008) | 5 lines
Ensure that we do not exceed the hold's maximum size with a single frame.
(closes issue #12047)
Reported by: fabianoheringer
Tested by: fabianoheringer
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thread-safe.
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the proper solution to the problem reported anyway.
(related to issue #12884)
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terminator
for the string to be copied into. This resulted in parse_request reading invalid
memory beyond the end of the string, and in some cases led to crashes. Thanks
to falves11 for providing the valgrind output which led to the closure of this issue.
(closes issue #12284)
Reported by: falves11
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r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | 15 lines
(closes issue #12302)
Reported by: pj
Tested by: murf
These changes will set a channel variable ~~EXTEN~~ just before generating code
for a switch, with the value of ${EXTEN}. The exten is marked as having a switch,
and ever after that, till the end of the exten, we substitute any ${EXTEN}
with ${~~EXTEN~~} instead in application arguments; (and the ${EXTEN: also).
The reason for this, is that because switches are coded using
separate extensions to provide pattern matching, and
jumping to/from these switch extensions messes up the ${EXTEN} value,
which blows the minds of users.
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Per comments from dimas:
1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones.
2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before.
3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best.
Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too.
4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used.
5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel.
This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality.
(closes issue #11968)
Reported by: dimas
Patches:
v2-11968-dsp.patch uploaded by dimas (license 88)
v4-11968-zap.patch uploaded by dimas (license 88)
Tested by: dimas, qwell
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to reload any file that references a given configuration file.
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r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines
Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet
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