[asterisk-commits] mnicholson: branch mnicholson/asttest r193830 - in /team/mnicholson/asttest: ...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon May 11 22:01:01 CDT 2009
Author: mnicholson
Date: Mon May 11 22:00:54 2009
New Revision: 193830
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=193830
Log:
Merged revisions 191411,191418-191419,191423,191489,191494,191560,191630,191700,191739,191775,191781,191785,191848,191884,191919,191955,191997,192032,192059,192096,192132,192171,192214,192279,192318,192357,192362,192387,192427,192430,192462,192525,192590,192634,192700,192736,192772,192808,192853,192861,192933,192938,193006,193077,193120,193194,193263,193274,193349,193387,193459,193461,193502,193545,193614,193678,193718,193756-193757 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r191411 | kpfleming | 2009-04-30 16:42:35 -0500 (Thu, 30 Apr 2009) | 8 lines
Add buffer and echo canceller control to CHANNEL() dialplan function for DAHDI channels
Adds ability for CHANNEL() dialplan function, when used on DAHDI channels,
to temporarily change the number of buffers and/or the buffer policy, and also
to enable, disable, or switch the echo canceller between FAX/data and voice
modes.
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r191418 | transnexus | 2009-05-01 04:50:11 -0500 (Fri, 01 May 2009) | 2 lines
Made security features optional.
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r191419 | file | 2009-05-01 09:58:59 -0500 (Fri, 01 May 2009) | 2 lines
Drop my IRC nickname.
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r191423 | seanbright | 2009-05-01 10:46:39 -0500 (Fri, 01 May 2009) | 21 lines
Blocked revisions 191041,191422 via svnmerge
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r191041 | seanbright | 2009-04-29 11:23:07 -0400 (Wed, 29 Apr 2009) | 6 lines
Fix a crash in app_queue with very long member lists.
A user reported via #asterisk that with very long lists of members, a crash
occurs in ast_strdupa, so just use a single buffer and ast_copy_string instead
of stack allocating copys of each interface name.
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r191422 | seanbright | 2009-05-01 11:42:48 -0400 (Fri, 01 May 2009) | 7 lines
Move the defintion of the a couple arrays out of loops.
According to Kevin, it is unspecified as to whether a variable defined inside
a block is allocated once by the compiler or for each pass through the block
(loops being the only interesting case), so just define these before we get
into our loop to be sure.
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r191489 | jpeeler | 2009-05-01 13:09:23 -0500 (Fri, 01 May 2009) | 15 lines
Merged revisions 191488 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009) | 9 lines
Fix DTMF not being sent to other side after a partial feature match
This fixes a regression from commit 176701. The issue was that
ast_generic_bridge never exited after the feature digit timeout had elapsed,
which prevented the queued DTMF from being sent to the other side.
This issue was reported to me directly.
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r191494 | tilghman | 2009-05-01 13:18:00 -0500 (Fri, 01 May 2009) | 4 lines
Set debug message back to DEBUG level.
(closes issue #15007)
Reported by: hulber
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r191560 | tilghman | 2009-05-01 15:01:21 -0500 (Fri, 01 May 2009) | 13 lines
Merged revisions 191559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009) | 6 lines
SIP Response 410 maps to cause code 22 (or 23), not 1.
(closes issue #14993)
Reported by: BigJimmy
Patches:
causepatch uploaded by BigJimmy (license 371)
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r191630 | mmichelson | 2009-05-02 05:46:02 -0500 (Sat, 02 May 2009) | 17 lines
Blocked revisions 191628-191629 via svnmerge
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r191628 | mmichelson | 2009-05-02 05:21:00 -0500 (Sat, 02 May 2009) | 8 lines
Move static buffers to outside for loops in app_chanspy.
Similar to seanbright's commit 191422, this moves some static buffers
to be defined outside of for loops since it is undefined if memory
will be re-used or if the stack will grow with each iteration of the
loop.
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r191629 | mmichelson | 2009-05-02 05:45:24 -0500 (Sat, 02 May 2009) | 3 lines
Kevin has informed me that thi sort of thing is not necessary.
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r191700 | seanbright | 2009-05-02 10:45:07 -0500 (Sat, 02 May 2009) | 1 line
Update copyright year to 2009
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r191739 | seanbright | 2009-05-02 11:43:59 -0500 (Sat, 02 May 2009) | 5 lines
Conditional include ioctl's to change EC policy based on DAHDI caps.
This feels like a sane change (wouldn't compile without this addition), but I'm
not intimately familiar with this code.
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r191775 | kpfleming | 2009-05-02 13:39:48 -0500 (Sat, 02 May 2009) | 5 lines
Fix an error in queue_log file rotation optimization code
This code was copy-and-pasted without properly changing references to event_rotate into queue_rotate, so under some conditions the log rotation would rotate queue_log even though it was not necessary.
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r191781 | mmichelson | 2009-05-02 13:54:52 -0500 (Sat, 02 May 2009) | 16 lines
Blocked revisions 191778 via svnmerge
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r191778 | mmichelson | 2009-05-02 13:48:20 -0500 (Sat, 02 May 2009) | 11 lines
Fix a bug which resulted from the Hebrew voicemail commit.
This fixes a case where a certain message could get played twice.
(closes issue #13155)
Reported by: greenfieldtech
Patches:
app_voicemail.c.multi-lang-patch uploaded by greenfieldtech (license 369)
Tested by: greenfieldtech
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r191785 | kpfleming | 2009-05-02 14:02:22 -0500 (Sat, 02 May 2009) | 8 lines
Remove rarely-used event_log/LOG_EVENT support
In discussions today at the Europe Asterisk Developer Meet-Up, we determined that
the event_log was used in only 9 places in the entire tree, and really was not needed
at all. The users have been converted to use LOG_NOTICE, or the messages have been
removed since other messages were already in place that provided the same information.
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r191848 | russell | 2009-05-02 16:15:18 -0500 (Sat, 02 May 2009) | 7 lines
Do a bit of code cleanup.
- convert handling of IE PLTYPEs to switch statements
- add braces to various small blocks
- remove a bit of trailing whitespace
- remove a couple of unnecessary ast_strdupa() uses
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r191884 | russell | 2009-05-03 09:05:10 -0500 (Sun, 03 May 2009) | 2 lines
Remove unnecessary compiler flag
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r191919 | kpfleming | 2009-05-03 09:28:59 -0500 (Sun, 03 May 2009) | 11 lines
Add 'bitflags'-style information elements to event framework
This patch add a new payload type for information elements, a set
of bit flags. The payload is transported as a 32-bit unsigned integer
but when matching is performed between events and subscribers,
the matching is done by using a bitwise AND instead of numeric value
comparison.
Review: http://reviewboard.asterisk.org/r/242/
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r191955 | kpfleming | 2009-05-04 04:57:36 -0500 (Mon, 04 May 2009) | 8 lines
Ensure that by default only one console channel driver is loaded
This configuration file was changed to ensure that only one console channel driver
(chan_oss) is loaded by default, but the change would only work if chan_console
was not built. Now it will work as expected; if chan_alsa or chan_console are built
and installed, they will not be loaded unless explicity requested.
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r191997 | kpfleming | 2009-05-04 07:52:44 -0500 (Mon, 04 May 2009) | 5 lines
Minor changes in test modules
Correct command description in test_sched.c and include asterisk/cli.h in test_skel.c, since it's highly unlikely that a test module will *not* want to provide CLI commands to execute the tests
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r192032 | eliel | 2009-05-04 10:35:35 -0500 (Mon, 04 May 2009) | 3 lines
Do not re-define _POSIX_C_SOURCE if it was already defined.
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r192059 | kpfleming | 2009-05-04 11:24:16 -0500 (Mon, 04 May 2009) | 5 lines
Ensure that astobj2 memory allocations are properly accounted for when MALLOC_DEBUG is used
This commit ensures that all astobj2 allocated objects are properly accounted for in MALLOC_DEBUG mode by passing down the file/function/line information from the module/function that actually called the astobj2 allocation function.
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r192096 | lmadsen | 2009-05-04 12:42:56 -0500 (Mon, 04 May 2009) | 4 lines
Commit documentation changes related to issue #14801.
(issue #14801)
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r192132 | tilghman | 2009-05-04 13:42:56 -0500 (Mon, 04 May 2009) | 6 lines
Pass libraries in LIBS, not LDFLAGS.
(closes issue #14671)
Reported by: Chainsaw
Patches:
asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by Chainsaw (license 723)
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r192171 | tilghman | 2009-05-04 14:29:13 -0500 (Mon, 04 May 2009) | 8 lines
Restore 'asyncagi break' command to 1.6.1 and higher.
(closes issue #14985)
Reported by: nikkk
Patches:
20090428__bug14985.diff.txt uploaded by tilghman (license 14)
20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: nikkk
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r192214 | dvossel | 2009-05-04 17:44:51 -0500 (Mon, 04 May 2009) | 17 lines
Merged revisions 192213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009) | 11 lines
global mohinterpret setting is ignored
mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers.
(closes issue #14728)
Reported by: dimas
Patches:
v1-14728.patch uploaded by dimas (license 88)
Tested by: dimas, dvossel
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r192279 | kpfleming | 2009-05-05 03:51:06 -0500 (Tue, 05 May 2009) | 5 lines
Ensure that string pools allocated to hold stringfields are properly accounted in MALLOC_DEBUG mode
This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function.
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r192318 | kpfleming | 2009-05-05 05:34:19 -0500 (Tue, 05 May 2009) | 5 lines
Properly account for memory allocated for channels and datastores
As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself.
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r192357 | kpfleming | 2009-05-05 08:18:21 -0500 (Tue, 05 May 2009) | 5 lines
Correct some flaws in the memory accounting code for stringfields and ao2 objects
Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that.
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r192362 | kpfleming | 2009-05-05 09:17:18 -0500 (Tue, 05 May 2009) | 5 lines
Add a more efficient way of allocating structures that use stringfields
This commit adds an API call that can be used to allocate a structure along with this stringfield storage in a single allocation.
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r192387 | file | 2009-05-05 09:22:47 -0500 (Tue, 05 May 2009) | 10 lines
Fix a bug with setting t38pt_udptl at the user or peer level.
If an incoming call authenticated as a user or peer and t38pt_udptl was
not set to yes in general then no UDPTL session would be present and any
T38 related things would fail. This commit changes it so that if after
authenticating T38 is enabled but no UDPTL session is present one will be
created.
(issue AST-215)
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r192427 | mattf | 2009-05-05 12:33:42 -0500 (Tue, 05 May 2009) | 1 line
Revert CPC patch for now, until I decide whether or not it all should be merged into libss7/1.0 (It's still in the bug13495 branch and in libss7/trunk)
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r192430 | file | 2009-05-05 12:46:51 -0500 (Tue, 05 May 2009) | 12 lines
Merged revisions 192429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5 lines
Fix a bug where the followme application would continue trying numbers after the caller hung up.
(closes issue #13624)
Reported by: sgenyuk
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r192462 | file | 2009-05-05 13:23:58 -0500 (Tue, 05 May 2009) | 15 lines
Merged revisions 192454 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8 lines
Fix an incorrect assumption that certain values on the channel will always exist when they may not.
The CDR code involved with bridges wrongly assumed that the currently executing application and data
values will always exist. It is possible for this to be false when call forwarding is involved.
(closes issue #14984)
Reported by: gincantalupo
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r192525 | seanbright | 2009-05-05 14:57:49 -0500 (Tue, 05 May 2009) | 18 lines
Merged revisions 192524 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue, 05 May 2009) | 11 lines
Fix Javascript error when using astman.js in Internet Explorer.
Internet Explorer (tested with 7.0) does not like trailing commas on constructs
like object initializers, so get rid of them to avoid some errors.
(closes issue #15026)
Reported by: rajnishgiri
Patches:
bug15026.patch uploaded by seanbright (license 71)
Tested by: seanbright
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r192590 | rmudgett | 2009-05-05 15:54:07 -0500 (Tue, 05 May 2009) | 5 lines
Fixed crashes from issue8824 review board channel locking changes.
The local struct ast_party_connected_line connected_caller variable
was uninitialized when the copy function was called.
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r192634 | file | 2009-05-06 08:34:35 -0500 (Wed, 06 May 2009) | 14 lines
Merged revisions 192633 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 lines
Update some old logic to stop both begin and end DTMF frames from reaching the core if rfc2833 is not enabled.
(closes issue #15036)
Reported by: dimas
Patches:
v1-15036.patch uploaded by dimas (license 88)
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r192700 | file | 2009-05-06 09:35:47 -0500 (Wed, 06 May 2009) | 5 lines
Fix an infinite loop with tab completion of CLI aliases that reference themselves.
(closes issue #15020)
Reported by: junky
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r192736 | file | 2009-05-06 11:09:27 -0500 (Wed, 06 May 2009) | 4 lines
Make the code that prevents an infinite loop from happening into a case insensitive check.
(thanks eliel)
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r192772 | tilghman | 2009-05-06 11:28:02 -0500 (Wed, 06 May 2009) | 7 lines
Add numbers in Urdu, the national language of Pakistan
(closes issue #15034)
Reported by: nasirq
Patches:
ast_say_number_full_ur-patch.c uploaded by nasirq (license 772)
urdu.ods uploaded by nasirq (license 772)
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r192808 | file | 2009-05-06 12:38:51 -0500 (Wed, 06 May 2009) | 10 lines
Fix a bug where a timer would be created but not acknowledged.
This scenario crept up if chan_iax2 was loaded with no configuration file present.
It would create a timer and tell it to go at an interval but the thread that normally
acknowledges it would not be created because no configuration file was present. The timer
will now be closed if no configuration file is present.
(closes issue #15014)
Reported by: madkins
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r192853 | jpeeler | 2009-05-06 17:02:46 -0500 (Wed, 06 May 2009) | 2 lines
If no extension was found in the pattern tree, don't crash.
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r192861 | jpeeler | 2009-05-06 17:17:27 -0500 (Wed, 06 May 2009) | 17 lines
Merged revisions 192858 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009) | 10 lines
Make ParkedCall application stop execution of the dialplan after hang up
Just changed park_exec to always return non-zero. I really wasn't entirely sure
at first if this was a bug. Decided it was since it would be surprising when
not using ParkedCall in the dialplan to hang up and have dialplan execution
continue.
(closes issue #14555)
Reported by: francesco_r
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r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009) | 17 lines
Merged revisions 192932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) | 10 lines
Eliminate repetition of fullcontact during reconstruction.
If the fullcontact field appears in both the sippeers and the
sipregs table, then during reconstruction of the field, it will
otherwise be doubled.
(closes issue #14754)
Reported by: Alexei Gradinari
Patches:
20090506__bug14754.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
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r192938 | tilghman | 2009-05-07 12:13:36 -0500 (Thu, 07 May 2009) | 6 lines
Send DTMF frame before playing back audio.
(closes issue #14858)
Reported by: barryf
Patches:
20090507__bug14858.diff.txt uploaded by tilghman (license 14)
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r193006 | tilghman | 2009-05-07 12:51:13 -0500 (Thu, 07 May 2009) | 7 lines
Second result should not contain data from the first result.
(closes issue #15039)
Reported by: jims
Patches:
20090506__issue15039.diff.txt uploaded by tilghman (license 14)
Tested by: jims
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r193077 | rmudgett | 2009-05-07 17:24:04 -0500 (Thu, 07 May 2009) | 12 lines
Merged revisions 193050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) | 5 lines
Give a more helpful message when an incoming call's dialed extension does not match.
Added the dialed extension and context to the chan_misdn messages warning
that the dialed number cannot be matched in the dialplan.
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r193120 | tilghman | 2009-05-07 18:42:28 -0500 (Thu, 07 May 2009) | 26 lines
Merged revisions 193119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009) | 19 lines
Fix Background within a Macro for FreePBX.
If the single digit DTMF is an extension in the specified context, then
go there and signal no DTMF. Otherwise, we should exit with that DTMF.
If we're in Macro, we'll exit and seek that DTMF as the beginning of an
extension in the Macro's calling context. If we're not in Macro, then
we'll simply seek that extension in the calling context. Previously,
someone complained about the behavior as it related to the interior of a
Gosub routine, and the fix (#14011) inadvertently broke FreePBX
(#14940). This change should fix both of these situations, but with the
possible incompatibility that if a single digit extension does not exist
(but a longer extension COULD have matched), it would have previously
gone immediately to the "i" extension, but will now need to wait for a
timeout.
(closes issue #14940)
Reported by: p_lindheimer
Patches:
20090420__bug14940.diff.txt uploaded by tilghman (license 14)
Tested by: p_lindheimer
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r193194 | kpfleming | 2009-05-08 09:06:15 -0500 (Fri, 08 May 2009) | 13 lines
Merged revisions 193193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) | 7 lines
Make absolute paths for logger channels work properly
(Note: This is not a new feature, it was previously undocumented and broken.)
The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf.
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r193263 | dvossel | 2009-05-08 09:52:19 -0500 (Fri, 08 May 2009) | 15 lines
Merged revisions 193262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009) | 9 lines
"misdn show config" segfaults asterisk, if no MSN lists
(closes issue #14976)
Reported by: alecdavis
Patches:
misdn_config.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, FabienToune
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r193274 | seanbright | 2009-05-08 10:18:40 -0500 (Fri, 08 May 2009) | 2 lines
Fix the spelling of UNAVAILABLE in func_devstate CLI completion.
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r193349 | mmichelson | 2009-05-08 14:50:44 -0500 (Fri, 08 May 2009) | 12 lines
Reset the members' call counts when resetting queue statistics.
This helps to prevent odd scenarios where a queue will claim to have
taken 0 calls, but the members appear to have taken a non-zero amount.
(closes issue #15068)
Reported by: sum
Patches:
patchreset.patch uploaded by sum (license 766)
Tested by: sum
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r193387 | dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines
TCP not matching valid peer.
find_peer() does not find a valid peer when using pvt->recv as the sockaddr_in argument. Because of the way TCP works, the port number in pvt->recv is not what we're looking for at all. There is currently only one place that find_peer searches for a peer using the sockaddr_in argument. If the peer is not found after using pvt->recv (works for UDP since the port number will be correct), a temp sockaddr_in struct is made using the Contact header in the sip_request. This has the correct port number in it.
Review: http://reviewboard.digium.com/r/236/
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r193459 | russell | 2009-05-09 06:30:15 -0500 (Sat, 09 May 2009) | 2 lines
Declare private data as static.
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r193461 | russell | 2009-05-09 06:33:09 -0500 (Sat, 09 May 2009) | 2 lines
Minor documentation update for ast_event_queue().
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r193502 | file | 2009-05-10 12:07:46 -0500 (Sun, 10 May 2009) | 2 lines
Fix a bug where receiving a control frame of subclass -1 would cause certain channels to get hung up.
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r193545 | lmadsen | 2009-05-11 13:01:44 -0500 (Mon, 11 May 2009) | 14 lines
Recorded merge of revisions 193544 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r193544 | lmadsen | 2009-05-11 13:35:17 -0400 (Mon, 11 May 2009) | 7 lines
Document CHANNEL(transfercapability) in CLI documentation.
(issue #15073)
Reported by: pkempgen
Patches:
20090511__issue15073.diff.txt uploaded by tilghman (license 14)
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r193614 | rmudgett | 2009-05-11 14:11:29 -0500 (Mon, 11 May 2009) | 19 lines
Merged revisions 193613 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines
Sent wrong message to clear a call we started if the other end has not responed yet.
In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet),
it is not allowed to clear the call with RELEASE_COMPLETE. It must be
cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer
to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b)
Patches:
chan-misdn-ccstate7.patch uploaded by customer.
JIRA ABE-1862
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r193678 | tilghman | 2009-05-11 14:32:13 -0500 (Mon, 11 May 2009) | 4 lines
Don't nullify an ast_str pointer.
(closes issue #15061)
Reported by: alecdavis
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r193718 | russell | 2009-05-11 17:04:40 -0500 (Mon, 11 May 2009) | 12 lines
Fix some timer state corruption.
In res_timer_timerfd, handle the case that set_rate gets called while a timer
is still in continuous mode. In this case, we want to remember the configured
rate, but not actually set it until continuous mode has been disabled.
Thanks to dvossel for finding and helping to debug the problem.
(closes issue #15080)
Reported by: dvossel
Tested by: dvossel
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r193756 | tilghman | 2009-05-11 17:50:47 -0500 (Mon, 11 May 2009) | 25 lines
Recorded merge of revisions 193755 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009) | 18 lines
Move 300 bytes around on the stack, to make more room for an extension buffer.
This allows more concurrent extensions to be copied for a single voicemail,
without creating a possibility of upsetting existing users, where a dialplan
could run out of stack space where it had run fine before. Alternatively,
we could have allocated off the heap, but that is a larger change and would
have increased the chance for instability introduced by this change.
This is really solved starting in 1.6.0.11, as the use of an ast_str buffer
allows an unlimited number of extensions (up to available memory). We
additionally create a new warning message when the buffer length is exceeded,
permitting administrators to see an issue after the fact, whereas previously
the list was silently truncated.
(closes issue #14739)
Reported by: p_lindheimer
Patches:
20090417__bug14739.diff.txt uploaded by tilghman (license 14)
Tested by: p_lindheimer
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r193757 | tilghman | 2009-05-11 18:04:14 -0500 (Mon, 11 May 2009) | 2 lines
Found and fixed a memory leak
................
Added:
team/mnicholson/asttest/doc/lang/urdu.ods
- copied unchanged from r193757, trunk/doc/lang/urdu.ods
Modified:
team/mnicholson/asttest/ (props changed)
team/mnicholson/asttest/CHANGES
team/mnicholson/asttest/Makefile
team/mnicholson/asttest/apps/app_dial.c
team/mnicholson/asttest/apps/app_directed_pickup.c
team/mnicholson/asttest/apps/app_followme.c
team/mnicholson/asttest/apps/app_forkcdr.c
team/mnicholson/asttest/apps/app_osplookup.c
team/mnicholson/asttest/apps/app_queue.c
team/mnicholson/asttest/apps/app_sms.c
team/mnicholson/asttest/apps/app_verbose.c
team/mnicholson/asttest/apps/app_voicemail.c
team/mnicholson/asttest/autoconf/ast_ext_tool_check.m4
team/mnicholson/asttest/channels/chan_dahdi.c
team/mnicholson/asttest/channels/chan_iax2.c
team/mnicholson/asttest/channels/chan_misdn.c
team/mnicholson/asttest/channels/chan_sip.c
team/mnicholson/asttest/channels/chan_unistim.c
team/mnicholson/asttest/channels/misdn_config.c
team/mnicholson/asttest/configs/logger.conf.sample
team/mnicholson/asttest/configs/modules.conf.sample
team/mnicholson/asttest/configs/osp.conf.sample
team/mnicholson/asttest/configure (contents, props changed)
team/mnicholson/asttest/configure.ac
team/mnicholson/asttest/funcs/func_channel.c
team/mnicholson/asttest/funcs/func_devstate.c
team/mnicholson/asttest/funcs/func_odbc.c
team/mnicholson/asttest/include/asterisk/astobj2.h
team/mnicholson/asttest/include/asterisk/autoconfig.h.in
team/mnicholson/asttest/include/asterisk/channel.h
team/mnicholson/asttest/include/asterisk/datastore.h
team/mnicholson/asttest/include/asterisk/event.h
team/mnicholson/asttest/include/asterisk/event_defs.h
team/mnicholson/asttest/include/asterisk/logger.h
team/mnicholson/asttest/include/asterisk/stringfields.h
team/mnicholson/asttest/main/asterisk.c
team/mnicholson/asttest/main/astobj2.c
team/mnicholson/asttest/main/audiohook.c
team/mnicholson/asttest/main/bridging.c
team/mnicholson/asttest/main/channel.c
team/mnicholson/asttest/main/datastore.c
team/mnicholson/asttest/main/event.c
team/mnicholson/asttest/main/features.c
team/mnicholson/asttest/main/logger.c
team/mnicholson/asttest/main/manager.c
team/mnicholson/asttest/main/pbx.c
team/mnicholson/asttest/main/say.c
team/mnicholson/asttest/main/utils.c
team/mnicholson/asttest/main/xml.c
team/mnicholson/asttest/pbx/pbx_spool.c
team/mnicholson/asttest/res/res_agi.c
team/mnicholson/asttest/res/res_clialiases.c
team/mnicholson/asttest/res/res_timing_timerfd.c
team/mnicholson/asttest/static-http/astman.js
team/mnicholson/asttest/tests/test_sched.c
team/mnicholson/asttest/tests/test_skel.c
Propchange: team/mnicholson/asttest/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/mnicholson/asttest/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/mnicholson/asttest/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Mon May 11 22:00:54 2009
@@ -1,1 +1,1 @@
-/trunk:1-191406
+/trunk:1-193829
Modified: team/mnicholson/asttest/CHANGES
URL: http://svn.asterisk.org/svn-view/asterisk/team/mnicholson/asttest/CHANGES?view=diff&rev=193830&r1=193829&r2=193830
==============================================================================
--- team/mnicholson/asttest/CHANGES (original)
+++ team/mnicholson/asttest/CHANGES Mon May 11 22:00:54 2009
@@ -11,8 +11,6 @@
--- Functionality changes from Asterisk 1.6.2 to Asterisk 1.6.3 -------------
------------------------------------------------------------------------------
-
-
SIP Changes
-----------
* Added preferred_codec_only option in sip.conf. This feature limits the joint
@@ -25,6 +23,10 @@
tlscertfile is searched for both public and private key.
* Added tlsclientmethod option to sip.conf. This allows the protocol for
outbound client connections to be specified.
+ * The sendrpid parameter has been expanded to include the options
+ 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
+ header to be sent (equivalent to setting sendrpid=yes) and setting
+ sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
Applications
------------
@@ -43,6 +45,30 @@
* Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
setting various connected line and redirecting party information.
* The CHANNEL() function now supports the "name" option.
+ * For DAHDI channels, the CHANNEL() dialplan function now
+ supports changing the channel's buffer policy (for the current
+ call only), using this syntax:
+
+ exten => s,n,Set(CHANNEL(buffers)=6,full)
+
+ This would change the channel to the 'full' buffer policy and
+ 6 (six) buffers. Possible options for this setting are the same
+ as those in chan_dahdi.conf.
+ * For DAHDI channels, the CHANNEL() dialplan function now allows
+ the dialplan to request changes in the configuration of the active
+ echo canceller on the channel (if any), for the current call only.
+ The syntax is:
+
+ exten => s,n,Set(CHANNEL(echocan_mode)=off)
+
+ The possible values are:
+
+ on - normal mode (the echo canceller is actually reinitalized)
+ off - disabled
+ fax - FAX/data mode (NLP disabled if possible, otherwise completely
+ disabled)
+ voice - voice mode (returns from FAX mode, reverting the changes that
+ were made when FAX mode was requested)
Queue changes
-------------
@@ -96,16 +122,9 @@
http://svn.digium.com/svn/thirdparty/mISDN/trunk
http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
-Taged versions of the modified mISDN code are available under:
+Tagged versions of the modified mISDN code are available under:
http://svn.digium.com/svn/thirdparty/mISDN/tags
http://svn.digium.com/svn/thirdparty/mISDNuser/tags
-
-SIP channel driver (chan_sip) changes
--------------------------------------------
- * The sendrpid parameter has been expanded to include the options
- 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
- header to be sent (equivalent to setting sendrpid=yes) and setting
- sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
Asterisk Manager Interface
--------------------------
@@ -120,6 +139,13 @@
across all .conf files. All affected sample.conf files have been modified to
reflect this change. Previous options such as 'sslenable' still work,
but options with the 'tls' prefix are preferred.
+
+Logger
+------
+ * The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
+ users of this channel in the tree have been converted to LOG_NOTICE or removed
+ (in cases where the same message was already generated to another channel).
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
------------------------------------------------------------------------------
Modified: team/mnicholson/asttest/Makefile
URL: http://svn.asterisk.org/svn-view/asterisk/team/mnicholson/asttest/Makefile?view=diff&rev=193830&r1=193829&r2=193830
==============================================================================
--- team/mnicholson/asttest/Makefile (original)
+++ team/mnicholson/asttest/Makefile Mon May 11 22:00:54 2009
@@ -306,7 +306,7 @@
MOD_SUBDIRS_MENUSELECT_TREE:=$(MOD_SUBDIRS:%=%-menuselect-tree)
ifneq ($(findstring darwin,$(OSARCH)),)
- ASTCFLAGS+=-D__Darwin__ -fnested-functions
+ ASTCFLAGS+=-D__Darwin__
SOLINK=-dynamic -bundle -undefined suppress -force_flat_namespace
else
# These are used for all but Darwin
Modified: team/mnicholson/asttest/apps/app_dial.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/mnicholson/asttest/apps/app_dial.c?view=diff&rev=193830&r1=193829&r2=193830
==============================================================================
--- team/mnicholson/asttest/apps/app_dial.c (original)
+++ team/mnicholson/asttest/apps/app_dial.c Mon May 11 22:00:54 2009
@@ -864,6 +864,8 @@
#endif
struct ast_party_connected_line connected_caller;
struct ast_str *featurecode = ast_str_alloca(FEATURE_MAX_LEN + 1);
+
+ ast_party_connected_line_init(&connected_caller);
if (single) {
/* Turn off hold music, etc */
if (!ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK))
Modified: team/mnicholson/asttest/apps/app_directed_pickup.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/mnicholson/asttest/apps/app_directed_pickup.c?view=diff&rev=193830&r1=193829&r2=193830
==============================================================================
--- team/mnicholson/asttest/apps/app_directed_pickup.c (original)
+++ team/mnicholson/asttest/apps/app_directed_pickup.c Mon May 11 22:00:54 2009
@@ -97,6 +97,7 @@
ast_debug(1, "Call pickup on '%s' by '%s'\n", target->name, chan->name);
connected_caller = target->connected;
+ ast_party_connected_line_init(&target->connected);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
ast_channel_update_connected_line(chan, &connected_caller);
@@ -105,7 +106,6 @@
ast_channel_unlock(chan);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
ast_channel_queue_connected_line_update(chan, &connected_caller);
- ast_party_connected_line_init(&target->connected);
ast_party_connected_line_free(&connected_caller);
if ((res = ast_answer(chan))) {
Modified: team/mnicholson/asttest/apps/app_followme.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/mnicholson/asttest/apps/app_followme.c?view=diff&rev=193830&r1=193829&r2=193830
==============================================================================
--- team/mnicholson/asttest/apps/app_followme.c (original)
+++ team/mnicholson/asttest/apps/app_followme.c Mon May 11 22:00:54 2009
@@ -887,7 +887,7 @@
if (winner)
break;
- if (!caller) {
+ if (!caller || ast_check_hangup(caller)) {
tpargs->status = 1;
ast_free(findme_user_list);
return;
Modified: team/mnicholson/asttest/apps/app_forkcdr.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/mnicholson/asttest/apps/app_forkcdr.c?view=diff&rev=193830&r1=193829&r2=193830
==============================================================================
--- team/mnicholson/asttest/apps/app_forkcdr.c (original)
+++ team/mnicholson/asttest/apps/app_forkcdr.c Mon May 11 22:00:54 2009
@@ -47,7 +47,7 @@
<optionlist>
<option name="a">
<para>Update the answer time on the NEW CDR just after it's been inited.
- The new CDR may have been answered already, the reset that forkcdr does
+ The new CDR may have been answered already. The reset that forkcdr does
will erase the answer time. This will bring it back, but the answer time
will be a copy of the fork/start time. It will only do this if the initial
cdr was indeed already answered.</para>
@@ -64,7 +64,8 @@
reset.</para>
</option>
<option name="e">
- <para>End the original CDR. Do this after all the necc. data.</para>
+ <para>End the original CDR. Do this after all the neccessry data is copied
+ from the original CDR to the new forked CDR.</para>
</option>
<option name="r">
<para>Do <emphasis>NOT</emphasis> reset the new cdr.</para>
Modified: team/mnicholson/asttest/apps/app_osplookup.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/mnicholson/asttest/apps/app_osplookup.c?view=diff&rev=193830&r1=193829&r2=193830
==============================================================================
--- team/mnicholson/asttest/apps/app_osplookup.c (original)
+++ team/mnicholson/asttest/apps/app_osplookup.c Mon May 11 22:00:54 2009
@@ -39,6 +39,7 @@
#include <osp/osp.h>
#include <osp/osputils.h>
+#include <osp/ospb64.h>
#include "asterisk/paths.h"
#include "asterisk/lock.h"
@@ -55,6 +56,7 @@
/* OSP Buffer Sizes */
#define OSP_INTSTR_SIZE ((unsigned int)16) /* OSP signed/unsigned int string buffer size */
#define OSP_NORSTR_SIZE ((unsigned int)256) /* OSP normal string buffer size */
+#define OSP_KEYSTR_SIZE ((unsigned int)1024) /* OSP certificate string buffer size */
#define OSP_TOKSTR_SIZE ((unsigned int)4096) /* OSP token string buffer size */
#define OSP_TECHSTR_SIZE ((unsigned int)32) /* OSP signed/unsigned int string buffer size */
#define OSP_UUID_SIZE ((unsigned int)16) /* UUID size */
@@ -127,7 +129,7 @@
char privatekey[OSP_NORSTR_SIZE]; /* OSP private key file name */
char localcert[OSP_NORSTR_SIZE]; /* OSP local cert file name */
unsigned int cacount; /* Number of cacerts */
- char cacerts[OSP_MAX_CERTS][OSP_NORSTR_SIZE]; /* Cacert file names */
[... 3625 lines stripped ...]
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