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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) | 3 lines
This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed.
Fixed for snuff-home on -dev channel.
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Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred
integer to indicate the number of pages transferred (so far) during the fax
session. The spandsp-0.0.6pre4 release removed the pages_transferred integer
and replaced it with two different integers - pages_tx and pages_rx. This
revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards
compatibility for previous spandsp releases.
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r177696 | dvossel | 2009-02-20 14:17:37 -0600 (Fri, 20 Feb 2009) | 8 lines
Fixes issue with undefined audio codecs in chan_iax2
During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.4 only audio codec bits 0-12 are defined, leaving bits 13-15 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-15 are not defined, these bits are never turned off. In trunk, bits 13-15 are defined, which means 1.4 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities.
(closes issue #14283)
Reported by: jcovert
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(closes issue #14231)
Reported by: jcovert
Patches:
20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14)
corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551)
Tested by: jcovert
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in copy_request
(issue #14478)
Reported by: erik_dedecker
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https://origsvn.digium.com/svn/asterisk/branches/1.4
Trunk was already pretty 8-bit clean; but I'm still
removing the --full from the flex command so everything
is uniform.
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r177540 | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines
This patch fixes a problem with 8-bit input to the ast_expr2 scanner.
The real culprit was the --full argument to flex
in the Makefile! This causes a 7-bit scanner to be
generated.
I reviewed the rules and found one rule where I needed
to specifically include 8-bit chars for a token.
I tested against the text supplied by ibercom, and
all looks very well.
This has been there a surprisingly long time!
(closes issue #14498)
Reported by: ibercom
Patches:
14498.patch uploaded by murf (license 17)
Tested by: murf
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009) | 7 lines
Fix up potential crashes, by reducing the sharing between interactive and non-interactive threads.
(closes issue #14253)
Reported by: Skavin
Patches:
20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
Tested by: Skavin
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trunk. Will do some
testing to make sure this works properly, but from reading the code, it does seem to work
as it should.
Blocked revisions 177450 via svnmerge
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r177450 | oej | 2009-02-19 19:58:57 +0100 (Tor, 19 Feb 2009) | 2 lines
Force a MWI notification after subscribe request. Reported by the Resiprocate dev team. Thanks!
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 lines
If we are able to create a speech structure unset the ERROR variable in case it was previously set.
(issue #LUMENVOX-13)
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asterisk-dev mailing list. Thanks!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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time ago.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) | 34 lines
This patch fixes a regression of sorts that was introduced in
rev 24425.
It basically fixes AST-190/ABE-1782.
What was wrong: the user has 6000 extensions in one context; and
then 6000 contexts, one per extension. The parser could only handle
about 4893 of the 6000 extens in the single context.
This was due to the regression I mentioned. To get rid of
shift/reduce conflicts, Luigi set up right-recursive lists
for globals, context elements, switch lists, and statements.
Right recursive lists got rid of the warnings, but instead, they
use up a tremendous amount of stack space when the lists are long.
I saw this a few years back, and resolved not to fix it until
someone complained. That day has arrived!
After the changes were made, I ran the regression test suite,
and there were no problems.
I took the test case the user provided, and added 100,000
extensions to the single context, that already had 6,000 extens
in it. (I'll see your 6, and raise you 100!) It takes a few minutes
to read it all in, check it and generate code for it, but no
problems.
So, I think I can say that fundamentally, there are no longer
any limits on the number of items you can place in contexts,
statement blocks, switches, or globals, beyond your virt mem
constraints.
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ensure that the samples are byte-swapped to network order if needed. also, when a smoother is operating on a format that has a sample rate other than 8000 samples per second, use the proper sample rate for computing delivery timestamps.
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action_bridge() and bridge_exec() both search for the channels to bridge to, and then immediately drop the lock. Instead, they should hold the lock until the masquerade is complete. This will guarantee the channel remains and prevent any other weirdness from occurring. In action_bridge() some more weirdness comes into play. Both channels are needlessly locked at the same time and perform the exact same logic. It makes sense from a coding organizational standpoint, but could cause a theoretical deadlock so I split the code up. There is an issue associated with this, but since its a rather complicated thing to reproduce I'm not certain this alone will close it.
issue# 14296
Review: http://reviewboard.digium.com/r/167/
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Several changes in PTLib have occurred requiring build time detection. Changes
accounted for include the library name change, config option change, install
location change, and a boolean type change which is handled by ast_ptlib.h.
Also, the sed check has been modified to properly work with autoconf >= 2.62.
(closes issue #14224)
Reported by: bergolth
Patches:
asterisk-autoconf-sed.patch uploaded by bergolth (license 661)
asterisk-pwlib-v3.patch uploaded by bergolth (license 661)
Tested by: jpeeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177162 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r177160 | jpeeler | 2009-02-18 14:06:45 -0600 (Wed, 18 Feb 2009) | 15 lines
Modify h323 to build against PTLib as well as the older PWLib
Several changes in PTLib have occurred requiring build time detection. Changes
accounted for include the library name change, config option change, install
location change, and a boolean type change which is handled by ast_ptlib.h.
Also, the sed check has been modified to properly work with autoconf >= 2.62.
(closes issue #14224)
Reported by: bergolth
Patches:
asterisk-autoconf-sed.patch uploaded by bergolth (license 661)
asterisk-pwlib-v3.patch uploaded by bergolth (license 661)
Tested by: jpeeler
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Enabling this option by default proved to be a bad idea, as the talker detection
is not very reliable. So, make it optional again, and off by default.
(issue #13801)
Reported by: justdave
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009) | 2 lines
Document the return value of the update method (as requested on -dev list)
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(closes issue #14497)
Reported by: vinsik
Patches:
chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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the amount available.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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hints are present.
Reason: when I re-engineered the merge_and_delete func to
reduce its lock time, I failed to notice that the
functions it calls still also do locking as before.
This leads to deadlocks on dialplan reloads, when
there are actually living, subscribed hints registered
in the system.
While the reporter come across this problem while using
AEL, I might note that these deadlocks should also happen
if extensions.conf were used.
Here I added these routines to pbx.c:
ast_add_extension_nolock
add_pri_lockopt
ast_add_extension2_lockopt
find_context
add_hint_nolock
All of the above routines are static and restricted
to be used only within pbx.c, and more specifically
within the merge_contexts_and_delete routine.
They are pretty much the same as their counterparts
except they don't lock contexts or hints.
Most of them now do the real work of their
name-alike, with optional locking via extra arguments,
and are called by their name-alike. The goal was to
have the original functions so they would behave
exactly as before.
Both PJ and I tested these fixes, and the deadlocking
problem is no longer encountered.
(closes issue #14357)
Reported by: pj
Patches:
14357.diff uploaded by murf (license 17)
Tested by: pj, murf
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The big problem here is that the 3rd argument provided in these uses of strncpy()
did not reserve a byte for the null terminator, leaving the potential for writing
one byte past the end of the buffer.
Aside from this, there were coding guidelines violations with regards to spacing,
as well as hard coded lengths being used instead of sizeof().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The previous implementation of T38 faxdetect resulted in both sides of the
call jumping to a fax extension when both sides had 't38pt_udptl=yes' and
'faxdetect=yes' in sip.conf and a 'fax' extension in the current context.
This revision will jump to a 'fax' extension on incoming calls only.
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This commit brings in the changes that were living out on the
svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch. codec_dahdi.c now
always uses signed linear as the simple codec so that a soft g729 codec will
not end up being preferred to the hardware codec. There are also changes to
allow codec_dahdi.c to feed packets to the hardware in the native sample size of
the codec. This solves problems with choppy audio when using G723.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines
Modify bridging to properly evaluate DTMF after first warning is played
The main problem is currently if the Dial flag L is used with a warning sound,
DTMF is not evaluated after the first warning sound. To fix this, a flag has
been added in ast_generic_bridge for playing the warning which ensures that if
a scheduled warning is missed, multiple warrnings are not played back (due to a
feature evaluation or waiting for digits). ast_channel_bridge was modified to
store the nexteventts in the ast_bridge_config structure as that information
was lost every time ast_channel_bridge was reentered, causing a hangup due to
incorrect time calculations.
(closes issue #14315)
Reported by: tim_ringenbach
Reviewed on reviewboard:
http://reviewboard.digium.com/r/163/
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exist for T38
This is required to create a UDPTL structure in create_addr_from_peer() to handle the
scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but
is defined the peer's context. I tested this patch by enabling t38pt_udptl in the
[general] section on one system and only enabling t38pt_udptl in a peer's context on
the system sending a fax. Without the patch, the sending system will fail to initiate
T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure".
When this patch is applied the sending side will successfully initiate T38 negotiation.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r176661 | tilghman | 2009-02-17 15:21:41 -0600 (Tue, 17 Feb 2009) | 9 lines
Backport change to 1.4:
Prior to masquerade, move the group definitions to the channel performing the
masq, so that the group count lingers past the bridge.
(closes issue #14275)
Reported by: kowalma
Patches:
20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
Tested by: kowalma
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1) Add module use count handling so that timing modules can be unloaded.
2) Implement unload_module() functions for the timing interface modules.
3) Allow multiple timing modules to be loaded, and use the one with the
highest priority value.
4) Report which timing module is being use in the "timing test" CLI command.
(closes issue #14489)
Reported by: russell
Review: http://reviewboard.digium.com/r/162/
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masq, so that the group count lingers past the bridge.
(closes issue #14275)
Reported by: kowalma
Patches:
20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
Tested by: kowalma
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch changes the scheduler to use a max-heap to store pending scheduler
entries instead of a fully sorted doubly linked list. When the number of
entries in the scheduler gets large, this will perform much better. For much
more detailed information on this change, see the review request.
Review: http://reviewboard.digium.com/r/160/
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Review: http://reviewboard.digium.com/r/160/
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A heap is a convenient data structure for implementing a priority queue.
Code from svn/asterisk/team/russell/heap/.
Review: http://reviewboard.digium.com/r/160/
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This patch includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.
The changes included are:
1) Remove the module res_indications. This included the critical functionality
that actually loaded the indications configuration. I have seen many people
have Asterisk problems because they accidentally did not have an
indications.conf present and loaded. Now, this code is in the core,
and Asterisk will fail to start without indications configuration.
There was one part of res_indications, the dialplan applications, which did
belong in a module, and have been moved to a new module, app_playtones.
2) Object management has been significantly changed. Tone zones are now
managed using astobj2, and it is no longer possible to crash Asterisk by
issuing a reload that destroys tone zones while they are in use.
3) The API documentation has been filled out.
4) The API has been updated to follow our naming conventions.
5) Various bits of code throughout the tree have been updated to account
for the API update.
6) Configuration parsing has been mostly re-written.
7) "Code cleanup"
The code is from svn/asterisk/team/russell/indications/.
Review: http://reviewboard.digium.com/r/149/
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(closes issue #14485)
Reported by: davevg
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The problem here is that the hint processing code was subscribed to the wrong
event type. So, it started processing state for a hint too soon, before the
device state cache had been updated.
Also, fix a similar bug in app_queue, as it was also subscribed to the wrong
event type.
(closes issue #14461)
Reported by: alecdavis
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families
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nonexistent columns.
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