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r109908 | murf | 2008-03-19 09:41:13 -0600 (Wed, 19 Mar 2008) | 72 lines
(closes issue #11442)
Reported by: tzafrir
Patches:
11442.patch uploaded by murf (license 17)
Tested by: murf
I didn't give tzafrir very much time to test this, but if he does
still have remaining issues, he is welcome to
re-open this bug, and we'll do what is called for.
I reproduced the problem, and tested the fix, so I hope I
am not jumping by just going ahead and committing the fix.
The problem was with what file_save does with templates;
firstly, it tended to print out multiple options:
[my_category](!)(templateref)
instead of
[my_category](!,templateref)
which is fixed by this patch.
Nextly, the code to suppress output of duplicate declarations
that would occur because the reader copies inherited declarations
down the hierarchy, was not working. Thus:
[master-template](!)
mastervar = bar
[template](!,master-template)
tvar = value
[cat](template)
catvar = val
would be rewritten as:
;!
;! Automatically generated configuration file
;! Filename: experiment.conf (/etc/asterisk/experiment.conf)
;! Generator: Manager
;! Creation Date: Tue Mar 18 23:17:46 2008
;!
[master-template](!)
mastervar = bar
[template](!,master-template)
mastervar = bar
tvar = value
[cat](template)
mastervar = bar
tvar = value
catvar = val
This has been fixed. Since the config reader 'explodes' inherited
vars into the category, users may, in certain circumstances, see
output different from what they originally entered, but it should
be both correct and equivalent.
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actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This does introduce a dependency on the GMime library for handling HTTP POSTs, but it is available in most distros.
If the library is present, then the compile flag for ENABLE_UPLOADS is enabled by default in menuselect.
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r109226 | mmichelson | 2008-03-17 17:05:49 -0500 (Mon, 17 Mar 2008) | 12 lines
Fix a logic flaw in the code that stores lock info which is displayed
via the "core show locks" command. The idea behind this section of code was
to remove the previous lock from the list if it was a trylock that had failed.
Unfortunately, instead of checking the status of the previous lock, we were referencing
the index immediately following the previous lock in the lock_info->locks array.
The result of this problem, under the right circumstances, was that the lock which
we currently in the process of attempting to acquire could "overwrite" the previous lock
which was acquired. While this does not in any way affect typical operation, it *could*
lead to misleading "core show locks" output.
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Reported by: mvanbaak
Tested by: murf, mvanbaak
Due to a bug that occurred when merge_contexts_and_delete scanned the "old" or existing contexts, and found a context
that doesn't exist in the new set, yet owned by a different registrar. The context is created in the new set, with the
old registrar, and and all the priorities and extens that have a different registrar are copied into it. But, not the
includes, ignorepats, and switches. I added code to do this immediately after the context is created.
This still leaves a logical hole in the code. If you define a context in two places, (eg. in extensions.conf and also
in extensions.ael), and they both have includes, but different in composition, no new context will be generated, and
therefore the 'old' includes, switches, and ignorepats will not be copied. I'd have added code to simply add any non-duplicates
into the 'new' context that had a different registrar, but there is one big complication: includes, and switches are definitely
order dependent. (ignorepats I'm not sure about). And we'll have to develop some sort of policy about how we
merge order dependent lists, especially if the intersection of the two sets is empty. (in other words, they do not have any
elements in common). Do the new go first, or the old? I've elected to punt this issue until a user complains. Hopefully,
this is pretty rare thing.
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r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) | 11 lines
Fix another issue that was causing crashes in chanspy. This introduces a new
datastore callback, called chan_fixup(). The concept is exactly like the
fixup callback that is used in the channel technology interface. This callback
gets called when the owning channel changes due to a masquerade. Before this
was introduced, if a masquerade happened on a channel being spyed on, the
channel pointer in the datastore became invalid.
(closes issue #12187)
(reported by, and lots of testing from atis)
(props to file for the help with ideas)
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Though this overflow is exploitable remotely, we are NOT issuing a security
advisory for this since in order to exploit the overflow, the attacker would
have to establish an authenticated manager session AND have the system privilege.
By gaining this privilege, the attacker already has more powerful weapons at his
disposal than overflowing a buffer with a malformed manager header, so the vulnerability
in this case really lies with the authentication method that allowed the attacker to
gain the system privilege in the first place.
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r108135 | russell | 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008) | 40 lines
(closes issue #12187, reported by atis, fixed by me after some brainstorming
on the issue with mmichelson)
- Update copyright info on app_chanspy.
- Fix a race condition that caused app_chanspy to crash. The issue was that
the chanspy datastore magic that was used to ensure that spyee channels did
not disappear out from under the code did not completely solve the problem.
It was actually possible for chanspy to acquire a channel reference out of
its datastore to a channel that was in the middle of being destroyed. That
was because datastore destruction in ast_channel_free() was done near the
end. So, this left the code in app_chanspy accessing a channel that was
partially, or completely invalid because it was in the process of being free'd
by another thread. The following sort of shows the code path where the race
occurred:
=============================================================================
Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
--------------------------------------||-------------------------------------
ast_channel_free() ||
- remove channel from channel list ||
- lock/unlock the channel to ensure ||
that no references retrieved from ||
the channel list exist. ||
--------------------------------------||-------------------------------------
|| channel_spy()
- destroy some channel data || - Lock chanspy datastore
|| - Retrieve reference to channel
|| - lock channel
|| - Unlock chanspy datastore
--------------------------------------||-------------------------------------
- destroy channel datastores ||
- call chanspy datastore d'tor ||
which NULL's out the ds' || - Operate on the channel ...
reference to the channel ||
||
- free the channel ||
||
|| - unlock the channel
--------------------------------------||-------------------------------------
=============================================================================
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r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines
Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait).
(closes issue #11945)
Reported by: xheliox
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caching would not work properly for users.conf and any other file read from
more than one place. I needed to add the filename which requested the config
file to get it to work properly.
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r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar 2008) | 11 lines
fix up various compiler warnings found with gcc-4.3:
- the output of flex includes a static function called 'input' that is not used, so for the moment we'll stop having the compiler tell us about unused variables in the flex source files (a better fix would be to improve our flex post-processing to remove the unused function)
- main/stdtime/localtime.c makes assumptions about signed integer overflow, and gcc-4.3's improved optimizer tries to take advantage of handling potential overflow conditions at compile time; for now, suppress these optimizations until we can fiure out if the code needs improvement
- main/udptl.c has some references to uninitialized variables; in one case there was no bug, but in the other it was certainly possibly for unexpected behavior to occur
- main/editline/readline.c had an unused variable
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r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008) | 8 lines
Fix another bug specifically related to asynchronous call origination. Once the
PBX is started on the channel using ast_pbx_start(), then the ownership of the
channel has been passed on to another thread. We can no longer access it in this
code. If the channel gets hung up very quickly, it is possible that we could
access a channel that has been free'd.
(inspired by BE-386)
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r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008) | 9 lines
Fix some bugs related to originating calls. If the code failed to start a PBX
on the channel (such as if you set a call limit based on the system's load
average), then there were cases where a channel that has already been free'd
using ast_hangup() got accessed. This caused weird memory corruption and
crashes to occur.
(fixes issue BE-386)
(much debugging credit goes to twilson, final patch written by me)
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Reported by: rizzo
Tested by: murf
Proposal of the changes to be made, and then an announcement of how they were accomplished:
http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html
and:
http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html
Here is a recap, file by file, of what I have done:
pbx/pbx_config.c
pbx/pbx_ael.c
All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.
We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.
pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and
then call merge_contexts_and_delete, which will merge (now) existing contexts and
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then
destroy the old dialplan.
chan_sip.c
chan_iax.c
chan_skinny.c
All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.
chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.
apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c
All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.
include/asterisk/pbx.h
ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create() interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.
include/asterisk/pval.h
ast_compile_ael2() interface changed to include the local hashtab table ptr.
main/features.c
For the sake of the parking context, we use ast_context_find_or_create().
main/pbx.c
I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.
refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.
Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.
Added some calls to ast_verb(3,...) for debug messages
Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.
find_or_create was upgraded to handle both local lists/tables as well as the globals.
context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables
ast_merge_contexts_and_delete() was heavily modified.
ast_add_extension2() was also upgraded to handle changes.
the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.
res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile
Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps. The main gotcha was I had to
include lock.h and hashtab.h in several places.
As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.
How's this for verbose commit messages?
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Fix a number of other places where the number of samples in a G722 frame was
not properly handled because of various reasons.
main/rtp.c:
- When a G722 frame is read from the smoother, the number of samples in the
frame must be divided by 2 before being sent out over the network. Even
though G722 is 16 kHz, an error in some previous spec has made it so that
we have to list the number of samples such as if it was 8 kHz.
main/file.c:
- When scheduling the next time to expect a frame, take into account that the
format of the file we're reading from may not be 8 kHz.
codecs/codec_g722.c:
- When converting from G722 to slinear, g722_decode() expects its samples
parameter to be in the silly (real samples / 2) format. Make it so.
- When converting from slinear to G722, properly set the number of samples in
the frame to be the number of bytes of output * 2.
formats/format_pcm.c:
- This format module handles G722, among a number of other formats. However,
the read() and seek() functions did not account for the fact that G722 has
2 samples per byte.
(closes issue #12130, reported by rickross, patched by me)
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r106437 | mmichelson | 2008-03-06 16:10:07 -0600 (Thu, 06 Mar 2008) | 8 lines
Quell an annoying message that is likely to print every single time that
ast_pbx_outgoing_app is called. The reason is that __ast_request_and_dial
allocates the cdr for the channel, so it should be expected that the channel
will have a cdr on it.
Thanks to joetester on IRC for pointing this out
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an attended transfer over AMI
(closes issue #10585)
Reported by: ornati
Patches:
atxfer-trunk-r90428.diff uploaded by ornati (license 210)
(with modifications from me)
Tested by: putnopvut
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r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines
Fix a bug that I just noticed in the RTP code. The calculation for setting the
len field in an ast_frame of audio was wrong when G.722 is in use. The len field
represents the number of ms of audio that the frame contains. It would have
set the value to be twice what it should be.
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r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines
When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson
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LIST instead of an RWLIST. The way this list works makes it such that
a RWLIST provides no additional benefit. Also, a mutex is needed for
use with the thread condition.
Merged revisions 105563 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) | 24 lines
Merge in some changes from team/russell/autoservice-nochans-1.4
These changes fix up some dubious code that I came across while auditing what
happens in the autoservice thread when there are no channels currently in
autoservice.
1) Change it so that autoservice thread doesn't keep looping around calling
ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition
so that the thread properly goes to sleep and does not wake up until a
channel is put into autoservice.
This actually fixes an interesting bug, as well. If the autoservice thread
is already running (almost always is the case), then when the thread goes
from having 0 channels to have 1 channel to autoservice, that channel would
have to wait for up to 1/2 of a second to have the first frame read from it.
2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no
channels and no fds to poll() on, such as was the case with the previous code
for the autoservice thread. In this case, the code would call alloca(0), and
pass the result as the first argument to poll(). In this case, the 2nd
argument to poll() specified that there were no fds, so this invalid pointer
shouldn't actually get dereferenced, but, this code makes it explicit and
ensures the pointers are NULL unless we have valid data to put there.
(related to issue #12116)
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r105560 | file | 2008-03-03 11:28:59 -0400 (Mon, 03 Mar 2008) | 7 lines
It is possible for no audio to pass between the current digit and next digit so expand logic that clears emulation to AST_FRAME_NULL.
(closes issue #11911)
Reported by: edgreenberg
Patches:
v1-11911.patch uploaded by dimas (license 88)
Tested by: tbsky
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r105409 | russell | 2008-02-29 17:34:32 -0600 (Fri, 29 Feb 2008) | 23 lines
Fix a major bug in autoservice. There was a race condition in the handling of
the list of channels in autoservice. The problem was that it was possible for
a channel to get removed from autoservice and destroyed, while the autoservice
thread was still messing with the channel. This led to memory corruption, and
caused crashes. This explains multiple backtraces I have seen that have
references to autoservice, but do to the nature of the issue (memory corruption),
could cause crashes in a number of areas.
(fixes the crash in BE-386)
(closes issue #11694)
(closes issue #11940)
The following issues could be related. If you are the reporter of one of these,
please update to include this fix and try again.
(potentially fixes issue #11189)
(potentially fixes issue #12107)
(potentially fixes issue #11573)
(potentially fixes issue #12008)
(potentially fixes issue #11189)
(potentially fixes issue #11993)
(potentially fixes issue #11791)
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r105116 | russell | 2008-02-28 16:23:05 -0600 (Thu, 28 Feb 2008) | 8 lines
Fix a bug in the lock tracking code that was discovered by mmichelson. The issue
is that if the lock history array was full, then the functions to mark a lock as
acquired or not would adjust the stats for whatever lock is at the end of the array,
which may not be itself. So, do a sanity check to make sure that we're updating
lock info for the proper lock.
(This explains the bizarre stats on lock #63 in BE-396, thanks Mark!)
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r104841 | mmichelson | 2008-02-27 15:49:20 -0600 (Wed, 27 Feb 2008) | 17 lines
Two fixes:
1. Make the list of ast_dial_channels a lockable list. This is because in some cases,
the ast_dial may exist in multiple threads due to asynchronous execution of its application, and
I found some cases where race conditions could exist.
2. Check in ast_dial_join to be sure that the channel still exists before attempting to lock it, since
it could have gotten hung up but the is_running_app flag on the ast_dial_channel may not have been
cleared yet.
(closes issue #12038)
Reported by: jvandal
Patches:
12038v2.patch uploaded by putnopvut (license 60)
Tested by: jvandal
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r105005 | qwell | 2008-02-28 13:20:10 -0600 (Thu, 28 Feb 2008) | 9 lines
Make pbx_exec pass an empty string into applications, if we get NULL.
This protects against possible segfaults in applications that may try
to use data before checking length (ast_strdupa'ing it, for example)
(closes issue #12100)
Reported by: foxfire
Patches:
12100-nullappargs.diff uploaded by qwell (license 4)
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r104593 | kpfleming | 2008-02-27 10:53:06 -0600 (Wed, 27 Feb 2008) | 8 lines
fallback to standard English prompts properly when using new prompt directory layout
(closes issue #11831)
Reported by: IgorG
Patches:
fallbacken.v1.diff uploaded by IgorG (license 20) (modified by me to improve code and conform rest of function to coding guidelines)
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automatically generated file like it used to be. This still needs to be there
for modules that have to check it to compile against multiple asterisk versions.
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r104102 | russell | 2008-02-25 17:19:05 -0600 (Mon, 25 Feb 2008) | 7 lines
Improve the lock tracking code a bit so that a bunch of old locks that threads
failed to lock don't sit around in the history. When a lock is first locked,
this checks to see if the last lock in the list was one that was failed to be
locked. If it is, then that was a lock that we're no longer sitting in a trylock
loop trying to lock, so just remove it.
(inspired by issue #11712)
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r104092 | qwell | 2008-02-25 14:49:42 -0600 (Mon, 25 Feb 2008) | 11 lines
Allow the use of #include and #exec in situations where the max include depth was only 1.
Specifically, this fixes using #include and #exec in extconfig.conf.
This was basically caused because the config file itself raises the include level to 1.
I opted not to raise the include limit, because recursion here could cause very bizarre behavior.
Pointed out, and tested by jmls
(closes issue #12064)
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r103786 | mmichelson | 2008-02-18 14:52:09 -0600 (Mon, 18 Feb 2008) | 10 lines
There was an invalid assumption when calculating the duration of a file that the filestream in question
was created properly. Unfortunately this led to a segfault in the situation where an unknown format was
specified in voicemail.conf and a voicemail was recorded. Now, we first check to be sure that the stream
was written correctly or else assume a zero duration.
(closes issue #12021)
Reported by: jakep
Tested by: putnopvut
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r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008) | 9 lines
When a SIP channel is being auto-destroyed, it's possible for it to still be
in bridge code. When that happens, we crash. Delay the RTP destruction until
the bridge is ended.
(closes issue #11960)
Reported by: norman
Patches:
20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: norman
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r103728 | russell | 2008-02-15 13:50:11 -0600 (Fri, 15 Feb 2008) | 4 lines
In the case that you try to directly reload a module has returned
AST_MODULE_LOAD_DECLINE, log a message indicating that the module is not fully
initialized and must be initialized using "module load".
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r103726 | russell | 2008-02-15 12:33:29 -0600 (Fri, 15 Feb 2008) | 6 lines
Don't attempt to execute the reload callback for a module that returned
AST_MODULE_LOAD_DECLINE. This fixes a crash that was reported against
chan_console in trunk.
(closes issue #11953, reported by junky, fixed by me)
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to indicate there was more data than there really was, causing the iterator read beyond the end
of the event structure. This led to invalid memory reads and potential crashes.
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Added ability to retrieve list of categories in a config file.
Added ability to retrieve the content of a particular category.
Added ability to empty a context.
Created new action to create a new file.
Updated delete action to allow deletion by line number with respect to category.
Added new action insert to add new variable to category at specified line.
Updated action newcat to allow new category to be inserted in file above another existing category.
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in the event API.
ast_event_iterator_get_ie_type should return the ie type in
host byte order. Furthermore, ast_event_get_ie_raw should already
have its ie type argument in host byte order since it could be called
externally (and it in fact is called in this way by ast_event_get_cached).
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(closes issue #8925)
About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI
commands in Asterisk 1.4 for the next version of their book, they documented
a lot of inconsistencies. This set of changes addresses all of these issues
and has been reviewed by Leif.
While this does introduce even more changes to the CLI command structure, it
makes everything consistent, which is the most important thing.
Thanks to all that helped with this one!
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r101601 | russell | 2008-01-31 17:10:06 -0600 (Thu, 31 Jan 2008) | 12 lines
Fix a couple of places where ast_frfree() was not called on a frame that came
from a translator. This showed itself by g729 decoders not getting released.
Since the flag inside the translator frame never got unset by freeing the frame
to indicate it was no longer in use, the translators never got destroyed, and
thus the g729 licenses were not released.
(closes issue #11892)
Reported by: xrg
Patches:
11892.diff uploaded by russell (license 2)
Tested by: xrg, russell
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r101222 | file | 2008-01-30 11:41:04 -0400 (Wed, 30 Jan 2008) | 4 lines
Fix an issue where if a frame of higher sample size preceeded a frame of lower sample size and ast_slinfactory_read was called with a sample size of the combined values or higher a crash would happen.
(closes issue #11878)
Reported by: stuarth
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r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28 Jan 2008) | 9 lines
Make some deadlock related fixes. These bugs were discovered and reported
internally at Digium by Steve Pitts.
- Fix up chan_local to ensure that the channel lock is held before the local
pvt lock.
- Don't hold the channel lock when executing the timing function, as it can
cause a deadlock when using chan_local. This actually changes the code back
to be how it was before the change for issue #10765. But, I added some other
locking that I think will prevent the problem reported there, as well.
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today on another mac. We ignore the return value of locking operations almost
everywhere in Asterisk. So, ignore these, as well, so Asterisk will actually
work on systems where this is occurring while I look into what the issue is.
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r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008) | 11 lines
When deleting a task from the scheduler, ignoring the return value could
possibly cause memory to be accessed after it is freed, which causes all
sorts of random memory corruption. Instead, if a deletion fails, wait a
bit and try again (noting that another thread could change our taskid
value).
(closes issue #11386)
Reported by: flujan
Patches:
20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, flujan, stuarth`
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as a channel variable BRIDGEPVTCALLID
This is important for call tracing in log files and CDRs, so that
the SIP callID can be traced along servers.
The CHANNEL dialplan function won't work here, since the outbound
channel is gone when we need the Call-ID.
Other channel drivers may now implement the same function :-),
but this patch only supports chan_sip.so.
Inspired by (issue #11816)
Reported by: ctooley
Patch by oej
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r99540 | tilghman | 2008-01-22 10:54:06 -0600 (Tue, 22 Jan 2008) | 7 lines
Ensure that we can get an address even when we don't have a default route.
(closes issue #9225)
Reported by: junky
Patches:
20080122__bug9225.diff.txt uploaded by Corydon76 (license 14)
Tested by: oej, loloski, sergee
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This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
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r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) | 9 lines
Revert adding the packed attribute, as it really doesn't make sense why that
would do any good. Fix the real bug, which is to do the check to see if the
frame came from a translator at the beginning of ast_frame_free(), instead of
at the end. This ensures that it always gets checked, even if none of the
parts of the frame are malloc'd, and also ensures that we aren't looking at
free'd memory in the case that it is a malloc'd frame.
(closes issue #11792, reported by explidous, patched by me)
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r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | 10 lines
Have IAX2 optimize the codec translation path just like chan_sip does it. If
the caller's codec is in our codec list, move it to the top to avoid transcoding.
(closes issue #10500)
Reported by: stevedavies
Patches:
iax-prefer-current-codec.patch uploaded by stevedavies (license 184)
iax-prefer-current-codec.1.4.patch uploaded by stevedavies (license 184)
Tested by: stevedavies, pj, sheldonh
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immediately at startup. Any commands in the startup_commands file in the Asterisk
config diretory will get executed.
(closes issue #11781)
Reported by: jamesgolovich
Patches:
asterisk-startupcmds.diff.txt uploaded by jamesgolovich (license 176)
-- With some changes by me.
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r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines
Commit a fix for some memory access errors pointed out by the valgrind2.txt
output on issue #11698.
The issue here is that it is possible for an instance of a translator to get
destroyed while the frame allocated as a part of the translator is still being
processed. Specifically, this is possible anywhere between a call to ast_read()
and ast_frame_free(), which is _a lot_ of places in the code. The reason this
happens is that the channel might get masqueraded during this time. During a
masquerade, existing translation paths get destroyed.
So, this patch fixes the issue in an API and ABI compatible way. (This one is
for you, paravoid!)
It changes an int in ast_frame to be used as flag bits. The 1 bit is still used
to indicate that the frame contains timing information. Also, a second flag has
been added to indicate that the frame came from a translator. When a frame with
this flag gets released and has this flag, a function is called in translate.c to
let it know that this frame is doing being processed. At this point, the flag gets
cleared. Also, if the translator was requested to be destroyed while its internal
frame still had this flag set, its destruction has been deffered until it finds out
that the frame is no longer being processed.
Admittedly, this feels like a hack. But, it does fix the issue, and I was not able
to think of a better solution ...
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r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6 lines
If the incoming RTP stream changes codec force the bridge to break if the other side does not support it.
(closes issue #11729)
Reported by: tsearle
Patches:
new_codec_patch_udiff.patch uploaded by tsearle (license 373)
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sample length with g722. It is _2_ samples per byte, not 1. This was all
over the place, and I believed it, and it is what caused me to take so long
to figure out what was broken.
- Update copyright information on codec_g722.
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This fix was made in favor of the proposed patch since it doesn't involve changing
a core codec define.
(closes issue #11722, reported and initially patched by caio1982, final patch by me)
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r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines
1) When we get a translated frame out, clone it, because if the
translator pvt is freed before we use the frame, bad things happen.
2) Getting a failure from ast_sched_delete means that the schedule
ID is currently running. Don't just ignore it.
(Closes issue #11698)
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r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1 line
This is a fix for 2 things: a problem Terry was having in OSX with null pointers, which was my fault, as I probably forgot to run the sed script last time I made mods. So, I moved the fix into the flex input itself. Then, I found when I used flex 2.5.33, that it was using __STDC_VERSION__, and that's not real good; so I added back in a DIFFERENT sed script to fix that little mess. Tested everything, a couple different ways. Hope I did no harm, at the least.
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Joshua Colp
Steve Murphy
Kevin P. Fleming
Russell Bryant
Terry Wilson
Mark Michelson
Michiel van Baak
Tilghman Lesher
Jason Parker
Olle Johansson
Jeff Peeler
James Golovich
Pari Nannapaneni