for the string to be copied into. This resulted in parse_request reading invalid
memory beyond the end of the string, and in some cases led to crashes. Thanks
to falves11 for providing the valgrind output which led to the closure of this issue.
(closes issue #12284)
Reported by: falves11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | 15 lines
(closes issue #12302)
Reported by: pj
Tested by: murf
These changes will set a channel variable ~~EXTEN~~ just before generating code
for a switch, with the value of ${EXTEN}. The exten is marked as having a switch,
and ever after that, till the end of the exten, we substitute any ${EXTEN}
with ${~~EXTEN~~} instead in application arguments; (and the ${EXTEN: also).
The reason for this, is that because switches are coded using
separate extensions to provide pattern matching, and
jumping to/from these switch extensions messes up the ${EXTEN} value,
which blows the minds of users.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Per comments from dimas:
1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones.
2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before.
3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best.
Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too.
4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used.
5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel.
This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality.
(closes issue #11968)
Reported by: dimas
Patches:
v2-11968-dsp.patch uploaded by dimas (license 88)
v4-11968-zap.patch uploaded by dimas (license 88)
Tested by: dimas, qwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines
Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) | 17 lines
(closes issue #11903)
Reported by: atis
Many thanks to atis for spotting this problem and reporting it.
The fix was to straighten out how items are placed on and removed
from the file stack. Regressions as well as the provided test case
helped to straighten out all code paths. valgrind was used to make
sure all memory allocated was freed.
Sorry for not solving this earlier. I got distracted.
Added the ntest23 regression test, which is mainly a copy of ntest22,
but with a few juicy errors thrown in, to replicate the kind of
error that atis spotted.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109357 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r108227 | mmichelson | 2008-03-12 16:16:28 -0500 (Wed, 12 Mar 2008) | 12 lines
Added a large comment before the AST_SCHED_DEL macro to explain its purpose as well as when
it is appropriate and when it is not appropriate to use it.
I also removed the part of the debug message that mentions that this is probably a bug because
there are some perfectly legitimate places where ast_sched_del may fail to delete an entry (e.g.
when the scheduler callback manually reschedules with a new id instead of returning non-zero to
tell the scheduler to reschedule with the same idea). I also raised the debug level of the debug
message in AST_SCHED_DEL since it seems like it could come up quite frequently since the macro
is probably being used in several places where it shouldn't be. Also removed the redundant line,
file, and function information since that is provided by ast_log.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r106704 | russell | 2008-03-07 11:16:58 -0600 (Fri, 07 Mar 2008) | 8 lines
Change a warning message to a debug message. This is happening quite frequently,
and it is not worth spamming users with these messages unless we are pretty confident
that it should never happen. As it stands today, it _will_ and _does_ happen and
until that gets cleaned up a reasonable amount on the development side, let's not
spam the logs of everyone else.
(closes issue #12154)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008) | 7 lines
Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log.
(closes issue #12140)
Reported by: slavon
Patches:
sch2.patch uploaded by slavon (license 288)
(Patch slightly modified by me)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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!)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105144 65c4cc65-6c06-0410-ace0-fbb531ad65f3