Russell Bryant [Mon, 10 Mar 2008 20:05:12 +0000 (20:05 +0000)]
Merged revisions 107158 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
Joshua Colp [Mon, 10 Mar 2008 20:00:21 +0000 (20:00 +0000)]
If we receive a 488 on a T38 request reinvite back to audio. As well reinvite across a bridge back to audio if one side doesn't negotiate to T38.
(closes issue #8677)
Reported by: alex-911
Mark Michelson [Mon, 10 Mar 2008 15:45:13 +0000 (15:45 +0000)]
app_queue has now been doxygenified thanks to snuffy! The ony thing I changed
was the way that locks are referenced, since the old 1.2 names were still used
in the comments.
Steve Murphy [Mon, 10 Mar 2008 14:55:21 +0000 (14:55 +0000)]
way back in July, in r.75706, a fix was made ot the strftime usages, which was good, but in this case, the check for a nil time was accidentally removed, and now it is restored, to keep timevals like '1969-12-31 17:00:00' from showing up in the cdrs. No idea what databases will do with this. No bugs filed as yet, but it felt like a bug.
don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down"
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.
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.
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.
Russell Bryant [Fri, 7 Mar 2008 17:17:30 +0000 (17:17 +0000)]
Merged revisions 106704 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
Russell Bryant [Fri, 7 Mar 2008 00:24:58 +0000 (00:24 +0000)]
Merge changes from team/russell/g722-sillyness ...
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)
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.
Donny Kavanagh [Thu, 6 Mar 2008 19:31:50 +0000 (19:31 +0000)]
trivial fix for an agi error when attempting to use EAGI on a dead/hungup channel, we now print an error that makes sense
given our removal of deadagi as an actual application.
Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things.
(closes issue #12148)
Reported by: jcomellas
document var_metric so no bugreports will come in when it's actually a configuration issue.
(issue #12151)
Reported and patched by: caio1982
1.4 patch by me
when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one
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)
Russell Bryant [Wed, 5 Mar 2008 01:54:16 +0000 (01:54 +0000)]
Merged revisions 105932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
Mark Michelson [Tue, 4 Mar 2008 23:24:56 +0000 (23:24 +0000)]
There are several places in manager.c where BUFSIZ is used for a buffer
which will contain nowhere near that amount of data. This makes these buffers
more reasonably sized.
Russell Bryant [Tue, 4 Mar 2008 20:36:16 +0000 (20:36 +0000)]
Fix some bugs in the SIP tcp helper thread.
- fix a spot where a lock wouldn't get unlocked in an error condition
- call ast_mutex_destroy() on the lock before freeing its memory
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
Russell Bryant [Tue, 4 Mar 2008 04:31:53 +0000 (04:31 +0000)]
Blocked revisions 105591 via svnmerge
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r105591 | russell | 2008-03-03 22:31:29 -0600 (Mon, 03 Mar 2008) | 4 lines
Backport a minor bug fix from trunk that I found while doing random code
cleanup. Properly break out of the loop when a context isn't found when
verify that includes are valid.
Russell Bryant [Mon, 3 Mar 2008 17:06:35 +0000 (17:06 +0000)]
Merged revisions 105568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r105568 | russell | 2008-03-03 11:05:16 -0600 (Mon, 03 Mar 2008) | 3 lines
Fix a potential memory leak of the local_pvt struct when ast_channel allocation
fails. Also, in passing, centralize the code necessary to destroy a local_pvt.
Russell Bryant [Mon, 3 Mar 2008 15:59:50 +0000 (15:59 +0000)]
3) In addition to merging the changes below, change trunk back to a regular
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.
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
Add a comment to describe some logic.
(closes issue #12120)
Reported by: flefoll
Patches:
chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license 244)
Terry Wilson [Sat, 1 Mar 2008 01:30:37 +0000 (01:30 +0000)]
Asterisk, when parking can drop rights a caller when a parking timeout occurs. Also, when doing built-in attended transfers, sometimes incorrectly passes rights from the transferrer to the transferee. This patch tries to fixes the parking issue and lays some groundwork for later fixing the transfer issue.
Russell Bryant [Sat, 1 Mar 2008 00:53:25 +0000 (00:53 +0000)]
Add a "devstate change" CLI command to control custom device states. Also,
do some additional code cleanup and improvement in passing.
(closes issue #12106)
Reported by: nizon
Patches:
devstate-patch.txt uploaded by nizon (license 415)
-- Updated to trunk, and tab completion added by me
Russell Bryant [Fri, 29 Feb 2008 23:36:46 +0000 (23:36 +0000)]
Merged revisions 105409 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
Philippe Sultan [Fri, 29 Feb 2008 14:15:03 +0000 (14:15 +0000)]
Remove unnecessary if statements before calling iks_delete (redundant check is
done inside iks_delete), thus making the code conform with coding guidelines.
Russell Bryant [Thu, 28 Feb 2008 22:39:26 +0000 (22:39 +0000)]
Merged revisions 105116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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!)
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.
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)
Tilghman Lesher [Thu, 28 Feb 2008 14:42:32 +0000 (14:42 +0000)]
Fix crash when configuration does not match hardware detection.
(closes issue #12096)
Reported by: mmickan
Patches:
chan_vpb.cc.diff uploaded by mmickan (license 400)
Russell Bryant [Wed, 27 Feb 2008 17:36:50 +0000 (17:36 +0000)]
Merged revisions 104625 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r104625 | russell | 2008-02-27 11:33:04 -0600 (Wed, 27 Feb 2008) | 4 lines
Fix a problem in ChanSpy where it could get stuck in an infinite loop without
being able to detect that the calling channel hung up.
(closes issue #12076, reported by junky, patched by me)
Use the lock (which already existed, it just wasn't used) on the updaters list to protect the contents instead of the overall module list lock.
(closes issue #12080)
Reported by: ChaseVenters
Joshua Colp [Wed, 27 Feb 2008 17:04:16 +0000 (17:04 +0000)]
After further discussion revert my previous commit for this. Currently in order to ensure devicestate is the expected value in another module (such as app_queue) then chan_sip must be loaded before hand.
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)
Russell Bryant [Wed, 27 Feb 2008 16:46:49 +0000 (16:46 +0000)]
Blocked revisions 104591 via svnmerge
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r104591 | russell | 2008-02-27 10:45:00 -0600 (Wed, 27 Feb 2008) | 4 lines
When we receive a known alarm, make sure that the unknown alarm flag is not still
set to make sure that when we come back out of alarm, it gets reported in the log
and manager interface (after discussion with tzafrir on the -dev list)
Joshua Colp [Wed, 27 Feb 2008 16:26:57 +0000 (16:26 +0000)]
When queueing up a device state change when the peer is loaded from the configuration give it a state of not in use. We have to do this because the channel technology may not yet be registered so the state could not be queried and would be considered invalid.
(closes issue #12087)
Reported by: liorm
projects / thirdparty / asterisk.git / log
