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${ noResults }
1184 Commits (57eaf9dd8f2860f3c27a6bde1ce178f7a039636c)
Author | SHA1 | Message | Date |
---|---|---|---|
|
bfed30d5dd |
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/branches/1.4@106704 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
cd703523db |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@106235 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
b350a97937 |
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/branches/1.4@106015 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
d564404d73 |
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/branches/1.4@105932 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
36bb1f9d46 |
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/branches/1.4@105674 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
547ac9f501 |
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/branches/1.4@105563 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
12e5fb358a |
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/branches/1.4@105116 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
bc56a84c58 |
Merge changes from team/russell/smdi-1.4
This commit brings in a significant set of changes to the SMDI support in Asterisk. There were a number of bugs in the current implementation, most notably being that it was very likely on busy systems to pop off the wrong message from the SMDI message queue. So, this set of changes fixes the issues discovered as well as introducing some new ways to use the SMDI support which are required to avoid the bugs with grabbing the wrong message off of the queue. This code introduces a new interface to SMDI, with two dialplan functions. First, you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access details in the message using the SMDI_MSG() function. A side benefit of this is that it now supports more than just chan_zap. For example, with this implementation, you can have some FXO lines being terminated on a SIP gateway, but the SMDI link in Asterisk. Another issue with the current implementation is that it is quite common that the station ID that comes in on the SMDI link is not necessarily the same as the Asterisk voicemail box. There are now additional directives in the smdi.conf configuration file which let you map SMDI station IDs to Asterisk voicemail boxes. Yet another issue with the current SMDI support was related to MWI reporting over the SMDI link. The current code could only report a MWI change when the change was made by someone calling into voicemail. If the change was made by some other entity (such as with IMAP storage, or with a web interface of some kind), then the MWI change would never be sent. The SMDI module can now poll for MWI changes if configured to do so. This work was inspired by and primarily done for the University of Pennsylvania. (also related to issue #9260) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@104119 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
638ef8c5d2 |
Change to the configure logic regarding IMAP. Prior to this commit, if you wished to configure
Asterisk with IMAP support, you would use the --with-imap configure switch in one of the following two ways: --with-imap=/some/directory would look in the directory specified for a UW IMAP source installation --with-imap would assume that you had imap-2004g installed in .. relative to the Asterisk source With this set of changes the two above options still work the same, but there are two new behaviors, too. --with-imap=system will assume that you have -libc-client.so where you store your shared objects and will attempt to find c-client headers in your include path either in the imap or c-client directory. If either of the two original methods of specifying the imap option should fail, then the check for --with-imap =system will be performed in addition. It is only after this "system" check that failure can happen. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@103698 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
0dafcac660 |
Cross-platform fix: OS X now deprecates the use of the daemon(3) API.
(closes issue #11908) Reported by: oej Patches: 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@102323 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
1191559147 |
Change detection of getifaddrs to use AST_C_COMPILE_CHECK, backported from trunk (as suggested by kpfleming)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@101894 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
df1dc7741d |
Compatibility fix for OpenWRT (reported by Brian Capouch via the mailing list)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@101772 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
ef78f25e8a |
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@100581 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
7060a6888d |
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` git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@100465 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
cc750dc9ce |
make these macros not assume that the only other field in the structure is 'argc'... this is true when someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable to define your own structure as long as it has the right fields
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@100264 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
cae4280341 |
Permit the user to specify number of seconds that a connection may remain idle,
which fixes a crash on reconnect with the MyODBC driver. (closes issue #11798) Reported by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@99341 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
d0d93be4f4 |
Remove the __ in front of the unused variable. This causes some compilers to freak out.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@99127 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
06d3c61a2e |
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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@99081 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
f0001ecf66 |
Since we're relying on the offset between the frame and the beginning of the translator
pvt struct, set the packed attribute to make sure we get to the right place. (potential fix for issue #11792) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@99079 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
6d0ee62540 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@99004 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
904f38a40a |
Add an unused pointer to the ast_channel struct. This makes the ast_channel structure
retain the same size as it had in previous 1.4 releases. Also, all of the offsets for members in the structure are still the same (except for the two pointers that got replaced for the new spy/whisper architecture.) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@98982 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
fa640604de |
Replace current spy architecture with backport of audiohooks. This should take care of current known spy issues.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@98972 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
a0c14dbda3 |
Add autoconf logic for speexdsp. Later versions use a separate library for some things so we need to use it if present in codec_speex.
(closes issue #11693) Reported by: yzg git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@98951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7a007060bd |
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 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@98943 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
45f3890926 |
Backport the ability to set the ToS bits on Linux when not running as root.
Normally, we would not backport features into 1.4, but, I was convinced by the justification supplied by the supplier of this patch. He pointed out that this patch removes a requirement for running as root, thus reducing the potential impacts of security issues. (closes issue #11742) Reported by: paravoid Patches: libcap.diff uploaded by paravoid (license 200) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@98265 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
00b8f86349 |
Fix a comment that is no longer true.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@97847 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
742fd8127c |
Remove pbx_kdeconsole from the tree. It hasn't worked in ages, and nobody has
complained. (closes issue #11706, reported by caio1982) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@97734 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
67df6e008d |
Use the constant that I really meant to use here ...
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@94829 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
5977f8d5be |
Convert the contexts lock to a read/write lock to resolve a deadlock. This
has a nice side benefit of improving performance. :) (closes issue #11609) (closes issue #11080) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@94466 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
612d0b8f45 |
Check for the existence of the soxmix application on the target platform and have
the result available in autoconfig.h. (part of issue #11589) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@94077 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
8057fb41cb |
Today is tomorrow's yesterday, and yesterday's tomorrow is today, and
tomorrow's tomorrow is the day after tomorrow, so who cares if you recycle anyway? If this confuses you, that's nothing compared to what this fixes. ;-) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@93336 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
627f595c58 |
In http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
rizzo brought up some issues related to the way that the metadata required for menuselect and the rest of the build system is extracted from the source files. Since I had a few hours to kill on an airplane today, I decided to improve this situation... so now the system caches the extracted metadata and uses it to build the menuselect 'tree' as much as it can. The result of this is that when a single source file is changed, only the metadata for that file needs to be extracted again, and the rest is used from the cache files. I also reduced the number of forked processes required to do the metadata extraction; it was actually possible to do most of what we needed in the Makefiles themselves without using any shell scripts at all! On my laptop, these changes resulted in an 80% decrease in the time required for the 'menuselect.makeopts' automatic check to occur after editing a single source file. While doing this work I also cleaned up a few minor things in the Makefiles, adding a check for 'awk' to the configure script and changed all remaining places we use 'grep' or 'awk' to use the ones found by the configure script, and changed the 'prep_tarball' script to build the menuselect metadata so that tarballs of Asterisk will include it and won't require the user to wait while it is extracted after unpacking. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@93180 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
c9f389a45d |
When compiling with DETECT_DEADLOCKS, don't spam the CLI with messages
about possible deadlocks. Instead just print the intended single message every five seconds. (closes issue 11537, reported and patched by dimas) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@92875 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b4b34f080d |
Test directly for the API that fixed AST-2007-026, to ensure that older
versions of PostgreSQL are no longer acceptable. (Closes issue #11526) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@92463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
7be75dbbc9 |
Fix another bug in the DEBUG_THREADS code. The ast_mutex_init() function had
the mutex attribute object marked as static. This means that multiple threads initializing locks at the same time could step on each other and end up with improperly initialized locks. (found when tracking down locking issues related to issue #11080) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@91828 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
168a6bed5a |
I love fixing lock related errors in the lock debugging code. That's about as
ironic as it gets in Asterisk programming land. Anyway, I spotted this bug while trying to track down why systems are locking up and acting weird in issue #11080. The mutex attribute object was marked as static in this function when it should not have been. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@91826 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
0522e3d872 |
Add a new module flag to indicate that a build sum is present. Modules built
against older Asterisk 1.4 headers will now load properly with just a warning indicating that they are old and may cause problems. (patch by paravoid) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@91501 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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95dce3bbba |
Make sure logger is reloaded at general reload in the cli.
(Discovered during Asterisk training in Portugal) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@91366 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
eccfefae2c |
Fix some crashes in chan_iax2 that were reported as happening on Mac systems.
It turns out that the problem was the Mac version of the ast_atomic_fetchadd_int() function. The Mac atomic add function returns the _new_ value, while this function is supposed to return the old value. So, the crashes happened on unreferencing objects. If the reference count was decreased to 1, ao2_ref() thought that it had been decreased to zero, and called the destructor. However, there was still an outstanding reference around. (closes issue #11176) (closes issue #11289) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@91070 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
fc232a9a27 |
Modify file.h to maintain API compatibility with earlier versions. If a recent
compiler is being used, then a warning will show up for any modules still using the old name "private" instead of "_private". (patch suggested by paravoid) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@91032 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
b8c8dee621 |
Solaris requires the inclusion of sys/loadavg.h for getloadavg().
Reported by: snuffy Patch by: snuffy,tilghman (Closes issue #11430) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@90753 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7b052b78e1 |
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop. This is accomplished by creating a datastore on the calling channel which has a linked list of all devices dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore is detached from the channel and destroyed. This change also introduces some side effects to the code which I shall enumerate here: 1. Datastore inheritance has been backported from trunk into 1.4 2. A large chunk of code has been removed from app_dial. This chunk is the section of code which handles the call forward case after the channel has been requested but before it has been called. This was removed because call-forwarding still works fine without it, it makes the code less error-prone should it need changing, and it made this set of changes much less painful to just have the forwarding handled in one place in each module. 3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore which is attached to the channel may be created and attached in either app_dial or app_queue, so they need a common place to find the datastore info. This approach was taken in case similar datastores are needed in the future, there will be a common place to add them. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@90735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b18d1bdd1a |
Preserve the indication currently playing on a channel when a masquerade operation happens. (issue #BE-88)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@90548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1f8caa100d |
Change the behavior of ao2_link(). Previously, in inherited a reference.
Now, it automatically increases the reference count to reflect the reference that is now held by the container. This was done to be more consistent with ao2_unlink(), which automatically releases the reference held by the container. It also makes it so it is no longer possible for a pointer to be invalid after ao2_link() returns. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@90348 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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82757dfaa1 |
Add some notes on the behavior of ao2_unlink() after a discussion with Tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@90310 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2c11f7d18e |
Use of "private" as a field name in a header file messes with C++ projects
Reported by: chewbacca Patch by: casper (Closes issue #11401) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@90155 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2fc83c3db1 |
This set of changes is to make some callerID handling thread-safe.
The ast_set_callerid() function needed to lock the channel. Also, the handlers for the CALLERID() dialplan function needed to lock the channel when reading or writing callerid values directly on the channel structure. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@90145 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1380790b60 |
Merge a change from team/russell/chan_refcount ...
This makes ast_stopstream() thread-safe. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@90142 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d0cd120d47 |
- update documentation for some of the goto functions to note that they
handle locking the channel as needed - update ast_explicit_goto() to lock the channel as needed git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89893 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0c3ec937ce |
If we get a codec offer using a well-known payload type, but using it for another
codec that we don't know, Asterisk did not remove that codec from the list. With this patch, we remove the codec from audio and video rtp objects and deny it ever existed. Thanks to lasse for testing. (closes issue #11376) Reported by: lasse Patches: bug11376.txt uploaded by oej (license 306) Tested by: lasse git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89630 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |