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${ noResults }
1392 Commits (0135ff19edd53c46c435f9de9eab57f42eba4a4a)
Author | SHA1 | Message | Date |
---|---|---|---|
|
415e95df9a |
Clarify documentation on how we store codec preference lists.
(closes issue #18397) Reported by: birgita git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@296990 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
d32582d42f |
The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@295790 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
7876a359ad |
Fix a deadlock in device state change processing.
Copied from some notes from the original author (Russell): Deadlock scenario: Thread 1: device state change thread Holds - rdlock on contexts Holds - hints lock Waiting on channels container lock Thread 2: SIP monitor thread Holds the "iflock" Holds a sip_pvt lock Holds channel container lock Waiting for a channel lock Thread 3: A channel thread (chan_local in this case) Holds 2 channel locks acquired within app_dial Holds a 3rd channel lock it got inside of chan_local Holds a local_pvt lock Waiting on a rdlock of the contexts lock A bunch of other threads waiting on a wrlock of the contexts lock To address this deadlock, some locking order rules must be put in place and enforced. Existing relevant rules: 1) channel lock before a pvt lock 2) contexts lock before hints lock 3) channels container before a channel What's missing is some enforcement of the order when you involve more than any two. To fix this problem, I put in some code that ensures that (at least in the code paths involved in this bug) the locks in (3) come before the locks in (2). To change the operation of thread 1 to comply, I converted the storage of hints to an astobj2 container. This allows processing of hints without holding the hints container lock. So, in the code path that led to thread 1's state, it no longer holds either the contexts or hints lock while it attempts to lock the channels container. (closes issue #18165) Reported by: antonio ABE-2583 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@294384 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
d5ad3e8246 |
Allow PRI to build properly when using --with-pri.
Use the directories found for the parent when using lib dependencies. (closes issue #17314) Reported by: tzafrir Patches: 17314-withdeps.diff uploaded by qwell (license 4) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@290750 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
8f6ca92a9d |
Change RFC2833 DTMF event duration on end to report actual elapsed time.
The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289797 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
b94a0e14a2 |
Solaris compatibility fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@288636 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
210ec76dba |
Underquoted definition of stringfield macro.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@288116 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
7fd5264698 |
Inherit CHANNEL() writes to both sides of a Local channel
Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286059 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
ab40f7e68d |
Fix Mac OS X build.
This also fixes a rather grievous calculation error for the offset of ast_fdset, which was masked on Linux and FreeBSD, because these platforms check the first 256 FDs regardless of the bitmask setting (due to backwards compatibility). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@285889 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
408cf5d2a2 |
Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows.
This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing a potential crash bug in all supported releases. (closes issue #17678) Reported by: russell Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select Review: https://reviewboard.asterisk.org/r/824/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@284478 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
0ebd53bba8 |
remove empty audiohook write list on channel
If a channel has an audiohook write list created on it, that list stays on the channel until the channel is destroyed. There is no reason to keep that list on the channel if it becomes empty. If it is empty that just means we are doing needless translating for every ast_read and ast_write. This patch removes the audiohook list from the channel once it is detected to be empty on either a read or write. If a audiohook is added back to the channel after this list is destroyed, the list just gets recreated as if it never existed to begin with. (closes issue #17630) Reported by: manvirr Review: https://reviewboard.asterisk.org/r/799/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@279945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
20454a45d2 |
Provide a default value for DAHDI_TRANSCODE so when DAHDI is not installed
menuselect doesn't get confused: Unknown value '' found in build_tools/menuselect-deps for DAHDI_TRANSCODE git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@279344 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
8618bacfd8 |
Do not queue up DTMF frames while a call is on hold.
(Fixes ABE-2110) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@278167 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
938b6264ff |
Document that a leading and trailing slash is expected for test categories.
Also, emit a warning if a test is registered without one of these. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@275021 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
14550f93d0 |
Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs.
(closes issue #17407) Reported by: pdf Patches: 20100527__issue17407.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/751/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273793 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
0dbe3c9921 |
Backport unit test API to 1.4.
Review: https://reviewboard.asterisk.org/r/750/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@272878 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
aafa3c7464 |
Remove an unnecessary assignment that causes a DEBUG_THREADS build failure on mac os x.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@271340 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
f311d0ac18 |
Fix a build problem on Mac OS X with DEBUG_THREADS enabled.
This set of changes was already in trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@271339 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
43cd1e6f6a |
Make the default install path appear to be /usr on Linux, instead of /usr/local.
Also, reorganize the options, so that they're more alphabetical. (closes issue #17013) Reported by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@267759 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
bc766d5b69 |
1.4 version of the dead PLC code fix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@267539 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
7e9a71f5d8 |
Merged revisions 265320,265467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines Add the FullyBooted AMI event It is possible to connect to the manager interface before all Asterisk modules are loaded. To ensure that an application does not send AMI actions that might require a module that has not yet loaded, the application can listen for the FullyBooted manager event. It will be sent upon connection if all modules have been loaded, or as soon as loading is complete. The event: Event: FullyBooted Privilege: system,all Status: Fully Booted Review: https://reviewboard.asterisk.org/r/639/ ........ r265467 | twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line Merge the rest of the FullyBooted patch ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@265570 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
ed585b5bb3 |
Don't hang up on a queue caller if the file we attempt to play does not exist.
This also fixes a documentation mistake in file.h that made my original attempt to correct this problem not work correctly. (closes issue #17061) Reported by: RoadKill git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@265089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
94238573fa |
Fix grammatical error in comment.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@264999 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
b91ff0498e |
Allow ast_safe_sleep to defer specific frames until after the sleep has concluded.
From reviewboard Background: A Digium customer discovered a somewhat odd bug. The setup is that parties A and B are bridged, and party A places party B on hold. While party B is listening to hold music, he mashes a bunch of DTMF. Party A takes party B off hold while this is happening, but party B continues to hear hold music. I could reproduce this about 1 in 5 times. The issue: When DTMF features are enabled and a user presses keys, the channel that the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read from the channel during the sleep, the frame is dropped. Thus the unhold indication is never made to the channel that was originally placed on hold. The fix: Originally, I discussed with Kevin possible ways of fixing the specific problem reported. However, we determined that the same type of problem could happen in other situations where ast_safe_sleep() is used. Using autoservice as a model, I modified ast_safe_sleep_conditional() to defer specific frame types so they can be re-queued once the sleep has finished. I made a common function for determining if a frame should be deferred so that there are not two identical switch blocks to maintain. Review: https://reviewboard.asterisk.org/r/674/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@264996 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
610c1dc6bf |
1.4 version of PLC fix.
Analogous to trunk revision 264452, but without the change to chan_sip since it is not necessary in this branch. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@264541 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
136073265c |
Internal timing is now on by default, if you're using DAHDI 2.3 or above.
The reason for ensuring DAHDI 2.3 or above is that this version ensures that a timer is always available, whereas in previous versions, it was possible for DAHDI to be loaded, but have no drivers to actually generate timing. If internal_timing was turned on in this circumstance, a complete lack of audio would result. This is the reason why internal_timing was not on by default. However, now that DAHDI ensures the availability of a timer, there is no reason for this setting to be off (and in fact, it solves a great many initial user problems). (closes issue #15932) Reported by: dimas Patches: 20100519__issue15932.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@264248 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
49c20ee9c5 |
Fixes crash in audiohook_write_list
The middle_frame in the audiohook_write_list function was being freed if a audiohook manipulator returned a failure. This is incorrect logic. This patch resolves this and adds detailed descriptions of how this function should work and why manipulator failures must be ignored. (closes issue #17052) Reported by: dvossel Tested by: dvossel (closes issue #16196) Reported by: atis Review: https://reviewboard.asterisk.org/r/623/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@260049 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
2b74f9be7b |
Allow application options with arguments to contain parentheses, through a variety of escaping techniques.
Fixes SWP-1194 (ABE-2143). Review: https://reviewboard.asterisk.org/r/604/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@257544 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
c94931c9a4 |
Fix DEBUG_THREADS issue with out-of-tree modules.
Take 2, without ABI breakage this time. Review: https://reviewboard.asterisk.org/r/588/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@254714 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
6eca8e3059 |
Add doxygen for acl.h
Review: https://reviewboard.asterisk.org/r/528 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@254552 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
8371df4e7b |
Revert revisions 254046 and 254098.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@254161 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
16a9a742a4 |
Allow out-of-tree modules to load, regardless of DEBUG_THREADS/DEBUG_CHANNEL_LOCKS differences.
This can be guaranteed by forcing the ABI to no longer change when these compiler flags are set. An unfortunate side-effect to this is that there is an ABI change here. However, there is some mitigation. Existing modules *will* fail to load since they would require functions that no longer exist. Review: https://reviewboard.asterisk.org/r/508/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@254046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
1b2dbf6c1a |
Revert API change in release branches
This re-renames ast_rtp_update_source to ast_rtp_new_source git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@253158 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
529e8af144 |
Merged revisions 252089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@252175 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
012c6fc5e2 |
Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script.
Changed after discussion on the -dev list about possible unnecessary build failures, due to checkouts/untars causing these special source files to possibly be newer than their resulting C files. This should additionally ensure that nobody need learn about extra Makefile arguments to ensure the proper files get rebuilt when changes are made to these special source files. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@242520 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
2418c3fe1e |
Add a flag to disable the Background behavior, for AGI users.
This is in a section of code that relates to two other issues, namely issue #14011 and issue #14940), one of which was the behavior of Background when called with a context argument that matched the current context. This fix broke FreePBX, however, in a post-Dial situation. Needless to say, this is an extremely difficult collision of several different issues. While the use of an exception flag is ugly, fixing all of the issues linked is rather difficult (although if someone would like to propose a better solution, we're happy to entertain that suggestion). (closes issue #16434) Reported by: rickead2000 Patches: 20091217__issue16434.diff.txt uploaded by tilghman (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: rickead2000 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@237405 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
f1e72bf7c7 |
Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces.
There was conditional code (based on build platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that was removed since it is fixed in newer versions of Solaris/OpenSolaris, but I am still running into it on Solaris 10 x86 so add a configure-time check for it. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@236585 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
8291fb5083 |
Correct CDR dispositions for BUSY/FAILED
This patch is simple in that it reorders the disposition defines so that the fix for issue 12946 works properly (the default CDR disposition was changed to AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all CDR records are written. The side effects of CDR changes are scary, so I'm documenting the test cases performed to attempt to catch any regressions. The following tests were all performed using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls B (busy) Hangup C Hangup A (Both SIP and features) A calls B A blind transfers to C Hangup C (Both SIP and features) A calls B A attended transfers to C Hangup C A calls B A attended transfers to C (SIP) C blind transfers to A (features) Hangup A All of the test scenario CDRs matched. The following tests were performed just with the patch to ensure proper operation (with unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) (closes issue #16180) Reported by: aatef Patches: bug16180.patch uploaded by jpeeler (license 325) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@235635 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
4dfd9d4800 |
Set a module load priority for format modules.
A recent change to app_voicemail made it such that the module now assumes that all format modules are available while processing voicemail configuration. However, when autoloading modules, it was possible that app_voicemail was loaded before the format modules. Since format modules don't depend on anything, set a module load priority on them to ensure that they get loaded first when autoloading. This version of the patch is specific to Asterisk 1.4 and 1.6.0. These versions did not already support module load priority in the module API. This adds a trivial version of this which is just a module flag to include it in a pass before loading "everything". Thanks to mmichelson for the review! (closes issue #16412) Reported by: jiddings Tested by: russell Review: https://reviewboard.asterisk.org/r/445/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@233782 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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6a680d9b28 |
Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list.
(closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@231614 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
a3887d4511 |
Perform limited bounds checking when destroying ast_mutex_t structures to make sure we don't try to use negative indices.
(closes issue #15588) Reported by: zerohalo Patches: 20090820__issue15588.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@228827 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
41f0b0de9c |
Fix documentation (pointed out by TheDavidFactor on #-dev)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@226304 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
217a115da8 |
detect ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi
* Set OSARCH to linux-gnu even if host_os is linux-gnueabi * When checking if we are Linux, check OSARCH rather than host_os The newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is tested for the value of 'linux-gnu' in one or two places in the tree. This patch also fixes the check libcap to check for $OSARCH rather than $host_os . See also: http://wiki.debian.org/ArmEabiPort git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@225957 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
6e8a455534 |
Fix documentation for ast_softhangup() and correct the misuse thereof.
(closes issue #16103) Reported by: majorbloodnok git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@225105 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
9d65850202 |
Isolate frames returned from a DSP instance or codec translator.
The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@224931 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
ff6a5575ad |
Make filestream frame handling safer by isolating frames before returning them.
This patch is related to a number of issues on the bug tracker that show crashes related to freeing frames that came from a filestream. A number of fixes have been made over time while trying to figure out these problems, but there re still people seeing the crash. (Note that some of these bug reports include information about other problems. I am specifically addressing the filestream frame crash here.) I'm still not clear on what the exact problem is. However, what is _very_ clear is that we have seen quite a few problems over time related to unexpected behavior when we try to use embedded frames as an optimization. In some cases, this optimization doesn't really provide much due to improvements made in other areas. In this case, the patch modifies filestream handling such that the embedded frame will not be returned. ast_frisolate() is used to ensure that we end up with a completely mallocd frame. In reality, though, we will not actually have to malloc every time. For filestreams, the frame will almost always be allocated and freed in the same thread. That means that the thread local frame cache will be used. So, going this route doesn't hurt. With this patch in place, some people have reported success in not seeing the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2) Tested by: aragon, russell (closes issue #15817) Reported by: zerohalo Tested by: zerohalo (closes issue #15845) Reported by: marhbere Review: https://reviewboard.asterisk.org/r/386/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222878 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
3e5979a040 |
fixes an ast_netsock_list memory leak.
ABE-1998 Review: https://reviewboard.asterisk.org/r/395/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222877 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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2ad7cb7e87 |
Fix ao2_iterator API to hold references to containers being iterated.
See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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96564de25e |
Change the SSRC by default when our media stream changes
Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221086 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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ea3f81a68f |
Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down.
This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h. (closes issue #15316) Reported by: vmarrone Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/362/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219136 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |