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${ noResults }
22358 Commits (5affceaa156250f240d742377576c17c6709e4f7)
Author | SHA1 | Message | Date |
---|---|---|---|
|
5affceaa15 |
func_curl: Fix leak of an ast_str in error handling code path.
........ Merged revisions 360413 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360414 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360415 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
d6d7f51476 |
chan_iax2: Use OBJ_NODATA to be a bit more explicit.
This is just a minor code cleanup change. These uses of ao2_callback() would never return anything since the callbacks always returned 0. However, be more explicit that no returned results are wanted by specifying OBJ_NODATA. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360369 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
0ec73946fa |
app_page: Fix a memory leak on every Page().
dial_list is a dynamically allocated array that is allocated at the beginning of Page() based on how many devices will be dialed. This was never being freed. ........ Merged revisions 360363 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360364 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
71a1541b0c |
app_jack: fix datastore memory leak in error handling path.
........ Merged revisions 360360 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360361 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360362 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
cad07b3800 |
Multiple revisions 360356-360357
........ r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012) | 6 lines expression parser: Fix (theoretical) memory leak. Fix a memory leak that is very unlikely to actually happen. If a malloc() succeeded, but the following strdup() failed, the memory from the original malloc() would be leaked. ........ r360357 | russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines Rebuild parsers. This is needed to include the last fix to main/ast_expr2.y. The changes look much bigger as this regeneration of the code was done with newer versions of flex and bison. ........ Merged revisions 360356-360357 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360358 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360359 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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721f92058f |
Make number not available presentation also set screening to network provided.
Q.951 indicates that when the presentation indicator is "Number not available due to interworking" for a number then the screening indicator field should be "Network provided". * Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE when the presentation is "Number not available due to interworking". This fix makes Asterisk consistent and it also makes it consistent with earlier branches as far as this presentation value is concerned. * Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions handle the "Number not available due to interworking" case better in sig_pri.c. This change is possible because the minimum required libpri version (v1.4.11) has the necessary defines in libpri.h. ........ Merged revisions 360309 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360310 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360311 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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df16bd973e |
Add missing initialization of update_redirecting in chan_sip.c
........ Merged revisions 360262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360263 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360264 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c6979ff581 |
Adds F option to Bridge application
Similar to dial and queue F option. (Closes issue ASTERISK-19282) Reported by: To Patches: bridge_f-v3.diff uploaded by To (license 6347) Review: https://reviewboard.asterisk.org/r/1825/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c5b3db1956 |
Kill off red blobs in most of main/*
Everything still compiled after making these changes, so I assume these whitespace-only changes didn't break anything (and shouldn't have). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360190 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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1d1c28ac4b |
Update install_prereq script to include missing GSM library for debian amd move SQLite3.
(closes issue ASTERISK-19367) Reported by: Andrew Latham Patches: debian_install_prereq.diff uploaded by Andrew Latham (license 5985) ........ Merged revisions 360138 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360139 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360140 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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ab6f40bd12 |
Also detect gmime 2.6
Also detect gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com> ........ Merged revisions 360087 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360098 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360137 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c88d1c8337 |
Ensure Asterisk sends a BYE when pending on the final response to a re-INVITE
When Asterisk detects a hangup and cannot send a BYE due to a pending INVITE, it sets the pendingbye flag and waits for the final response to that INVITE. When the response is received, it transmits the BYE. If, however, that INVITE request is a pending re-INVITE, it needs to first send a CANCEL request to terminate the pending re-INVITE. In that circumstance, Asterisk was, in some scenarios, clearing the pendingbye flag after processing the CANCEL request and not checking for a pending BYE when receiving the final 487 response to the INVITE. This patch ensures that if the pendingbye flag is set, it is honored regardless of the nature of the INVITE request currently in flight. (closes issue ASTERISK-19365) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license 6283) Review: https://reviewboard.asterisk.org/r/1807 ........ Merged revisions 360086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360088 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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6ff8f14865 |
Prevent Echo() from relaying control, null, and modem frames
Echo()'s description states that it echoes audio, video, and DTMF except for # while it actually echoes any frame that it receives other than DTMF #. This was causing frame storms in the test suite in some circumstances where Echo() was attached to both ends of a pair of local channels and control frames were being periodically generated. Echo()'s behavior and description have been modifed so that it only echoes media and non-# DTMF frames. ........ Merged revisions 360033 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360034 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360036 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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3a231e090f |
chan_iax2: Correct spelling of 'Port' header in IAX2 PeerStatus AMI Events
The PeerStatus event for IAX2 channels currently includes a header named Post which should have been Port. Post was removed and the AMI version has been updated to 1.3. ........ Merged revisions 359982 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359983 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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334f13d8b8 |
Allow AMI action callback to be reentrant.
Fix AMI module reload deadlock regression from ASTERISK-18479 when it tried to fix the race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. Locking the ao2 object guaranteed that there were no active callbacks that mattered when ast_manager_unregister() was called. Unfortunately, this causes the deadlock situation. The patch stops locking the ao2 object to allow multiple threads to invoke the callback re-entrantly. There is no way to guarantee a module unload will not crash because of an active callback. The code attempts to minimize the chance with the registered flag and the maximum 5 second delay before ast_manager_unregister() returns. The trunk version of the patch changes the API to fix the race condition correctly to prevent the module code from unloading from memory while an action callback is active. * Don't hold the lock while calling the AMI action callback. (closes issue ASTERISK-19487) Reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1818/ Review: https://reviewboard.asterisk.org/r/1820/ ........ Merged revisions 359979 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359980 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359981 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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3714e8b1e5 |
Convert MuteAudio documentation to XML.
* Added missing error exits with cause in manager_mutestream(). * Cleaned up manager_mutestream() and func_mute_write(). * Some whitespace and comment cleanup. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359942 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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0399daaa2e |
Prevent chanspy from binding to zombie channels
This patch addresses a bug with chanspy on local channels which roughly 50% of the time would create a situation where chanspy can latch onto a zombie channel, keeping the zombie alive forever and causing the channel doing the spying to never be able to hang up. (closes issue ASTERISK-19493) Reported by: lvl Review: https://reviewboard.asterisk.org/r/1819/ ........ Merged revisions 359892 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359898 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359905 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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dd4a3b1825 |
Simplify some code in ast_app_run_sub().
* Remove unnnecessary const from const char * const var declaration in the ast_app_run_macro() and ast_app_run_sub() prototypes. The second const is unnecessary. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359904 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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827f2eae92 |
Revert the pre-dial addition.
The code may be just fine, but it had not received a "ship it!" on review board yet. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359857 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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9ac6938e09 |
Missed lastinvite CSeq int to uint32_t change
from Review: https://reviewboard.asterisk.org/r/1699/ ........ Merged revisions 359809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359810 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359811 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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d6e1c619d4 |
Fix warning from commit r359705 (predial options for app_dial)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359772 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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cca1f9f48a |
Fix remotely exploitable stack overflow in HTTP manager
There exists a remotely exploitable stack buffer overflow in HTTP digest authentication handling in Asterisk. The particular method in question is only utilized by HTTP AMI. When parsing the digest information, the length of the string is not checked when it is copied into temporary buffers allocated on the stack. This patch fixes this behavior by parsing out pre-defined key/value pairs and avoiding unnecessary copies to the stack. (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested by: Matt Jordan ........ Merged revisions 359706 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359707 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359708 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c65b41f57a |
Add options PreDial options 'b' and 'B' to app_dial
* Added 'b' and 'B' options to Dial. These options will allow you to run last-minute dialplan on the caller and callee channels while the Dial application is executing, but before the call is started. For example you can use the 'b' option to run dialplan on the callee channel to get the name of the newly created channel right away. Review: https://reviewboard.asterisk.org/r/1229/ (closes issue: ASTERISK-19548) Reported by: Mark Murawski Tested by: Mark Murawski, Stefan Schmidt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359705 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c61d49d5cc |
Fix remotely exploitable stack overrun in Milliwatt
Milliwatt is vulnerable to a remotely exploitable stack overrun when using the 'o' option. This occurs due to the milliwatt_generate function not accounting for AST_FRIENDLY_OFFSET when calculating the maximum number of samples it can put in the output buffer. This patch resolves this issue by taking into account AST_FRIENDLY_OFFSET when determining the maximum number of samples allowed. Note that at no point is remote code execution possible. The data that is written into the buffer is the pre-defined Milliwatt data, and not custom data. (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by Russell Bryant (license 6283) Note that this patch was written by Russell, even though Matt uploaded it ........ Merged revisions 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ Merged revisions 359656 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359694 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359704 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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31462e7bd6 |
Remove unused variable ‘srch’
Missed on the previous commit git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359651 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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e9703da1d5 |
Add missing connected line macro calls to initial dial for Dial and Queue apps.
The connected line interception macros do not get executed when the outgoing channel is initially created and that channel's caller-id is implicitly imported into the incoming channel's connected line data. If you are using the interception macros, you would expect that they get run for every change to a channel's connected line information outside of normal dialplan execution. Review: https://reviewboard.asterisk.org/r/1817/ ........ Merged revisions 359609 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359620 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359644 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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831af9fbc7 |
Remove some dead code found in _sip_show_peers()
Review: https://reviewboard.asterisk.org/r/1696/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359607 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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44434bf1cf |
chan_iax2: Fix use of uninitialized sockaddr_in in try_transfer().
Initialize a struct sockaddr_in in try_transfer() so that the code isn't (potentially) trying to read from it while uninitialized. ........ Merged revisions 359558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359559 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359560 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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3b0eb28d86 |
chan_gtalk: Fix potential use of uninitialized variable.
Avoid potential use of idroster in gtalk_alloc() before it has been initialized. ........ Merged revisions 359508 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359509 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359510 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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45205716d7 |
app_chanisavail: Fix use of uninitialized variable.
Ensure that status is set before it is used by resetting it during each loop iteration. This could have resulted in incorrect results from this app. ........ Merged revisions 359486 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359491 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359495 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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69f19a5225 |
udptl: Ensure fec[] in udptl_build_packet() is initialized.
Scan results indicated that this array could be used uninitialized. At a quick look, it looks correct. In any case, initializing it is a Good Thing (tm). ........ Merged revisions 359457 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359458 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359459 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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28881524dc |
app.h: Always initialize AST_DECLARE_APP_ARGS().
This patch ensures that the struct defined by AST_DECLARE_APP_ARGS() is always fully initialized. I'm not sure if this fixes any real bugs, but it silences a bunch of warnings from coverity, and is generally a good thing to do anyway. ........ Merged revisions 359452 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359454 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359456 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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9b31bd3cd8 |
Fix deadlock potential with some ast_indicate/ast_indicate_data calls.
Calling ast_indicate()/ast_indicate_data() with the channel lock held can result in a deadlock with a local channel because of how local channels need to avoid deadlock. ........ Merged revisions 359451 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359453 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359455 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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a699bb72ad |
Add tests for main/jitterbuf.c
This patch adds unit tests for main/jitterbuf.c. This includes checking for the following: * Nominal insertion and retrieval of frames * Insertion and retrieval of frames where the frames are inserted out of order with respect to the previous frame * Insertion and retrieval of frames where some number of frames that would occur in the expected sequence are instead dropped * Insertion and retrieval of frames with an arrival time that does not occur at the same rate as the surrounding frames * Resynchronization of the jitter buffer when an inserted frame breaks the resynchronization threshold * Overfilling of the jitter buffer For each of the tests, both JB_TYPE_VOICE and JB_TYPE_CONTROL permutations exist. Review: https://reviewboard.asterisk.org/r/1815 (issue: ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359406 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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a22b6f6e4b |
Three copies of the file contents in channel_internal.h are a bit excessive.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359360 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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40289b63db |
Fix incorrect jitter buffer overflow due to missed resynchronizations
When a change in time occurs, such that the timestamps associated with frames being placed into an adaptive jitter buffer (implemented in jitterbuf.c) are significantly different then the previously inserted frames, the jitter buffer checks to see if it needs to be resynched to the new time frame. If three consecutive packets break the threshold, the jitter buffer resynchs itself to the new timestamps. This currently only occurs when history is calculated, and hence only on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other hand, are never passed to the history calculations. Because of this, if the jump in time is greater then the maximum allowed length of the jitter buffer, the JB_TYPE_CONTROL frames are dropped and no resynchronization occurs. Alterntively, if the overfill logic is not triggered, the JB_TYPE_CONTROL frame will be placed into the buffer, but with a time reference that is not applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger the overflow logic until reads from the jitter buffer reach the errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames are unlikely to occur in multiples, it perform the resynchronization on any JB_TYPE_CONTROL frame that breaks the resynch threshold. Note that this only impacts chan_iax2, as other consumers of the adaptive jitter buffer use the abstract jitter buffer API, which does not use JB_TYPE_CONTROL frames. Review: https://reviewboard.asterisk.org/r/1814/ (closes issue ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw (license 5722) ........ Merged revisions 359356 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359358 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359359 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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2019a7e6b9 |
Fix Dial m and r options and forked calls generating warnings for voice frames.
When connected line support was added, the wait_for_answer() variable single changed its meaning slightly. Unfortunately, the places where single was used did not necessarily get updated to reflect that change. Also audio/video frames were sent to all forked calls when the endpoints were never made compatible. * Don't pass audio/video media frames when the channels have not been made compatible. * Added handling of AST_CONTROL_SRCCHANGE to app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD because that frame can also pass a requested MOH class. (closes issue ASTERISK-16901) Reported by: Chris Gentle (closes issue ASTERISK-17541) Reported by: clint Review: https://reviewboard.asterisk.org/r/1805/ ........ Merged revisions 359344 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359355 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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6df0ae5c1f |
Force non-inlining of ao2_iterator_destroy when TEST_FRAMEWORK is enabled
In r357272, astobj2 was changed to automatically enable REF_DEBUG when the TEST_FRAMEWORK flag was enabled. Unfortunately, some compilers (gcc 4.5.1 at least) will attempt to inline ao2_iterator_destroy in handle_astobj2_test. This by itself is not a problem; unfortunately, the compiler believes that there is a code path wherein an object allocated on the stack will be free'd. As warnings are treated as errors, this prevents compilation of astobj2. This patch works around that by adding the noinline attribue to ao2_iterator_destroy, but only if the TEST_FRAMEWORK flag is enabled. Preventing inlining is only needed for the test method defined in astobj2, which is also only enabled if TEST_FRAMEWORK is enabled. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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00b270833f |
Fix bogus reads/writes of console log levels in asterisk.c
This patch updates the NUMLOGLEVELS define in logger.h to 32, to match the fact that logger.c implements 32 log levels (because of the custom log level stuff). asterisk.c uses this define to size an array of levels per remote console. This array is modified in ast_console_toggle_loglevel(), which is called by the "logger set level" CLI command. While the documentation for the CLI command doesn't make it terribly obvious, you can use this CLI command to toggle a custom log level on a remote console, as well. However, doing so led to an invalid array index in asterisk.c. This array is read from any time a log message is written to a console. So, all custom log level messages resulted in a bogus read if a remote console was connected. ........ Merged revisions 359259 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359260 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359261 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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6c9f009b6d |
Fix invalid reads/writes due to incorrect sizeof().
These few places in the code used sizeof() on h_addr in struct hostent. This is sizeof(char *). The correct way to get the size of this address is to use h_length. This error would result in reads/writes of 8 bytes instead of 4 on 64-bit machines. ........ Merged revisions 359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359212 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359213 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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6ac425df31 |
Fix inaccurate sizeof() in sched.c.
This code just needed sizeof(int), not sizeof(int *). ........ Merged revisions 359157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359162 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359166 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c1d9194482 |
Fix incorrect sizeof() in astman.
........ Merged revisions 359116 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359117 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359118 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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5ad03ac4a1 |
Fix incorrect usage of sizeof() in res_crypto.
In this case, just remove the memset(). There was a redundant memset that is done correctly just 2 lines later. ........ Merged revisions 359110 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359114 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359115 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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b58f44b0e9 |
Fix broken usage of sizeof() in res_adsi.
........ Merged revisions 359088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359091 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359095 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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9410f85699 |
Fix incorrect sizeof() usage in features.c.
This didn't actually result in a bug anywhere, luckily. The only place where the result of these memcpys was used is in app_dial, and the only field that it read out of ast_call_feature was the first one, which is an int, so these memcpys always copied just enough to avoid a problem. ........ Merged revisions 359069 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359072 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359075 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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1b3cbdacd7 |
Fix incorrect sizeof() on a pointer in MD5Final().
........ Merged revisions 359059 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359060 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359061 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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6ec5c103d6 |
Don't use a buffer after it goes out of scope.
's' is set to 'workspace'. Make sure 'workspace' doesn't go out of scope while the reference to it via 's' is still used. ........ Merged revisions 359056 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359057 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359058 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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14edd30fd2 |
Blocked revisions 359054
........ Dump cache of published events when a node joins the cluster. Also use a more reliable method for stopping the poll() thread. ........ Merged revisions 359053 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359055 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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4585000039 |
Remove chan_usbradio and app_rpt.
These modules are being maintained outside of the tree and have been for a long time now, so it doesn't make sense to keep them here. Review: https://reviewboard.asterisk.org/r/1764/ ........ Merged revisions 359050 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359051 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359052 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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128c9109b0 |
Add missing channel_internal.h
...again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359011 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |