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${ noResults }
22201 Commits (3979ff87b4397d9f6bc58f7a2e0c4530cfb69c5d)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
3979ff87b4 |
Add tests for Digium phone support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@361228 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
|
738a78e8c1 |
apps changes necessary for Digium phone support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@361216 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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94a6534af5 |
Add Digium phone changes for the funcs directory.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@361207 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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5f612abf55 |
Add Digium phone changes for the include/asterisk directory
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@361206 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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87ea1deb44 |
Add phone changes for message.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@361200 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c343f0be25 |
Update everything in the main directory for Digium phone additions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@361199 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c71f0ac4b1 |
Update sample config files.
This just involved updating manager.conf.sample to have the new "message" write permission. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@361178 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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344c613492 |
Create a Digium phones branch of Asterisk 10.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@361140 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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4c5dcf2da5 |
Make the MeetMeAdmin N command (mute all nonadmins) not mute admins
(Closes Issue ASTERISK-19335) Reported by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/ ........ Merged revisions 361090 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@361091 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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bb61612873 |
Fix the display of documentation for Transfer
This came up while fixing documentation generation for many other cases where the argument separator was not being displayed properly. Now that it is displayed properly, it shows up in the wrong place for Transfer since the '/' is only required if Tech is present. (related to issue ASTERISK-18168) ........ Merged revisions 361040 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@361041 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d3ec89d9b6 |
Stop sending out RTCP if RTP is inactive
This change prevents Asterisk from sending RTCP receiver reports during a remote bridge since it is no longer receiving media and should not be reporting anything. (related to ASTERISK-19366) ........ Merged revisions 360987 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360993 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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e9591a724d |
Fix logger deadlock on Asterisk shutdown.
The logger_thread() had an exit path that failed to release the logmsgs list lock. * Make logger_thread() exit path unlock the logmsgs list lock. * Made ast_log() not queue any messages to the logmsgs list if the close_logger_thread flag is set. (issue ASTERISK-19463) Reported by: Matt Jordan ........ Merged revisions 360933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360934 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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59a886e25f |
Fix potential race condition during call pickup.
Prior to this patch, a connected line update was queued during
call pickup and then an answer frame was queued. The original
caller would presumably then have his connected line updated
and then the call would be answered.
In actuality, the answer frame was not how the call ended up
being answered. Rather, an odd section in app_dial that checks
if the called channel's state is up.
The result is that the order of the connected line update and
the answer were variable. In most cases, this wasn't actually
a bad thing. However, if the 'I' option was passed to dial, the
connected line update would be inhibited.
The fix is to queued the connected line after the answer frame is
queued. This way the race in app_dial is between two
conditions resulting in an answer. This way the connected line
update occurs after the answer every time.
(closes issue ASTERISK-19183)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Mark Michelson
Patches:
ASTERISK-19183.patch uploaded by Mark Michelson (license 5049)
........
Merged revisions 360884 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360885 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
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4477d99b29 |
Improve accuracy of identifying information sent in dialog-info SIP NOTIFY requests.
This change makes use of connected party information in addition to caller ID in order
to populate local and remote XML elements in the dialog-info NOTIFYs.
(closes issue ASTERISK-16735)
Reported by: Maciej Krajewski
Tested by: Maciej Krajewski
Patches:
local_remote_hint2.diff uploaded by Mark Michelson (license 5049)
........
Merged revisions 360862 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
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d10ae278e4 |
Destroy configs when they are no longer used
https://reviewboard.asterisk.org/r/1834/ ........ Merged revisions 360712 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360717 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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9829106be8 |
Make a debug message regarding subscription changes more accurate.
I was getting confused during some testing why Asterisk was saying that a subscription was being added when it was clearly being removed. This fixes that confusion. ........ Merged revisions 360625 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b4a1888fd7 |
Updates config with bootstrap where I changed configure.ac in r360488
(issue ASTERISK-17842) Reported by: Bryon Clark ........ Merged revisions 360574 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360575 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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8d8d0724b3 |
Fix BETTER_BACKTRACES library detection for Fedora/RedHat/CentOS
(closes ASTERISK-17842) Reported by: Bryon Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch uploaded by Bryon Clark (license 6157) ........ Merged revisions 360488 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360489 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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55e50324bc |
Update CHANGES for r360471
........ Merged revisions 360474 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360476 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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98658b0670 |
Blocked revisions 360473
........ Fix Asterisk version typo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360475 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b735edd891 |
Increase verbosity level for ast_verb messages
While this does not fix the issue of the CLI being flooded by 'doing dnsmgr_lookup' messages, increasing the verbosity level above 5 should help minimize it. ........ Merged revisions 360471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360472 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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5c85347063 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360414 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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2bcf95931e |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b41c6c6eef |
app_jack: fix datastore memory leak in error handling path.
........ Merged revisions 360360 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360361 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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e55b86fca9 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360358 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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94fb536135 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360310 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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6b2201e80a |
Add missing initialization of update_redirecting in chan_sip.c
........ Merged revisions 360262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360263 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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751c365b96 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360139 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1ba59f0188 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360098 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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de2f65ce68 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0d6c4ddd6d |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360034 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b60963eab9 |
chan_iax2: Emit Port alongside Post in PeerStatus AMI Event.
The PeerStatus event for IAX2 channels currently includes a header named Post which should have been Port. So include Port along with Post when emitting the event. We'll remove Post in trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359982 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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01b999f833 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359980 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ba1a6fc5f7 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359898 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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078fd3f925 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359810 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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522df4c5f8 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359707 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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e10005d38f |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359694 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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3fc64ea06b |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359620 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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22b75fa244 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359559 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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be35984b09 |
Blocked revisions 359534
........ chan_gtalk: Fix use of uninitialized vars in config handling. Fix potential use of context, parkinglot, and prefs before they are initialized. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359537 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d0cb19147c |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359509 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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51dcdcd897 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359491 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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f892e9152d |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359458 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1ebd1d352f |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359454 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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43100ddd82 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359453 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b7254422e9 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359358 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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46f24f0596 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359355 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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cca76a24b6 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359260 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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40cc905d86 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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cfb780a5f5 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359162 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |