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${ noResults }
22376 Commits (745fcdbffe207d1cbbfabac3c8e8ea1a33df6a48)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Mark Murawki
|
745fcdbffe |
Fix dev-mode compiler warning about gnu_printf
(related to ASTERISK-19575) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361039 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Murawki
|
e4252eac10 |
Allow the Hangup manager action to match channels by regex
* Hangup now can take a regular expression as the Channel option. If you want to hangup multiple channels, use /regex/ as the Channel option. Existing behavior to hanging up a single channel is unchanged, but if you pass a regex, the manager will send you a list of channels back that were hung up. (closes issue ASTERISK-19575) Reported by: Mark Murawski Tested by: Mark Murawski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361038 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Kinsey Moore
|
9cc6f2c59e |
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 ........ Merged revisions 360993 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360994 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Richard Mudgett
|
6a540e9087 |
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 ........ Merged revisions 360934 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360935 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Mark Michelson
|
314d459317 |
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
........
Merged revisions 360885 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360886 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Mark Michelson
|
cc2366bca0 |
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
........
Merged revisions 360863 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Richard Mudgett
|
fb796aac06 |
Misc changes to make astobj2 enhancement diffs easier to follow.
* Rename astobj2 API parameter funcname to func. * Rename astobj2 API iterator parameter to iter. * Update some documentation for OBJ_MULTIPLE. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360827 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Jonathan Rose
|
655a8d4420 |
Introducing the log message unique call identifiers feature
Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360787 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Jonathan Rose
|
d501c2ea2d |
undoing 360785 due to merging mistake
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360786 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Jonathan Rose
|
bf994f0e04 |
Introducing the log message unique call identifiers feature
Log messages will now display a call number that they are tied to (ordered for calls based on when they started). This feature is made to be minimally invasive without requiring changes to many of the existing log messages. These IDs won't show up for verbose messages on CLI (but they will in log files) This is currently in phase II of production, see more about this feature on the wiki -- https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging Review: https://reviewboard.asterisk.org/r/1823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Terry Wilson
|
dd9405db05 |
Fix setting CDR variables in the hangup extension
A previous CDR fix for setting CDR variables during a bridge via custom dialplan features broke setting CDR variables in the hangup extension. This patch fixes the issue. Review: https://reviewboard.asterisk.org/r/1794/ ........ Merged revisions 358978 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 358989 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360724 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Mark Michelson
|
01cc64585e |
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 ........ Merged revisions 360672 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360673 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Richard Mudgett
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38e892b370 |
Add global ao2 array container.
Global ao2 objects must always exist after initialization because there is no access control to obtain another reference to the global object. It is expected that module configuration could use these new API calls to replace an active configuration parameter object with an updated configuration parameter object. With these new API calls, the global object could be replaced, removed, or referenced without the risk of someone using a stale global object pointer. Review: https://reviewboard.asterisk.org/r/1824/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360627 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
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8611bea122 |
Attempt to be more helpful when using a bad ao2 object pointer.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360626 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Jonathan Rose
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65f56cda7a |
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 ........ Merged revisions 360575 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360576 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Paul Belanger
|
8331a89892 |
Blocked revisions 360476
........ 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/trunk@360537 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Paul Belanger
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dea8936f89 |
Convert ast_verb() to ast_debug() and increase log level
Rather then flood the CLI with verbose messages, we've changed the level to debug. This will help keep the CLI clean. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Jonathan Rose
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337f933977 |
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 ........ Merged revisions 360489 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360490 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Russell Bryant
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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 |
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Russell Bryant
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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 |
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Russell Bryant
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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 |
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Russell Bryant
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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 |
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Russell Bryant
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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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Richard Mudgett
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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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Richard Mudgett
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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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Jonathan Rose
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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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Kinsey Moore
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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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Jonathan Rose
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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 |
Tzafrir Cohen
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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 |
Matthew Jordan
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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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Kinsey Moore
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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 |
Sean Bright
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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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Richard Mudgett
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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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Richard Mudgett
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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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Jonathan Rose
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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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Richard Mudgett
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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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Mark Michelson
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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 |
Alec L Davis
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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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Mark Murawki
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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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Matthew Jordan
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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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Mark Murawki
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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 |
|
Matthew Jordan
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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 |
|
Paul Belanger
|
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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Richard Mudgett
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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 |
|
Paul Belanger
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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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Russell Bryant
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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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Russell Bryant
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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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Russell Bryant
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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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Russell Bryant
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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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Russell Bryant
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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 |