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r335510 | russell | 2011-09-13 02:24:34 -0500 (Tue, 13 Sep 2011) | 22 lines
Merged revisions 335497 via svnmerge from
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r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011) | 15 lines
Fix a crash in res_ais.
This patch resolves a crash observed in a load testing environment that
involved the use of the res_ais module. I observed some crashes where
the event delivery callback would get called, but the length parameter
incidcating how much data there was to read was 0. The code assumed
(with good reason I would think) that if this callback got called, there
was an event available to read. However, if the rare case that there's
nothing there, catch it and return instead of blowing up.
More specifically, the change always ensure that the size of the received
event in the cluster is always big enough to be a real ast_event.
Review: https://reviewboard.asterisk.org/r/1423/
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r335346 | kmoore | 2011-09-12 09:22:15 -0500 (Mon, 12 Sep 2011) | 17 lines
Merged revisions 335341 via svnmerge from
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r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) | 10 lines
Ensure frames are not written to dialed channel if ringback is requested
When a single channel was dialed and there was media to be forwarded to the
calling channel, the media was written without regard for ringback causing
silence to be heard in some circumstances. This regression was introduced
when the meaning of "single" changed to mean only the number of channels
dialed.
(closes issue ASTERISK-18083)
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r335323 | oej | 2011-09-12 15:47:13 +0200 (Mån, 12 Sep 2011) | 19 lines
Merged revisions 335319 via svnmerge from
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r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12 lines
Lock the peer->mvipvt to avoid crashes with SIP history enabled
After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt,
which cause issues with SIP history additions in combination with the max limit for
number of history entries.
Review: https://reviewboard.asterisk.org/r/1373/
(closes issue ASTERISK-18288)
Thanks to irrot for peer review. Work with this bug funded by IPvision AS
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r335321 | kmoore | 2011-09-12 08:27:04 -0500 (Mon, 12 Sep 2011) | 16 lines
Merged revisions 335320 via svnmerge from
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r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) | 9 lines
Prevent IAX2 from getting IPv6 addresses via DNS
IAX2 does not support IPv6 and getting such addresses from DNS can cause error
messages on the remote end involving bad IPv4 address casts in the presence of
IPv6/IPv4 tunnels. This patch ensures that IAX2 will not encounter IPv6
addresses via DNS queries.
(closes issue ASTERISK-18090)
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r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
Merged revisions 335064 via svnmerge from
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r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then
the 484 Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete
application dialplan logic was automatically triggered; now, explicit
dialplan usage of the application is required.
Additionally, this patch adds a new AST_CONTOL_FRAME type called
AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame,
it is an indication that the dialplan expects more digits back from the
device. If the device supports overlap dialing it should attempt to
notify the device that the dialplan is waiting for more digits; otherwise,
it can handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288)
Reported by: Mikael Carlsson
Tested by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/1416/
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r334954 | rmudgett | 2011-09-08 17:28:56 -0500 (Thu, 08 Sep 2011) | 17 lines
Merged revisions 334953 via svnmerge from
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r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011) | 10 lines
Fix crash with res_fax when MALLOC_DEBUG and "core stop gracefully" are used.
Asterisk crashes if MALLOC_DEBUG is enabled when res_fax tries to
unregister its logger level.
* Make ast_logger_unregister_level() use ast_free() instead of free().
When MALLOC_DEBUG is enabled, ast_free() does not degenerate into a call
to free(). Therefore, if you allocated memory with a form of ast_malloc
you must free it with ast_free.
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r334841 | rmudgett | 2011-09-07 14:33:38 -0500 (Wed, 07 Sep 2011) | 17 lines
Merged revisions 334840 via svnmerge from
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r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011) | 10 lines
Fix AMI action Park crash.
* Made AMI action Park not say anything to the parker channel (AMI header
Channel2) since the AMI action is a third party parking the call. (This
is a change in behavior that cannot be preserved without a lot of effort.)
* Made not play pbx-parkingfailed if the Park 's' option is used.
JIRA AST-660
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This permits the list of codecs to be specified in one configuration line,
instead of two or more, generally with the aim of either allowing all codecs
with the exception of a few or disallowing most but permitting a few.
Review: https://reviewboard.asterisk.org/r/1411/
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r334357 | rmudgett | 2011-09-02 16:08:16 -0500 (Fri, 02 Sep 2011) | 26 lines
Merged revisions 334355 via svnmerge from
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r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011) | 19 lines
MusicOnHold has extra unref which may lead to memory corruption and crash.
The problem happens when a call is disconnected and you had started a MOH
class that does not use the files mode. If you define REF_DEBUG and
recreate the problem, it will announce itself with the following warning:
Attempt to unref mohclass 0xb70722e0 (default) when only 1 ref remained,
and class is still in a container!
* Fixed moh_alloc() and moh_release() functions not handling the
state->class reference consistently.
(closes issue ASTERISK-18346)
Reported by: Mark Murawski
Patches:
jira_asterisk_18346_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Mark Murawski
Review: https://reviewboard.asterisk.org/r/1404/
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r334297 | rmudgett | 2011-09-02 12:15:08 -0500 (Fri, 02 Sep 2011) | 46 lines
Merged revisions 334296 via svnmerge from
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r334296 | rmudgett | 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011) | 39 lines
Fix potential memory allocation failure crashes in config.c.
* Added required checks to the returned memory allocation pointers to
prevent crashes.
* Made ast_include_rename() create a replacement ast_variable list node if
the new filename is longer than the available space. Fixes potential
crash and memory leak.
* Factored out ast_variable_move() from ast_variable_update() so
ast_include_rename() can also use it when creating a replacement
ast_variable list node.
* Made the filename stuffed at the end of the struct a minimum allocated
size in ast_variable_new() in case ast_include_rename() changes the stored
filename.
* Constify struct char pointers pointing to strings stuffed at the end of
the struct for: ast_variable, cache_file_mtime, and ast_config_map.
* Factored out cfmtime_new() to remove inlined code and allow some struct
pointers to become const.
* Removed the list lock from struct cache_file_mtime that was never used.
* Added doxygen comments to several structure elements and better
documented what strings are stuffed at the struct end char array.
* Reworked ast_config_text_file_save() and set_fn() to handle allocation
failure of the include file scratch pad object tracking blank lines.
* Made ast_config_text_file_save() fn[] declared with PATH_MAX to ensure
it is long enough for any filename with path. Also reduced the number of
container fileset buckets from a rediculus 180,000 to 1023.
JIRA AST-618
Review: https://reviewboard.asterisk.org/r/1378/
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r334230 | tilghman | 2011-09-01 12:30:19 -0500 (Thu, 01 Sep 2011) | 25 lines
Merged revisions 334229 via svnmerge from
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r334229 | tilghman | 2011-09-01 12:28:09 -0500 (Thu, 01 Sep 2011) | 18 lines
Create a local alias for ast_odbc_clear_cache.
As a function pointer, the reference has to be resolved at load time
irrespective of the RTLD_LAZY flag. Creating a local alias solves
this problem, because the structure is initialized with that local
function pointer, while the actual function can remain lazily linked
until runtime.
The reason why this is important is because we lazily load function
references during the module loading process, in order to obtain
priority values for each module, ensuring that modules are loaded in
the correct order. Previous to this change, when this module was
initially loaded, the module loader would emit a symbol resolution
error, because of the above requirement.
Closes ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by
Walter Doekes, patch by me)
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* Make check_rtp_timeout() remember the values returned by
ast_rtp_instance_get_timeout(), ast_rtp_instance_get_hold_timeout(), and
ast_rtp_instance_get_keepalive() instead of repeatedly calling them.
(closes issue ASTERISK-18319)
Reported by: Rob Gagnon
Patches:
issue-18319-trunk-r333066.diff (License #6159) patch uploaded by Rob Gagnon
Review: https://reviewboard.asterisk.org/r/1377/
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r334013 | rmudgett | 2011-08-31 11:00:49 -0500 (Wed, 31 Aug 2011) | 30 lines
Merged revisions 334012 via svnmerge from
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r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011) | 23 lines
No DAHDI channel available for conference, user introduction disabled.
The following error will consistently occur when trying to dial into a
MeetMe conference when the server does not have DAHDI hardware installed:
app_meetme.c: No DAHDI channel available for conference, user introduction
disabled (is chan_dahdi loaded?)
While chan_dahdi is loaded correctly during compilation and install of
Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf
configuration file in /etc/asterisk is not created by FreePBX if hardware
does not exist, causing MeetMe to be unable to open a DAHDI pseudo
channel.
* Allow chan_dahdi to create a pseudo channel when there is no
chan_dahdi.conf file to load.
(closes issue ASTERISK-17398)
Reported by: Preston Edwards
Patches:
jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
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r334010 | rmudgett | 2011-08-31 10:23:11 -0500 (Wed, 31 Aug 2011) | 50 lines
Merged revisions 334009 via svnmerge from
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r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011) | 43 lines
Call pickup race leaves orphaned channels or crashes.
Multiple users attempting to pickup a call that has been forked to
multiple extensions either crashes or fails a masquerade with a "bad
things may happen" message.
This is the scenario that is causing all the grief:
1) Pickup target is selected
2) target is marked as being picked up in ast_do_pickup()
3) target is unlocked by ast_do_pickup()
4) app dial or queue gets a chance to hang up losing calls and calls
ast_hangup() on target
5) SINCE A MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with
ast_channel_masquerade(), ast_hangup() completes successfully and the
channel is no longer in the channels container.
6) ast_do_pickup() then calls ast_channel_masquerade() to schedule the
masquerade on the dead channel.
7) ast_do_pickup() then calls ast_do_masquerade() on the dead channel
8) bad things happen while doing the masquerade and in the process
ast_do_masquerade() puts the dead channel back into the channels container
9) The "orphaned" channel is visible in the channels list if a crash does
not happen.
This patch does the following:
* Made ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up channel
and not release the channel lock until that has happened.
* Made __ast_channel_masquerade() not setup a masquerade if either channel
has AST_FLAG_ZOMBIE set.
* Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer work.
(closes issue ASTERISK-18222)
Reported by: Alec Davis
Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer
(closes issue ASTERISK-18273)
Reported by: Karsten Wemheuer
Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer
Review: https://reviewboard.asterisk.org/r/1400/
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r334007 | kmoore | 2011-08-31 10:19:30 -0500 (Wed, 31 Aug 2011) | 14 lines
Merged revisions 334006 via svnmerge from
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r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) | 7 lines
Correct an AMI protocol violation with SIPshowpeer
The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are
ended by using \r\n this confuses any interfacing script.
(closes issue ASTERISK-17486)
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r333961 | may | 2011-08-31 01:21:53 +0400 (Wed, 31 Aug 2011) | 11 lines
Merged revisions 333947 via svnmerge from
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r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5 lines
cleanups in ACF/ARJ GK replies processing
fixed long (24 sec) pause if acf/arj proccessed
before ast_cond_wait called to wait this
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r333962 | may | 2011-08-31 01:53:42 +0400 (Wed, 31 Aug 2011) | 3 lines
security fix. really drop call if signalling addr is not same as socket
addr
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