Ensure that astobj2 memory allocations are properly accounted for when MALLOC_DEBUG is used
This commit ensures that all astobj2 allocated objects are properly accounted for in MALLOC_DEBUG mode by passing down the file/function/line information from the module/function that actually called the astobj2 allocation function.
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Ensure that by default only one console channel driver is loaded
This configuration file was changed to ensure that only one console channel driver
(chan_oss) is loaded by default, but the change would only work if chan_console
was not built. Now it will work as expected; if chan_alsa or chan_console are built
and installed, they will not be loaded unless explicity requested.
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Fix an error in queue_log file rotation optimization code
This code was copy-and-pasted without properly changing references to event_rotate into queue_rotate, so under some conditions the log rotation would rotate queue_log even though it was not necessary.
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Move static buffers to outside for loops in app_chanspy.
Similar to seanbright's commit 191422, this moves some static buffers
to be defined outside of for loops since it is undefined if memory
will be re-used or if the stack will grow with each iteration of the
loop.
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r191629 | mmichelson | 2009-05-02 05:45:24 -0500 (Sat, 02 May 2009) | 3 lines
Kevin has informed me that thi sort of thing is not necessary.
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Fix DTMF not being sent to other side after a partial feature match
This fixes a regression from commit 176701. The issue was that
ast_generic_bridge never exited after the feature digit timeout had elapsed,
which prevented the queued DTMF from being sent to the other side.
This issue was reported to me directly.
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Sean Bright [Fri, 1 May 2009 16:26:32 +0000 (16:26 +0000)]
Fix a crash in app_queue with very long member lists.
A user reported via #asterisk that with very long lists of members, a crash
occurs in ast_strdupa, so just use a single buffer and ast_copy_string instead
of stack allocating copys of each interface name.
Change working directory to / under certain conditions.
If backgrounding and no core will be produced, then changing the directory
won't break anything; likewise, if the CWD isn't accessible by the current
user, then a core wasn't possible anyway.
(closes issue #14831)
Reported by: chris-mac
Patches:
20090428__bug14831.diff.txt uploaded by tilghman (license 14)
20090430__bug14831.diff.txt uploaded by tilghman (license 14)
Tested by: chris-mac
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Build XML documention from *only* the source files that have docs in them
Change the build process so that doc/core-en_US.xml is dependent solely on the source files that have documentation in them, not on all source files.
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We never, ever want these files to processed automatically, because we store the output files in Subversion and users should never need to rebuild them.
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Fix 'inconsistent line endings' when autoconf 2.63 is used
Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings
This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway.
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Resolve a crash in res_smdi when used with chan_dahdi.
When chan_dahdi goes to get an SMDI message, it provides no search criteria.
It just grabs the next message that arrives. This code was written with the
SMDI dialplan functions in mind, since that is now the preferred method of
using SMDI. However, this broke support of it being used from chan_dahdi.
Allow for a position to be specified when entering a queue.
This would allow for one to add a caller to a specific place in the
queue instead of just placing the caller in the back every time. To help
facilitate some interesting manipulations, a new channel variable called
QUEUEPOSITION has been added. When a caller is removed from a queue, his
position in that queue is stored in the QUEUEPOSITION variable. One such
strategy an administrator can employ is to allow for the removal of a caller
from one queue followed by the insertion of the same caller into a separate
queue in the same position.
Labels are sometimes (most of the time?) NULL for extensions.
(closes issue #14895)
Reported by: chris-mac
Patches:
20090423__bug14895__2.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
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If both sides of a Local channel were hung up at around the same time it was
possible for one thread to destroy the local private structure and have the other thread
immediately try to remove the already freed structure from the local channel list.
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There seems to be a bug with old versions of g++ that doesn't allow a structure
member to use the name list. Rename list member to group_list in ast_group_info
and change the few places it is used.
Previously, packetization settings were ignored and now they are not. A new
config option 'autoframing' has been added to mirror the way chan_sip handles
it. Turning on the autoframing option (available both as a global option or per
peer) overrides the local settings with the remote packetization settings.
Testing was performed with varying packetization levels with the following
codecs: ulaw, alaw, gsm, and g729.
(closes issue #12415)
Reported by: pj
Patches:
2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7),
modified by me
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Do not continue to receive DTMF, when the channel is hungup and about to be destroyed.
(closes issue #14858)
Reported by: barryf
Patches:
20090421__bug14858.diff.txt uploaded by tilghman (license 14)
Tested by: barryf
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Fixes segfault when switching UDP to TCP in sip.conf after reload.
If transport in sip.conf is switched from UDP to TCP, Asterisk segfaults right after issuing a sip reload. The problem is the socket type is changed to TCP but the fd may still be present for UDP. Later, when the TCP session should be created or set using an existing one, it isn't because the old file descriptor is still present. Now every time transport is changed during a sip.conf reload, the file descriptor is set to -1, signifying it must be created or found.
Add check in configure script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h
This allows config.c to compile when linked against uclibc that does not support these parameters
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AEL was not handling the case of a device hint containing an @ symbol, which
caused parking hints (e.g. hint(park:exten@context)) to error out the parser.
This patch makes AEL treat the @ the same way it treats colon and ampersand
now, meaning the characters are included in verbatim.
Clean up problem with manager implementation of mmap where it was not testing against MAP_FAILED response.
Got rid of shadowed variable used in processign the mmap results.
Change test of mmap results to compare against MAP_FAILED
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Fix a bug with non-UDP connections that caused dialogs to not get freed.
This issue crept up because of a reference count issue on non-UDP based dialogs.
The dialog reference count was increased when transmitting a packet reliably but never
decreased. This caused the dialog structure to hang around despite being unlinked from
the dialogs container.
Fixed autologoff in agents.conf not working when agent logs in via AgentLogin app
An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening.
Prevent a crash when SIP blonde transferring an unbridged call.
If one attempts to use the attended transfer button on a SIP phone
to transfer an unbridged call (such as a call to an IVR) but hangs
up while the target of the transfer is still ringing, we need to not
crash.
The problem was that ast_hangup was called from outside the channel
thread.
Fix a bug where a value used to create the channel name was bogus.
This commit fixes the scenario where an incoming call is authenticated
using a peer entry. Previously the channel name was created using either
the username setting from the sip.conf entry or the IP address that the
call came from. Now the channel name will be created using the peer name
itself. This commit will not change the way the channel name is generated
for users or friends.
(closes issue #14256)
Reported by: Nick_Lewis
Patches:
chan_sip.c-chname.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis, file
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1. Differentiate between literal characters in an extension
and characters that should be treated as a pattern match. Prior to
these fixes, an extension such as NNN would be treated as a pattern,
rather than a literal string of N's.
2. Fixed the logic used when matching an extension with a bracketed
expression, such as 2[5-7]6.
3. Removed all areas of code that were executed when NOT_NOW was
#defined. The code in these areas had the potential to crash, for
one thing, and the actual intent of these blocks seemed counterproductive.
4. Fixed many many coding guidelines problems I encountered while looking
through the corresponding code.
5. Added failure cases and warning messages for when duplicate extensions
are encountered.
6. Miscellaneous fixes to incorrect or redundant statements.
National prefix inserted even when caller ID not available
When the caller ID is restricted, the expected behavior is for the caller id to be blank. In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank.
Make the cancellation of the dial timeout on a call forward optional.
This introduces the 'z' option to app_dial. With it set, a call forward
will cancel any timeout originally set for this instance of the Dial
application.
This is the companion commit to libpri r732. Service messages are now supported
for switch types 4ess/5ess. A new option service_message_support has been added
to chan_dahdi.conf and is noted in the sample config file. The service message
support is turned off by default. The current implementation relies on AstDB
to keep track of channel state, which allows the statuses to be preserved
across Asterisk restarts. Below is a description of the storage format.
The state and reason for the service state are in the form <state>:<reason>,
where:
<state> ::= { 'O' } // 'O' – Out Of Service
<reason> ::= { '0' | '1' | '2' | '3' }, where:
'0' – No reason (backwards compatibility)
'1' – NEAR END
'2' – FAR END
'3' – both NEAR and FAR END
The new CLI commands to handle channel service state are:
pri service disable channel <chan>
pri service enable channel <chan>
Many people contributed to the development of this functionality. Because I
entered at the very end I do not know the exact history. Special thanks to
all who moved the bug forward one way or another:
cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7,
tilghman, lmadsen, and especially dhubbard (he answered lots of my questions
and did a large portion of the work)
Fix a bug with the change I made yesterday to outbound proxy support.
Per discussion with oej on IRC we need the actual IP address, not the
outbound proxy IP address, in the sa field. Upon further inspection
this should make the behaviour of all other uses of the outbound proxy
in the code.
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Fix a bug where using an outbound proxy would cause the local address to be 127.0.0.1.
Copy the outbound proxy IP address into the SIP dialog structure as the IP address we will
be sending to. This has to be done because the logic that determines what local IP address to use
in the SIP messages is not aware of an outbound proxy being in place. It only knows what IP address
we are sending to.
This allows for you to change the From header for outgoing MWI
NOTIFY requests. Prior to this, the best you could do was to
set a callerid in the general section of sip.conf. The problem
was that this was used for all outbound requests, not just
MWI NOTIFY requests.
Handle a SIP race condition (reinvite before an ACK) properly.
RFC 5047 explains the proper course of action to take if a
reINVITE is received before the ACK from a previous invite
transaction. What we are to do is to treat the reINVITE as
if it were both an ACK and a reINVITE and process it normally.
Later, when we receive the ACK we had been expecting, we will
ignore it since its CSeq is less than the current iseqno of
the sip_pvt representing this dialog.
Add ability for dialplan execution to continue when caller hangs up.
The F option to app_dial has been modified to accept no parameters and perform
the above functionality. I don't see anywhere else that is doing function
overloading, but this really is the best place for this operation because:
- It makes it close to the 'g' option in the argument list which provides
similar functionality.
- The existing code to support the current F option provides a very
convienient location to add this new feature.
Race condition between ast_cli_command() and 'module unload' could cause a deadlock.
Add lock timeouts to avoid this potential deadlock.
(closes issue #14705)
Reported by: jamessan
Patches:
20090320__bug14705.diff.txt uploaded by tilghman (license 14)
Tested by: jamessan
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Fix a crash in res_musiconhold when using cached realtime moh.
The moh_register function links an mohclass and then immediately
unrefs the class since the container now has a reference. The problem
with using realtime music on hold is that the class is allocated,
registered, and started in one fell swoop. The refcounting logic
resulted in the count being off by one. The same problem did not
happen when using a static config because the allocation and registration
of an mohclass is a separate operation from starting moh. This also did
not affect non-cached realtime moh because the classes are not registered
at all.
I also have modified res_musiconhold to use the _t_ variants of the ao2_
functions so that more info can be gleaned when attempting to trace the
refcounts. I found this to be incredibly helpful for debugging this issue
and there's no good reason to remove it.
Permit zero-length text messages in SIP.
(Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal")
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add a dedicated log channel for modules to be able report security-related events, so that they can be fed into external processes for analysis and possible mitigation efforts
(inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members)
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Add timer for features so that backup bridge config can go away
The biggest change done here was elimination of the backup_config for use with
features. Previously, the bridging code upon detecting a feature would set the
start time of the bridge to the start time of the feature. Then after the
feature had either expired or timed out the start time would be reset to the
true bridge start time from the backup_config. Now, the time differences are
calculated with respect to the newly added feature_start_time timeval instead.
There should be no behavior changes from the previous functionality aside from
the bridge timing being unaffected by either valid or partial feature matches.
Previously the timing would be increased by the length of time configured for
featuredigittimeout, which was probably never noticed.
Fix a small logical error when loading moh classes.
We were unconditionally incrementing the number of mohclasses
registered. However, we should actually only increment if the
call to moh_register was successful.
While this probably has never caused problems, I noticed it
and decided to fix it anyway.
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Make a couple of changes with regards to a new message printed in ast_read().
"ast_read() called with no recorded file descriptor" is a new message added
after a bug was discovered. Unfortunately, it seems there are a bunch of places
that potentially make such calls to ast_read() and trigger this error message
to be displayed. This commit does two things to help to make this message appear
less.
First, the message has been downgraded to a debug level message if dev mode is
not enabled. The message means a lot more to developers than it does to end users,
and so developers should take an effort to be sure to call ast_read only when
a channel is ready to be read from. However, since this doesn't actually cause an
error in operation and is not something a user can easily fix, we should not spam
their console with these messages.
Second, the message has been moved to after the check for any pending masquerades.
ast_read() being called with no recorded file descriptor should not interfere with
a masquerade taking place.
This could be seen as a simple way of resolving issue #14723. However, I still want
to try to clear out the existing ways of triggering this message, since I feel that
would be a better resolution for the issue.
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projects / thirdparty / asterisk.git / log
