Richard Mudgett [Fri, 14 Dec 2012 21:35:44 +0000 (21:35 +0000)]
app_queue: Revert bad ringinuse=no patch.
With the option ringinuse=no set, the patch committed for ASTERISK-16115
causes non-SIP queue members to never be called because the device state
is checked after a channel is created to determine if the member is busy.
These queue members always get the "Member %s is busy, cannot dial"
message.
Most channel drivers other than chan_sip use the default device state
handling. The default device-state state is considered in use or unknown
if the channel exists or not respectively.
Damien Wedhorn [Fri, 14 Dec 2012 01:55:43 +0000 (01:55 +0000)]
Fix skinny to recognise vmexten in general section of conf
Fixup the vmexten so if globally set in general section will be honored by
chan_skinny. Also get rid of the 'global_' part of variable name to match
regexten.
Richard Mudgett [Thu, 13 Dec 2012 21:28:15 +0000 (21:28 +0000)]
confbridge: Fix MOH on simultaneous user entry to a new conference.
When two users entered a new conference simultaneously, one of the callers
hears MOH. This happened if two unmarked users entered simultaneously and
also if a waitmarked and a marked user entered simultaneously.
* Created a confbridge internal MOH API to eliminate the inlined MOH
handling code. Note that the conference mixing bridge needs to be locked
when actually starting/stopping MOH because there is a small window
between the conference join unsuspend MOH and actually joining the mixing
bridge.
* Created the concept of suspended MOH so it can be interrupted while
conference join announcements to the user and DTMF features can operate.
* Suspend any MOH until the user is about to actually join the mixing
bridge of the conference. This way any pre-join file playback does not
need to worry about MOH.
* Made post-join actions only play deferred entry announcement files.
Changing the user/conference state during that time is not protected or
controlled by the state machine.
Damien Wedhorn [Thu, 13 Dec 2012 21:25:31 +0000 (21:25 +0000)]
Minor fixes for chan_skinny
Whitespace, change SUBSTATE_ONHOOK to correct SKINNY_ONHOOK and
correct len of 2 strcmp in skinny_setdebug(). (see opticron's review
on https://reviewboard.asterisk.org/r/2240/)
........
Merged revisions 377991 from http://svn.asterisk.org/svn/asterisk/branches/11
Sean Bright [Thu, 13 Dec 2012 21:20:32 +0000 (21:20 +0000)]
Make generate_exchange_uuid() always return the passed ast_str pointer.
I changed this code earlier to return NULL if it wasn't able to generate a UUID,
whereas the earlier code would always return the ast_str that was passed in.
Switch back to returning the ast_str, only set it to the empty string instead if
UUID generation fails. We still do a validity check later which will catch this
and blow up if necessary.
Sean Bright [Thu, 13 Dec 2012 15:37:55 +0000 (15:37 +0000)]
Use the UUID API to generate and validate UUIDs for res_calendar_exchange.
Currently the res_calendar_exchange module uses its own method of generating
UUIDs using ast_random(). Now that we have a UUID API we should use that
instead.
Brent Eagles [Thu, 13 Dec 2012 15:22:27 +0000 (15:22 +0000)]
This change adds a SIP peer configuration feature to allow the peer's
configured codecs to take precedence on an outgoing call.
This change introduces a new peer configuration property named
'ignore_requested_pref' that causes the requested codec to be ignored when
determining the preferred codec for an outgoing call leg. The consequence is
that Asterisk's usual efforts to prefer avoiding transcoding can be overridden
on a peer-by-peer basis where appropriate.
Kinsey Moore [Thu, 13 Dec 2012 14:28:57 +0000 (14:28 +0000)]
Ensure Min-SE is included in outbound INVITEs
Asterisk now includes Min-SE in outbound INVITEs when the value is not
90 (the default) and session timers are not disabled. This has the
effect of Asterisk following RFC4028 more closely with regard to 422
responses and preventing situations in which Asterisk would be forced
to temporarily accept a call to tear it down based on a Session-Expires
below the locally configured Min-SE.
(issue SWP-5051)
Review: https://reviewboard.asterisk.org/r/2222/ Reported-by: Kinsey Moore Patch-by: Kinsey Moore
........
Merged revisions 377946 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 377947 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 377948 from http://svn.asterisk.org/svn/asterisk/branches/11
Michael L. Young [Wed, 12 Dec 2012 04:43:18 +0000 (04:43 +0000)]
Convert Dynamic Features Buffer To Use ast_str
Currently, the buffer for the dynamic features list is set to a fixed size of
128. If the list is bigger than that, it results in the dynamic feature(s) not
being recognized.
This patch changes the buffer from a fixed size to a dynamic one.
(closes issue ASTERISK-20680)
Reported by: Clod Patry
Tested by: Michael L. Young
Patches:
asterisk-20680-dynamic-features-v2.diff
uploaded by Michael L. Young (license 5026)
Mark Michelson [Wed, 12 Dec 2012 00:02:31 +0000 (00:02 +0000)]
Fix a potential deadlock in chan_sip during transfers.
The issue comes from the fact that transfers may perform
a redirecting update on a channel. The issue is that lock
inversion between the channel and its tech_pvt occurs since
the channel lock is released during the transfer process.
The fix is to move when the redirecting update occurs to a
place where neither the tech_pvt or the channel is locked so
that the two can be locked in the proper order.
(closes issue ASTERISK-20708)
reported by Mark Michelson
patches:
ASTERISK-20708-3.patch uploaded by Mark Michelson (License #5049)
Tested by:
Tim Ringenbach at Asteria Solutions Group
........
Merged revisions 377910 from http://svn.asterisk.org/svn/asterisk/branches/11
Mark Michelson [Tue, 11 Dec 2012 20:53:34 +0000 (20:53 +0000)]
Fix crash that can occur if CLI registration fails for an aliased command.
A recent memory leak fix in main/cli.c causes an ast_cli_entry's command
field to be freed and NULLed if ast_cli_register() fails. res_clialiases
was ignoring the return value of ast_cli_register() and was then passing
the NULL command off to a a hash function. This resulted in a crash.
The fix is not to ignore the erroneous return value. If ast_cli_register()
fails, then we do not continue trying to process the current alias.
........
Merged revisions 377840 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 377842 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 377843 from http://svn.asterisk.org/svn/asterisk/branches/11
Richard Mudgett [Tue, 11 Dec 2012 02:13:37 +0000 (02:13 +0000)]
Cleanup indications on exit.
* Made ast_unregister_indication_country() unlink the found tone zone
before selecting a new default_tone_zone to make it impossible to select
the tone zone being unregistered again.
* Ringcadence is no longer parsed twice in store_config_tone_zone().
* Cleanup CLI commands and destroy default_tone_zone on exit.
Kinsey Moore [Mon, 10 Dec 2012 16:56:37 +0000 (16:56 +0000)]
Ensure ReceiveFax provides a CED tone via T.38
When using res_fax_digium, the T.38 CED tone was not being provided
properly which would cause some incoming faxes to fail. This was not an
issue with res_fax_spandsp since it does not strictly honor the
send_ced flag and sends the CED tone whenever receiving a T.38 fax.
(closes issue FAX-343) Reported-by: Benjamin Tietz Patch-by: Kinsey Moore
........
Merged revisions 377655 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 377656 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 377657 from http://svn.asterisk.org/svn/asterisk/branches/11
Kinsey Moore [Mon, 10 Dec 2012 14:45:52 +0000 (14:45 +0000)]
Handle Session-Expires less than local Min-SE in 200 OK
Ensure that a call is immediately torn down if a Session-Expires value
received in a 200 OK is less than the local Min-SE. This also prevents
Asterisk from allowing calls with Session-Expires below the
RFC4028-mandated minimum (90s).
(closes issue ASTERISK-20653)
Review: https://reviewboard.asterisk.org/r/2237/ Patch-by: Kinsey Moore
........
Merged revisions 377623 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 377624 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 377625 from http://svn.asterisk.org/svn/asterisk/branches/11
Fix code to send in both rx and tx open stream messages correct codecs. Found that on phase 0/1 phones wrong codecs cause to no audio in some situations.
(issue ASTERISK-20183)
........
Merged revisions 377591 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 377592 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 377593 from http://svn.asterisk.org/svn/asterisk/branches/11
Tilghman Lesher [Mon, 10 Dec 2012 01:41:50 +0000 (01:41 +0000)]
Improve documentation by making all of the colors used readable,
no matter what the background color is.
Dark blue on a black background is unreadable, as is yellow on a
light background. This patch turns on the bright attribute for
colors when on a dark background and turns *off* the bright
attribute when the -W command line option is used (indicating a
_light_ background). This ensures that text is readable in both
cases.
Richard Mudgett [Sat, 8 Dec 2012 00:30:40 +0000 (00:30 +0000)]
Fix order of SIP allow/disallow in MySQL contrib script.
Using the contrib sippeers.sql script to create the sippeers MySQL table
would result in being unable to place calls if you set the disallow value
to all.
(closes issue ASTERISK-20756)
Reported by: Andre Luis
Patches:
sippeers.patch patch uploaded by Andre Luis
........
Merged revisions 377431 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 377432 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 377433 from http://svn.asterisk.org/svn/asterisk/branches/11
Kinsey Moore [Fri, 7 Dec 2012 22:08:48 +0000 (22:08 +0000)]
codec_dahdi: Fix output of "transcoder show" CLI command.
In r306010 "Asterisk media architecture conversion - no more format
bitfields", the logic for incrementing encoders and decoders when
opening transcoder channels was changed without making the corresponding
change when decrementing encoder / decoder channels. The result being
that when a channel was destroyed, codec_dahdi couldn't properly tell if
it was an encoder or decoder, and the default case is to assume it was a
decoder.
This could result in negative numbers for decoders in use like in:
VOIP6*CLI> transcoder show
2/-2 encoders/decoders of 92 channels are in use.
Russell Bryant [Thu, 6 Dec 2012 15:13:37 +0000 (15:13 +0000)]
Minor code cleanup in named_acl.c.
This patch makes a few little cleanups to named_acl.c. A couple non-public
functions were made static and an opening brace for a function was moved to
its own line, per the coding guidelines.
Matthew Jordan [Thu, 6 Dec 2012 14:26:13 +0000 (14:26 +0000)]
Fix memory leak in 'manager show event' when command entered incorrectly
When the CLI command 'manager show event' was run incorrectly and its usage
instructions returned, a reference to the event container was leaked. This
would prevent the container from being reclaimed when Asterisk exits. We now
properly decrement the count on the ao2 object using the nifty RAII_VAR macro.
Thanks to Russell for helping me stumble on this, and Terry for writing that
ridiculously helpful macro.
........
Merged revisions 377319 from http://svn.asterisk.org/svn/asterisk/branches/11
Jonathan Rose [Wed, 5 Dec 2012 17:17:06 +0000 (17:17 +0000)]
res_srtp: Fix a crash caused by srtp_dealloc on an already dealloced session
When srtp_create fails, the session may be dealloced or just not alloced. At
the same time though, the session pointer might not be set to NULL in this
process and attempting to srtp_dealloc it again will cause a segfault. This
patch checks for failure of srtp_create and sets the session pointer to NULL
if it fails.
Joshua Colp [Wed, 5 Dec 2012 16:51:58 +0000 (16:51 +0000)]
Fix a SIP request memory leak with TLS connections.
During the TLS re-work in chan_sip some TLS specific code was moved
into a separate function. This function operates on a copy of the
incoming SIP request. This copy was never deinitialized causing a
memory leak for each request processed.
This function is now given a SIP request structure which it can use
to copy the incoming request into. This reduces the amount of memory
allocations done since the internal allocated components are reused
between packets and also ensures the SIP request structure is
deinitialized when the TLS connection is torn down.
(closes issue ASTERISK-20763)
Reported by: deti
........
Merged revisions 377257 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 377258 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 377259 from http://svn.asterisk.org/svn/asterisk/branches/11
Richard Mudgett [Wed, 5 Dec 2012 01:11:26 +0000 (01:11 +0000)]
confbridge: Fix several small issues.
* Made func_confbridge_helper() allow an empty value when setting options.
You previously could not Set(CONFBRIDGE(user,pin)=) and clear the
configured pin from the dialplan.
* Made func_confbridge_helper() handle its datastore better if multiple
threads attempt to set the first CONFBRIDGE option value on the channel.
* Made the func_confbridge_helper() only output one diagnostic message
concerning the option.
* Made the bridge video_mode able to repeatedly change in the config file
and CONFBRIDGE dialplan function. The video_mode option values are an
enum and not independent of each other.
* Made handle_cli_confbridge_show_bridge_profile() better handle the
video_mode option.
* Simplified datastore handling code in conf_find_user_profile() and
conf_find_bridge_profile().
Richard Mudgett [Mon, 3 Dec 2012 23:00:08 +0000 (23:00 +0000)]
Cleanup ast_run_atexits() atexits list.
* Convert atexits list to a mutex instead of a rd/wr lock. The lock is
only write locked.
* Move CLI verbose Asterisk ending message to where AMI message is output
in really_quit() to avoid further surprises about using stuff already
shutdown.
Joshua Colp [Mon, 3 Dec 2012 14:56:36 +0000 (14:56 +0000)]
Fix an RTP instance reference count leak in chan_motif.
When setting up an RTP instance the RTCP portion of the instance
keeps a reference to the instance itself. In order to release this
reference and stop RTCP the stop API call must be called before
destroying the instance.
Olle Johansson [Mon, 3 Dec 2012 14:46:02 +0000 (14:46 +0000)]
Move functions to AFTER the block of forward declarations of functions.
It was a mess. The first part of chan_sip.c is constants, declarations, structures and stuff,
then forward declarations and then actual code. It's still a mess, but a bit less messy ;-)
Joshua Colp [Sat, 1 Dec 2012 00:47:42 +0000 (00:47 +0000)]
Tweak extension used for incoming calls received on Motif.
Based on feedback from numerous individuals this patch tweaks incoming calls
to first look for an extension with the name of the endpoint. If no such extension
exists the call will silently fall back to the "s" extension as it previously
did.
........
Merged revisions 376983 from http://svn.asterisk.org/svn/asterisk/branches/11
Mark Michelson [Fri, 30 Nov 2012 16:56:53 +0000 (16:56 +0000)]
Fix potential crashes during SIP attended transfers.
The principal behind this patch is simple. During a transfer,
we manipulate channels that are owned by a separate thread than
the one we currently are running in, so it makes sense that we
need to grab a reference to the channels so that they cannot
disappear out from under us.
In the wild, crashes were sometimes seen when the transferring
party would hang up the call before the transfer target answered
the call. The most common place to see the crash occur was when
attempting to send a connected line update to the transferer
channel.
(closes issue ASTERISK-20226)
Reported by Jared Smith
Patches:
ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
Tested by: Jared Smith
........
Merged revisions 376901 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376916 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376917 from http://svn.asterisk.org/svn/asterisk/branches/11
Michael L. Young [Thu, 29 Nov 2012 21:58:41 +0000 (21:58 +0000)]
Improve Code Readability And Fix Setting natdetected Flag
For 1.8, 10, 11 and trunk we are are improving the code readability.
For 11 and trunk, auto nat detection was added. The natdetected flag was being
set to 1 when the host address in the VIA header did not specifiy a port. This
patch fixes this by setting the port on the temporary sock address used to
SIP_STANDARD_PORT in order for the sock address comparison to work properly.
(closes issue ASTERISK-20724)
Reported by: Michael L. Young
Patches:
asterisk-20724-set-port-v2.diff uploaded by Michael L. Young (license 5026)
Pedro Kiefer [Thu, 29 Nov 2012 16:44:42 +0000 (16:44 +0000)]
Fix chan_sip websocket payload handling
Websocket by default doesn't return an ast_str for the payload received. When
converting it to an ast_str on chan_sip the last character was being omitted,
because ast_str functions expects that the given length includes the trailing
0x00. payload_len only has the actual string length without counting the
trailing zero.
For most cases this passed unnoticed as most of SIP messages ends with \r\n.
* Adds the following CLI commands to control MALLOC_DEBUG reporting of
unreleased malloc memory when Asterisk is shut down.
memory atexit list on
memory atexit list off
memory atexit summary byline
memory atexit summary byfunc
memory atexit summary byfile
memory atexit summary off
* Made check all remaining allocated region blocks atexit for fence
violations.
* Increased the allocated region hash table size by about three times. It
still isn't large enough considering the number of malloced blocks
Asterisk uses.
* Made CLI "memory show allocations anomalies" use
regions_check_all_fences().
Jonathan Rose [Wed, 28 Nov 2012 16:47:44 +0000 (16:47 +0000)]
manager: Make challenge work with allowmultiplelogin=no
Prior to this patch, challenge would yield a multiple logins error if used
without providing the username (which isn't really supposed to be an argument
to challenge) if allowmultiplelogin was set to no because allowmultiplelogin
finds a user with a zero length login name. This check is simply disabled for
the challenge action when the username is empty by this patch.
(closes issue ASTERISK-20677)
Reported by: Vladimir
Patches:
challenge_action_nomultiplelogin.diff uploaded by Jonathan Rose (license 6182)
........
Merged revisions 376725 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376726 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376727 from http://svn.asterisk.org/svn/asterisk/branches/11
Richard Mudgett [Wed, 28 Nov 2012 00:13:10 +0000 (00:13 +0000)]
Fix extension matching with the '-' char.
The '-' char is supposed to be ignored by the dialplan extension matching.
Unfortunately, it's treatment is not handled consistently throughout the
extension matching code.
* Made the old exten matching code consistently ignore '-' chars.
* Made the old exten matching code consistently handle case in the
matching.
* Made ignore empty character sets.
* Fixed ast_extension_cmp() to return -1, 0, or 1 as documented. The only
user of it in pbx_lua.c was testing for -1. It was originally returning
the strcmp() value for less than which is not usually going to be -1.
* Fix character set sorting if the sets have the same number of characters
and start with the same character. Character set [0-9] now sorts before
[02-9a] as originally intended.
* Updated some extension label and priority already in use warnings to
also indicate if the extension is aliased.
Richard Mudgett [Tue, 27 Nov 2012 20:39:51 +0000 (20:39 +0000)]
Remove unnecessary channel module references.
* Removed call to ast_module_user_hangup_all() in res_config_mysql.c since
it is effectively a noop. No channels can attach a reference to that
module.
* Removed call to ast_module_user_hangup_all() in app_celgenuserevent.c.
The caller of unload_module() has already called it.
* Removed redundant channel module references in pbx_dundi.c. The
registered dialplan function callback dispatchers for the read/read2/write
callbacks already reference the module before calling.
* pbx_dundi: Moved unregistering CLI commands, DUNDi switch, and dialplan
functions to the first thing the unload_module() does. This will reduce
the chance of new channels using DUNDi services while the module is being
torn down.
........
Merged revisions 376657 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376658 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376659 from http://svn.asterisk.org/svn/asterisk/branches/11
Matthew Jordan [Fri, 23 Nov 2012 00:02:23 +0000 (00:02 +0000)]
Re-initialize logmsgs mutex upon logger initialization to prevent lock errors
Similar to the patch that moved the fork earlier in the startup sequence to
prevent mutex errors in the recursive mutex surrounding the read/write thread
registration lock, this patch re-initializes the logmsgs mutex. Part of the
start up sequence before forking the process into the background includes
reading asterisk.conf; this has to occur prior to the call to daemon in order
to read startup parameters. When reading in a conf file, log statements can
be generated. Since this can't be avoided, the mutex instead is
re-initialized to ensure a reset of any thread tracking information.
This patch also includes some additional debugging to catch errors when
locking or unlocking the recursive mutex that surrounds locks when the
DEBUG_THREADS build option is enabled. DO_CRASH or THREAD_CRASH will
cause an abort() if a mutex error is detected.
Alec L Davis [Tue, 20 Nov 2012 17:39:11 +0000 (17:39 +0000)]
Reduce CLI spam of "Extension Changed" device state messages.
Asterisk 11 follows RFC3265 that states that after every subscribe or resubscribe a notify should be sent.
Thus the console if filled continuously with the following after every subscribe;
== Extension Changed 8512[phones] new state IDLE for Notify User cisco1
In Asterisk 1.8 only changes would be sent. Thus only when a device state changed was anything emitted to the console.
fix:
Only print to console when device state isn't forced.
Matthew Jordan [Mon, 19 Nov 2012 02:14:54 +0000 (02:14 +0000)]
Fix uninitialized in this function error
With some versions of gcc, n_buckets will be flagged as being uninitialized
before use. While its technically impossible (since the switch statement,
even without a default, accounts for all possibilities), we'll initialize the
variable to 0 anyway.
Matthew Jordan [Sun, 18 Nov 2012 20:27:45 +0000 (20:27 +0000)]
Reorder startup sequence to prevent lockups when process is sent to background
Although it is very rare and timing dependent, the potential exists for the
call to 'daemon' to cause what appears to be a deadlock in Asterisk during
startup. This can occur when a recursive mutex is obtained prior to the
daemon call executing. Since daemon uses fork to send the process into the
background, any threading primitives are unsafe to re-use after the call.
Implementations of pthread recursive mutexes are highly likely to store the
thread identifier of the thread that previously obtained the mutex. If
the mutex was locked prior to the fork, a subsequent unlock operation will
potentially fail as the thread identifier is no longer valid. Since the
mutex is still locked, all subsequent attempts to grab the mutex by other
threads will block.
This behavior exhibited itself most often when DEBUG_THREADS was enabled, as
this compile time option surrounds the mutexes in Asterisk with another
recursive mutex that protects the storage of thread related information. This
made it much more likely that a recursive mutex would be obtained prior to
daemon and unlocked after the call.
This patch does the following:
a) It backports a patch from Asterisk 11 that prevents the spawning of the
localtime monitoring thread. This thread is now spawned after Asterisk has
fully booted.
b) It re-orders the startup sequence to call daemon earlier during Asterisk
startup. This limits the potential of threading primitives being accessed
by initialization calls before daemon is called.
c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called.
Developers should send error messages directly to stderr prior to daemon,
as calls to ast_log may access recursive mutexes that store thread related
information.
d) It reorganizes when thread local storage is created for storing lock
information during the creation of threads. Prior to this patch, the
read/write lock protecting the list of threads in ast_register_thread would
utilize the lock in the thread local storage prior to it being initialized;
this patch prevents that.
On a very related note, this patch will *greatly* improve the stability of the
Asterisk Test Suite.
Matthew Jordan [Sun, 18 Nov 2012 14:31:32 +0000 (14:31 +0000)]
Add a test event that reports changes in ConfBridge state
This patch adds a test event to ConfBridge that reports transitions between
states in ConfBridge. This is used by tests in the Asterisk Test Suite
that verify state changes based on the entering/leaving of conference
participants.
........
Merged revisions 376414 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376415 from http://svn.asterisk.org/svn/asterisk/branches/11
David M. Lee [Fri, 16 Nov 2012 00:08:00 +0000 (00:08 +0000)]
Migrate hashtest/hashtest2 to be unit tests.
Both hashtest and hashtest2 are manual testing apps that thrash hash
tables (hashtab and ao2 containers, respectively), by spinning up
several threads that randomly insert, delete, lookup and iterate over
the hash table. If the app doesn't crash, the hash table probably passes
the test. Those utils are not a part of the typical Asterisk build, so
they do not usually get compiled. This all makes them less that useful.
This patch removes those manual test programs and replaces them with
Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It also
attempts to make the tests more deterministic.
* Rather than spinning up some number of threads that operate on the
hash table randomly, spin up four threads that concurrenly add,
remove, lookup and iterate over the hash table.
* Each thread checks the state of the hash table both during and after
execution, and indicates a test failure if things are not as expected.
* Each thread times out after 60 seconds to prevent deadlocking the unit
test run.
(closes issue ASTERISK-20505)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2189/
........
Merged revisions 376306 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376315 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376339 from http://svn.asterisk.org/svn/asterisk/branches/11
Jonathan Rose [Thu, 15 Nov 2012 23:10:13 +0000 (23:10 +0000)]
app_meetme: Fix channels lingering when hung up under certain conditions
Channels would get stuck and MeetMe would repeatedly display an Unable
to write frame to channel error in the conf_run function if hung up
during certain sound prompts such as during user count announcements.
This patch fixes that by reintroducing a hangup check in the meetme's
main loop (also in conf_run).
(closes issue ASTERISK-20486)
Reported by: Michael Cargile
Review: https://reviewboard.asterisk.org/r/2187/
Patches:
meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan Rose (license 6182)
........
Merged revisions 376307 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376308 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376310 from http://svn.asterisk.org/svn/asterisk/branches/11
Brent Eagles [Thu, 15 Nov 2012 14:35:01 +0000 (14:35 +0000)]
Patch to prevent stopping the active generator when it is not the silence
generator.
This patch introduces an internal helper function to safely check whether the
current generator is the one that is expected before deactivating it. The
current externally accessible ast_channel_stop_generator() function has been
modified to be implemented in terms of the new function.
(closes issue ASTERISK-19918)
Reported by: Eduardo Abad
........
Merged revisions 376217 from http://svn.asterisk.org/svn/asterisk/branches/11
Rusty Newton [Thu, 15 Nov 2012 02:29:40 +0000 (02:29 +0000)]
Patch to play correct sound file when a voicemail's urgent status is removed
We were attempting to play "vm-urgent-removed", which didn't exist. Now we play "vm-marked-nonurgent" which exists
and is the correct sound file. Previous behavior was silence and a warning on the CLI.
(issue ASTERISK-20280)
(closes issue ASTERISK-20280)
Reported by: Tomo Takebe
Tested by: Rusty Newton
Patches:
asterisk20280.patch uploaded by Rusty Newton (license 5829)
........
Merged revisions 376262 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376263 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376264 from http://svn.asterisk.org/svn/asterisk/branches/11
Richard Mudgett [Wed, 14 Nov 2012 19:55:39 +0000 (19:55 +0000)]
Fix call files when astspooldir is relative.
Future dated call files are ignored when astspooldir is relative to the
current directory. The queue_file() assumed that the qdir needed to be
prepended if the given filename did not start with a '/'. If astspooldir
is relative it is not going to start from the root directory obviously so
it will not start with a '/'. The filename used in queue_file()
ultimately results in qdir prepended multiple times.
* Made queue_file() not prepend qdir if the filename contains a '/'.
(closes issue ASTERISK-20593)
Reported by: James Le Cuirot
Patches:
0004-Fix-future-call-files-from-relative-directories.patch (license #6439) patch uploaded by James Le Cuirot
........
Merged revisions 376232 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 376233 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 376234 from http://svn.asterisk.org/svn/asterisk/branches/11
Jonathan Rose [Tue, 13 Nov 2012 19:42:13 +0000 (19:42 +0000)]
chan_sip: Add SubscribeContext field to SIPshowpeer AMI response
The new field is will show up within the response if the requested peer has a
subscribe context set.
(closes issue ASTERISK-20626)
Reported by: Jaco Kroon
Patches:
asterisk-sip-ami-SubscrContext.patch uploaded by jkroon (license 5671)
-with modifications by jrose to conform to style guidelines
Review: https://reviewboard.asterisk.org/r/2195/
Joshua Colp [Mon, 12 Nov 2012 20:46:51 +0000 (20:46 +0000)]
Properly check if the "Context" and "Extension" headers are empty in a ShowDialPlan action.
The code which handles the ShowDialPlan action wrongly assumed that a non-NULL return value
from the function which retrieves headers from an action indicates that the header has a
value. This is incorrect and the contents must be checked to see if they are blank.
Michael L. Young [Mon, 12 Nov 2012 20:18:47 +0000 (20:18 +0000)]
Fix Dynamic Hints Variable Substition - Underscore Problem
When adding a dynamic hint, if an extension contains an underscore no variable
subsitution is being performed.
This patch changes from checking if the extension contains an underscore to
checking if the extension begins with an underscore.
(closes issue ASTERISK-20639)
Reported by: Steven T. Wheeler
Tested by: Steven T. Wheeler, Michael L. Young
Patches:
asterisk-20639-dynamic-hint-underscore.diff
uploaded by Michael L. Young (license 5026)
projects / thirdparty / asterisk.git / log
