Let's say you have a template T with variable VAR1 = ON and you have a
context C(T) that doesn't specify VAR1. If you read C, the effective value
of VAR1 is ON. Now you change T VAR1 to OFF and call
ast_config_text_file_save. The current behavior is that the file gets
re-written with T/VAR1=OFF but C/VAR1=ON is added. Personally, I think this
is a bug. It's preserving the effective state of C even though I didn't
specify C/VAR1 in th first place. I believe the behavior should be that if
I didn't specify C/VAR1 originally, then the effective value of C/VAR1 should
continue to follow the inherited state. Now, if I DID explicitly specify
C/VAR1, the it should be preserved even if the template changes.
Even though I think the existing behavior is a bug, it's been that way forever
so I'm not changing it. Instead, I've created ast_config_text_file_save2()
that takes a bitmask of flags, one of which is to preserve the effective context
(the current behavior). The original ast_config_text_file_save calls *2 with
the preserve flag. If you want the new behavior, call *2 directly without a
flag.
I've also updated Manager UpdateConfig with a new parameter
'PreserveEffectiveContext' whose default is 'yes'. If you want the new behavior
with UpdateConfig, set 'PreserveEffectiveContext: no'.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4297/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After the initial DTMF atxfer call attempt to the transfer target fails to
answer during a blonde transfer, the recall callback channels do not get
setup with information from the initial transferrer channel. As a result,
the recall callback to the transferrer does not have callid, channel
variables, datastores, accountcode, peeraccount, COLP, and CLID setup. A
similar situation happens with the recall callback to the transfer target
but it is less visible. The recall callback to the transfer target does
not have callid, channel variables, datastores, accountcode, peeraccount,
and COLP setup.
* Added missing information to the recall callback channels before
initiating the call. callid, channel variables, datastores, accountcode,
peeraccount, COLP, and CLID
* Set callid of the transferrer channel on the DTMF atxfer controller
thread attended_transfer_monitor_thread().
* Added missing channel unlocks and props unref to off nominal paths in
attended_transfer_properties_alloc().
ASTERISK-23841 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/4259/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The QUEUESTART log entry has historically acted like a fully booted event
for the queue_log file. When the QUEUESTART entry was posted to the log
was broken by the change made by ASTERISK-15863.
* Made post the QUEUESTART queue_log entry when Asterisk fully boots.
This restores the intent of that log entry and happens after realtime has
had a chance to load.
AST-1444 #close
Reported by: Denis Martinez
Review: https://reviewboard.asterisk.org/r/4282/
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The named ACL code incorrectly destroyed the config options information if loading
of the configuration file failed at startup. This would result in reloading
also failing even if a valid configuration file was put in place.
ASTERISK-23733 #close
Reported by: Richard Kenner
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After a blond transfer (start attended and hang up)
to a destination that also hangs up without answer,
the Local;1 channel was leaked and would show up on
core show channels. This was happening because the
attended state blond_nonfinal_enter() resetting the
props->transfer_target to null while releasing it's
own reference, which would later prevent props from
releasing another reference during destruction. The
change made here is simply to not assign the target
to NULL.
ASTERISK-24513 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4262/
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The AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT flag was created to prevent bridges from
prematurely acting on orphaned channels in bridges. The problem with the AMI
redirect action was that it was setting this flag on channels based on the presence
of a PBX, not whether the channel was in a bridge. Whether a channel has a PBX
is irrelevant, so the condition has been altered to check if the channel is in a
bridge.
ASTERISK-24536 #close
Reported by Niklas Larsson
Review: https://reviewboard.asterisk.org/r/4268
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r413586 (1.8) various casts were added to silence gcc 4.10 warnings.
Those fixes included things like:
-out += sprintf(out, "%%%02X", (unsigned char) *ptr);
+out += sprintf(out, "%%%02X", (unsigned) *ptr);
That works for low ascii characters, but for the high range that yields
e.g. FFFFFFC3 when C3 is expected.
This changeset:
- fixes those casts to use the 'hh' unsigned char modifier instead
- consistently uses %02x instead of %2.2x (or other non-standard usage)
- adds a few 'h' modifiers in various places
- fixes a 'replcaes' typo
- dev/urandon typo (in 13+ patch)
Review: https://reviewboard.asterisk.org/r/4263/
ASTERISK-24619 #close
Reported by: Stefan27 (on IRC)
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No functionality change. Just move the definition of ast_module_reload
from _private.h to module.h so it can be public.
Also removed the include of _private.h from manager.c since ast_module_load
was the only reason for including it.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4251/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch started with David Lee's patch at
https://reviewboard.asterisk.org/r/2826/ and includes a regression fix
introduced by the ASTERISK-22455 patch.
The initialization of a mutex's lock tracking structure was not protected
in a critical section. This is fine for any mutex that is explicitly
initialized, but a static mutex may have its lock tracking double
initialized if multiple threads attempt the first lock simultaneously.
* Added a global mutex to properly serialize initialization of the lock
tracking structure. The painful global lock can be mitigated by adding a
double checked lock flag as discussed on the original review request.
* Defer lock tracking initialization until first use.
* Don't be "helpful" and initialize an uninitialized lock when
DEBUG_THREADS is enabled. Debug code is not supposed to fix or change
normal code behavior. We don't need a lock initialization race that would
force a re-setup of lock tracking. Lock tracking already handles
initialization on first use.
* Properly handle allocation failures of the lock tracking structure.
* No need to initialize tracking data in __ast_pthread_mutex_destroy()
just to turn around and destroy it.
The regression introduced by ASTERISK-22455 is the result of manipulating
a pthread_mutex_t struct outside of the pthread library code. The
pthread_mutex_t struct seems to have a global linked list pointer member
that can get changed by other threads. Therefore, saving and restoring
the contents of a pthread_mutex_t struct is a bad thing.
Thanks to Thomas Airmont for finding this obscure regression.
* Don't overwrite the struct ast_lock_track.reentr_mutex member to restore
tracking data in __ast_cond_wait() and __ast_cond_timedwait(). The
pthread_mutex_t struct must be treated as a read-only opaque variable.
Miscellaneous other items fixed by this patch:
* Match ast_suspend_lock_info() with ast_restore_lock_info() in
__ast_cond_timedwait().
* Made some uninitialized lock sanity checks return EINVAL and try a
DO_THREAD_CRASH.
* Fix bad canlog initialization expressions.
ASTERISK-24614 #close
Reported by: Thomas Airmont
Review: https://reviewboard.asterisk.org/r/4247/
Review: https://reviewboard.asterisk.org/r/2826/
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When shutting down Asterisk the codecs are cleaned up. As a result anything
attempting to get a codec based on ID or details will find that no codec
exists. This currently occurs when determining the sample count of a frame.
This code did not take this situation into account.
This change fixes this by getting the codec directly from the format and
eliminates the lookup. This is both faster and also provides a guarantee
that the codec will exist and will be valid.
ASTERISK-24604 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4260/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When connecting to the remote console, an id string
is first provided that consts of the hostname, pid,
and version. This is parsed by the remote instance
using a buffer that may be too short, and can allow
a buffer overrun because it is not terminated. This
patch adds termination and a larger buffer.
Review: https://reviewboard.asterisk.org/r/4182/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When endpoints with direct_media enabled, behind a firewall (Asterisk on a
separate network) and were bridged sometimes Asterisk would send the ip
address of the firewall in the sdp to one of the phones in the reinvite
resulting in one way audio. When sending the reinvite Asterisk will retrieve
the media address from the associated rtp instance, but if frames were being
read this can be overwritten with another address (in this case the
firewall's). This patch ensures that Asterisk uses the original device
address when using direct media.
ASTERISK-24563
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4216/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This corrects several bugs that currently exist in the stasis
application code.
* After a masquerade, the resulting channels have channel topics that
do not match their uniqueids
** Masquerades now swap channel topics appropriately
* StasisStart and StasisEnd messages are leaked to observer
applications due to being published on channel topics
** StasisStart and StasisEnd publishing is now properly restricted
to controlling apps via app topics
* Race conditions exist where StasisStart and StasisEnd messages due to
a masquerade may be received out of order due to being published on
different topics
** These messages are now published directly on the app topic so this
is now a non-issue
* StasisEnds are sometimes missing when sent due to masquerades and
bridge swaps into and out of Stasis()
** This was due to StasisEnd processing adjusting message-sent flags
after Stasis() had already exited and Stasis() had been re-entered
** This was corrected by adjusting these flags prior to sending the
message while the initial Stasis() application was still shutting
down
Review: https://reviewboard.asterisk.org/r/4213/
ASTERISK-24537 #close
Reported by: Matt DiMeo
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On a 32-bit system, a type of intmax_t will result in a compilation warning
when formatted as a 'long int'. Use the format specifier of %jd (which was
what was used originally in manager.c) to format the JSON extracted integer
on both 32-/64-bit systems.
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This patch fixes a race condition between the raising of test AMI events (which
drive many tests in the Asterisk Test Suite) and other AMI events. Prior to
this patch, the Stasis messages published to the test topic were not forwarded
to the AMI topic. Instead, the code in manager had a dedicated handler for test
messages that was independent of the topics forwarded to the AMI topic. This
results in no synchronization between the test messages and the rest of the
Stasis messages published out over AMI. In some test with very tight timing
constraints, this can result in out of order messages and spurious test
failures. Properly forwarding the Test Suite topic to the AMI topic ensures
that the messages are synchronized properly.
This patch does that, and moves the message handling to the Stasis definition
of the Test Suite message in test.c as well.
Review: https://reviewboard.asterisk.org/r/4221/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, all Stasis subscriptions would receive a dedicated
thread for servicing published messages. In contrast, prior to r400178
(see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
shared a thread pool. It was discovered during some initial work on Stasis
that, for a low subscription count with high message throughput, the
threadpool was not as performant as simply having a dedicated thread per
subscriber.
For situations where a subscriber receives a substantial number of messages
and is always present, the model of having a dedicated thread per subscriber
makes sense. While we still have plenty of subscriptions that would follow
this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
the following two categories:
* Large number of subscriptions, specifically those tied to endpoints/peers.
* Low number of messages. Some subscriptions exist specifically to coordinate
a single message - the subscription is created, a message is published, the
delivery is synchronized, and the subscription is destroyed.
In both of the latter two cases, creating a dedicated thread is wasteful (and
in the case of a large number of peers/endpoints, harmful). In those cases,
having shared delivery threads is far more performant.
This patch adds the ability of a subscriber to Stasis to choose whether or not
their messages are dispatched on a dedicated thread or on a threadpool. The
threadpool is configurable through stasis.conf.
Review: https://reviewboard.asterisk.org/r/4193
ASTERISK-24533 #close
Reported by: xrobau
Tested by: xrobau
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The static buffer for codecs when invoking the "core show channeltype" CLI command
did not have enough room for all codecs. This has been extended so it does.
ASTERISK-24542 #close
Reported by: snuffy
patches:
channeltype-tech.diff submitted by snuffy (license 5024)
Review: https://reviewboard.asterisk.org/r/4204/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When shutting down Asterisk that has an active AMI connection, you get
several "failed to extend from %d to %d" messages because use of the
EVENT_FLAG_SHUTDOWN attempts to add all AMI permission strings to the
event.
* Created MAX_AUTH_PERM_STRING to use when creating stack based struct
ast_str variables used with the authority_to_str() and
user_authority_to_str() functions instead of a variety of magic numbers
that could be too small.
* Added a special check for EVENT_FLAG_SHUTDOWN to authority_to_str() so
it will not attempt to add all permission level strings.
Review: https://reviewboard.asterisk.org/r/4200/
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As a result of https://reviewboard.asterisk.org/r/3305, res_sorcery_realtime
was tossing database fields that didn't have an exact match to a sorcery
registered field. This broke the ability to use regexes as field names which
manifested itself as a failure of res_pjsip_phoneprov_provider which uses
this capability. It also broke handling of fields that start with '@' in
realtime but I don't think anyone noticed.
This patch does the following...
* Modifies ast_sorcery_fields_register to pre-compile the name regex.
* Modifies ast_sorcery_is_object_field_registered to test the regex if it
exists instead of doing an exact strcmp.
* Modifies res_pjsip_phoneprov_provider with a few tweaks to get it to work
with realtime.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4185/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428544 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r428165, two bugs were introduced:
* Prior to entering the features retry loop, the buffer that holds the
collected digits is wiped. However, this inadvertently wipes out the
first collected digit on the first pass through, which is obtained
in ast_stream_and_wait. This caused all of the features tests to fail.
* If ast_app_dtget returns a hangup (-1), the loop would retry incorrectly.
If we detect a hangup, we have to stop trying the feature.
This patch fixes both issues.
Review: https://reviewboard.asterisk.org/r/4196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This allows for a configurable number of attempts for a transferer
to dial an extension to transfer the call to. For Asterisk 13, the
default values are such that upgrading between versions will not
cause a behaivour change. For trunk, though, the defaults will be
changed to be more user-friendly.
Review: https://reviewboard.asterisk.org/r/4167
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From reviewboard:
"During blind transfer testing, it was noticed that tests were failing
occasionally because the ARI blind transfer event was not being sent.
After investigating, I detected a race condition in the blind transfer
code. When blind transferring a single channel, the actual transfer
operation (i.e. removing the transferee from the bridge and directing
them to the proper dialplan location) is queued onto the transferee
bridge channel. After queuing the transfer operation, the blind transfer
Stasis message is published. At the time of publication, snapshots of
the channels and bridge involved are created. The ARI subscriber to the
blind transfer Stasis message then attempts to determine if the bridge
or any of the involved channels are subscribed to by ARI applications.
If so, then the blind transfer message is sent to the applications. The
way that the ARI blind transfer message handler works is to first see
if the transferer channel is subscribed to. If not, then iterate over
all the channel IDs in the bridge snapshot and determine if any of
those are subscribed to. In the test we were running, the lone
transferee channel was subscribed to, so an ARI event should have been
sent to our application. Occasionally, though, the bridge snapshot did
not have any channels IDs on it at all. Why?
The problem is that since the blind transfer operation is handled by a
separate thread, it is possible that the transfer will have completed and
the channels removed from the bridge before we publish the blind transfer
Stasis message. Since the blind transfer has completed, the bridge on
which the transfer occurred no longer has any channels on it, so the
resulting bridge snapshot has no channels on it. Through investigation of
the code, I found that attended transfers can have this issue too for the
case where a transferee is transferred to an application."
The fix employed here is to decouple the creation of snapshots for the transfer
messages from the publication of the transfer messages. This way, snapshots
can be created to reflect what they are at the time of the transfer operation.
Review: https://reviewboard.asterisk.org/r/4135
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk - in res_rtp_asterisk - only understands a single RTCP report info
block. When the RTCP information was refactored in the RTP Engine to be pushed
over the Stasis message bus, I put in the hooks into the engine to handle
multiple RTCP report info blocks, in the hope that a future RTP implementation
would be able to provide that data. Unfortunately, res_rtp_asterisk has a
tendency to "lie":
(1) It will send RTCP reports with a reception_report_count greater than 1
(which is pulled directly from the RTCP packet itself, so that part is
correct)
(2) It will only provide a single report block
When the rtp_engine goes to convert this to a JSON blob, hilarity ensues as it
looks for a report block that doesn't exist.
This patch updates the rtp_engine to be a bit more skeptical about what it is
presented with. While this could also be fixed in res_rtp_asterisk, this patch
prefers to fix it in the engine for two reasons:
(1) The engine is designed to work with multiple RTP implementation, and hence
having it be more robust is a good thing (tm)
(2) res_rtp_asterisk's handling of RTCP information is "fun". It should report
the correct reception_report_count; ideally it should also be giving us all
of the blocks - but it is *definitely* not designed to do that. Going down
that road is a non-trivial effort.
Review: https://reviewboard.asterisk.org/r/4158/
ASTERISK-24489 #close
Reported by: Gregory Malsack
Tested by: Gregory Malsack
ASTERISK-24498 #close
Reported by: Beppo Mazzucato
Tested by: Beppo Maazucato
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a config file is read, an unescaped semicolon signals comments which are
stripped from the value before it's stored. Escaped semicolons are then
unescaped and become part of the value. Both of these behaviors are normal
and expected. When the config is serialized either by 'dialplan save' or
AMI/UpdateConfig however, the now unescaped semicolons are written as-is.
If you actually reload the file just saved, the unescaped semicolons are
now treated as start of comments.
Since true comments are stripped on read, any semicolons in
ast_variable.value must have been escaped originally. This patch
re-escapes semicolons in ast_variable.values before they're written to
file either by 'dialplan save' or config/ast_config_text_file_save which
is called by AMI/UpdateConfig. I also fixed a few pre-existing formatting
issues nearby in pbx_config.c
Tested-by: George Joseph
ASTERISK-20127 #close
Review: https://reviewboard.asterisk.org/r/4132/
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When merging from 12 to 13 there were conflicts,
I mistakenly had the loop run ast_closestream(others[0])
when it should be ast_closestream(others[x]).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a channel is imparted to a bridge, the invocation of the function may
provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart,
the caller must assume that ownership has passed to the function, as in all
paths the function destroys the struct prior to returning (as its purpose is
to configure the behavior of the channel while in the bridge). On one off
nominal path - where the channel already has a PBX thread - the struct was not
being destroyed.
This patch fixes that glitch.
ASTERISK-24437 #close
Reported by: Scott Griepentrog
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