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When the publication of attended transfer messages were pushed to another
thread, some subtle race conditions were introduced with the CEL unit tests.
This patch fixes one of them, and pushes the other to ASTERISK-22367, which
already exists to fix another bouncy CEL unit test.
In particular, this patch fixes the test_cel_attended_transfer_bridges_link
test, and defers the test_cel_attended_transfer_bridges_swap test to the
aforementioned JIRA issue.
ASTERISK-22367
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The UW IMAP library is instrinsically not thread-safe, and relies upon higher
level applications to guarantee thread safety. For the most part, this is
provided by the vms object, which provides locking for individual streams.
Unfortunately, this is not sufficient for calls to mail_open which create the
IMAP stream. mail_open can, on some systems, call into a UW IMAP specific
function for determining the address of a system based on a hostname,
ip_nametoaddr.
In the ip6_unix implementation of this function, static variables are used
to hold parsing buffers. This can cause a crash if multiple threads attempt
to convert a hostname to an address at the same time. Locking on a single
mail stream is not sufficient to prevent simultaneous access to these static
variables.
In the IMAP library, this function can be called from the mail_open and
imap_status functions. As the imap_status function is not used by
app_voicemail, locking on access to mail_open is sufficient to prevent
any mangling of the buffers.
Review: https://reviewboard.asterisk.org/r/4188/
ASTERISK-24516 #close
Reported by: David Duncan Ross Palmer
Tested by: David Duncan Ross Palmer
patches:
ASTERISK-24516.diff uploaded by David Duncan Ross Palmer (License 6660)
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Tested-by: George Joseph
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When merging the changes up stream in r428687, I missed the fact that the
signature for stasis_message_type_create was changed. This patch fixes
the compilation issues introduced by that merge.
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This patch makes a small rearrangement to only do dialplan lookups during
loopback switches if the pattern matches. Prior to this patch, the dialplan
lookups were always performed, even when the result would be discarded.
Dialplan lookups can be very costly if remote switches - like DUNDi - are
present. In those cases extension matching is sped up considerably, making
the issue of lost digits more manageable.
As collateral damage, 6 trailing spaces were killed.
Review: https://reviewboard.asterisk.org/r/4211
ASTERISK-24577 #close
Reported by: Birger Harzenetter
patches:
ast-loopback.patch uploaded by Birger Harzenetter (License 5870)
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of a transfer.
There are two methods within res_pjsip_refer for keeping track of the state of a transfer.
The first is a framehook which looks at frames passing by to determine the state. The second
subscribes to know when the channel joins a bridge. In the case when the channel joins the
bridge the framehook is *NOT* removed and this prevents the native RTP bridging technology
from getting used.
This change gets the channel and if it still exists remove the framehook.
Review: https://reviewboard.asterisk.org/r/4218/
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Add
const char *ast_variable_find_in_list(const struct ast_variable *list,
const char *variable);
ast_variable_find() requires a config category to search whereas
ast_variable_find_in_list() just needs the root list element which is
useful if you don't have a category.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4217/
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While troubleshooting other things I realized there were no pjsip cli
commands for identify. This patch adds them. It also also fixes a
reference leak when a 'show endpoint' displayed identifies and properly
sets the return code if load_module can't allocate a cli formatter structure.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4212/
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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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of a second of the recording.
The Record dialplan function trims 1/4 of a second from the end of recordings in case
they are terminated because of DTMF. When hanging up, however, you don't want this to happen.
This change makes it so on hangup this does not occur.
ASTERISK-24530 #close
Reported by: Ben Smithurst
patches:
app_record_v2.diff submitted by Ben Smithurst (license 6529)
Review: https://reviewboard.asterisk.org/r/4201/
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command.
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/
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Fix the failing /channels/features/test_features_channel_dtmf unit test.
DTMF emulation does not work without a stream of packets to prod the
emulation code.
Review: https://reviewboard.asterisk.org/r/4199/
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Any partially collected DTMF digits for a DTMF hook need to be pushed into
the bridge when a channel leaves the bridging system as if there were a
timeout.
Review: https://reviewboard.asterisk.org/r/4199/
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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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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/
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Prior to this commit, the address family of the first item in an ACL
was used to compare all incoming traffic. This could lead to traffic
of other IP address families bypassing ACLs.
ASTERISK-24469 #close
Reported by Matt Jordan
Patches:
ASTERISK-24469-11.diff uploaded by Matt Jordan (License #6283)
AST-2014-012
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The DB dialplan function when executed from an external protocol (for instance
AMI), could result in a privilege escalation.
Asterisk now inhibits the DB function from being executed from an external
interface if the live_dangerously option is set to no.
ASTERISK-24534
Reported by: Gareth Palmer
patches: submitted by Gareth Palmer (license 5169)
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The biggest problem this patch fixes is that ACLs weren't previously being
loaded when the res_pjsip_acl module was loaded. Yikes. In addition, the
ACL options contact_permit and contact_acl were effectively interpreted as
contact_deny and this patch fixes that as well.
AST-1418 #close
Reported by: Thomas Thompson
Review: https://reviewboard.asterisk.org/r/4120/
ASTERISK-24531 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4171/
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Confbridge dialplan function permission escalation via AMI and inappropriate
class authorization on the ConfbridgeStartRecord action. The CONFBRIDGE dialplan
function when executed from an external protocol (for instance AMI), could
result in a privilege escalation. Also, the AMI action “ConfbridgeStartRecord”
could also be used to execute arbitrary system commands without first checking
for system access.
Asterisk now inhibits the CONFBRIDGE function from being executed from an
external interface if the live_dangerously option is set to no. Also, the
“ConfbridgeStartRecord” AMI action is now only allowed to execute under a
user with system level access.
ASTERISK-24490
Reported by: Gareth Palmer
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res_pjsip_refer.
The implementation of INVITE with Replaces in res_pjsip_refer did not expect them to
occur in-dialog. As a result it would incorrectly attempt to hang up a channel it
thought was under its control. In reality the channel would be under the control of
another thread. When the other thread accessed the channel it would be accessing freed
memory and could crash.
This change makes res_pjsip_refer not act on an in-dialog INVITE with Replaces.
ASTERISK-24528 #close
Reported by: Joshua Colp
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a CANCEL has been received.
Due to the serialized architecture of chan_pjsip there exists a race condition where a CANCEL may
be received and processed before responses (such as 180 Ringing, 183 Session Progress, and 200 OK)
are sent. Since the session is in an unexpected state PJSIP will assert when this is attempted.
This change makes it so that these responses are not sent on disconnected sessions.
ASTERISK-24471 #close
Reported by: yaron nahum
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ast_string_fields_copy relies on the fact that
__ast_string_field_release_active never previously
zeroed pool->used, so keeping the existing pointer
was "ok". Now that existing pools can be reset to
'empty', it is important to set each field to
__ast_string_field_empty after releasing the memory.
ASTERISK-24535 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4186/
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Accessing members of struct ast_str outside of the string manipulation API
routines is invalid since struct ast_str is supposed to be treated as
opaque.
Review: https://reviewboard.asterisk.org/r/4194/
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Optimistic SRTP is the ability to enable SRTP but not have it be
a fatal requirement. If SRTP can be used it will be, if not it won't be.
This gives you a better chance of using it without having your sessions
fail when it can't be.
Encrypt all the things!
Review: https://reviewboard.asterisk.org/r/3992/
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There is no guarantee that when we get a Refer-To that it will be NULL terminated.
As the URI parsing function requires it to be we now NULL terminate it.
Additionally parsing the Refer-To as a 'To' header is needless and it can
simply be done as a URI. This also fixes a problem where certain Refer-To headers
would not be parsed as a 'To' header causing the REFER to fail.
ASTERISK-24508 #close
Reported by: Beppo Mazzucato
Review: https://reviewboard.asterisk.org/r/4187/
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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
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If transmit_refer is called when p->refer is already allocated,
it leaks the previous allocation. Updated code to always free
previous allocation during a new allocation. Also instead of
checking if we have a previous allocation, always create a
clean record.
ASTERISK-15242 #close
Reported by: David Woolley
Review: https://reviewboard.asterisk.org/r/4160/
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When r428077 was made for ASTERISK-24522, it failed to take into account users
who are neither wait_marked nor end_marked. These users are *also* supposed to
hear the 'leader has left the conference' message. Granted, this behaviour is
a bit odd; however, that is how it used to work... and behaviour changes are
not good.
This patch ensures that if there are any 'normal' users present when the last
marked user leaves the conference, the message will still be played to them.
Note that this regression was caught by the Asterisk Test Suite's
confbridge_nominal test, which has a quirky combination of users.
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tests/test_cel: Fix CEL unit tests
This is a backport of the test_cel portion of r427870, which was not applied to
the 12 branch. This fixes the compilation issues with the CEL unit tests
introduced by the API changes needed to fix publication of blind transfer
messages.
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Consider the following:
- A marked user in a conference
- One or more end_marked only users in the conference
When the marked users leaves, we will be in the conf_state_multi_marked state.
This currently will traverse the users, kicking out any who have the end_marked
flags. When they are kicked, a full ast_bridge_remove is immediately called on
the channels. At this time, we also unilaterally set the need_prompt flag.
When the need_prompt flag is set, we then playback a sound to the bridge
informing everyone that the leader has left; however, no one is left in the
bridge. This causes some odd behaviour for the end_marked users - they are
stuck waiting for the bridge to be unlocked. This results in them waiting for
5 or 6 seconds of dead air before hearing that they've been kicked.
Unfortunately, we do have to keep the bridge locked while we're playing back
the 'leader-has-left' prompt. If there are any wait_marked users in the
conference, this behaviour can't be easily changed - but we do make the case
of the end_marked users better with this patch.
Review: https://reviewboard.asterisk.org/r/4184/
ASTERISK-24522 #close
Reported by: Matt Jordan
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The AOR value may contain the name of an AOR or a full SIP URI.
Checking if the AOR exists can't be done as a result of this.
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and it does not exist.
ASTERISK-24499 #close
Reported by: Rusty Newton
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For chan_motif the direct return value of the underlying config options framework
was passed back. This can relay various states which the module loader would not
interpet as success. It has been changed so only on errors will it report back
an error.
For chan_pjsip the code implemented a dummy reload function which always
returned an error. This has been removed as all configuration is held within
res_pjsip instead.
ASTERISK-23651 #close
Reported by: Rusty Newton
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and normal expiration period.
This change enforces the requirements in PJSIP for session timer configuration. The minimum
expiration period must be 90 seconds or higher and the normal expiration period can not
be lower than the minimum expiration period. If either of these were done the code would
assert at session setup time.
ASTERISK-24336 #close
Reported by: Leon Rowland
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This patch adds microsecond precision when inserting a CEL record into a table
with an "eventtime" column of type timestamp, instead of second precision. The
documentation (configs/cel_odbc.conf.sample) was already saying that the
eventtime column included microseconds precision, but that was not the case.
Also, without this patch, if you had a table with an "eventtime" column of
type varchar, you had millisecond precision. With this patch, you also get
microsecond precision in this case.
Review: https://reviewboard.asterisk.org/r/3980
ASTERISK-24283 #close
Reported by: Etienne Lessard
patches:
cel_odbc_time_precision.patch uploaded by Etienne Lessard (License 6394)
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If the test fails due to memory allocation errors, we may as well attempt to
unlock the bridge on the way out.
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ASTERISK-24279 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4109/
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When sending the USERNAME attribute in an RTP STUN
response, the implementation in append_attr_string
passed the actual length, instead of padding it up
to a multiple of four bytes as required by the RFC
3489. This change adds separate variables for the
string and padded attributed lengths, and performs
padding correctly.
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/4139/
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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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Currently if the leader of a conference bridge leaves any participant
that has musiconhold enabled will not hear the "leader has left" sound.
This is because musiconhold is started and THEN the sound is played.
This change makes it so that the sound is played and THEN musiconhold
is started. This provides a better experience for users as they may not
have known previously why they went back to musiconhold.
Review: https://reviewboard.asterisk.org/r/4177/
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This is the Asterisk 13 version of the patch. The main difference is in the pubsub
code since it was completely refactored between Asterisk 12 and 13.
Review: https://reviewboard.asterisk.org/r/4175
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When using a non-default sorcery wizard (in this instance realtime) for
outbound registrations and after adding in an appropriate call to
ast_sorcery_apply_config() (since it is missing) Asterisk will crash after
a stack overflow occurs due to the code infinitely recursing. The fix entails
removing the outbound registration state dependency from the outbound
registration sorcery object and instead keeping an in memory container that
can be used to lookup the state when needed.
ASTERISK-24514
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4164/
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This change corrects message ordering in cases where a channel-related
message can be received after a Stasis/ARI application has received the
StasisEnd message. The StasisEnd message was being passed to
applications directly without waiting for the channel topic to empty.
As a result of this fix, other bugs were also identified and fixed:
* StasisStart messages were also being sent directly to apps and are
now routed through the stasis message bus properly
* Masquerade monitor datastores were being removed at the incorrect
time in some cases and were causing StasisEnd messages to not be sent
* General refactoring where necessary for the above
* Unsubscription on StasisEnd timing changes to prevent additional
messages from following the StasisEnd when they shouldn't
A channel sanitization function pointer was added to reduce processing
and AO2 lookups.
Review: https://reviewboard.asterisk.org/r/4163/
ASTERISK-24501 #close
Reported by: Matt Jordan
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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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Add missing reference cleanup for newly created bridge.
ASTERISK-24281
Reported by: Stefan Engström
Review: https://reviewboard.asterisk.org/r/4154/
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If during the operation of adding an extension a priority is added but
fails it is possible for the extension to be freed but still exist in
the PBX core. If this occurs subsequent lookups may try to access the
extension and end up in freed memory.
This change removes the extension from the PBX core when the priority
addition fails and then frees the extension.
ASTERISK-24444 #close
Reported by: Leandro Dardini
Review: https://reviewboard.asterisk.org/r/4162/
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When DONT_OPTIMIZE is enabled with dev-mode, it causes a shadow compilation
to be done with output to /dev/null. This can cause errors with coverage
when GCC attempts to write to /dev/null.gcno. This change disables
coverage for the shadow compilation.
ASTERISK-24502 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4151/
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Fix reference leak that happens if (session && !blastaway).
ASTERISK-24505 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4153/
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