If the last line of a file does not have a terminating EOL sequence, we
potentially add garbage to the value returned from the FILE() function.
There is no overflow potential here as we are reading from a buffer of a
known size, we are just reading too much of it.
ASTERISK-26481 #close
Change-Id: I50dd4fcf416fb3c83150040a1a79a59d9eb1ae01
MySql 8.0.1 replaced the "my_bool" type with "bool" so an #if
was added to use "bool" with MYSQL_VERSION_ID >= 80001.
ASTERISK-28604
Change-Id: I66a28d8f0011e33774edee13a6f8efd2302bb920
If the "max_retries" option is set to 0 then upon failure no
further attemps are made, so explicitly document the behavior.
ASTERISK-28602
Change-Id: I1e30daae9dd6c49ce18744164214d3def505acbf
Calling ne_uri_parse allocates memory that needs to be freed with a
corresponding call to ne_uri_free.
ASTERISK-28572 #close
Change-Id: I8a6834da27000a6807d89cb7a157b2a88fcb5e61
This change ensures that the module isn't unloaded when a
WebSocket is open. Previously it was possible to unload the
module manually or during shutdown which could cause a crash
when any active WebSockets were terminated.
ASTERISK-28585
Change-Id: I85c71ab112f99875b586419a34c08c8b34c14c5c
During execution "pjsip show channelstats" cli command by an
external module asterisk crashed. It seems this is a separate
thread running to fetch and print rtp stats. The crash happened on
the ao2_lock method, just before it going to read the rtp stats on
a rtp instance. According to gdb backtrace log, it seems the
session media was already cleaned up at that moment.
ASTERISK-28578
Change-Id: Ia918bfa3c8119060633e39b14852e8d983e0417e
If you specify multiple formats in voicemail.conf, eg. "format = gsm|wav"
and are using realtime ODBC backend, only the first format gets stored
in the database. So when you forward a message later on, there is a bug
generating the email, related to the stored format (GSM) being different
than the desired email format (WAV) specified for the user. Sox can
handle this, but Asterisk needs to tell sox exactly what to do.
ASTERISK-22192
Change-Id: I7321e7f7e7c58adbf41dd4fd7191c887b9b2eafd
PostgreSQL 12 finally removed column adsrc from table pg_catalog.pg_attrdef
(column default values), which has been deprecated since version 8.0.
Since then, the official/correct/supported way to retrieve the column
default value from the catalog is function pg_catalog.pg_get_expr().
This change breaks compatibility with pre-8.0 PostgreSQL servers,
but has reached end-of-support more than a decade ago.
cdr_pgsql and res_config_pgsql still have support for pre-7.3
servers, but cleaning that up is perhaps a topic for a major release,
not this bugfix.
ASTERISK-28571
Change-Id: I834cb3addf1937e19e87ede140bdd16cea531ebe
Fixes a failure in /main/taskprocesor unit test, only occurring in Ubuntu.
Newer versions of GCC require variable initialization.
Change-Id: I2994d8aab9307a8c2c7330584f287a27144a580c
When creating an unsolicited MWI aggregate subscription it was possible for
the subscription object to be double unref'ed. This patch removes the explicit
unref as it is not needed since the RAII_VAR will handle it at function end.
Less concerning there was also a bug that could potentially allow the aggregate
subscription object to be added to the unsolicited container twice. This patch
ensures it is added only once.
ASTERISK-28575
Change-Id: I9ccfdb5ea788bc0c3618db183aae235e53c12763
Issues in pjproject 2.9 caused us to revert some of their changes
as a work around. This introduced another issue where pjproject
wouldn't build with older gcc versions such as that found on
CentOS 6. This commit replaces the reverts with the official
fixes for the original issues and allows pjproject to be built
on CentOS 6 again.
ASTERISK-28574
Reported-by: Niklas Larsson
Change-Id: I06f8507bea553d1a01b0b8874197d35b9d47ec4c
Here's the basic scenario that occurred when executing an AMI fast originate
while at the same time something else locks the channels container, and also
wants a lock on the dialed channel:
1. pbx_outgoing_attempt obtains a lock on a dialed channel
2. concurrently another thread obtains a lock on the channels container, and
subsequently requests a lock on the dialed channel. It waits on #1. For
instance, "core show channel <dialed channel"
3. the outgoing call does not fail, but ends before the pbx_outgoing_attempt
function exits
4. pbx_outgoing_attempt function exits, the outgoing structure destructs, and
attempts to hang up the dialed channel
5. hang up tries to obtain the channels container lock, but can't due to #2.
6. Asterisk is deadlocked.
The solution was to allow the pbx_outgoing_exec function to "steal" ownership
of the dialed channel, and handle hanging it up. The channel now is either hung
up prior to it being potentially locked by the initiating thread, or if locked
the hang up takes place in a different thread, thus alleviating the deadlock.
ASTERISK-28561
patches:
iliketrains.diff submitted by Joshua Colp (license 5000)
Change-Id: I51b42b92dde8f2215b69bb509e28667ee3a3853a
This reverts commit fd2e8d0da7.
Reason for revert: Problematic for users who store their voicemail
on network storage devices, or share voicemail storage between
multiple Asterisk instances.
ASTERISK-28567 #close
Change-Id: I3ff4ca983d8e753fe2971f3439bd154705693c41
On shutdown it's possible for the unsolicited mwi container to be freed before
other dependent threads are done using it. This patch ensures this can no
longer happen by wrapping the container in an ao2_global object. The solicited
container was also changed too.
ASTERISK-28552
Change-Id: I8f812286dc19a34916acacd71ce2ec26e1042047
Both res_pjsip and res_pjsip_mwi made use of serializer pools. However, they
both implemented their own serializer pool functionality that was pretty much
identical in each of the source files. This patch removes the duplicated code,
and uses the new 'ast_serializer_pool' object instead.
Additionally res_pjsip_mwi enables a shutdown group on the pool since if the
timing was right the module could be unloaded while taskprocessor threads still
needed to execute, thus causing a crash.
Change-Id: I959b0805ad024585bbb6276593118be34fbf6e1d
Serializer pools have previously existed in Asterisk. However, for the most
part the code has been duplicated across modules. This patch abstracts the
code into an 'ast_serializer_pool' object. As well the code is now centralized
in serializer.c/h.
In addition serializer pools can now optionally be monitored by a shutdown
group. This will prevent the pool from being destroyed until all serializers
have completed.
Change-Id: Ib1e906144b90ffd4d5ed9826f0b719ca9c6d2971
Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows
the on-hold behavior to be controlled on a per-call basis
ASTERISK-28542 #close
Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8
There are some warning messages which are not informative without endpoint:
"No registered subscribe handler for event presence.winfo"
"No registered publish handler for event presence"
This patch adds an endpoint name to these messages.
Change-Id: Ia2811ec226d8a12659b4f9d4d224b48289650827
Added "like" support for 'core show taskprocessors'. Now you
can specify a specific set of taskprocessors (or just one) by
adding the keyword "like" to the above command, followed by
your search criteria.
Change-Id: I021e740201e9ba487204b5451e46feb0e3222464
pbx_extension_helper takes two 'context' arguments. One (con) is a
pointer directly to a 'struct ast_context' and the other (context) is
the name of the context. In all cases, one of these arguments is NULL
and the other is non-NULL.
Functions that are ultimately called by pbx_extension_helper expect that
'context' will be non-NULL, so we set it unconditionally on entry into
this function.
ASTERISK-28534 #close
Change-Id: Ifbbc5e71440afd80efd441f7a9d72e8b10b6f47d
If a permanent contact URI associated with an AOR is invalid, we add a
Contact header to REGISTER responses with a NULL URI, causing a crash.
ASTERISK-28463 #close
Change-Id: Id2b643e58b975bc560aab1c111e6669d54db9102
This change adds an option, moh_passthrough, that when enabled will pass
hold and unhold requests through using a SIP re-invite. When placing on
hold a re-invite with sendonly will be sent and when taking off hold a
re-invite with sendrecv will be sent. This allows remote servers to handle
the musiconhold instead of the local Asterisk instance being responsible.
Review: https://reviewboard.asterisk.org/r/4103/
Change-Id: Ib6294e906e577e1a4245cb1f058d3976ff484c52
