When the configuration section scheme of chan_dahdi.conf is used (keyword
dahdichan instead of channel) all setvar= options are completely ignored.
No variable defined this way appears in the created DAHDI channels.
* Move the clearing of setvar values to after the deferred processing of
dahdichan.
AST-1378 #close
Reported by: Guenther Kelleter
Patch by: Guenther Kelleter
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For the featdmf signaling mode the incoming MF Caller-ID information is
formatted as follows: *${CALLERID(ani2)}${CALLERID(ani)}#*${EXTEN}#
Rather than discarding the ani2 digits, populate the CALLERID(ani2) value
with what is received instead.
AST-1368 #close
Reported by: Denis Martinez
Patches:
extract_ani2_for_featdmf_v11.patch (license #5621) patch uploaded by Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429783 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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously when SRTP was enabled on a channel it was not possible
to switch to T.38 as no crypto attributes would be present.
This change makes it so it is now possible. If a T.38 re-invite
comes in SRTP is terminated since in practice you can't encrypt
a UDPTL stream. Now... if we were doing T.38 over RTP (which
does exist) then we'd have a chance but almost nobody does that so
here we are.
ASTERISK-24449 #close
Reported by: Andreas Steinmetz
patches:
udptl-ignore-srtp-v2.patch submitted by Andreas Steinmetz (license 6523)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429632 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Verbose message displayed when a file is played back via 'stream file'
was formatted differently than other playbacks:
* It didn't include the channel name
* It didn't include the channel language
It does, however, include the playback offset as well as any escape digits.
That information was kept; however, this patch updates the formatting to more
closely match the Verbose messages displayed when a file is played back by
'control stream file', Playback, ControlPlayback, or any other file playback
operation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Frames with a payload length of 0 were incorrectly handled in res_http_websocket.
Provided a frame with a payload had been received prior it was possible for a double
free to occur. The realloc operation would succeed (thus freeing the payload) but be
treated as an error. When the session was then torn down the payload would be
freed again causing a crash. The read function now takes this into account.
This change also fixes assumptions made by users of res_http_websocket. There is no
guarantee that a frame received from it will be NULL terminated.
ASTERISK-24472 #close
Reported by: Badalian Vyacheslav
Review: https://reviewboard.asterisk.org/r/4220/
Review: https://reviewboard.asterisk.org/r/4219/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When repeatedly starting/stopping a Monitor on a channel, the accumulated
in/out sample counts are never reset to 0. This can cause inadvertent jumps
in the recordings, as the code in the channel core will determine incorrectly
that a jump in the recorded file position should occur. Setting the sample
counts to 0 simply reflects the initial state a Monitor should be in when it
is started, as this is the initial count that would be on the channels at that
time.
ASTERISK-24573 #close
Reported by: Nuno Borges
patches:
24573.patch uploaded by Nuno Borges (License 6116)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the app_meetme module is loaded without its configuration file, the
module settings aren't initialized. In particular, this impacts the use
of logging realtime members. This patch guarantees that we always set the
default module settings on initial load.
Review: https://reviewboard.asterisk.org/r/4242/
ASTERISK-24572 #close
Reported by: Nuno Borges
patches:
24572.patch uploaded by Nuno Borges (License 6116)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428653 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
Merged revisions 428402 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
Merged revisions 428331 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Under load it was possible for the bridging API, and thus ConfBridge, to get
channels that may have hung up stuck in it. This is because handling of state
transitions for a bridged channel within a bridge was not protected and simply
set the new state without regard to the existing state. If the existing state
had been hung up this would get overwritten.
This change adds locking to protect changing of the state and also
takes into consideration the existing state.
ASTERISK-24440 #close
Reported by: Ben Klang
Review: https://reviewboard.asterisk.org/r/4173/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In chan_agent, a '*' is used by default to terminate a bridge with a caller.
This can lead to all sorts of problems if '*' is used by a feature in
features.conf, as the chan_agent disconnect '*' may be detected first.
This patch adds a documentation snippet to features.conf so that users who
attempt to use features with agents know of the potential conflict.
ASTERISK-20402 #close
Reported by: Matt Riddell
patches:
features.conf.diff uploaded by Matt Riddell (License 5023)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r227276, a while loop was turned into a for loop. Unfortunately, a portion
of the while loop was left in the code such that, when a static gateway is
encountered in the list of MGCP gateways, the next gateway would be skipped.
At best, we would simply flip past a gateway; at worst, this could lead to a
crash.
ASTERISK-24500 #close
Reported by: Xavier Hienne
patches:
chan_mgcp.patch uploaded by Xavier Hienne (License 6657)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When UCS2 character encoding is used, one symbol in national language can be
expanded to 4 bytes. The current buffer used for receiving message in
do_monitor_phone is 256 bytes, which is not large enough for incoming messages.
For example:
* AT+CMGR phone response prefix
'+CMGR: "REC UNREAD","+7**********",,"14/10/29,13:31:39+12"\r\n' - 60 bytes
* SMS body with UCS2 encoding (max) - 280 bytes
* AT+CMGR phone response suffix '\r\n\r\nOK\r\n' - 8 bytes
* Terminating null character - 1 byte
This results in a needed buffer size of 349 bytes. Hence, this patch opts for a
350 byte buffer.
ASTERISK-24468 #close
Reported by: Dmitriy Bubnov
patches:
chan_mobile-1_8.diff uploaded by Dmitriy Bubnov (License 6651)
chan_mobile-trunk.diff uploaded by Dmitry Bubnov (License 6651)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
fn_wrapper only adds a reference to the format's module if the file
was able to be opened. If not this causes an unmatched
ast_module_unref in filestream_destructor. Move ast_module_ref to
get_stream.
ASTERISK-24492 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4149/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427464 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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fix cleanup in __ast_play_and_record where others[x] may be leaked.
This was caught where prepend != NULL && outmsg != NULL, once
realfile[x] == NULL any further others[x] would be leaked. A cleanup
block was also added for prepend != NULL && outmsg == NULL.
11+: Fix leak of ast_writestream recording_fs in
app_voicemail:leave_voicemail.
ASTERISK-24476 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4138/
........
Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When r426594 was made, it did not take into account a unit test that verified
that the function properly populated the unsupported buffer. The function
would previously memset the buffer if it detected it had any contents; since
this function can now be called iteratively on successive headers, the unit
tests would now fail. This patch updates the unit tests to reset the buffer
themselves between successive calls, and updates the documentation of the
function to note that this is now required.
........
Merged revisions 426858 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426860 65c4cc65-6c06-0410-ace0-fbb531ad65f3
set_member_value has a couple leaks to references in the variable q
found through testsuite tests/queues/set_penalty. Also remove the
REF_DEBUG_ONLY_QUEUES compiler declaration, this is no longer possible
with the updated REF_DEBUG code.
ASTERISK-24466 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4125/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In update_messages_by_imapuser(), messages were appended to a finite
array which resulted in a crash when an IMAP mailbox contained more
than 256 entries. This memory is now dynamically increased as needed.
Observe that this patch adds a bunch of XXX's to questionable code. See
the review (url below) for more information.
ASTERISK-24190 #close
Reported by: Nick Adams
Tested by: Nick Adams
Review: https://reviewboard.asterisk.org/r/4126/
........
Merged revisions 426691 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When frames are translated by a fax gateway they need to be freed. The
existing call to ast_frfree was unreachable. This change reorganizes
fax_gateway_framehook to ensure that ast_frfree is called when needed.
ASTERISK-24457 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4115/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
