Walter Doekes [Mon, 10 Feb 2014 16:34:27 +0000 (16:34 +0000)]
res_config_pgsql: Fix ast_update2_realtime calls.
Fix so multiple updates from a single call works (add missing ',').
Remove bogus ast_free's that weren't supposed to be there.
Moved a few spaces for readability.
Kinsey Moore [Mon, 10 Feb 2014 15:28:16 +0000 (15:28 +0000)]
ConfBridge: Correct prompt playback target
Currently, when the first marked user enters the conference that
contains waitmarked users, a prompt is played indicating that the user
is being placed into the conference. Unfortunately, this prompt is
played to the marked user and not the waitmarked users which is not
very helpful.
This patch changes that behavior to play a prompt stating
"The conference will now begin" to the entire conference after adding
and unmuting the waitmarked users since the design of confbridge is not
conducive to playing a prompt to a subset of users in a conference in
an asynchronous manner.
(closes issue PQ-1396)
Review: https://reviewboard.asterisk.org/r/3155/
Reported by: Steve Pitts
Tzafrir Cohen [Sun, 9 Feb 2014 15:52:58 +0000 (15:52 +0000)]
chan_dahdi: handle DAHDI_EVENT_REMOVED on a pri D-Channel
When a DAHDI device is removed at run-time it sends the event
DAHDI_EVENT_REMOVED on each channel. This is intended to signal the
userspace program to close the respective file handle, as the driver of
the device will need all of them closed to properly clean-up.
This event has long since been handled in chan_dahdi (chan_zap at the
time). However the event that is sent on a D-Channel of a "PRI" (ISDN)
span simply gets ignored.
This commit adds handling for closing the file descriptor (and shutting
down the span, while we're at it).
It also adds a CLI command 'pri destroy span <N>' to destroy the span
and its DAHDI channels.
Richard Mudgett [Fri, 7 Feb 2014 17:56:57 +0000 (17:56 +0000)]
chan_iax2: Block unnecessary control frames to/from the wire.
Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later)
results in an unexpected call disconnect. The problem happens because
newer values in the enum ast_control_frame_type are not consistent between
the branch versions of Asterisk.
For example:
1) v1.4 calls v1.8 (or later) using IAX2
2) v1.8 answers and sends a connected line update control frame. (on v1.8
AST_CONTROL_CONNECTED_LINE = 22)
3) v1.4 receives the control frame as an end-of-q (on v1.4
AST_CONTROL_END_OF_Q = 22)
4) v1.4 disconnects the call once the receive queue becomes empty.
Several things are done by this patch to fix the problem and attempt to
prevent it from happening again in the future:
* Added a warning at the definition of enum ast_control_frame_type about
how to add new control frame values.
* Made block sending and receiving control frames that have no reason to
go over the wire.
* Extended the connectedline iax.conf parameter to also include the
redirecting information updates.
* Updated the connectedline iax.conf parameter documentation to include a
notice that the parameter must be "no" when the peer is an Asterisk v1.4
instance.
Tzafrir Cohen [Fri, 7 Feb 2014 13:06:35 +0000 (13:06 +0000)]
indications.conf: add stutter tone; end properly
* If the "stutter" (voicemail indication) tone is indeed a stutter tone,
and it ends with a constant tone, make sure that it is the dial tone.
This was done for India (in), Mexico (mx) and the Philippines (ph).
* If no "stutter" tone exists for a country, provide one. This was done for
Spain (es), Malaysia (my) and Venezuela (ve).
Rusty Newton [Wed, 5 Feb 2014 22:59:39 +0000 (22:59 +0000)]
formats/format_wav: enhancing log message "Not a wav file" to be clear on what is supported
Modifying the log message to be more specific as to what is supported. Specifically it seems format_wav supports only PCM encoded versions with a lower-case '.wav' extension.
(closes issues ASTERISK-22310)
Reported by: Jim Credland
Review: https://reviewboard.asterisk.org/r/3188/
........
Merged revisions 407511 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Wed, 5 Feb 2014 20:37:12 +0000 (20:37 +0000)]
Logger: Fix handling of absolute paths
This fixes path handling for log files so that an extra / is not
appended to the file path when the path is absolute (begins with /).
This would previously result in different but functionally equivalent
paths in the output of 'logger show channels'.
........
Merged revisions 407455 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Joshua Colp [Tue, 4 Feb 2014 02:20:52 +0000 (02:20 +0000)]
res_clialiases: Fix crash when reloading and re-aliasing an alias that is in use.
The code assumed that unregistering the alias would always succeed while in
practice this is not actually true. A common case is the "reload" command itself.
If the cli_aliases.conf configuration file was changed and reload executed the
command would fail to unregister and ultimately point to freed memory.
The reload process now checks whether unregistering succeeded or not and if not
the old CLI alias is retained.
(closes issue ASTERISK-19773)
Reported by: Joel Vandal
Corey Farrell [Sat, 1 Feb 2014 00:23:42 +0000 (00:23 +0000)]
app_stack: protect against missing parameters to STACK_PEEK and LOCAL_PEEK
STACK_PEEK requires 2 parameters and LOCAL_PEEK requires 1 parameter. This
protects against situations where those parameters are blank or missing by
logging an error and returning.
(closes issue ASTERISK-23220)
Reported by: James Sharp
........
Merged revisions 407100 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Fri, 31 Jan 2014 23:28:30 +0000 (23:28 +0000)]
app_dial: Allow macro/gosub pre-bridge execution to occur on priorities
The parsing for the destination of the macro/gosub uses the '^' character to
separate out context, extension, and priority. However, the logic for the
macro/gosub execution was written such that it would only do the actual
macro/gosub jump if a '^' character existed. This doesn't apply when the
macro/gosub jump occurs in a priority/priority label. This patch changes
the logic so that the parsing still occurs, but the jump will occur even
for priorities/priority labels.
Corey Farrell [Thu, 30 Jan 2014 20:28:40 +0000 (20:28 +0000)]
res_rtp_asterisk & udptl: fix port selection to work with SELinux restrictions
ast_bind to a port reserved for another program by SELinux causes
errno == EACCES. This caused random failures when binding rtp or
udptl sockets. Treat EACCES as a non-fatal error, try next port.
Asterisk's RADIUS module currently build against libradiusclient-ng, but this
project has been superseeded by libfreeradius-client. The API is 99% compatible
except that the header name has changed, the library name has changed, and
the configuration file location has changed.
(closes issue ASTERISK-22980)
Reported by: Jeremy Lainé
Patches:
freeradius-client.patch uploaded by sharky (license 6561)
........
Merged revisions 406801 from http://svn.asterisk.org/svn/asterisk/branches/1.8
rtp_engine: improved handling of get_rtp_info failure
In ast_rtp_instance_make_compatible(), after a failure of
channel tech call get_rtp_info() to return peer_instance,
the null pointer would be passed to ao2_ref, producing an
error that looked like a refernce counting problem but is
not. This patch corrects that and adds helpful LOG_ERROR
messages to indicate which failure path occurred.
Russell Bryant [Mon, 27 Jan 2014 20:35:25 +0000 (20:35 +0000)]
Allow nested #includes in extconfig.conf
extconfig.conf was hard-coded to not allow nested includes for some reason.
The code has been this way since a patch was merged for ASTERISK-3333 (revision
4889), which was a significant update to this code ("Merge config updates").
I can't figure out any good reason why this should be limited. This patch just
removes the limit and uses the default nesting depth limit.
Russell Bryant [Mon, 27 Jan 2014 01:14:19 +0000 (01:14 +0000)]
Protect ast_filestream object when on a channel
The ast_filestream object gets tacked on to a channel via
chan->timingdata. It's a reference counted object, but the reference
count isn't used when putting it on a channel. It's theoretically
possible for another thread to interfere with the channel while it's
unlocked and cause the filestream to get destroyed.
Use the astobj2 reference count to make sure that as long as this code
path is holding on the ast_filestream and passing it into the file.c
playback code, that it knows it's valid.
Richard Mudgett [Fri, 24 Jan 2014 23:07:08 +0000 (23:07 +0000)]
CEL: Protect data structures during reload and shutdown.
The CEL data structures need to be protected during a configuration reload
and shutdown. Asterisk crashed during a shutdown because CEL events were
still in flight and the CEL data structures were already destroyed.
* Protected the appset and linkedids ao2 containers using the reload_lock.
As a result appset, linkedids, and held objects don't need a lock.
* Added NULL checks before use of the appset and linkedids ao2 containers
in case the CEL module is already shutdown.
* Fixed overloading of the linkedids held objects reference count. During
shutdown any held objects would be leaked.
* Fixed memory leak of linkedids held objects if the LINKEDID_END is not
being tracked. The objects in the linkedids container were not removed if
the LINKEDID_END event is not used.
* Added access protection to the appset container during the CLI "cel show
status" command.
* Made CEL config reload not set defaults if the cel.conf file is invalid.
Richard Mudgett [Fri, 24 Jan 2014 17:54:18 +0000 (17:54 +0000)]
manager: Protect data structures during shutdown.
Occasionally, the manager module would get an "INTERNAL_OBJ: bad magic
number" error on a "core restart gracefully" command if an AMI connection
is established.
* Added ao2_global_obj protection to the sessions global container.
* Fixed the order of unreferencing a session object in session_destroy().
* Removed unnecessary container traversals of the white/black filters
during session_destructor().
pbx.c: Pre-initialize timezone to avoid crash on destroy
In ast_build_timing, initialize the timezone value to NULL
in order to avoid deferencing an uninitialized value later
when calling ast_destroy_timing. The timezone value could
be uninitialized if ast_build_timing were to fail due to a
zero length time string.
(closes issue ASTERISK-22861)
Reported by: Sebastian Murray-Roberts
Review: https://reviewboard.asterisk.org/r/3134/
Patches:
ast_build_timing-initialize-timezone.patch uploaded by coreyfarrell (license 5909)
........
Merged revisions 406241 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Wed, 22 Jan 2014 19:31:12 +0000 (19:31 +0000)]
ConfBridge: Fix channel parameter documentation
Confbridge AMI and CLI commands for mute, unmute, and setting the
single video source can accept channel prefixes in lieu of a full
channel name, but documentation states only that it is required and is
a channel name. This corrects the documentation.
Kinsey Moore [Wed, 22 Jan 2014 18:30:18 +0000 (18:30 +0000)]
chan_sip: Decline image streams on unsupported transports
This change allows chan_sip to decline individual image streams over
unsupported transports in the SDP of the 200 response. Previously,
an image stream offer with RTP/AVP as the transport would cause
chan_sip to respond with a 488.
Rusty Newton [Fri, 17 Jan 2014 15:40:37 +0000 (15:40 +0000)]
Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027
Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.
(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
transferred.patch uploaded by Jeremy Laine (license 6561)
hyphen.patch uploaded by Jeremy Laine (license 6561)
sip.conf.sample.patch uploaded by Eugene (license 6360)
........
Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kevin Harwell [Thu, 16 Jan 2014 18:57:43 +0000 (18:57 +0000)]
res_fax: check_modem_rate() returned incorrect rate for V.27
According to the new standard for V.27 and V.32 they are able to transmit
at a bit rate of 4,800 or 9,600. The check_mode_rate function needed to be
updated to reflect this. Also, because of this change the default 'minrate'
value was updated to be 4800.
(closes issue ASTERISK-22790)
Reported by: Paolo Compagnini
Patches:
res_fax.txt uploaded by looserouting (license 6548)
........
Merged revisions 405656 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Joshua Colp [Wed, 15 Jan 2014 16:35:30 +0000 (16:35 +0000)]
cel_manager: Don't crash if configuration file is invalid.
The cel_manager module did not properly handle the case where the
configuration file was invalid. The module will now output a warning
message and disable itself if this occurs.
Reported by: Bryan Walters
........
Merged revisions 405581 from http://svn.asterisk.org/svn/asterisk/branches/1.8
chan_sip: No BYE message sent after INVITE with Replaces
Setting channel state DOWN is an unnecessary step that was
only being done in handle_invite_replaces(). This changes
that by removing the call and reducing locking.
(closes issue ASTERISK-23010)
Reported by: Ryan Tilton
Review: https://reviewboard.asterisk.org/r/3116/
........
Merged revisions 405486 from http://svn.asterisk.org/svn/asterisk/branches/1.8
chan_sip: fix Local From tag on outbound register regression
In ASTERISK-12117, an improvement to insure consistant local from tags
on outbound registrations resulted in an undesirable behavior - caused
by leftover unexpired sip_pvt dialogs (with the previous cseq number),
resulting in many uncessary REGISTER requests. Instead of significant
rework of transmit_register(), this change deletes the dialogs after a
200 OK response indiciating a successful registration, keeping the old
dialogs from interfering with normal operation.
(closes issue ASTERISK-22946)
Reported by: Stephan Eisvogel
Review: https://reviewboard.asterisk.org/r/3109/
........
Merged revisions 405433 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Tue, 14 Jan 2014 17:26:35 +0000 (17:26 +0000)]
verbosity: Fix performance of console verbose messages.
The per console verbose level feature as previously implemented caused a
large performance penalty. The fix required some minor incompatibilities
if the new rasterisk is used to connect to an earlier version. If the new
rasterisk connects to an older Asterisk version then the root console
verbose level is always affected by the "core set verbose" command of the
remote console even though it may appear to only affect the current
console. If an older version of rasterisk connects to the new version
then the "core set verbose" command will have no effect.
* Fixed the verbose performance by not generating a verbose message if
nothing is going to use it and then filtered any generated verbose
messages before actually sending them to the remote consoles.
* Split the "core set debug" and "core set verbose" CLI commands to remove
the per module verbose support that cannot work with the per console
verbose level.
* Added a silent option to the "core set verbose" command.
* Fixed "core set debug off" tab completion.
* Made "core show settings" list the current console verbosity in addition
to the root console verbosity.
* Changed the default verbose level of the 'verbose' setting in the
logger.conf [logfiles] section. The default is now to once again follow
the current root console level. As a result, using the AMI Command action
with "core set verbose" could again set the root console verbose level and
affect the verbose level logged.
Matthew Jordan [Tue, 14 Jan 2014 15:32:16 +0000 (15:32 +0000)]
chan_sip: Hangup transferer/transferee when transfer to Parking fails
When performing a SIP transfer to a Park extension, if the Park fails, chan_sip
will currently not hang up either the transferer or the transfer target. This
results in the channels being orphaned with no thread to service frames,
resulting in stuck channels.
This patch immediately hangs up the two channels if a Park fails.
Matthew Jordan [Mon, 13 Jan 2014 21:45:35 +0000 (21:45 +0000)]
res/Makefile: alias dist-clean to distclean
A 'make distclean' is equivalent to 'make dist-clean' in the top most Makefile.
This patch updates the res/Makefile to recognize both distclean and dist-clean.
Note that this is needed for removing build.mak, which can run into problems
if the source file of Asterisk or its path is changed after build.mak is
generated.
Kevin Harwell [Thu, 9 Jan 2014 16:49:55 +0000 (16:49 +0000)]
res_rtp_asterisk: Fails to resume WebRTC call from hold
In ast_rtp_ice_start if the ice session create check list failed, start check
was never initiated and ice_started was never set to true. Upon re-entering
the function (for instance, [un]hold) it would try to create the check list
again with duplicate remote candidates.
Fixed so that if the create check list fails the necessary data structures
are properly re-initialized for any subsequent retries.
Note, it was decided to not stop ice support (by calling ast_rtp_ice_stop) on a
check list failure because it possible things might still work. However, a
debug message was added to help with any future troubleshooting.
Matthew Jordan [Thu, 9 Jan 2014 15:41:31 +0000 (15:41 +0000)]
app_confbridge: Fix crash caused when waitmarked/marked users leave together
When waitmarked users join a ConfBridge, the conference state is transitioned
from EMPTY -> INACTIVE. In this state, the users are maintined in a waiting
users list. When a marked user joins, the ConfBridge conference transitions
from INACTIVE -> MULTI_MARKED, and all users are put onto the active list of
users. This process works correctly.
When the marked user leaves, if they are the last marked user, the MULTI_MARKED
state does the following:
(1) It plays back a message to the bridge stating that the leader has left the
conference. This requires an unlocking of the bridge.
(2) It moves waitmarked users back to the waiting list
(3) It transitions to the appropriate state: in this case, INACTIVE
However, because it plays the prompt back to the bridge before moving the users
and before finishing the state transition, this creates a race condition: with
the bridge unlocked, waitmarked users who leave the conference (or are kicked
from it) can cause a state transition of the bridge to another state before
the conference is transitioned to the INACTIVE state. This causes the state
machine to get a bit wonky, often leading to a crash when the MULTI_MARKED state
attempts to conclude its processing.
This patch fixes this problem:
(1) It prevents kicked users from being kicked again. That's just a nicety.
(2) More importantly, it fixes the race condition by only playing the prompt
once the state has transitioned correctly to INACTIVE. If waitmarked users
sneak out during the prompt being played, no harm no foul.
Review: https://reviewboard.asterisk.org/r/3108/
Note that the patch committed here is essentially the same as uploaded by
Simon Moxon on ASTERISK-22740, with the addition of the double kick prevention.
(closes issue AST-1258)
Reported by: Steve Pitts
(closes issue ASTERISK-22740)
Reported by: Simon Moxon
patches:
ASTERISK-22740.diff uploaded by Simon Moxon (license 6546)
Kinsey Moore [Tue, 7 Jan 2014 20:45:15 +0000 (20:45 +0000)]
Add the missing part of r400140
When the patch to add retry-on-forbidden-response was committed, part
of the patch for chan_sip was not committed which caused the feature to
be entirely nonfunctional. This corrects the code in question.
Kevin Harwell [Fri, 3 Jan 2014 21:58:17 +0000 (21:58 +0000)]
cel_pgsql: module not correctly reloading
Upon reload the module unconditionally "unloaded" the module (freeing memory
and setting pointers to NULL) and then when attempting a "load" if the config
file had not changed then nothing would be reinitialized.
By moving the "unload" to occur conditionally (reload only) after an attempted
configuration load, but before module "loading" alleviates the issue. The module
now loads/unloads/reloads correctly.
Kevin Harwell [Fri, 3 Jan 2014 18:27:25 +0000 (18:27 +0000)]
app_meetme: compiler warning
Fixed a compiler warning (errors in 'dev-mode') given by gcc version 4.8.1.
The one in app_meetme involved the 'sizeof-pointer-memaccess'
(see: http://gcc.gnu.org/gcc-4.8/porting_to.html) warning. Fixed so
it would no longer issue a warning and can compile again in 'dev-mode'.
func_strings: use memmove to prevent overlapping memory on strcpy
When calling REPLACE() with an empty replace-char argument, strcpy
is used to overwrite the the matching <find-char>. However as the
src and dest arguments to strcpy must not overlap, it causes other
parts of the string to be overwritten with adjacent characters and
the result is mangled. Patch replaces call to strcpy with memmove
and adds a test suite case for REPLACE.
Kevin Harwell [Tue, 31 Dec 2013 21:26:00 +0000 (21:26 +0000)]
cel_pgsql: deadlock on unload and core_event_dispatcher
A deadlock can happen between a thread unloading or reloading the cel_pgsql
module and the core_event_dispatcher taskprocessor thread. Description of
what is happening:
Thread 1 (for example, a netconsole thread):
a "module reload cel_pgsql" is launched
the thread enter the "my_unload_module" function (cel_pgsql.c)
the thread acquire the write lock on psql_columns
the thread enter the "ast_event_unsubscribe" function (event.c)
the thread try to acquire the write lock on ast_event_subs[sub->type]
the taskprocessor pop a CEL event
the thread enter the "handle_event" function (event.c)
the thread acquire the read lock on ast_event_subs[sub->type]
the thread callback the "pgsql_log" function (cel_pgsql.c), since it's a subscriber of CEL events
the thread try to acquire a read lock on psql_columns
res_fax.c: crash on framehook with no dsp in fax detect
In fax_detect_framehook() a null pointer reference can occur where a
voice frame is processed but no dsp is attached to the fax detection
structure. The code block that rejects frames that detection cannot
be processed on is checking for dsp but falls through when it should
instead return, as this change implements.
When Asterisk is shut down, the astdb_atexit() function releases
(finalize) the previously initiated (prepared) SQL statements in
sqlite3. Another thread making a subsequent request can cause a
crash in sqlite3. This patch eliminates that issue by resetting
the statement pointer after it is released/cleared. The sqlite3
code detects the null pointer, and aborts the operation cleanly.
Alexandr Anikin [Thu, 19 Dec 2013 08:15:45 +0000 (08:15 +0000)]
Handle temporary failures on gk registration
Introduce new 'stopped' state for gk client and restart gk client
on failures
Remove ooh323 stack command lock as it is not need now.
(closes issue ASTERISK-21960)
Reported by: Dmitry Melekhov
Patches:
ASTERISK-21960.patch
ASTERISK-21960-stacklockup-2.patch
Tested by: Dmitry Melekhov
David M. Lee [Mon, 16 Dec 2013 17:14:14 +0000 (17:14 +0000)]
security: Inhibit execution of privilege escalating functions
This patch allows individual dialplan functions to be marked as
'dangerous', to inhibit their execution from external sources.
A 'dangerous' function is one which results in a privilege escalation.
For example, if one were to read the channel variable SHELL(rm -rf /)
Bad Things(TM) could happen; even if the external source has only read
permissions.
Execution from external sources may be enabled by setting
'live_dangerously' to 'yes' in the [options] section of asterisk.conf.
Although doing so is not recommended.
pbx.c: put copy of ast_exten.data on stack to prevent memory corruption
During dialplan execution in pbx_extension_helper(), the contexts global
read lock prevents link list corruption, but was released with a pointer
to the ast_exten and data later used in variable substitution. Instead,
this patch removes pbx_substitute_variables() and locates a copy of the
ast_exten data on the stack before releasing the lock, where ast_exten
could get free'd by another thread performing a module reload.
(issue AST-1179)
Reported by: Thomas Arimont
(issue AST-1246)
Reported by: Alexander Hömig
Review: https://reviewboard.asterisk.org/r/3055/
........
Merged revisions 403862 from http://svn.asterisk.org/svn/asterisk/branches/1.8
app_sms: BufferOverflow when receiving odd length 16 bit message
This patch prevents an infinite loop overwriting memory when
a message is received into the unpacksms16() function, where
the length of the message is an odd number of bytes.
(closes issue ASTERISK-22590)
Reported by: Jan Juergens
Tested by: Jan Juergens
........
Merged revisions 403853 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Russell Bryant [Wed, 11 Dec 2013 19:14:52 +0000 (19:14 +0000)]
Reset peer outboundproxy on sip.conf reload
If you set a peer's outboundproxy and then removed it from the config,
this would not get picked up in a config reload. This patch fixes that
by resetting it in set_peer_defaults().
Matthew Jordan [Mon, 9 Dec 2013 03:11:05 +0000 (03:11 +0000)]
res_fax_spandsp: Always init T.38 session to avoid crashes during state change
Prior to this patch, res_fax_spandsp was conservative with how it initialized
the spandsp T.38 context. It would only initialize it if the driver thought
the current state was a T.38 fax. While this works fine in nominal situations,
in certain off nominal situations, res_fax_spandsp can believe that a T.38
fax will not occur when in fact one has started. In particular, this was
discovered when res_fax would fall back to audio after timing out on a T.38
upgrade. The SIP channel driver would continue to retry the re-INVITE and -
if the remote end responded after res_fax timed out with a 200 OK - a T.38
frame would be delivered to the res_fax stack when it no longer expected it.
As it turns out, there does not appear to be any downside to always
initializing the T.38 context, other than the actual memory allocation.
Since that avoids this off nominal situation (and others which are equally
likely hard to predict), this is the safest way to avoid this problem.
Much thanks to Torrey as well for providing a scenario that reproduces this
issue.
Alexandr Anikin [Mon, 2 Dec 2013 17:55:49 +0000 (17:55 +0000)]
Check and reject non-digits e164 values on peers and general sections in ooh323.conf
Regenerate e164 endpoint list on reload ooh323
(issue ASTERISK-22901)
Reported by: Cyril CONSTANTIN
Patches:
ASTERISK-22901.patch
Joshua Colp [Fri, 22 Nov 2013 17:11:07 +0000 (17:11 +0000)]
translate: Move freeing of frame to after it is used.
When translating from one format to another it is possible
to inform the translation function that the source frame should
be freed. This was previously done immediately but shortly
afterwards the frame that was freed was accessed and used again.
This change moves code around a bit so that the frame is now
freed after it has been completely used.
Kinsey Moore [Tue, 12 Nov 2013 15:00:36 +0000 (15:00 +0000)]
chan_dahdi: Fix crash during caller ID read
Asterisk will sometimes core dump during caller id read on analog
channels due to a negative return value from the read() in
my_get_callerid that slips through as a negative length argument to
callerid_feed() if the errno returned by DAHDI is ELAST. This change
ensures that the negative return is treated properly even when it is
ELAST.
(closes issue ASTERISK-22746)
Reported by: Michael Walton
Patches:
chan_dahdi_cid_crash_fix.r401410.patch uploaded by Michael Walton (License 6502)
........
Merged revisions 402708 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Mon, 11 Nov 2013 15:35:22 +0000 (15:35 +0000)]
app_queue: Honor penalty limits of 0
In the current app_queue code from 1.8 up to trunk the upper and lower
penalties can be set to 0 but the value is interpreted to be disabled
instead of actually setting limits. This is especially evident if min
and max limits are set to 0 and members with penalties of 0 and 1 are
in the queue since the member with penalty 1 will still receive calls.
This patch adjusts the special disabled value to be INT_MAX instead of
0.
(closes issue ASTERISK-20862)
Review: https://reviewboard.asterisk.org/r/2995/
Reported by: Schmooze Com
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Merged revisions 402645 from http://svn.asterisk.org/svn/asterisk/branches/1.8
chan_sip: keep same local (from) tag for outgoing register requests
For outbound register requests the tag on the From line was
updated every 20 seconds prior to a successful registration
and also once for each registration renewal. That behavior
can possibly cause the registration to be denied because of
the different tag, and is not aligned with the intention of
RFC 3261 8.1.3.5 "... request constitutes a new transaction
and SHOULD have the same value of the Call-ID, To, and From
of the previous request...". This updates chan_sip to have
a field to keep the local tag in the registration structure
and use that tag for registration requests where the callid
is also unchanged.
Kevin Harwell [Tue, 5 Nov 2013 15:11:07 +0000 (15:11 +0000)]
Recorded merge of revisions 402468 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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chan_sip: notify dialog info ignores presentation indicator in callerid
The presentation indicator in a callerid (e.g. set by dialplan function
Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog Info Notifies
are generated during extension monitoring. Added a check to make sure the
name and/or number presentations on the callee (remote identity) are set to
allow. If they are restricted then "anonymous" is used instead.
(closes issue AST-1175)
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2976/
Kevin Harwell [Mon, 4 Nov 2013 20:52:58 +0000 (20:52 +0000)]
chan_sip: notify dialog info ignores presentation indicator in callerid
The presentation indicator in a callerid (e.g. set by dialplan function
Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog Info Notifies
are generated during extension monitoring. Added a check to make sure the
name and/or number presentations on the callee (remote identity) are set to
allow. If they are restricted then "anonymous" is used instead.
(closes issue AST-1175)
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2976/
Richard Mudgett [Sat, 2 Nov 2013 02:11:03 +0000 (02:11 +0000)]
confbridge: Separate user muting from system muting overrides.
The system overrides the user muting requests when MOH is playing or a
waitmarked user is waiting for a marked user to join. System muting
overrides interfere with what the user may wish the muting to be when the
system override ends.
* User muting requests are now independent of the system muting overrides.
The effective muting is now the logical or of the user request and system
override.
* Added a Muted column to the CLI "confbridge list <conference>" command.
* Added a Muted header to the AMI ConfbridgeList action ConfbridgeList
event.
Richard Mudgett [Fri, 1 Nov 2013 23:52:45 +0000 (23:52 +0000)]
config: Allow ConfBridge DTMF menus to have '#' as the first digit.
ConfBridge allows custom DTMF menus to be created in the confbridge.conf
file by assigning a DTMF key sequence to a sequence of actions as follows:
DTMF-sequence = action,action...
Unfortunately, the normal config file processing code interprets an
initial '#' character as starting a directive such as #include.
* Add the ability to escape the first non-blank character in a config line
so the '#' character can be used without triggering the directive
processing code.
(closes issue AFS-2)
(closes issue ASTERISK-22478)
Reported by: Nicolas Tanski
Patches:
jira_asterisk_22478_v11.patch (license #5621) patch uploaded by rmudgett (modified)
Kinsey Moore [Fri, 1 Nov 2013 12:31:49 +0000 (12:31 +0000)]
chan_sip: Fix RTCP port for SRFLX ICE candidates
This corrects one-way audio between Asterisk and Chrome/jssip as a
result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX
ICE candidates. This also exposes an ICE component enumeration to
extract further details from candidates.
(closes issue ASTERISK-21383)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/2967/
Matthew Jordan [Thu, 31 Oct 2013 15:59:50 +0000 (15:59 +0000)]
core/loader: Don't call dlclose in a while loop
For awhile now, we've noticed continuous integration builds hanging on CentOS 6
64-bit build agents. After resolving a number of problems with symbols, strange
locks, and other shenanigans, the problem has persisted. In all cases, gdb
shows the Asterisk process stuck in loader.c on one of the infinite while loops
that calls dlclose repeatedly until success.
The documentation of dlclose states that it returns 0 on success; any other
value on error. It does not state that repeatedly calling it will eventually
clear those errors. Most likely, the repeated calls to dlclose was to force a
close by exhausting the references on the library; however, that will never
succeed if:
(a) There is some fundamental error at work in the loaded library that
precludes unloading it
(b) Some other loaded module is referencing a symbol in the currently loaded
module
This results in Asterisk sitting forever.
Since we have matching pairs of dlopen/dlclose, this patch opts to only call
dlclose once, and log out as an ERROR if dlclose fails to return success. If
nothing else, this might help to determine why on the CentOS 6 64-bit build agent
things are not closing successfully.
Rusty Newton [Tue, 29 Oct 2013 23:42:45 +0000 (23:42 +0000)]
Updates for 1.4.25 core sounds and 1.4.14 extra sounds, plus new en_GB language set
The new sound packages relate to issues: ASTERISK-22544, ASTERISK-22411, ASTERISK-21413, ASTERISK-20782
Modified sounds/Makefile for the new sound versions and to account for the new en_GB language set.
Matthew Jordan [Tue, 29 Oct 2013 12:49:53 +0000 (12:49 +0000)]
Remove some spammy debug messages; improve clarity of others
Debug messages aren't free. Even when the debug level is sufficiently low such
that the messages are never evaluated, there is a cost to having to parse
Asterisk logs that contain debug messages that (a) fail to convey sufficient
information or (b) occur so frequently as to be next to meaningless. Based on
having to stare at lots of DEBUG messages, this patch makes the following
changes:
* channel.c: When copying variables from a parent channel to a child channel,
specify the channels involved. Do not log anything for a variable that is not
inherited; the fact that it doesn't have an _ or __ already signifies that it
won't be inherited.
* pbx.c: Specify what function evaluation has occurred that created the result.
* translate.c: Bump up the translator path messages to 10. I've never once had
to use these debug messages, and for each format that is registered (on
startup) and unregistered (on shutdown) the entire f^2 matrix is logged out.
For short tests in the Asterisk Test Suite, this should make finding the
actual test much easier.
* xmldoc.c: The debug message that 'blah' is not found in the tree is expected.
Often, description elements - which are not required - are not provided.
This debug message adds no additional value, as it is not indicative of an
error or helpful in debugging which element did not contain a 'blah' element
as a child. If an element is supposed to contain a child element, then that
XML tree should have failed validation in the first place.
Michael L. Young [Mon, 28 Oct 2013 14:50:12 +0000 (14:50 +0000)]
chan_sip: Clarify 'Forcerport' Setting Displayed When Running "sip show peers"
While looking at ASTERISK-22236, Walter Doekes pointed out that when running
"sip show peers", the setting being displayed can be confusing. The display of
"N" used to mean NAT (i.e. yes). The NAT setting has gone through many
different changes resulting in the display of different characters to try and
convey what the current setting is for 'Forcerport' (A for Auto and Forcerport
is currently on, a for Auto but Forcerport is off, Y for yes, and N for no).
During the initial code review to try and clarify these settings (especially
since "N" no longer meant what it used to mean in prior versions of Asterisk),
Mark Michelson suggested using the full space available to display the settings
which helped to make the settings very clear. That was a great suggestion.
Therefore, this patch does the following:
* The column for 'Forcerport' now will show: Auto (Yes), Auto (No), Yes, or No.
* A column for the 'Comedia' setting has been added. It too will display the
setting in a non-cryptic way: Auto (Yes), Auto (No), Yes, or No.
* UPGRADE.txt has been updated to document this change.
(closes issue ASTERISK-22728)
Reported by: Walter Doekes
Tested by: Michael L. Young
Patches:
asterisk-forcerport-display-clarification_v3.diff
uploaded by Michael L. Young (license 5026)
rtp_engine: fix rtp payloads copy and improve argument names
In function ast_rtp_instance_early _bridge_make_compatible the
use of instance 0/1 as arguments doesn't clearly communicate a
direction that the copying of payloads from the source channel
to the destination channel will occur, making it more probable
to have the arguments to ast_rtp_codecs_payloads_copy() put in
the reverse order. This patch renames the arguments with _dst
and _src suffixes and corrects the copy direction.
(closes issue ASTERISK-21464)
Reported by: Kevin Stewart
Review: https://reviewboard.asterisk.org/r/2894/
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Merged revisions 402000 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Test shows rtpmap:119 being copied per this change, but is not in sip invite
pbx.c: fix confused match caller id that deleted exten still in hash
This fixes a bug where a zero length callerid match adjacent to a no
match callerid extension entry would be deleted together, which then
resulted in hashtable references to free'd memory. A third state of
the matchcid value has been added to indicate match to any extension
which allows enforcing comparison of matchcid on/off without errors.
Kevin Harwell [Fri, 25 Oct 2013 16:05:55 +0000 (16:05 +0000)]
chan_sip: Allow a sip peer to accept both AVP and AVPF calls
Adapts the behaviour of avpf to only impact the format of outgoing calls. For
inbound calls, both AVP and AVPF calls will be accepted regardless of the value
of avpf in the configuration.
Kevin Harwell [Thu, 24 Oct 2013 20:44:09 +0000 (20:44 +0000)]
Logging: Logging types ignored after specifying a verbose level
If one specified a verbose level within a logging facility in
logger.conf then any component after it was ignored. Fixed so
all values are correctly read.
(closes issue ASTERISK-22456)
Reported by: Kevin Harwell
projects / thirdparty / asterisk.git / log
