Documentation for how to add custom headers/content to notifies created
with the PJSIPNotify manager action was a little sparse and it also
wasn't vetting application of Content-length headers like its chan_sip
equivalent was (so two Content-length headers could be applied... and
PJSIP determines the content length anyway, so it just opens people up
for error). This patch also flips the variable order so that the
variables are interpreted in the same order as they are put in the AMI
action.
Review: https://reviewboard.asterisk.org/r/3587/
........
Merged revisions 415658 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415659 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is a re-do of r414122.
When r414122 was merged, a major problem with it was uncovered. UNBRIDGE soft
hangup flags have a catastrophic effect on the pbx core if they leak out from
the bridge layer: the channel gets hung up. With the number of threads
involved in a blind transfer, and with the initial patch, it was likely that
this would occur. This caused a large number of test failures
This patch is nearly identical with the one proposed in r414122, save for the
following changes:
- We explicitly clear the UNBRIDGE flag when setting an after goto on a
channel in a bridge
- Defensively, if we encounter an UNBRIDGE flag in the pbx core, we handle it
https://reviewboard.asterisk.org/r/3585/
........
Merged revisions 415443 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415444 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If using the custom URI parsing code (not external uriparser lib) and there
was no query parameters the resulting pointer would be NULL and then an
attempt was made to subtract from it. The pointer is now set to a valid
value if there is no query parameter(s).
Also, in the 'ast_uri_make_host_with_port' function when setting the terminator
on the resulting string it was writing it one past the end of allocated memory.
It now writes the string terminator appropriately.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The twisted logic determining if a config file should be reloaded was
mostly broken and disabled. The incorrect test that ASTERISK-23383 fixed
actually reenabled the broken logic. The incorrect test was causing the
timestamp to always be cleared which caused config files with includes to
always be reloaded.
* Made wildcard includes always cause a reload. Determining if a file was
deleted cannot be determined without restructuring the cache to determine
if any files are missing from the last files actually loaded. Also
without refactoring config_text_file_load(), the glob loop couldn't check
more than one file for changes anyway.
* Made remove the cache entry if the file no longer exists when trying to
get its timestamp or it is no longer a regular file. This fixes the
corner case where the file was loaded, then deleted, then the config
reloaded, then the file restored with the same timestamp, and then the
config reloaded again.
* Made remove the cache entry include list when actually loading the file.
This gets rid of any stale includes the file had from the last time the
file was loaded.
ASTERISK-23683 #close
Reported by: tootai
Review: https://reviewboard.asterisk.org/r/3575/
........
Merged revisions 415225 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 415229 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 415230 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added a websocket server client in Asterisk. Asterisk has a websocket server,
but not a client. The ability to have Asterisk be able to connect to a websocket
server can potentially be useful for future work (for instance this could allow
ARI to connect back to some external system, although more work would be needed
in order to incorporate that).
Also a couple of things to note - proxy connection support has not been
implemented and there is limited http response code handling (basically, it is
connect or not).
Also added an initial new URI handling mechanism to core. Internet type URI's
are parsed into a data structure that contains pointers to the various parts of
the URI.
(closes issue ASTERISK-23742)
Reported by: Kevin Harwell
Review: https://reviewboard.asterisk.org/r/3541/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new channel function TALK_DETECT that, when set on a
channel, causes events indicating the start/stop of talking on a channel to be
emitted to both AMI and ARI clients.
The function allows setting both the silence threshold (the length of silence
after which we decide no one is talking) as well as the talking threshold (the
amount of energy that counts as talking). Parameters can be updated on a channel
after talk detection has been enabled, and talk detection can be removed at
any time.
The events raised by the function use a nomenclature similar to existing AMI/ARI
events.
For AMI: ChannelTalkingStart/ChannelTalkingStop
For ARI: ChannelTalkingStarted/ChannelTalkingFinished
Review: https://reviewboard.asterisk.org/r/3563/
#ASTERISK-23786 #close
Reported by: Matt Jordan
........
Merged revisions 414934 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses some aesthetic issues in Asterisk. These are all just
minor tweaks to improve the look of the CLI when used in a variety of
settings. Specifically:
* A number of chatty verbose messages were removed or demoted to DEBUG
messages. Verbose messages with a verbosity level of 5 or higher were -
if kept as verbose messages - demoted to level 4. Several messages
that were emitted at verbose level 3 were demoted to 4, as announcement
of dialplan applications being executed occur at level 3 (and so the
effects of those applications should generally be less).
* Some verbose messages that only appear when their respective 'debug'
options are enabled were bumped up to always be displayed.
* Prefix/timestamping of verbose messages were moved to the verboser
handlers. This was done to prevent duplication of prefixes when the
timestamp option (-T) is used with the CLI.
* Verbose magic is removed from messages before being emitted to
non-verboser handlers. This prevents the magic in multi-line verbose
messages (such as SIP debug traces or the output of DumpChan) from
being written to files.
* _Slightly_ better support for the "light background" option (-W) was
added. This includes using ast_term_quit in the output of XML
documentation help, as well as changing the "Asterisk Ready" prompt to
bright green on the default background (which stands a better chance of
being displayed properly than bright white).
Review: https://reviewboard.asterisk.org/r/3547/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a channel is destroyed (such as via ast_channel_release in off nominal
paths in core_unreal), it will attempt to free (via ast_free) the channel tech
pvt. This is problematic for a few reasons:
1. The channel tech pvt is an ao2 object in core_unreal. Free'ing the pvt
directly is no good.
2. The channel tech pvt's reference count is dropped just prior to calling
ast_channel_release, resulting in the pvt's destruction. Hence, the
channel destructor is free'ing an invalid pointer.
This patch keeps the dropping of the reference count, but sets the pvt to
NULL on the channel prior to releasing it. This models what would occur if the
channel was hung up directly.
........
Merged revisions 414542 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
User events can now be generated from ARI. Events can be signalled with
arbitrary json variables, and include one or more of channel, bridge, or
endpoint snapshots. An application must be specified which will receive
the event message (other applications can subscribe to it). The message
will also be delivered via AMI provided a channel is attached. Dialplan
generated user event messages are still transmitted via the channel, and
will only be received by a stasis application they are attached to or if
the channel is subscribed to.
This change also introduces the multi object blob mechanism used to send
multiple snapshot types in a single message. The dialplan app UserEvent
was also changed to use multi object blob, and a new stasis message type
created to handle them.
ASTERISK-22697 #close
Review: https://reviewboard.asterisk.org/r/3494/
........
Merged revisions 414405 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes res_corosync such that it works with Asterisk 12. This
restores the functionality that was present in previous versions of
Asterisk, and ensures compatibility with those versions by restoring the
binary message format needed to pass information from/to them.
The following changes were made in the core to support this:
* The event system has been partially restored. All event definition and
event types in this patch were pulled from Asterisk 11. Previously, we had
hoped that this information would live in res_corosync; however, the
approach in this patch seems to be better for a few reasons:
(1) Theoretically, ast_events can be used by any module as a binary
representation of a Stasis message. Given the structure of an ast_event
object, that information has to live in the core to be used universally.
For example, defining the payload of a device state ast_event in
res_corosync could result in an incompatible device state representation
in another module.
(2) Much of this representation already lived in the core, and was not
easily extensible.
(3) The code already existed. :-)
* Stasis message types now have a message formatter that converts their
payload to an ast_event object.
* Stasis message forwarders now handle forwarding to themselves. Previously
this would result in an infinite recursive call. Now, this simply creates a
new forwarding object with no forwards set up (as it is the thing it is
forwarding to). This is advantageous for res_corosync, as returning NULL
would also imply an unrecoverable error. Returning a subscription in this
case allows for easier handling of message types that are published directly
to an aggregate topic that has forwarders.
Review: https://reviewboard.asterisk.org/r/3486/
ASTERISK-22912 #close
ASTERISK-22372 #close
........
Merged revisions 414330 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes issues with direct media bridges that occur after a blind
transfer. These issues were caught by the (currently failing)
pjsip/transfers/blind_transfer/caller_direct_media test.
The test currently fails primarily for two reasons:
(1) When Bob and Charlie (the transfer target and the transfer destination)
enter a bridge together, the framehook remains on the transfer target
channel until both channels are in the bridge. As it consumes voice frames,
the initial bridge type is a simple bridge. The framehook is removed when
both channels are in the bridge; however, this does not currently cause the
bridging framework to re-evaluate the bridge. This patch adds a
AST_SOFTHANGUP_UNBRIDGE poke to the transfer target channel when a
framehook is removed so the bridge can re-evaluate itself.
(2) When a channel leaves a native RTP bridge, it may be leaving due to being
hung up. Sending a re-INVITE to a channel that is about to be hung up is
not nice - in fact, there's a good chance we'll send the BYE request before
the channel has had a chance to send back a 200 OK. To be somewhat nicer,
this patch adds a function to channel.h that allows the bridging framework
to query for exactly why a channel is leaving a bridge via the channel's
soft hangup flags. This allows it to only send the re-INVITE if there's a
chance the channel will survive the native bridging experience.
Review: https://reviewboard.asterisk.org/r/3535/
........
Merged revisions 414122 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the past framehooks have had no capability to determine what frame types a hook
is actually interested in consuming. This has meant that code has had to assume they
want all frames, thus preventing native bridging.
This change adds a callback which allows a framehook to be queried for whether it
is consuming a frame of a specific type. The native RTP bridging module has also
been updated to take advantange of this, allowing native bridging to occur when
previously it would not.
ASTERISK-23497 #comment Reported by: Etienne Lessard
ASTERISK-23497 #close
Review: https://reviewboard.asterisk.org/r/3522/
........
Merged revisions 413681 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the past framehooks have had no capability to determine what frame types a hook
is actually interested in consuming. This has meant that code has had to assume they
want all frames, thus preventing native bridging.
This change adds a callback which allows a framehook to be queried for whether it
is consuming a frame of a specific type. The native RTP bridging module has also
been updated to take advantange of this, allowing native bridging to occur when
previously it would not.
ASTERISK-23497 #comment Reported by: Etienne Lessard
ASTERISK-23497 #close
Review: https://reviewboard.asterisk.org/r/3522/
........
Merged revisions 413650 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed early exit in sip_msg_send() not destroying the message iterator.
* Made ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy()
tolerant of a NULL iter parameter in case ast_msg_var_iterator_init()
fails.
* Made ast_msg_var_iterator_destroy() clean up any current message data
ref.
* Made struct ast_msg_var_iterator, ast_msg_var_iterator_init(),
ast_msg_var_iterator_next(), ast_msg_var_unref_current(), and
ast_msg_var_iterator_destroy() use iter instead of i.
* Eliminated RAII_VAR usage in res_pjsip_messaging.c:vars_to_headers().
........
Merged revisions 413139 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 413142 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When bridge locking was added for bridge snapshot creation, some
locations where bridge locking was added were not guaranteed to
actually have a bridge and locking NULL AO2 objects tends to cause
segfaults. This ensures that NULL bridges aren't locked.
........
Merged revisions 413073 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Backport -r411687 and fix the fix because content_length is the length of
out plus the length of the file controlled by fd.
When a response has an out content length of 0, fwrite would be called to
write a buffer with no data in it. This resulted in the following classic
error message:
[Apr 3 11:49:17] ERROR[26421] http.c: fwrite() failed: Success
This patch makes it so that we only attempt to write the content of out if
the out string is non-zero.
........
Merged revisions 412922 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 412923 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 412924 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For some odd reason, loading app_mixmonitor was fine, but res_monitor was not.
This patch fixes a set of issues related to func_periodic_hook exporting the
beep functions that gets res_monitor working again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds the TCP_NODELAY option to accepted connections on the HTTP
server built into Asterisk. This option disables the Nagle algorithm
which controls queueing of outbound data and in some cases can cause
delays on receipt of response by the client due to how the Nagle
algorithm interacts with TCP delayed ACK. This option is already set on
all non-HTTP AMI connections and this change would cover standard HTTP
requests, manager HTTP connections, and ARI HTTP requests and
websockets in Asterisk 12+ along with any future use of the HTTP
server.
Review: https://reviewboard.asterisk.org/r/3466/
........
Merged revisions 412745 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 412748 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 412749 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When ASTERISK-23265/ASTERISK-23320 was fixed, it inadvertently led to realtime
features breaking. This was due to features loading prior to realtime. This
patch fixes this by loading features after loading dynamic modules.
ASTERISK-23487 #close
Reported by: Denis
Tested by: Denis
........
Merged revisions 412698 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Restore the reason value set by pbx_outgoing_attempt() to use
AST_CONTROL_xxx values as all the consumers were expecting rather than
cause codes.
* Fixed the dial routines to set cause codes for more than just
ast_request() so pbx_outgoing_attempt() reason codes will function.
* Fix inconsistent locked_channel return status in pbx_outgoing_attempt().
The chanel may not have been locked or the channel may have been a stale
pointer.
* Fixed the OutgoingSpoolFailed channel to run dialplan whenever the
dialing fails for an originate exten and 1 < synchronous.
* Fix incorrect ast_cond_wait() usage in pbx_outgoing_attempt().
Indroduced by issue ASTERISK-22212 patch.
* Made struct pbx_outgoing use the ao2 lock instead of its own lock for
the cond wait mutex. No sense in having two locks associated with the
same struct when only one is needed.
Review: https://reviewboard.asterisk.org/r/3421/
........
Merged revisions 412581 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed ast_channel_publish_dial_forward() not locking the forwarded
channel when taking the channel snapshot.
* Fixed app_dial.c:do_forward() using the wrong channel to get the
original call forwarding string.
* Removed unnecessary locking when calling ast_channel_publish_dial() and
ast_channel_publish_dial_forward() in app_dial and app_queue. Holding
channel locks when calling ast_channel_publish_dial_forward() with a
forwarded channel could result in pausing the system while the stasis bus
completes processsing a forwarded channel subscription.
Review: https://reviewboard.asterisk.org/r/3451/
........
Merged revisions 412579 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds a tones URI type to the playback resource. The tone can be specified by
name (from indications.conf) or by a tone pattern. In addition, tonezone can
be specified in the URI (by appending ;tonezone=<zone>). Tones must be
stopped manually in order for a stasis control to move on from playback of
the tone. Tones may be paused, resumed, restarted, and stopped. They may
not be rewound or fast forwarded (tones can't be controlled in a way that
lets you skip around from note to note and pausing and resuming will also
restart the tone from the beginning). Tests are currently in development
for this feature (https://reviewboard.asterisk.org/r/3428/).
(closes issue ASTERISK-23433)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3427/
........
Merged revisions 412535 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Remove unused RAII_VAR() declarations. The compiler cannot catch these
because the cleanup function "references" the unused variable. Some
actually allocated and released resources that were never used.
* Fixed some whitespace issues in stasis_bridges.c.
........
Merged revisions 412399 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The failing assertion ensures that the final snapshot gets generated so
CDR records can get finalized. The only place where a channel staging
snapshot flag could be left set is in chan_sip.c:handle_request_bye().
The function could return before clearing the flag because the channel
could dissappear while the function had to have the channel unlocked.
* Fixed handle_request_bye() channel snapshot staging coverage area to not
have a return in the middle of it and be unable to clear the staging flag.
* Pushed the channel snapshot staging coverage area into
ast_rtp_instance_set_stats_vars() to ensure that the staging is not
interrutped.
* Made callers of ast_rtp_instance_set_stats_vars() not call it with any
channels or channel driver private locks held to eliminate the deadlock
potential. The callers must hold references to the passed in channel and
rtp objects.
* Eliminated sip_hangup() trying to get the bridge peer. It is futile at
this point because the channel could never be in a bridge.
Review: https://reviewboard.asterisk.org/r/3431/
........
Merged revisions 412385 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
(1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables
REF_DEBUG globally throughout Asterisk.
(2) The ref debug log file is now created in the AST_LOG_DIR directory.
Every run will now blow away the previous run (as large ref files
sometimes caused issues). We now also no longer open/close the file
on each write, instead relying on fflush to make sure data gets written
to the file (in case the ao2 call being performed is about to cause a
crash)
(3) It goes with a comma delineated format for the ref debug file. This
makes parsing much easier. This also now includes the thread ID of the
thread that caused ref change.
(4) A new python script instead for refcounting has been added in the
contrib/scripts folder.
(5) The old refcounter implementation in utils/ has been removed.
Review: https://reviewboard.asterisk.org/r/3377/
........
Merged revisions 412114 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 412115 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 412153 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The masquerade supertest frequently fails because either the local channel
chain doesn't completely optimize out or the DTMF handshake doesn't
completely get accross. Local channel optimization requires frames
flowing to trigger when optimization can happen. When optimization
happens the media frame that triggered the optimization is dropped.
Sending DTMF requires frames to flow in the other direction for timing
purposes while sending nothing. If internal timing is not enabled when
MOH is playing, Asterisk switches to received timing when an audio frame
is received. With optimization dropping media frames and MOH not sending
frames unless it receives frames, occasionaly there are no more frames
being passed and the test fails.
* The asterisk command line -I option and the asterisk.conf
internal_timing option are removed. Asterisk now always uses internal
timing when needed if any timing module is loaded. The issue
ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken
if other internal timing modules besides DAHDI are used. The
ast_read_generator_actions() now only does received timing if it has no
choice for frame generators like MOH, silence, and playback streaming.
* Cleaned up some code dealing with frame generators in
ast_deactivate_generator(), generator_write_format_change(),
ast_activate_generator(), and ast_channel_stop_silence_generator().
* Removed ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and
ast_opt_internal_timing.
ASTERISK-22846 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3414/
........
Merged revisions 411715 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 411716 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 411717 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Assert if a channel is destroyed but has the snapshot staging flag set.
In this case the final channel destruction snapshot would never get taken.
* Assert if what we just got out of the stasis cache is not what we were
looking for. This assert would have saved several days searching for a
bug and a lot of my hair.
* Assert if the music on hold message posts could not find the associated
channel. A crash will happen later when manager tries to send the MOH AMI
message. This assert catches the problem when the stasis message is
posted instead of by the thread processing the defective message.
* Always generate a backtrace when an ast_assert() fails.
Review: https://reviewboard.asterisk.org/r/3411/
........
Merged revisions 411701 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411702 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a response has a content length of 0, fwrite would be called to write a
buffer with no data in it. This resulted in the following classic error
message:
[Apr 3 11:49:17] ERROR[26421] http.c: fwrite() failed: Success
This patch makes it so that we only attempt to write out the content if the
calculated content_length is non-zero.
........
Merged revisions 411687 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit contains several changes to sorcery:
1) Application of sorcery configuration based on module name is automatically performed
when sorcery is opened for a module.
2) Sorcery will not attempt to apply the same wizard to an object type more than once.
3) Sorcery gives more exact results when attempting to apply a wizard, whether as the
default or based on configuration.
Sorcery unit tests still pass for me after making these changes.
Review: https://reviewboard.asterisk.org/r/3326
........
Merged revisions 411159 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch works around a problem with the HTTP body
being dropped from the response to a specific client
and under specific circumstances:
a) Client request comes from node.js user agent
"Shred" via use of swagger-client library.
b) Asterisk and Client are *not* on the same
host or TCP/IP stack
In testing this problem, it has been determined that
the write of the HTTP body is lost, even if the data
is written using low level write function. The only
solution found is to instruct the TCP stack with the
shutdown function to flush the last write and finish
the transmission. See review for more details.
ASTERISK-23548 #close
(closes issue ASTERISK-23548)
Reported by: Sam Galarneau
Review: https://reviewboard.asterisk.org/r/3402/
........
Merged revisions 411462 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 411463 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 411465 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On graceful shutdown, sorcery wizards are all killed off, but it is
possible for sorcery instances to still have dangling pointers after
this, possibly causing a crash. Giving the sorcery instances a reference
to their wizards ensures that the wizard reference will remain valid for
the lifetime of the sorcery instance.
Review: https://reviewboard.asterisk.org/r/3401
........
Merged revisions 411295 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix memory leak in ast_unreal_new_channels(). Made it generate the ;2
uniqueid on a stack variable instead of mallocing it.
* Made send error response to ARI and AMI requests instead of just logging
excessive uniqueid length and allowing truncation. action_originate() and
ari_channels_handle_originate_with_id().
* Fixed minor truncating uniqueid hole when generating the ;2 uniqueid
string length. Created public and internal lengths of uniqueid. The
internal length can handle a max public uniqueid plus an appended ;2.
* free() and ast_free() are NULL tolerant so they don't need a NULL test
before calling.
* Made use better struct initialization format instead of the position
dependent initialization format. Also anything not explicitly initialized
in the struct is initialized to zero by the compiler.
* Made ast_channel_internal_set_fake_ids() use the safer
ast_copy_string() instead of strncpy().
Review: https://reviewboard.asterisk.org/r/3371/
........
Merged revisions 410949 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a request was received with a Content-type of json,
the body was sent for json parsing - even if it was zero
length. This resulted in ARI requests failing that were
valid, such as a channel DELETE with no parameters. The
code has now been changed to skip json parsing with zero
content length.
(closes issue SWP-6748)
Reported by: Samuel Galarneau
Review: https://reviewboard.asterisk.org/r/3360/
........
Merged revisions 410858 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the CDR core, every channel should either be filtered out (due to being an
'internal' channel used as an implementation detail, such as playing media
back into a bridge) or it should get a CDR. Even if that CDR ends up being
discarded, we still give the channel a CDR in case we end up needing it. If we
hit a situation where a channel does not have a CDR, we should blow up in
-dev-mode. Asserts are appropriate for that.
This patch adds those asserts, as they would have quickly caught the error
fixed by r410814.
........
Merged revisions 410861 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change enables DNS client support within PJSIP. System
nameservers are automatically discovered using res_init or
res_ninit. If this fails then PJSIP will resort to using
gethostbyname for resolution.
By enabling this support we gain SRV support, failover, and
weight support.
(closes issue ASTERISK-23435)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3343/
........
Merged revisions 410795 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410796 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Callerid checksum-ing was being handled incorrectly here. When the checksum is
calculated to be 0x00, it will perform 0x100-0x00 which results in 0x100. This
value will then fail the otherwise correct callerid message.
This patch changes the logic to simply add the calculated checksum to the
transmitted 2's compliment checksum.
Review: https://reviewboard.asterisk.org/r/3356/
(closes issue ASTERISK-23488)
........
This is a merge of merged revisions 410750 410747 from http://svn.asterisk.org/svn/asterisk/branches/12
I didn't want a broken patch to be comitted to trunk so I pre-merge merged them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Playing back a file to a channel in an ARI bridge would attempt to wait until
the playback concluded before returning. The method used involved signaling the
waiting thread in the ARI custom playback function.
The problem with this is that there were some corner cases that were not accounted for:
* If a bridge channel could not be found, then we never would attempt the playback but
would still attempt to wait for the playback to complete.
* If the bridge playfile action failed to queue, we would still attempt to wait for the
playback to complete.
* If the bridge playfile action were queued but some circumstance caused the playback
not to occur (the bridge dies, the channel is removed from the bridge), then we would
never be notified.
The solution to this is to move the waiting logic into the bridge code. A new bridge
API function is added to queue a synchronous action on a bridge. The waiting thread
is notified when the queued frame has been freed, either due to an error occurring
or due to successful playback. As a failsafe, the waiting thread has a 10 minute
timeout just in case there is a frame leak somewhere.
Review: https://reviewboard.asterisk.org/r/3338
........
Merged revisions 410673 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Realtime backends' update and store callbacks return the number of rows affected,
or -1 if there was a failure. There were a couple of issues:
* The config API was treating 0 as a successful return, and positive values as
a failure. Now the config API treats anything >= 0 as a success.
* res_sorcery_realtime was treating 0 as a successful return from the store
procedure, and any positive values as a failure. Now sorcery treats anything
> 0 as a success. It still considers 0 a "failure" since there is no change
to report to observers.
Review: https://reviewboard.asterisk.org/r/3341
........
Merged revisions 410592 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The syncing thread sleeps for a second before waiting to be
told to attempt to sync again. If a signal were sent during this
sleeping period, we would end up having to wait until the next
sync signal occurred in order to sync up the astdb.
This code rearrangement also ensures that any pending transactions
will be synced prior to Asterisk shutting down.
Patches: db_sync.patch by John Hardin (License #6512)
........
Merged revisions 410556 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 410559 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made res_musiconhold.c always post the MusicOnHoldStart/MusicOnHoldStop
events when it actually starts/stops the music streams. This allows the
events to always happen when MOH starts/stops. The event posting code was
moved to the MOH alloc/release routines.
* Made channel_do_masquerade() stop any MOH on the original channel before
masquerading so the original channel will get a stop event with correct
information.
* Cleaned up a couple odd codings in moh_files_alloc() and moh_alloc()
dealing with the music state variable.
(issue ASTERISK-23311)
Reported by: Benjamin Keith Ford
Review: https://reviewboard.asterisk.org/r/3306/
........
Merged revisions 410493 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Created the 'pjsip show channel' and 'pjsip show contact' commands.
Refactored out the hated ast_hashtab. Replaced with ao2_container.
Cleaned up function naming. Internal only, no public name changes.
Cleaned up whitespace and brace formatting in cli code.
Changed some NULL checking from "if"s to ast_asserts.
Fixed some register/unregister ordering to reduce deadlock potential.
Fixed ast_sip_location_add_contact where the 'name' buffer was too short.
Fixed some self-assignment issues in res_pjsip_outbound_registration.
(closes issue ASTERISK-23276)
Review: http://reviewboard.asterisk.org/r/3283/
........
Merged revisions 410287 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The config option help information has always parsed the <see-also> tags in the
XML documentation. Unfortunately, it just never bothered displaying them on
the CLI. With this patch, when you execute 'config show help [module] [obj]
[option]', it will display what other options are useful to you.
(closes issue ASTERISK-22008)
Reported by: Richard Mudgett
........
Merged revisions 410209 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In order to prevent confusion over the allow and disallow
list of codecs being the same an option for registering a
field as an alias is added. The alias field will be read
from the configuration file, but afterwards is not listed
as a known field. With disallow set as an alias, the CLI
command pjsip show endpoint # will list the allow= field,
but not the disallow field.
(closes issue ASTERISK-23092)
Review: https://reviewboard.asterisk.org/r/3193/
........
Merged revisions 410190 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A stasis cache entry now contains more than a single message/snapshot. It
contains messages/snapshots for the local entity as well as any remote
entities that post to the cached item. In addition callbacks can be
supplied when the cache is created to compute and post the aggregate
message/snapshot representing all entities stored in the cache entry.
* All stasis messages now have an eid to indicate what entity posted it.
* The stasis cache enhancements allow device state to cache and aggregate
the device states from local and remote entities in a single operation.
The cached aggregate device state is available immediately after it is
posted to the stasis bus. This improves performance by eliminating a
cache dump and associated ao2 container traversals to calculate the
aggregate state.
(closes issue ASTERISK-23204)
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3281/
........
Merged revisions 410184 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Much needed was a way to assign id to objects on creation, and
much change was necessary to accomplish it. Channel uniqueids
and linkedids are split into separate string and creation time
components without breaking linkedid propgation. This allowed
the uniqueid to be specified by the user interface - and those
values are now carried through to channel creation, adding the
assignedids value to every function in the chain including the
channel drivers. For local channels, the second channel can be
specified or left to default to a ;2 suffix of first. In ARI,
bridge, playback, and snoop objects can also be created with a
specified uniqueid.
Along the way, the args order to allocating channels was fixed
in chan_mgcp and chan_gtalk, and linkedid is no longer lost as
masquerade occurs.
(closes issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3191/
........
Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch creates the AST_SORCERY dialplan function which allows someone to
retrieve any value from a sorcery-based config file. It's similar to
AST_CONFIG.
The creation of the function itself was fairly straightforward but it required
changes to the underlying sorcery infrastructure that rippled into individual
sorcery objects. The changes stemmed from inconsistencies in how sorcery
created ast_variable objectsets from sorcery objects and the inconsistency
in how individual objects used that feature especially when it came to
parameters that can be specified multiple times like contact in aor and match
in identify. You can read more here...
http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html
So, what this patch does, besides actually creating the AST_SORCERY function,
is the following...
* Creates ast_variable_list_append which is a helper to append one ast_variable
list to another.
* Modifies the ast_sorcery_object_field_register functions to accept the
already-defined sorcery_fields_handler callback.
* Modifies ast_sorcery_objectset_create to accept a parameter indicating return
type preference...a single ast_variable with all values concatenated or an
ast_variable list with multiple entries. Also fixed a few bugs.
* Modifies individual sorcery object implementations to use the new function
definition of the ast_sorcery_object_field_register functions.
* Modifies location.c and res_pjsip_endpoint_identifier_ip.c to implement
sorcery_fields_handler handlers so they return multiple occurrences as an
ast_variable_list.
* Added a whole bunch of tests to test_sorcery.
(closes issue ASTERISK-22537)
Review: http://reviewboard.asterisk.org/r/3254/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Transport TOS values were interpreted as DSCP values without being documented
as such. Endpoint TOS values (tos_audio/tos_video) behaved normally as TOS
values have historically. This patch makes the transport TOS values behave as
TOS values and makes all TOS values readable as string values (e.g. AF11).
In addition, alembic scripts have been updated to use the proper field types
for all TOS/COS values.
(issue ASTERISK-23235)
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3304/
........
Merged revisions 410028 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Remove some unnecessary RAII_VAR() usage.
* Made the struct stasis_subscription ao2 object use the ao2 lock instead
of a redundant join_lock in the struct for ast_cond_wait().
* Removed locks on some ao2 objects that don't need the lock.
* Made the topic pool entries container use the ao2 template functions.
* Add some missing allocation failure checks.
* Add missing cleanup in off nominal path of dispatch_message().
........
Merged revisions 409270 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is possible to pre-load pbx_config. As a result, pbx_config - which will
load and parse the dialplan - will attempt to use various dialplan components,
such as device state providers and presence state providers, prior to them
being initialized by the core. This would lead to a crash, as the components
had not created their Stasis cache entries.
This patch moves a number of core component initializations before the module
pre-load. This guarantees that if someone does pre-load pbx_config - or other
pbx modules - that the Stasis caches for the various core components are
created.
(closes issue ASTERISK-23320)
Reported by: xrobau
(closes issue ASTERISK-23265)
Reported by: Andrew Nagy
Tested by: Andrew Nagy, Rusty Newton
........
Merged revisions 408855 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk didn't support the dynamic payload change in rtp mapping in the 200
OK response.
Scenario:
Asterisk sends the INVITE proposing alaw and telephone-event, it proposes
rtpmap:101 for telephone-event. Peer responds with 2xx, it answers with
alaw and telephone-event also, but it proposes a different rtpmap number
(rtpmap:103) for telephone-event.
Expected Behaviour:
Asterisk should honour the rtpmapping in the response and send DTMF packets
using 103 as payload type for DTMF.
Actual Behaviour: Asterisk sends DTMF packets using payload type 101.
With this patch asterisk now supports changes that can occur in the rtp mapping
in the response.
(closes issue ASTERISK-23279)
Reported by: NITESH BANSAL
Review: https://reviewboard.asterisk.org/r/3225/
Patches:
dynamic_payload_change.patch uploaded by nbansal (license 6418)
........
Merged revisions 408729 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 408730 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fixed off-nominal json ref counting issue with using the following API
calls: ast_json_object_set() and ast_json_array_append().
* Fixed off-nominal error reporting in ast_ari_endpoints_list().
* Fixed some miscellaneous off-nominal json ref counting issues in
report_receive_fax_status() and dial_to_json().
........
Merged revisions 408713 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Updated the code to check to see if MOH is playing on the transferor and if
so then start it on the channel that replaces it during a masquerade.
Example scenario of the problem:
Alice calls Bob and then Bob begins the attended transfer process into a queue.
Upon going on hold Alice hears music and so does Bob once he is in the queue.
Bob then transfers Alice into the queue and then music for Alice stops even
though she should be hearing it since has now replaced Bob in the queue.
The problem that was occurring is that once the channel was masqueraded the app
(queues, confbridge, etc...) had no way of knowing that the channel had just
been swapped out thus it did not start music for the present channel.
Credit to Olle Johansson for pointing me in the right direction on this issue.
(closes issue ASTERISK-19499)
Reported by: Timo Teräs
Review: https://reviewboard.asterisk.org/r/3226/
........
Merged revisions 408642 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 408643 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 408644 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In order to retrieve an arbitrary sorcery instance from a dialplan function
(or any place else) there needs to be a registry of sorcery instances.
ast_sorcery_init now creates a hashtab as a registry.
ast_sorcery_open now checks the hashtab for an existing sorcery instance
matching the caller's module name. If it finds one, it bumps the
refcount and returns it. If not, it creates a new sorcery instance,
adds it to the hashtab, then returns it.
ast_sorcery_retrieve_by_module_name is a new function that does a hashtab
lookup by module name. It can be called by the future dialplan function.
res_pjsip/config_system needed a small change to share the main res_pjsip
sorcery instance.
tests/test_sorcery was updated to include a test for the registry.
(closes issue ASTERISK-22537)
Review: http://reviewboard.asterisk.org/r/3184/
........
Merged revisions 408518 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is highly unlikely, but - at least in Asterisk 12 - theoretically possible
to load Asterisk with no dialplan whatsoever. If that occurs, and some other
module (that is not a pbx module) attempts to merge its contexts into the
dialplan, the existing merge routine will crash. This is because it is not
insane, and rightly believes that you provided some sort of dialplan,
somewhere.
This patch will gracefully merge the contexts in such a case. Note that this
is highly unlikely to occur in 1.8/11, as features will most likely provide
some dialplan via parking. However, in Asterisk 12, parking is now provided
by res_parking, and hence may create its dialplan later.
(closes issue ASTERISK-23297)
Reported by: CJ Oster
Review: https://reviewboard.asterisk.org/r/3222
........
Merged revisions 408200 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 408201 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 408220 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
URI's are supposed to be case sensitive and all
lower case. In practice some portions of URI's
in ARI are case insensitive and others are not,
such as TECH, which in one instance would match
a lower case name and in another would not. In
this patch, the ast_endpoint_lastest_snapshot()
function is modified to change the TECH portion
to full upper case before lookup. This resolves
the discrepancy noted by the reporter. However
I chose to avoid forcing the /ari prefix of the
URI's to be lower case for now. Except for the
two cases here, all URI's should be lower case,
unless they are part of a resource name or id.
Review: https://reviewboard.asterisk.org/r/3211/
Reported by: Zane Conkle
(closes issue ASTERISK-23125)
........
Merged revisions 408140 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ast_format_sdp_parse and ast_format_sdp_generate
the check checks for a valid interface and function
were potentially confusing, and hid an error in the
test of the presence of the function that is called
later. This patch clears up and corrects the test.
Review: https://reviewboard.asterisk.org/r/3208/
(closes issue ASTERISK-23098)
Reported by: marcelloceschia
Patches:
main_format.patch uploaded by marcelloceschia (license 6036)
ASTERISK-23098.patch uploaded by coreyfarrell (license 5909)
........
Merged revisions 408137 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 408138 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The old code depended on undefined va_arg behaviour: calling a function
twice with the same va_list parameter and expecting it to continue where
it left off. The changed code behaves like the manpage says it should.
Also added a bunch of early returns to trap errors (e.g. OOM) instead of
crashing.
The problem was found by Julian Lyndon-Smith. The deviant behaviour on
the raspberry PI also uncovered another bug (fixed in r407875) in the
res_config_pgsql.so driver.
Reported by: jmls
Tested by: jmls
Review: https://reviewboard.asterisk.org/r/3201/
........
Merged revisions 407968 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a first stab at tweaking the performance profile of the scheduler. Removing
the hashtab usage removes an extra memory allocation when scheduling something and
makes it so rescheduling does not incur any memory allocation at all.
Review: https://reviewboard.asterisk.org/r/3199/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When formatting an optional IE, the value is, of course, optional. As such, it
is entirely appropriate for ast_json_object_get to return NULL. If that occurs,
we now simply skip the IE that was requested, as it was not provided by the
entity that raised the event.
Thanks to George Joseph (gtjoseph) for catching this and reporting it in
#asterisk-dev
........
Merged revisions 407750 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change allows timing implementation data to be stored directly
on the timer itself thus removing the requirement for many
implementations to do a container lookup for the same information.
This means that API calls into timing implementations can directly
access the information they need instead of having to find it.
Review: https://reviewboard.asterisk.org/r/3175/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The appdocsxml.dtd specifies that a "required" attribute in a parameter may
have a value of yes, no, true, or false. On some systems, specifying "False"
instead of "false" would cause a validation error. This patch fixes the casing
to explicitly match the DTD.
........
Merged revisions 407676 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch brings CDR processing further in line with r407085. During some dial
operations, the application would not be locked to the Dial application and
would instead continue to show the previously known application. In particular,
this would occur when a Parked call would time out. This was due to a previous
snapshot already locking the application to Park - processing this in a Dial
Begin allows the Dial application to reassert its rightful place.
(CDRs. Ugh.)
But hooray for the Parked Call tests for catching this in the Asterisk Test
Suite.
........
Merged revisions 407166 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes a number of small-ish problems that were noticed when
witnessing the records that the FreePBX dialplan produces:
(1) Mid-call events (as well as privacy options) have the ability to change the
overall state of the Dial operation after the called party answers. This
means that publishing the DialEnd event when the called party is premature;
we have to wait for the execution of these subroutines to complete before
we can signal the overall status of the DialEnd. This patch moves that
publication and adds handlers for the mid-call events.
(2) The AST_FLAG_OUTGOING channel flag is cleared if an after bridge goto
datastore is detected. This flag was preventing CDRs from being recorded
for all outbound channels that had a 'continue' option enabled on them by
the Dial application.
(3) The CDR engine now locks the 'Dial' application as being the CDR
application if it detects that the current CDR has entered that app. This
is similar to the logic that is done for Parking. In general, if we entered
into Dial, then we want that CDR to record the application as such - this
prevents pre-dial handlers, mid-call handlers, and other shenaniganry
from changing the application value.
(4) The CDR engine now checks for the AST_SOFTHANGUP_HANGUP_EXEC in more places
to determine if the channel is in hangup logic or dead. In either case, we
don't want to record changes in the channel.
(5) The default option for "endbeforehexten" has been changed to "yes". In
general, you don't want to see CDRs in the 'h' exten or in hangup logic.
Since the semantics of that option changed in 12, it made sense to update
the default value as well.
(6) Finally, because we now have the ability to synchronize on the messages
published to the CDR topic, on shutdown the CDR engine will now synchronize
to the messages currently in flight. This helps to ensure that all
in-flight CDRs are written before shutting down.
(closes issue ASTERISK-23164)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3154
........
Merged revisions 407084 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an 'ExtensionStatus' event was raised it included the status as a
numerical value, but did not include a text description of the status.
Added a 'StatusText' field to the event which is a string representation
of the extension status. Also added this to the 'Extension State' command
response.
(closes issue ASTERISK-23154)
Reported by: Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 cel_backends, cel_dialstatus_store, and cel_linkedids ao2
containers with a global ao2 object wrapper.
* Added NULL checks before use of the cel_backends, cel_dialstatus_store,
and cel_linkedids ao2 containers in case the CEL module is already
shutdown.
* Fixed overloading of the cel_linkedids held objects reference count.
During shutdown any held objects would be leaked.
* Fixed memory leak of cel_linkedids held objects if the LINKEDID_END is
not being tracked. The objects in the cel_linkedids container were not
removed if the LINKEDID_END event is not used.
* Added access protection to the cel_backends container during the CLI
"cel show status" command.
* Made cel_backends, cel_dialstatus_store, and cel_linkedids use the
standard ao2 callback templates for the hash and cmp functions.
* Eliminated unnecessary uses of RAII_VAR().
* Made ast_cel_engine_init() cleanup alocated resources on failure.
(closes issue AST-1253)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3128/
........
Merged revisions 406417 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 406418 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 406465 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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().
(closes issue AST-1242)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3144/
........
Merged revisions 406341 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 406342 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds improvements to support for allow=all in
pjsip.conf so that it functions as intended. Previously,
the allow/disallow socery configuration would set & clear
codecs from the media.codecs and media.prefs list, but if
all was specified the prefs list was not updated. Then a
call would fail when create_outgoing_sdp_stream() created
an SDP with no audio codecs.
A new function ast_codec_pref_append_all() is provided to
add all codecs to the prefs list - only those not already
on the list. This enables the configuration to specify a
codec preference, but still add all codecs, and even then
remove some codecs, as shown in this example:
allow = ulaw, alaw, all, !g729, !g723
Also, the display order of allow in cli output is updated
to match the configuration by using prefs instead of caps
when generating a human readable string.
Finally, a change to create_outgoing_sdp_stream() skips a
codec when it does not have a payload code instead of the
call failing.
(closes issue ASTERISK-23018)
Reported by: xrobau
Review: https://reviewboard.asterisk.org/r/3131/
........
Merged revisions 405875 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405876 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(closes issue AST-1252)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3114/
........
Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In https://reviewboard.asterisk.org/r/3057/, applications and functions that
manipulate CDRs were made to interact over Stasis. This was done to
synchronize manipulations of CDRs from the dialplan with the updates the
engine itself receives over the message bus.
This change rested on a faulty premise: that messages published to the CDR
topic or to a topic that forwards to the CDR topic are synchronized with the
messages handled by the CDR topic subscription in the CDR engine. This is not
the case. There is no ordering guaranteed for two messages published to the
same topic; ordering is only guaranteed if a message is published to the same
subscriber.
Stasis was modified in r405311 to allow a publisher to synchronize on the
subscriber. This patch uses that API to synchronize the CDR publishers with
the CDR engine message router, which maintains the overall topic subscription.
(closes issue ASTERISK-22884)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3099/
........
Merged revisions 405312 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds an API call to Stasis that allows a publisher to publish a
stasis message that will not return until a specific subscriber handles the
message. Since a subscriber can have their own forwarding topic which orders
messages from many topics, this allows a publisher who knows of that subscriber
to synchronize to that subscriber regardless of the forwarding relationships
between topics.
This is of particular use for dialplan applications that need to synchronize
on a particular subscriber's handling of a message.
(issue ASTERISK-22884)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3099/
........
Merged revisions 405311 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
AMI action UserEvent event response would include the action header in its
keyvalue pairs list. Adjusted the start of the header loop to skip over the
action part.
(closes issue ASTERISK-22899)
Reported by: outtolunc
Patches:
svn_manager.c.skip_action.diff.txt uploaded by outtolunc (license 5198)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk does not support any of the transfer encodings specified in
HTTP/1.1, other than the default "identity" encoding.
According to RFC 2616:
A server which receives an entity-body with a transfer-coding it does
not understand SHOULD return 501 (Unimplemented), and close the
connection. A server MUST NOT send transfer-codings to an HTTP/1.0
client.
This patch adds the 501 Unimplemented response, instead of the hard work
of actually implementing other recordings.
This behavior is especially problematic for Node.js clients, which use
chunked encoding by default.
(closes issue ASTERISK-22486)
Review: https://reviewboard.asterisk.org/r/3092/
........
Merged revisions 404565 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Implements the following cli commands:
pjsip list aors
pjsip list auths
pjsip list channels
pjsip list contacts
pjsip list endpoints
pjsip show aor(s)
pjsip show auth(s)
pjsip show channels
pjsip show endpoint(s)
Also...
Minor modifications made to the AMI command implementations to facilitate
reuse.
New function ast_variable_list_sort added to config.c and config.h to implement
variable list sorting.
(issue ASTERISK-22610)
patches:
pjsip_cli_v2.patch uploaded by george.joseph (License 6322)
........
Merged revisions 404480 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch updates the log messages to include descriptive
names for event types. This is an improvement over having
only cryptic type numbers.
(closes issue ASTERISK-22909)
Reported by: outtolunc
Review: https://reviewboard.asterisk.org/r/3081/
Patches:
svn_security_events.c.names.diff.txt uploaded by outtolunc (license 5198)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For the explanation, here is a copy-paste of the review board explanation:
Initially, it was discovered that performing an attended transfer of a
multiparty bridge with a PJSIP channel would cause a deadlock. A PBX thread
started a masquerade and reached the point where it was calling the fixup()
callback on the "original" channel. For chan_pjsip, this involves pushing a
synchronous task to the session's serializer. The problem was that a task ahead
of the fixup task was also attempting to perform a channel masquerade. However,
since masquerades are designed in a way to only allow for one to occur at a
time, the task ahead of the fixup could not continue until the masquerade
already in progress had completed. And of course, the masquerade in progress
could not complete until the task ahead of the fixup task had completed.
Deadlock.
The initial fix was to change the fixup task to be asynchronous. While this
prevented the deadlock from occurring, it had the frightful side effect of
potentially allowing for tasks in the session's serializer to operate on a
zombie channel.
Taking a step back from this particular deadlock, it became clear that the
problem was not really this one particular issue but that masquerades
themselves needed to be addressed. A PJSIP attended transfer operation calls
ast_channel_move(), which attempts to both set up and execute a masquerade. The
problem was that after it had set up the masquerade, the PBX thread had swooped
in and tried to actually perform the masquerade. Looking at changes that had
been made to Asterisk 12, it became clear that there never is any time now that
anyone ever wants to set up a masquerade and allow for the channel thread to
actually perform the masquerade. Everyone always is calling ast_channel_move(),
performs the masquerade itself before returning.
In this patch, I have removed all blocks of code from channel.c that will
attempt to perform a masquerade if ast_channel_masq() returns true. Now, there
is no distinction between setting up a masquerade and performing the
masquerade. It is one operation. The only remaining checks for
ast_channel_masq() and ast_channel_masqr() are in ast_hangup() since we do not
want to interrupt a masquerade by hanging up the channel. Instead, now
ast_hangup() will wait for a masquerade to complete before moving forward with
its operation.
The ast_channel_move() function has been modified to basically in-line the
logic that used to be in ast_channel_masquerade(). ast_channel_masquerade() has
been killed off for real. ast_channel_move() now has a lock associated with it
that is used to prevent any simultaneous moves from occurring at once. This
means there is no need to make sure that ast_channel_masq() or
ast_channel_masqr() are already set on a channel when ast_channel_move() is
called. It also means the channel container lock is not pulling double duty by
both keeping the container locked and preventing multiple masquerades from
occurring simultaneously.
The ast_do_masquerade() function has been renamed to do_channel_masquerade()
and is now internal to channel.c. The function now takes explicit arguments of
which channels are involved in the masquerade instead of a single channel.
While it probably is possible to do some further refactoring of this method, I
feel that I would be treading dangerously. Instead, all I did was change some
comments that no longer are true after this changeset.
The other more minor change introduced in this patch is to res_pjsip.c to make
ast_sip_push_task_synchronous() run the task in-place if we are already a SIP
servant thread. This is related to this patch because even when we isolate the
channel masquerade to only running in the SIP servant thread, we would still
deadlock when the fixup() callback is reached since we would essentially be
waiting forever for ourselves to finish before actually running the fixup. This
makes it so the fixup is run without having to push a task into a serializer at
all.
(closes issue ASTERISK-22936)
Reported by Jonathan Rose
Review: https://reviewboard.asterisk.org/r/3069
........
Merged revisions 404356 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change is in preparation for external MWI support.
Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context. The
only exception is the legacy hasvoicemail users.conf option. The legacy
option will only work for app_voicemail mailboxes. The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.
chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier. chan_sip just stored and
compared the two components. chan_dahdi actually used the box
information.
The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number. As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.
Review: https://reviewboard.asterisk.org/r/3072/
........
Merged revisions 404348 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(closes issue AST-1265)
Reported by: Alexander Hömig
(closes issue ASTERISK-22350)
Reported by: Birger "WIMPy" Harzenetter
Review: https://reviewboard.asterisk.org/r/3078/
........
Merged revisions 404344 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 404345 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When doing the rework of the CDR engine that pushed all of the logic into cdr.c
and made it respond to changes in channel state over Stasis, we knew that
accessing the CDR engine from the dialplan would be "slightly"
non-deterministic. Dialplan threads would be accessing CDRs while Stasis
threads would be updating the state of said CDRs - whereas in the past,
everything happened on the dialplan threads. Tests have shown that "slightly"
is in reality "very".
This patch synchronizes things by making the dialplan applications/functions
that manipulate CDRs do so over Stasis. ForkCDR, NoCDR, ResetCDR, CDR, and
CDR_PROP now all use Stasis to send their requests over to the CDR engine,
and synchronize on the channel Stasis topic via a subscription so that they
return their values/control to the dialplan at the appropriate time.
While going through this, the following changes were also made:
* DISA, which can reset the CDR when a user successfully authenticates, now
just uses the ResetCDR app to do this. This prevents having to duplicate
the same Stasis synchronization logic in that application.
* Answer no longer disables CDRs. It actually didn't work anyway - calling
DISABLE on the channel's CDR doesn't stop the CDR from getting the Answer
time - it just kills all CDRs on that channel, which isn't what the caller
would intend.
(closes issue ASTERISK-22884)
(closes issue ASTERISK-22886)
Review: https://reviewboard.asterisk.org/r/3057/
........
Merged revisions 404294 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Original commit message by mmichelson (asterisk 12 r403311):
"This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such."
The above was initially committed and then reverted at r403398. The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed. Fixed by unreffing the channels.
Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel. Fixed by
unlocking "other->chan"
(closes issue ASTERISK-22709)
Reported by: John Bigelow
........
Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Bridges have two new optional properties, a creator and a name.
Certain consumers of bridges will automatically provide bridges that
they create with these properties. Examples include app_bridgewait,
res_parking, app_confbridge, and app_agent_pool. In addition, a name
may now be provided as an argument to the POST function for creating
new bridges via ARI.
(closes issue AFS-47)
Review: https://reviewboard.asterisk.org/r/3070/
........
Merged revisions 404042 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404043 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Framehooks can be used in a reactive manner to execute specific logic
when a frame is received with a certain type and payload. Since it is
possible for framehooks to provide frames it was possible for this
reactive framehook to be unaware of frames it is looking for.
This change makes it so that when framehooks return a modified frame
the code will now re-iterate (from the beginning) and call any
previous framehooks that have not provided a modified frame themselves.
Review: https://reviewboard.asterisk.org/r/3046/
........
Merged revisions 404027 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
Also, the ABI was changed to something more reasonable, since Asterisk
12 does not yet have a public release.
(closes issue ASTERISK-22905)
Review: http://reviewboard.digium.internal/r/432/
........
Merged revisions 403913 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 403917 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 403959 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change adds an event for when an originated call is redirected to
another target. This event contains the original channel and the newly
created channel. If a stasis subscription exists on the original originated
channel for a stasis application then a new subscription will also be
created on the stasis application to the redirected channel. This allows
the application to follow the call path completely.
(closes issue ASTERISK-22719)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/3054/
........
Merged revisions 403808 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There were still a few cases in which ATTENDEDTRANSFER and BLINDTRANSFER
wouldn't be set on channels involved with blind and attended transfers.
This would happen with features that were initialized by channel driver
specific mechanisms in multiparty calls. This patch resolves those cases
while attempted to keep the behavior for setting those variables as
consistent as possible.
(closes issue AFS-24)
Review: https://reviewboard.asterisk.org/r/3040/
........
Merged revisions 403781 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The change contains a slightly adjusted patch that was on the issue
(submitted by kmoore). A fix was made by adding in a bridge lock
while calling bridge_start/stop from the framehook callback. Since
the framehook callback is not called from the bridging core the bridge
is not locked, but needs to be before calling bridge_start.
(closes issue ASTERISK-22749)
Reported by: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/3066/
Patches:
lock_inversion.diff uploaded by kmoore (license 6273)
........
Merged revisions 403767 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added the ability to specify channel variables when creating/originating a
channel in ARI. The variables are sent in the body of the request and should
be formatted as a single level JSON object. No nested objects allowed.
For example: {"variable1": "foo", "variable2": "bar"}.
(closes issue ASTERISK-22872)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3052/
........
Merged revisions 403752 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* The voicemail registration/unregistration functions now take a struct of
callbacks instead of a lengthy parameter list of callbacks.
* The voicemail registration/unregistration functions now prevent a
competing module from interfering with an already registered callback
supplying module.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan
to use the CHANNEL function on a chan_pjsip channel to obtain run-time
information about the channel from the PJSIP channel driver and the PJSIP
stack. This includes:
* RTP information, including source/destination media addresses, whether or
not the media is secure, held, and other properties.
* RTCP information. This includes sets of parseable information, as well as
individual statistic attriutes.
* PJSIP information. This includes URIs, local/remote signalling addresses,
whether or not the signalling is secure, and other properties.
* The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT
function to obtain more detailed endpoint information.
Review: https://reviewboard.asterisk.org/r/3038/
........
Merged revisions 403618 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Eliminated shadowing of the __ast_sorcery_apply_config() name parameter
causing confusion.
* Fix potential crash from sorcery.conf user input in
__ast_sorcery_apply_config() if the user supplied a malformed config line
that is missing the sorcery object type name.
* Remove redundant test in __ast_sorcery_apply_config(). !config and
config == CONFIGS_STATUS_FILEMISSING are identical.
........
Merged revisions 403541 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403544 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The snapshot process for endpoints uses the channel ids present
on the endpoint itself. Without keeping a reference it was possible
for the strings to be freed underneath any consumer of an endpoint
snapshot.
A reference is now held by the snapshot to the channel ids and
released when the snapshot is destroyed.
(issue ASTERISK-22801)
Reported by: Matt Jordan
........
Merged revisions 403542 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Make ast_sorcery_observer_remove() accept a const callbacks struct.
* Make ast_sorcery_observer_remove() tolerant of the sorcery parameter
being NULL. Now it can be called within a module unload routine if the
sorcery initialization fails.
* Fix ast_sorcery_observer_add() to fail if the container link fails.
........
Merged revisions 403324 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
........
Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The patch allows ARI to parse request parameters from an incoming JSON
request body, instead of requiring the request to come in as query
parameters (which is just weird for POST and DELETE) or form
parameters (which is okay, but a bit asymmetric given that all of our
responses are JSON).
For any operation that does _not_ have a parameter defined of type
body (i.e. "paramType": "body" in the API declaration), if a request
provides a request body with a Content type of "application/json", the
provided JSON document is parsed and searched for parameters.
The expected fields in the provided JSON document should match the
query parameters defined for the operation. If the parameter has
'allowMultiple' set, then the field in the JSON document may
optionally be an array of values.
(closes issue ASTERISK-22685)
Review: https://reviewboard.asterisk.org/r/2994/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Created a data model and implemented functionality for an ARI device state
resource. The following operations have been added that allow a user to
manipulate an ARI controlled device:
Create/Change the state of an ARI controlled device
PUT /deviceStates/{deviceName}&{deviceState}
Retrieve all ARI controlled devices
GET /deviceStates
Retrieve the current state of a device
GET /deviceStates/{deviceName}
Destroy a device-state controlled by ARI
DELETE /deviceStates/{deviceName}
The ARI controlled device must begin with 'Stasis:'. An example controlled
device name would be Stasis:Example. A 'DeviceStateChanged' event has also
been added so that an application can subscribe and receive device change
events. Any device state, ARI controlled or not, can be subscribed to.
While adding the event, the underlying subscription control mechanism was
refactored so that all current and future resource subscriptions would be
the same. Each event resource must now register itself in order to be able
to properly handle [un]subscribes.
(issue ASTERISK-22838)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3025/
........
Merged revisions 403134 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Created the following AMI commands and corresponding events for res_pjsip:
PJSIPShowEndpoints - Provides a listing of all pjsip endpoints and a few
select attributes on each.
Events:
EndpointList - for each endpoint a few attributes.
EndpointlistComplete - after all endpoints have been listed.
PJSIPShowEndpoint - Provides a detail list of attributes for a specified
endpoint.
Events:
EndpointDetail - attributes on an endpoint.
AorDetail - raised for each AOR on an endpoint.
AuthDetail - raised for each associated inbound and outbound auth
TransportDetail - transport attributes.
IdentifyDetail - attributes for the identify object associated with
the endpoint.
EndpointDetailComplete - last event raised after all detail events.
PJSIPShowRegistrationsInbound - Provides a detail listing of all inbound
registrations.
Events:
InboundRegistrationDetail - inbound registration attributes for each
registration.
InboundRegistrationDetailComplete - raised after all detail records have
been listed.
PJSIPShowRegistrationsOutbound - Provides a detail listing of all outbound
registrations.
Events:
OutboundRegistrationDetail - outbound registration attributes for each
registration.
OutboundRegistrationDetailComplete - raised after all detail records
have been listed.
PJSIPShowSubscriptionsInbound - A detail listing of all inbound subscriptions
and their attributes.
Events:
SubscriptionDetail - on each subscription detailed attributes
SubscriptionDetailComplete - raised after all detail records have
been listed.
PJSIPShowSubscriptionsOutbound - A detail listing of all outboundbound
subscriptions and their attributes.
Events:
SubscriptionDetail - on each subscription detailed attributes
SubscriptionDetailComplete - raised after all detail records have
been listed.
(issue ASTERISK-22609)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2959/
........
Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change prevents channels used as implementation details from
leaking out to ARI. It does this by preventing creation of JSON blobs
of channel snapshots created from those channels and sanitizing JSON
blobs of bridge snapshots as they are created. This introduces a
framework for excluding information from output targeted at Stasis
applications on a consumer-by-consumer basis using channel sanitization
callbacks which could be extended to bridges or endpoints if necessary.
This prevents unhelpful error messages from being generated by
ast_json_pack.
This also corrects a bug where BridgeCreated events would not be
created.
(closes issue ASTERISK-22744)
Review: https://reviewboard.asterisk.org/r/2987/
Reported by: David M. Lee
........
Merged revisions 403069 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403070 65c4cc65-6c06-0410-ace0-fbb531ad65f3