asterisk/asterisk.git
4 years agoMerge "app_dial.c: Force COLP update if outgoing channel name changed."
Matt Jordan [Tue, 29 Sep 2015 12:26:20 +0000 (07:26 -0500)]
Merge "app_dial.c: Force COLP update if outgoing channel name changed."

4 years agoMerge "translate: Fix transcoding while different in frame size."
Matt Jordan [Mon, 28 Sep 2015 21:24:19 +0000 (16:24 -0500)]
Merge "translate: Fix transcoding while different in frame size."

4 years agoMerge "app_dial.c: Factor out a connected line update routine."
Joshua Colp [Mon, 28 Sep 2015 19:07:05 +0000 (14:07 -0500)]
Merge "app_dial.c: Factor out a connected line update routine."

4 years agoMerge "res/res_stasis: Fix accidental subscription to 'all' bridge topic"
Matt Jordan [Mon, 28 Sep 2015 13:36:09 +0000 (08:36 -0500)]
Merge "res/res_stasis: Fix accidental subscription to 'all' bridge topic"

4 years agoMerge "Scripts: check file versions of Asterisk and dependencies"
Joshua Colp [Mon, 28 Sep 2015 11:50:12 +0000 (06:50 -0500)]
Merge "Scripts: check file versions of Asterisk and dependencies"

4 years agores/res_stasis: Fix accidental subscription to 'all' bridge topic
Matt Jordan [Mon, 28 Sep 2015 01:45:50 +0000 (20:45 -0500)]
res/res_stasis: Fix accidental subscription to 'all' bridge topic

When b99a7052621700a1aa641a1c24308f5873275fc8 was merged, subscribing to a
NULL bridge will now cause app_subscribe_bridge to implicitly subscribe to
all bridges. Unfortunately, the res_stasis control loop did not check that
a bridge changing on a channel's control object was actually also non-NULL.
As a result, app_subscribe_bridge will be called with a NULL bridge when a
channel leaves a bridge. This causes a new subscription to be made to the
bridge. If an application has also subscribed to the bridge, the application
will now have two subscriptions:
(1) The explicit one created by the app
(2) The implicit one accidentally created by the control structure

As a result, the 'BridgeDestroyed' event can be sent multiple times. This
patch corrects the control loop such that it only subscribes an application
to a new bridge if the bridge pointer is non-NULL.

ASTERISK-24870

Change-Id: I3510e55f6bc36517c10597ead857b964463c9f4f

4 years agoScripts: check file versions of Asterisk and dependencies
Scott Griepentrog [Fri, 4 Sep 2015 18:51:46 +0000 (13:51 -0500)]
Scripts: check file versions of Asterisk and dependencies

To help in diagnosing mismatched modules and libraries, this
script scans for version, repository, and source information
and reports what is found.

ASTERISK-25376 #close
Reported by: Ashley Sanders

Change-Id: Ib0642d0fb96712476f59760d6d137a24633fe2d6

4 years agoapp_dial.c: Force COLP update if outgoing channel name changed.
Richard Mudgett [Thu, 24 Sep 2015 17:59:08 +0000 (12:59 -0500)]
app_dial.c: Force COLP update if outgoing channel name changed.

* When a call is answered and the outgoing channel name has changed then
force a connected line update because the channel is no longer the same.
The channel was masqueraded into by another channel.  This is usually
because of a call pickup.

Note: Forwarded calls are handled in a controlled manner so the original
channel name is replaced with the forwarded channel.

ASTERISK-25423
Reported by: John Hardin

Change-Id: I2e01f7a698fbbc8c26344a59c2be40c6cd98b00c

4 years agoapp_dial.c: Factor out a connected line update routine.
Richard Mudgett [Thu, 24 Sep 2015 17:37:07 +0000 (12:37 -0500)]
app_dial.c: Factor out a connected line update routine.

Replace inlined code with update_connected_line_from_peer().

ASTERISK-25423
Reported by: John Hardin

Change-Id: Ia14f18def417645cd7fb453e1bdac682630a5091

4 years agoapp_dial.c: Remove some no-op code.
Richard Mudgett [Wed, 23 Sep 2015 22:41:35 +0000 (17:41 -0500)]
app_dial.c: Remove some no-op code.

Change-Id: Ice1884a94315d3cb7e3bbd47a9fba76a27276c54

4 years agoMerge "logger: Prevent duplicate dynamic channels from being added."
Joshua Colp [Fri, 25 Sep 2015 15:57:40 +0000 (10:57 -0500)]
Merge "logger: Prevent duplicate dynamic channels from being added."

4 years agologger: Prevent duplicate dynamic channels from being added.
Mark Michelson [Wed, 23 Sep 2015 19:02:15 +0000 (14:02 -0500)]
logger: Prevent duplicate dynamic channels from being added.

There was a problem observed where the "logger add channel" CLI command
would allow for a channel with the same name to be added multiple times.
This would result in each message being written out to the same file
multiple times.

The problem was due to the difference in how logger channel filenames
are stored versus the format they are allowed to be presented when they
are added. For instance, if adding the logger channel "foo" through the
CLI, the result would be a logger channel with the file name
/var/log/asterisk/foo being stored. So when trying to add another "foo"
channel, "foo" would not match "/var/log/asterisk/foo" so we'd happily
add the duplicate channel.

The fix presented here is to introduce two new methods in the logger
code:
 * make_filename(): given a logger channel name, this creates the
   filename for that logger channel.
 * find_logchannel(): given a logger channel name, this calls
   make_filename() and then traverses the list of logchannels in order
   to find a match.

This change has made use of make_filename() and find_logchannel()
throughout to more consistently behave.

ASTERISK-25305 #close
Reported by Mark Michelson

Change-Id: I892d52954d6007d8bc453c3cbdd9235dec9c4a36

4 years agoDo not swallow frames on channels leaving bridges.
Mark Michelson [Thu, 24 Sep 2015 19:49:46 +0000 (14:49 -0500)]
Do not swallow frames on channels leaving bridges.

When leaving a bridge, indications on a channel could be swallowed by
the internal indication logic because it appears that the channel is on
its way to be hung up anyway. One such situation where this is
detrimental is when channels on hold are redirected out of a bridge. The
AST_CONTROL_UNHOLD indication from the bridging code is swallowed,
leaving the channel in question to still appear to be on hold.

The fix here is to modify the logic inside ast_indicate_data() to not
drop the indication if the channel is simply leaving a bridge. This way,
channels on hold redirected out of a bridge revert to their expected "in
use" state after the redirection.

ASTERISK-25418 #close
Reported by Mark Michelson

Change-Id: If6115204dfa0551c050974ee138fabd15f978949

4 years agoMerge "ARI: Add the ability to subscribe to all events"
Matt Jordan [Wed, 23 Sep 2015 17:56:36 +0000 (12:56 -0500)]
Merge "ARI: Add the ability to subscribe to all events"

4 years agoMerge "res/res_stasis_device_state: Allow for subscribing to 'all' device state"
Matt Jordan [Wed, 23 Sep 2015 17:56:31 +0000 (12:56 -0500)]
Merge "res/res_stasis_device_state: Allow for subscribing to 'all' device state"

4 years agoMerge "ARI: Add events for Contact and Peer Status changes"
Matt Jordan [Wed, 23 Sep 2015 17:56:26 +0000 (12:56 -0500)]
Merge "ARI: Add events for Contact and Peer Status changes"

4 years agoapp_page.c: Fix crash when forwarding with a predial handler.
Richard Mudgett [Tue, 22 Sep 2015 22:08:49 +0000 (17:08 -0500)]
app_page.c: Fix crash when forwarding with a predial handler.

Page uses the async method of dialing with the dial API.  When a call gets
forwarded there is no calling channel available.  If the predial handler
was set then the calling channel could not be put into auto-service
for the forwarded call because it doesn't exist.  A crash is the result.

* Moved the callee predial parameter string processing to before the
string is passed to the dial API rather than having the dial API do it.
There are a few benefits do doing this.  The first is the predial
parameter string processing doesn't need to be done for each channel
called by the dial API.  The second is in async mode and the forwarded
channel is to have the predial handler executed on it then the
non-existent calling channel does not need to be present to process the
predial parameter string.

* Don't start auto-service on a non-existent calling channel to execute
the predial handler when the dial API is in async mode and forwarding a
call.

ASTERISK-25384 #close
Reported by: Chet Stevens

Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981

4 years agoMerge "core/logging: Fix logging to more than one syslog channel"
Matt Jordan [Tue, 22 Sep 2015 18:20:39 +0000 (13:20 -0500)]
Merge "core/logging: Fix logging to more than one syslog channel"

4 years agoARI: Add the ability to subscribe to all events
Matt Jordan [Fri, 4 Sep 2015 17:25:07 +0000 (12:25 -0500)]
ARI: Add the ability to subscribe to all events

This patch adds the ability to subscribe to all events. There are two possible
ways to accomplish this:
(1) On initial WebSocket connection. This patch adds a new query parameter,
    'subscribeAll'. If present and True, Asterisk will subscribe the
    applications to all ARI events.
(2) Via the applications resource. When subscribing in this manner, an ARI
    client should merely specify a blank resource name, i.e., 'channels:'
    instead of 'channels:12354'. This will subscribe the application to all
    resources of the 'channels' type.

ASTERISK-24870 #close

Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6

4 years agoMerge "app_record: RECORDED_FILE variable not being populated"
Matt Jordan [Tue, 22 Sep 2015 12:40:35 +0000 (07:40 -0500)]
Merge "app_record: RECORDED_FILE variable not being populated"

4 years agoMerge "pbx: Update device and presence state when changing a hint extension."
Joshua Colp [Tue, 22 Sep 2015 10:29:52 +0000 (05:29 -0500)]
Merge "pbx: Update device and presence state when changing a hint extension."

4 years agoapp_record: RECORDED_FILE variable not being populated
Kevin Harwell [Mon, 21 Sep 2015 23:06:15 +0000 (18:06 -0500)]
app_record: RECORDED_FILE variable not being populated

The RECORDED_FILE variable is empty unless a '%d' is specified in the filename.
This patch makes it so the variable is always set to the filename.

ASTERISK-25410 #close

Change-Id: I4ec826d8eb582ae2ad184e717be8668b74d37653

4 years agoMerge "astfd: Adds a timestamp for each entry."
Joshua Colp [Mon, 21 Sep 2015 13:43:45 +0000 (08:43 -0500)]
Merge "astfd: Adds a timestamp for each entry."

4 years agocore/logging: Fix logging to more than one syslog channel
Elazar Broad [Mon, 21 Sep 2015 13:16:46 +0000 (09:16 -0400)]
core/logging: Fix logging to more than one syslog channel

Currently, Asterisk will log to the last configured syslog
channel in logger.conf. This is due to the fact that the
final call to openlog() supersedes all of the previous calls.
This commit removes the call to openlog() and passes the
facility to ast_log_vsyslog(), along with utilizing the
LOG_MAKEPRI macro to ensure that the message is routed to
the correct facility and with the correct priority.

ASTERISK-25407 #close
Reported by: Elazar Broad
Tested by: Elazar Broad

Change-Id: Ie2a2416bc00cce1b04e99ef40917c2011953ddd2

4 years agores/res_stasis_device_state: Allow for subscribing to 'all' device state
Matt Jordan [Fri, 4 Sep 2015 17:24:57 +0000 (12:24 -0500)]
res/res_stasis_device_state: Allow for subscribing to 'all' device state

This patch adds support for subscribing to all device state changes. This is
done either by subscribing to an empty device, e.g., 'eventSource=deviceState:',
or by the WebSocket connection specifying that it wants all state in the
system.

ASTERISK-24870

Change-Id: I9cfeca1c9e2231bd7ea73e45919111d44d2eda32

4 years agoARI: Add events for Contact and Peer Status changes
Matt Jordan [Fri, 4 Sep 2015 02:19:21 +0000 (21:19 -0500)]
ARI: Add events for Contact and Peer Status changes

This patch adds support for receiving events regarding Peer status changes
and Contact status changes. This is particularly useful in scenarios where
we are subscribed to all endpoints and channels, where we often want to know
more about the state of channel technology specific items than a single
endpoint's state.

ASTERISK-24870

Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9

4 years agoMerge "main/config_options: Check for existance of internal object before derefing"
Matt Jordan [Mon, 21 Sep 2015 13:08:36 +0000 (08:08 -0500)]
Merge "main/config_options: Check for existance of internal object before derefing"

4 years agoMerge "app_queue: AgentComplete event has wrong reason"
Matt Jordan [Sat, 19 Sep 2015 21:26:33 +0000 (16:26 -0500)]
Merge "app_queue: AgentComplete event has wrong reason"

4 years agoastfd: Adds a timestamp for each entry.
Alexander Traud [Sat, 19 Sep 2015 17:49:46 +0000 (19:49 +0200)]
astfd: Adds a timestamp for each entry.

Now with menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", a timestamp is
shown with each file descriptor. This helps to debug leaked UDP/TCP ports on
long-lived servers, for example.

ASTERISK-25405 #close

Change-Id: I968339e5155a512eba1032a5263f1ec8b5e1f80b

4 years agoMerge "app_queue: Crash when transferring"
Matt Jordan [Sat, 19 Sep 2015 14:15:37 +0000 (09:15 -0500)]
Merge "app_queue: Crash when transferring"

4 years agoMerge "CHAOS: res_pjsip_diversion avoid crash if allocation fails"
Matt Jordan [Sat, 19 Sep 2015 14:15:21 +0000 (09:15 -0500)]
Merge "CHAOS: res_pjsip_diversion avoid crash if allocation fails"

4 years agoMerge "dr_adaptive_odbc.c, cel_odbc.c, cel_pgsql.c: REFACTOR Macro LENGTHEN_BUF"
Matt Jordan [Sat, 19 Sep 2015 14:15:02 +0000 (09:15 -0500)]
Merge "dr_adaptive_odbc.c, cel_odbc.c, cel_pgsql.c: REFACTOR Macro LENGTHEN_BUF"

4 years agoMerge "chan_sip: Fix From header truncation for extremely long CALLERID(name)."
Joshua Colp [Sat, 19 Sep 2015 13:31:52 +0000 (08:31 -0500)]
Merge "chan_sip: Fix From header truncation for extremely long CALLERID(name)."

4 years agoMerge "CHAOS: avoid crash if string create fails"
Joshua Colp [Sat, 19 Sep 2015 13:22:45 +0000 (08:22 -0500)]
Merge "CHAOS: avoid crash if string create fails"

4 years agopbx: Update device and presence state when changing a hint extension.
Joshua Colp [Wed, 16 Sep 2015 13:22:35 +0000 (10:22 -0300)]
pbx: Update device and presence state when changing a hint extension.

When changing a hint extension without removing the hint first the
device state and presence state is not updated. This causes the state
of the hint to be that of the previous extension and not the current
one. This state is kept until a state change occurs as a result of
something (presence state change, device state change).

This change updates the hint with the current device and presence
state of the new extension when it is changed. Any state callbacks
which may have been added before the hint extension is changed are
also informed of the new device and presence state if either have
changed.

ASTERISK-25394 #close

Change-Id: If268f1110290e502c73dd289c9e7e7b27bc8432f

4 years agoCHAOS: avoid crash if string create fails
Scott Griepentrog [Thu, 17 Sep 2015 21:34:41 +0000 (16:34 -0500)]
CHAOS: avoid crash if string create fails

Validate string buffer allocation before using them.

ASTERISK-25323

Change-Id: Ib9c338bdc1e53fb8b81366f0b39482b83ef56ce0

4 years agodr_adaptive_odbc.c, cel_odbc.c, cel_pgsql.c: REFACTOR Macro LENGTHEN_BUF
Rodrigo Ramírez Norambuena [Fri, 11 Sep 2015 06:52:15 +0000 (03:52 -0300)]
dr_adaptive_odbc.c, cel_odbc.c, cel_pgsql.c: REFACTOR Macro LENGTHEN_BUF

Remove repeated code on macro of assigned buffer to SQL vars

Change-Id: Icb19ad013124498e172ea1d0b29ccd0ed17deef0

4 years agochan_sip: Fix From header truncation for extremely long CALLERID(name).
Walter Doekes [Thu, 17 Sep 2015 09:52:09 +0000 (11:52 +0200)]
chan_sip: Fix From header truncation for extremely long CALLERID(name).

The CALLERID(num) and CALLERID(name) and other info are placed into the
`char from[256]` in initreqprep. If the name was too long, the addr-spec
and params wouldn't fit.

Code is moved around so the addr-spec with params is placed there first,
and then fitting in as much of the display-name as possible.

ASTERISK-25396 #close

Change-Id: I33632baf024f01b6a00f8c7f35c91e5f68c40260

4 years agoCHAOS: res_pjsip_diversion avoid crash if allocation fails
Richard Mudgett [Thu, 17 Sep 2015 21:59:18 +0000 (16:59 -0500)]
CHAOS: res_pjsip_diversion avoid crash if allocation fails

Validate ast_malloc buffer returned before using it in
set_redirecting_value().

ASTERISK-25323

Change-Id: I15d2ed7cb0546818264c0bf251aa40adeae83253

4 years agoapp_queue: AgentComplete event has wrong reason
Kevin Harwell [Thu, 17 Sep 2015 21:47:33 +0000 (16:47 -0500)]
app_queue: AgentComplete event has wrong reason

When a queued caller transfers an agent to another extension sometimes the
raised AgentComplete event has a reason of "caller" and sometimes "transfer".
Since a transfer has taken place this should always be transfer. This occurs
because sometimes the stasis hangup event arrives before the transfer event
thus writing a different reason out.

With this patch, when a hangup event is received during a transfer it will
check to see if the channel that is hanging up is part of a transfer. If so
it will return and let the subsequently received transfer event handler take
care of the cleanup.

ASTERISK-25399 #close

Change-Id: Ic63c49bd9a5ed463ea7a032fd2ea3d63bc81a50d

4 years agoPJSIP: avoid crash when getting rtp peer
Scott Griepentrog [Thu, 17 Sep 2015 18:09:56 +0000 (13:09 -0500)]
PJSIP: avoid crash when getting rtp peer

Although unlikely, if the tech private is returned as
a NULL, chan_pjsip_get_rtp_peer() would crash.

ASTERISK-25323

Change-Id: Ie231369bfa7da926fb2b9fdaac228261a3152e6a

4 years agoapp_queue: Crash when transferring
Kevin Harwell [Thu, 17 Sep 2015 16:31:15 +0000 (11:31 -0500)]
app_queue: Crash when transferring

During some transfer scenarios involving queues Asterisk would sometimes
crash when trying to obtain a channel snapshot (could happen on caller or
member channels). This occurred because the underlying channel had already
disappeared when trying to obtain the latest snapshot.

This patch adds a reference to both the member and caller channels that
extends to the lifetime of the queue'd call, thus making sure the channels
will always exist when retrieving the latest snapshots.

ASTERISK-25185 #close
Reported by: Etienne Lessard

Change-Id: Ic397fa68fb4ff35fbc378e745da9246a7b552128

4 years agores_pjsip_pubsub: Eliminate race during initial NOTIFY.
Mark Michelson [Wed, 16 Sep 2015 22:36:32 +0000 (17:36 -0500)]
res_pjsip_pubsub: Eliminate race during initial NOTIFY.

There is a slim chance of a race condition occurring where two threads
can both attempt to manipulate the same area.

Thread A can be handling an incoming initial SUBSCRIBE request. Thread A
lets the specific subscription handler know that the subscription has
been established.

At this point, Thread B may detect a state change on the subscribed
resource and queue up a notification task on Thread C, the subscription
serializer thread.

Now Thread A attempts to generate the initial NOTIFY request to send to
the subscriber at the same time that Thread C attempts to generate a
state change NOTIFY request to send to the subscriber.

The result is that Threads A and C can step on the same memory area,
resulting in a crash. The crash has been observed as happening when
attempting to allocate more space to hold the body for the NOTIFY.

The solution presented here is to queue the subscription establishment
and initial NOTIFY generation onto the subscription serializer thread
(Thread C in the above scenario). This way, there is no way that a state
change notification can occur before the initial NOTIFY is sent, and if
there is a quick succession of NOTIFYs, we can guarantee that the two
NOTIFY requests will be sent in succession.

Change-Id: I5a89a77b5f2717928c54d6efb9955e5f6f5cf815

4 years agotranslate: Fix transcoding while different in frame size.
Alexander Traud [Fri, 28 Aug 2015 20:42:23 +0000 (22:42 +0200)]
translate: Fix transcoding while different in frame size.

When Asterisk translates between codecs, each with a different frame size (for
example between iLBC 30 and Speex-WB), too large frames were created by
ast_trans_frameout. Now, ast_trans_frameout is called with the correct frame
length, creating several frames when necessary. Affects all transcoding modules
which used ast_trans_frameout: GSM, iLBC, LPC10, and Speex.

ASTERISK-25353 #close

Change-Id: I2e229569d73191d66a4e43fef35432db24000212

4 years agoscheduler: Use queue for allocating sched IDs.
Mark Michelson [Thu, 10 Sep 2015 22:19:26 +0000 (17:19 -0500)]
scheduler: Use queue for allocating sched IDs.

It has been observed that on long-running busy systems, a scheduler
context can eventually hit INT_MAX for its assigned IDs and end up
overflowing into a very low negative number. When this occurs, this can
result in odd behaviors, because a negative return is interpreted by
callers as being a failure. However, the item actually was successfully
scheduled. The result may be that a freed item remains in the scheduler,
resulting in a crash at some point in the future.

The scheduler can overflow because every time that an item is added to
the scheduler, a counter is bumped and that counter's current value is
assigned as the new item's ID.

This patch introduces a new method for assigning scheduler IDs. Instead
of assigning from a counter, a queue of available IDs is maintained.
When assigning a new ID, an ID is pulled from the queue. When a
scheduler item is released, its ID is pushed back onto the queue. This
way, IDs may be reused when they become available, and the growth of ID
numbers is directly related to concurrent activity within a scheduler
context rather than the uptime of the system.

Change-Id: I532708eef8f669d823457d7fefdad9a6078b99b2

4 years agomain/config_options: Check for existance of internal object before derefing
Matt Jordan [Fri, 4 Sep 2015 02:15:13 +0000 (21:15 -0500)]
main/config_options: Check for existance of internal object before derefing

Asterisk can load and register an object type while still having an invalid
sorcery mapping. This can cause an issue when a creation call is invoked.
For example, mis-configuring PJSIP's endpoint identifier by IP address mapping
in sorcery.conf will cause the sorcery mechanism to be invalidated; however, a
subsequent ARI invocation to create the object will cause a crash, as the
internal type may not be registered as sorcery expects.

Merely checking for a NULL pointer here solves the issue.

Change-Id: I54079fb94a1440992f4735a9a1bbf1abb1c601ac

4 years agoMerge "res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route"
Matt Jordan [Fri, 11 Sep 2015 21:13:53 +0000 (16:13 -0500)]
Merge "res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route"

4 years agoMerge "chan_sip.c: Validation on module reload"
Matt Jordan [Fri, 11 Sep 2015 17:40:43 +0000 (12:40 -0500)]
Merge "chan_sip.c: Validation on module reload"

4 years agochan_sip.c: Validation on module reload
Rodrigo Ramírez Norambuena [Sat, 22 Aug 2015 02:50:31 +0000 (22:50 -0400)]
chan_sip.c: Validation on module reload

Change validation on reload module because now used the cli function for
reload. The sip_reload() function never fail and ever return NULL for this
reason on reload() now use the call the sip_reload() and return
AST_MODULE_LOAD_SUCCESS.

This problem is dectected on reload by PUT method on ARI, getting always
404 http code when the module is reloaded.

ASTERISK-25325 #close
Reporte by: Rodrigo Ramírez Norambuena

Change-Id: I41215877fb2cfc589e0d4d464000cf6825f4d7fb

4 years agores_pjsip_pubsub.c: Mark ast_sip_create_subscription() as not used.
Richard Mudgett [Fri, 21 Aug 2015 22:39:01 +0000 (17:39 -0500)]
res_pjsip_pubsub.c: Mark ast_sip_create_subscription() as not used.

Change-Id: I2b8db18eac36c01a5c7eb9467699124e203fd093

4 years agores_pjsip_pubsub.c: Add some notification comments.
Richard Mudgett [Wed, 9 Sep 2015 17:24:31 +0000 (12:24 -0500)]
res_pjsip_pubsub.c: Add some notification comments.

Change-Id: Ie62ff1f4b7adc1a12fa0303f53926af249b25e20

4 years agores_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response.
Richard Mudgett [Fri, 21 Aug 2015 23:01:59 +0000 (18:01 -0500)]
res_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response.

We should not try to send a SIP response message because we may be
restoring a persistent subscription where we are not responding to a SIP
request.

Change-Id: Id89167ef90320c5563f37e632db0dda6cb9e7dec

4 years agores_pjsip_pubsub.c: Fix off-nominal memory leak.
Richard Mudgett [Fri, 21 Aug 2015 22:40:39 +0000 (17:40 -0500)]
res_pjsip_pubsub.c: Fix off-nominal memory leak.

Fix off-nominal visited vector leak in build_resource_tree().

Change-Id: If0399c7941c9c0b1038bcfb7b9a371760977831c

4 years agores_pjsip_pubsub.c: Fix one byte buffer overrun error.
Richard Mudgett [Fri, 21 Aug 2015 20:26:30 +0000 (15:26 -0500)]
res_pjsip_pubsub.c: Fix one byte buffer overrun error.

ast_sip_pubsub_register_body_generator() did not account for the null
terminator set by sprintf() in the allocated output buffer.

Change-Id: I388688a132e479bca6ad1c19275eae0070969ae2

4 years agores_pjsip_pubsub.c: Use ast_alloca() instead of alloca().
Richard Mudgett [Fri, 21 Aug 2015 20:25:11 +0000 (15:25 -0500)]
res_pjsip_pubsub.c: Use ast_alloca() instead of alloca().

Change-Id: Ia396096b4fedc2874649ca11137612c3f55e83e3

4 years agores_pjsip_pubsub.c: Add missing error return in load_module().
Richard Mudgett [Fri, 21 Aug 2015 16:04:08 +0000 (11:04 -0500)]
res_pjsip_pubsub.c: Add missing error return in load_module().

Change-Id: I15debd0f717f16ee2f78e7f56151c3b3b97b72fc

4 years agores_pjsip/location.c: Use the builtin ao2_callback() match function instead.
Richard Mudgett [Fri, 21 Aug 2015 16:03:02 +0000 (11:03 -0500)]
res_pjsip/location.c: Use the builtin ao2_callback() match function instead.

Change-Id: I364906d6d2bad3472929986704a0286b9a2cbe3f

4 years agores_pjsip: Copy default_from_user to avoid crash.
Mark Michelson [Thu, 10 Sep 2015 14:49:45 +0000 (09:49 -0500)]
res_pjsip: Copy default_from_user to avoid crash.

The default_from_user retrieval function was pulling the
default_from_user from the global configuration struct in an unsafe way.
If using a database as a backend configuration store, the global
configuration struct is short-lived, so grabbing a pointer from it
results in referencing freed memory.

The fix here is to copy the default_from_user value out of the global
configuration struct.

Thanks go to John Hardin for discovering this problem and proposing the
patch on which this fix is based.

ASTERISK-25390 #close
Reported by Mark Michelson

Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c

4 years agores/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route
Matt Jordan [Thu, 10 Sep 2015 13:39:21 +0000 (08:39 -0500)]
res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route

We will only rewrite the Contact header if there is no Record-Route header in
the received request. If a malfunctioning proxy places a Record-Route header
into a REGISTER request, we will decide that we shouldn't update the IP/port
in the Contact header, and we will end up storing a contact with an AoR that
contains the NAT'd IP address.

While it is nice to have the proxy *not* send a Record-Route in a REGISTER
request, it's also a good idea to not process the header in a non-dialog
message. This patch updates the code to explicitly ignore the Record-Route
header in REGISTER requests.

ASTERISK-25387 #close

Change-Id: I4bd3bcccc4003d460cc354d986b0dea2e433ef3f

4 years agoMerge "ParkAndAnnounce: Add variable inheritance"
Joshua Colp [Thu, 10 Sep 2015 12:25:02 +0000 (07:25 -0500)]
Merge "ParkAndAnnounce: Add variable inheritance"

4 years agoMerge "chan_ooh323: Add ProgressIndicator IE with inband info available"
Matt Jordan [Thu, 10 Sep 2015 00:12:09 +0000 (19:12 -0500)]
Merge "chan_ooh323: Add ProgressIndicator IE with inband info available"

4 years agoMerge "pjsip: avoid possible crash req_caps allocation failure"
Joshua Colp [Wed, 9 Sep 2015 22:22:22 +0000 (17:22 -0500)]
Merge "pjsip: avoid possible crash req_caps allocation failure"

4 years agochan_ooh323: Add ProgressIndicator IE with inband info available
Alexander Anikin [Wed, 9 Sep 2015 21:46:44 +0000 (01:46 +0400)]
chan_ooh323: Add ProgressIndicator IE with inband info available

Add ProgressIndicator IE with inband info present to Progress and
Alerting Q.931 message

ASTERISK-25227 #close
Reported by: Alexandr Dranchuk

Change-Id: I326ad13cb1db9a72b3fd902bafed3c28a3684203

4 years agopjsip: avoid possible crash req_caps allocation failure
Scott Griepentrog [Tue, 8 Sep 2015 15:35:57 +0000 (10:35 -0500)]
pjsip: avoid possible crash req_caps allocation failure

Make certain that the pjsip session has not failed to
allocate the format capabilities structure, which can
otherwise cause a crash when referenced.

ASTERISK-25323

Change-Id: I602790ba12714741165e441cc64a3ecde4cb5750

4 years agoMerge "res_pjsip: Use hash for contact object identity instead of Contact URI."
Joshua Colp [Wed, 9 Sep 2015 10:53:02 +0000 (05:53 -0500)]
Merge "res_pjsip: Use hash for contact object identity instead of Contact URI."

4 years agoMerge "res_rtp_asterisk: Add more ICE debugging"
Matt Jordan [Tue, 8 Sep 2015 21:33:29 +0000 (16:33 -0500)]
Merge "res_rtp_asterisk: Add more ICE debugging"

4 years agoMerge "Core/General: Add #ifdef needed on FreeBSD."
Joshua Colp [Tue, 8 Sep 2015 21:11:00 +0000 (16:11 -0500)]
Merge "Core/General: Add #ifdef needed on FreeBSD."

4 years agores_rtp_asterisk: Add more ICE debugging
David M. Lee [Fri, 4 Sep 2015 21:33:39 +0000 (16:33 -0500)]
res_rtp_asterisk: Add more ICE debugging

In working through a recent ICE negotiation bug, I found the debug
logging in res_rtp_asterisk to be lacking. This patch adds a number of
debug and warning statements that were helpful.

Change-Id: I950c6d8f13a41f14b3d6334b4cafe7d4e997be80

4 years agoMerge "res/res_pjsip: Purge contacts when an AoR is deleted"
Joshua Colp [Tue, 8 Sep 2015 19:03:42 +0000 (14:03 -0500)]
Merge "res/res_pjsip: Purge contacts when an AoR is deleted"

4 years agores_pjsip: Use hash for contact object identity instead of Contact URI.
Joshua Colp [Tue, 8 Sep 2015 12:21:11 +0000 (09:21 -0300)]
res_pjsip: Use hash for contact object identity instead of Contact URI.

In the wild it is possible for Contact URIs to be quite long as
parameters can exist on them. This can present a problem when storing
them in the AstDB as the URI is used as part of the object name and
there is a fixed length limit for the AstDB. This will cause
the contact to not get stored.

This change uses the MD5 hash of the Contact URI as part of the
object name instead. This has a fixed length which is guaranteed
to not exceed the AstDB length limit.

ASTERISK-25295 #close

Change-Id: Ie8252a75331ca00b41b9f308f42cc1fbdf701a02

4 years agochan_ooh323: call ast_rtp_instance_stop on ooh323_destroy
Alexander Anikin [Mon, 7 Sep 2015 18:19:41 +0000 (22:19 +0400)]
chan_ooh323: call ast_rtp_instance_stop on ooh323_destroy

    Call ast_rtp_instance_stop on ooh323_destroy to free resources
    allocated by rtp instance

    ASTERISK-25299 #close
    Report by: Alexandr Dranchuk

Change-Id: I455096bd7da016b871afe90af86067c2c7c9f33f

4 years agores/res_pjsip: Purge contacts when an AoR is deleted
Matt Jordan [Mon, 7 Sep 2015 16:15:59 +0000 (11:15 -0500)]
res/res_pjsip: Purge contacts when an AoR is deleted

When an AoR is deleted by an external mechanism, such as through ARI, we
currently do not remove dynamic contacts that were created for that AoR as a
result of a received REGISTER request. As a result, re-creating the AoR will
cause the dynamic contact to be interpreted as a persistent contact, leading
to some rather strange state being created for the contacts/endpoints.

This patch adds a sorcery observer for the 'aor' object. When a delete is
issued on the underlying sorcery object, the observer is called, and all
contacts created and persisted in sorcery for that AoR are also removed. Note
that we don't want to perform this action when an AO2 object that is an AoR is
destroyed, as the AoR can still exist in the backing storage (and we would
thus be removing valid contacts from an AoR that still "exists".)

ASTERISK-25381 #close

Change-Id: I6697e51ef6b2858b5d63401f35dc378bb0f90328

4 years agoMerge "endpoint snapshot: avoid second cleanup on alloc failure"
Matt Jordan [Sat, 5 Sep 2015 23:48:26 +0000 (18:48 -0500)]
Merge "endpoint snapshot: avoid second cleanup on alloc failure"

4 years agoMerge "channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id"
Matt Jordan [Sat, 5 Sep 2015 23:43:50 +0000 (18:43 -0500)]
Merge "channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id"

4 years agoMerge "res_pjsip: Change default from user value."
Joshua Colp [Sat, 5 Sep 2015 20:56:58 +0000 (15:56 -0500)]
Merge "res_pjsip: Change default from user value."

4 years agoMerge "Fix when remote candidates exceed PJ_ICE_MAX_CAND"
Joshua Colp [Sat, 5 Sep 2015 20:42:37 +0000 (15:42 -0500)]
Merge "Fix when remote candidates exceed PJ_ICE_MAX_CAND"

4 years agochannels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id
Matt Jordan [Sat, 5 Sep 2015 19:58:41 +0000 (14:58 -0500)]
channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id

This patch adds a new option to the CHANNEL function that allows for the
extraction of the SIP call-id. It is used in conjunction with the 'pjsip'
option, and will return the Call-ID of the INVITE request that established
the PJSIP channel.

ASTERISK-25352

Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a

4 years agoFix when remote candidates exceed PJ_ICE_MAX_CAND
David M. Lee [Fri, 4 Sep 2015 21:06:39 +0000 (16:06 -0500)]
Fix when remote candidates exceed PJ_ICE_MAX_CAND

We were passing the wrong count into pj_ice_sess_create_check_list(),
causing the create to fail if we ever received more than PJ_ICE_MAX_CAND
candidates.

Change-Id: I0303d8e1ecb20a8de9fe629a3209d216c4028378

4 years agores_pjsip: Change default from user value.
Mark Michelson [Fri, 4 Sep 2015 19:40:38 +0000 (14:40 -0500)]
res_pjsip: Change default from user value.

When Asterisk sends an outbound SIP request, if there is no direct
reason to place a specific value for the username in the From header,
Asterisk would generate a UUID. For example, this would happen when
sending outbound OPTIONS requests when qualifying or when sending
outbound INVITE requests when originating (if no explicit caller ID were
provided). The issue is that some SIP providers reject these sorts of
requests with a "Name too long" error response.

This patch aims to fix this by changing the default outbound username in
From headers to "asterisk". This value can be overridden by changing the
default_from_user option in the global options if desired.

ASTERISK-25377 #close
Reported by Mark Michelson

Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190

4 years agoParkAndAnnounce: Add variable inheritance
Jonathan Rose [Thu, 3 Sep 2015 19:07:35 +0000 (14:07 -0500)]
ParkAndAnnounce: Add variable inheritance

In Asterisk 11, the announcer channel would receive channel variables
from the channel being parked by means of normal channel inheritance.
This functionality was lost during the big res_parking project in
Asterisk 12. This patch restores that functionality.

ASTERISK-25369 #close
Review: https://gerrit.asterisk.org/#/c/1180/

Change-Id: Ie47e618330114ad2ea91e2edcef1cb6f341eed6e

4 years agoendpoint snapshot: avoid second cleanup on alloc failure
Scott Griepentrog [Fri, 4 Sep 2015 14:26:46 +0000 (09:26 -0500)]
endpoint snapshot: avoid second cleanup on alloc failure

In ast_endpoint_snapshot_create(), a failure to init the
string fields results in two attempts to ao2_cleanup the
same pointer.  Removed RAII_VAR to eliminate problem.

ASTERISK-25375 #close
Reported by: Scott Griepentrog

Change-Id: If4d9dfb1bbe3836b623642ec690b6d49b25e8979

4 years agores/pjsip: Mark WSS transport as secure
Martin Tomec [Fri, 4 Sep 2015 10:33:47 +0000 (12:33 +0200)]
res/pjsip: Mark WSS transport as secure

Pjsip is refusing to use unsecure transport with "sips" in url.
WSS should be considered as secure transport.

ASTERISK-24602 #comment Partially fixed by setting WSS as secure

Change-Id: Iddac406c6deba6240c41a603b8859dfefe1a5353

4 years agoCore/General: Add #ifdef needed on FreeBSD.
Guido Falsi [Tue, 1 Sep 2015 15:16:55 +0000 (17:16 +0200)]
Core/General: Add #ifdef needed on FreeBSD.

pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED on FreeBSD
too.

ASTERISK-25310 #close
Reported by: Guido Falsi

Change-Id: Iae6befac9028b5b9795f86986a4a08a1ae6ab7c4

4 years agores_pjsip: Fix contact refleak on stateful responses.
Mark Michelson [Wed, 2 Sep 2015 22:26:14 +0000 (17:26 -0500)]
res_pjsip: Fix contact refleak on stateful responses.

When sending a stateful response, creation of the transaction can fail,
most commonly because we are trying to create a transaction from a
retransmitted request. When creation of the transaction fails, we end up
leaking a reference to a contact that was bumped when the response was
created.

This patch adds the missing deref and fixes the reference leak.

Change-Id: I2f97ad512aeb1b17e87ca29ae0abacb4d6395f07

4 years agopbx: Fix crash when issuing "core show hints" with long pattern match.
Joshua Colp [Wed, 2 Sep 2015 17:41:10 +0000 (14:41 -0300)]
pbx: Fix crash when issuing "core show hints" with long pattern match.

When issuing the "core show hints" CLI command a combination of both
the hint extension and context is created. This uses a fixed size
buffer expecting that the extension will not exceed maximum extension
length. When the extension is actually a pattern match this constraint
does not hold true, and the extension may exceed the maximum extension
length. In this case extra characters are written past the end of the
fixed size buffer.

This change makes it so the construction of the combined hint extension
and context can not exceed the size of the buffer.

ASTERISK-25367 #close

Change-Id: Idfa1b95d0d4dc38e675be7c1de8900b3f981f499

4 years agores_pjsip_pubsub: re-re-fix persistent subscription storage.
Mark Michelson [Tue, 1 Sep 2015 14:05:34 +0000 (09:05 -0500)]
res_pjsip_pubsub: re-re-fix persistent subscription storage.

A recent change to res_pjsip_pubsub switched to using pjsip_msg_print as
a means of writing an appropriate packet to persistent storage. While
this partially solved the issue, it had its own problems.
pjsip_msg_print will always add a Content-Length header to the message
it prints. Frequent restarts of Asterisk can result in persistent
subscriptions being written with five or more Content-Length headers. In
addition, sometimes some apparent corruption of individual headers could
be seen.

This aims to fix the problem by not running a parsed message through an
interpreter but rather by taking the raw message and saving it. The
logic for what to save is going to be different depending on whether a
SUBSCRIBE was received from the wire or if it was pulled from
persistence. When receiving a packet from the wire, when using a
streaming transport, the rdata->pkt_info.packet may contain multiple SIP
messages or fragments. However, the rdata->msg_info.msg_buf will always
contain the current SIP message to be processed. When pulling from
persistence, though, the rdata->msg_info.msg_buf will be NULL since no
transport actually handled the packet. However, since we know that we
will always ever pull one SIP message from persistence, we are free to
save directly from rdata->pkt_info.packet instead.

ASTERISK-25365 #close
Reported by Mark Michelson

Change-Id: I33153b10d0b4dc8e3801aaaee2f48173b867855b

4 years agoMerge "taskprocessor: Fix race condition between unreferencing and finding."
Matt Jordan [Sat, 29 Aug 2015 17:31:58 +0000 (12:31 -0500)]
Merge "taskprocessor: Fix race condition between unreferencing and finding."

4 years agoMerge "res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items."
Matt Jordan [Sat, 29 Aug 2015 17:30:59 +0000 (12:30 -0500)]
Merge "res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items."

4 years agotaskprocessor: Fix race condition between unreferencing and finding.
Joshua Colp [Sat, 29 Aug 2015 15:36:35 +0000 (12:36 -0300)]
taskprocessor: Fix race condition between unreferencing and finding.

When unreferencing a taskprocessor its reference count is checked
to determine if it should be unlinked from the taskprocessors
container and its listener shut down. In between the time when the
reference count is checked and unlinking it is possible for
another thread to jump in, find it, and get a reference to it. If
the thread then uses the taskprocessor it may find that it is not
in the state it expects.

This change locks the taskprocessors container during almost the
entire unreference operation to ensure that any other thread which
may attempt to find the taskprocessor has to wait.

ASTERISK-25295

Change-Id: Icb842db82fe1cf238da55df92e95938a4419377c

4 years agores_pjsip_sdp_rtp: Fix multiple keepalive scheduled items.
Joshua Colp [Sat, 29 Aug 2015 01:22:45 +0000 (22:22 -0300)]
res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items.

The keepalive support in res_pjsip_sdp_rtp currently assumes
that a stream will only be negotiated once. This is false.
If the stream is replaced and later added back it can be
negotiated again causing multiple keepalive scheduled items
to exist. This change explicitly deletes the existing
keepalive scheduled item before adding the new one.

The res_pjsip_sdp_rtp module also does not stop RTP
keepalives or timeout timer if the stream has been
replaced. This change adds a callback to the session media
interface to allow a media stream to be stopped without
the resources being destroyed. This allows the scheduled
items and RTP to be stopped when the stream no longer
exists.

ASTERISK-25356 #close

Change-Id: Ibe6a7cc0927c87326fd5f1c0d4ad889dbfbea1de

4 years agosched: ast_sched_del may return prematurely due to spurious wakeup
Joshua Colp [Sat, 29 Aug 2015 00:57:14 +0000 (21:57 -0300)]
sched: ast_sched_del may return prematurely due to spurious wakeup

When deleting a scheduled item if the item in question is currently
executing the ast_sched_del function waits until it has completed.
This is accomplished using ast_cond_wait. Unfortunately the
ast_cond_wait function can suffer from spurious wakeups so the
predicate needs to be checked after it returns to make sure it has
really woken up as a result of being signaled.

This change adds a loop around the ast_cond_wait to make sure that
it only exits when the executing task has really completed.

ASTERISK-25355 #close

Change-Id: I51198270eb0b637c956c61aa409f46283432be61

4 years agores_pjsip_session: Don't invoke session supplements twice for BYE requests.
Joshua Colp [Thu, 27 Aug 2015 17:26:09 +0000 (14:26 -0300)]
res_pjsip_session: Don't invoke session supplements twice for BYE requests.

When a BYE request is received the PJSIP invite session implementation
creates and sends a 200 OK response before we are aware of it. This
causes the INVITE session state callback to be called into and ultimately
the session supplements run on the BYE request. Once this response has
been sent the normal transaction state callback is invoked which
invokes the session supplements on the BYE request again. This can
be problematic in particular with res_pjsip_rfc3326 as it may
attempt to update the hangup cause code on the channel while it is
in the process of being hung up.

This change makes it so the session supplements are only invoked
once by the INVITE session state callback.

ASTERISK-25318 #close

Change-Id: I69c17df55ccbb61ef779ac38cc8c6b411376c19a

4 years agoMerge "res_pjsip: Add common ast_sip_get_host_ip API."
Joshua Colp [Thu, 27 Aug 2015 20:41:54 +0000 (15:41 -0500)]
Merge "res_pjsip: Add common ast_sip_get_host_ip API."

4 years agoMerge "Chaos: make hangup NULL tolerant"
Mark Michelson [Thu, 27 Aug 2015 19:53:45 +0000 (14:53 -0500)]
Merge "Chaos: make hangup NULL tolerant"

4 years agoChaos: handle failed allocation in get_media_encryption_type
Scott Griepentrog [Wed, 26 Aug 2015 20:26:00 +0000 (15:26 -0500)]
Chaos: handle failed allocation in get_media_encryption_type

If the ast_strndup() call fails to allocate a copy of the
transport string for parsing, fail gracefully.

ASTERISK-25323
Reported by: Scott Griepentrog

Change-Id: Ia4b905ce6d03da53fea526224455c1044b1a5a28

4 years agoChaos: make hangup NULL tolerant
Scott Griepentrog [Wed, 26 Aug 2015 19:25:42 +0000 (14:25 -0500)]
Chaos: make hangup NULL tolerant

In chan_pjsip_new, if allocation of the pvt
structure fails, ast_hangup is called.  But
it was written to assume pvt was valid, and
this change corrects that.

ASTERISK-25323
Reported by: Scott Griepentrog

Change-Id: I5f47860fe9cee4cd56abd3f79b108678ab72cc87

4 years agochan_sip: Allow call pickup to set the hangup cause.
Joshua Colp [Wed, 26 Aug 2015 10:40:32 +0000 (07:40 -0300)]
chan_sip: Allow call pickup to set the hangup cause.

The call pickup implementation in chan_sip currently sets the channel
hangup cause to "normal clearing" if call pickup is successfully
performed. This action overwrites the "answered elsewhere" hangup cause
set by the call pickup code and can result in the SIP device in
question showing a missed call when it should not.

This change sets the hangup cause to "normal clearing" as a
default initially but allows the call pickup to change it as
needed.

ASTERISK-25346 #close

Change-Id: I00ac2c269cee9e29586ee2c65e83c70e52a02cff

4 years agores_pjsip: Add common ast_sip_get_host_ip API.
Joshua Colp [Tue, 25 Aug 2015 12:17:34 +0000 (09:17 -0300)]
res_pjsip: Add common ast_sip_get_host_ip API.

Modules commonly used the pj_gethostip function for retrieving the
IP address of the host. This function does not cache the result and may
result in a DNS lookup occurring, or additional work. If the DNS
server is unreachable or network issues arise this can cause the
pj_gethostip function to block for a period of time.

This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string
function which does the same thing but caches the host IP address at
module load time. This results in no additional work being done each
time the local host IP address is needed.

ASTERISK-25342 #close

Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e

4 years agoMerge "res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced"
Mark Michelson [Mon, 24 Aug 2015 22:16:48 +0000 (17:16 -0500)]
Merge "res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced"

4 years agoMerge "bridge: Kick channel from bridge if hung up during action."
Mark Michelson [Mon, 24 Aug 2015 22:12:06 +0000 (17:12 -0500)]
Merge "bridge: Kick channel from bridge if hung up during action."