Currently when requesting a channel the native formats of the
calling channel are provided to the core for usage when dialing
the outbound channel. This occurs without holding the channel lock
or keeping a reference to the formats. This is problematic as
the channel driver may end up changing the formats during this time.
In the case of chan_sip this happens when an SDP negotiation
completes.

This change makes it so app_dial keeps a reference to the native
formats of the calling channel which guarantees that they will
remain valid for the period of time needed.

ASTERISK-25172 #close

Change-Id: I2f0a67bd0d5d14c3bdbaae552b4b1613a283f0db

Change-Id: I39ae612746d892d2dbe86f3ff2d7027fa1da57f7

Change-Id: I399cb9d61bbba706b48c98e0bf75e98984cd9a9e

* Break some long lines.

* Fix doxygen comment.

Change-Id: I8f12ba6822f84d5e7bb575280270cd7e2fefb305

After completing an attended transfer the transfer target channel was not being
hung up after leaving the bridge. Added an explicit softhangup to hangup said
channel, but only if it was previously bridged.

ASTERISK-24782 #close
Reported by: John Bigelow

Change-Id: Idde9543d56842369384a5e8c00d72a22bbc39ada

Change-Id: I82e6e388e3688aebe0783f16c9e0800a747584b5

On reload, previously allowed codecs were not removed. Therefore, it was not
possible to remove codecs while Asterisk was running. Furthermore, newly added
codecs got appended behind the previous codecs. Therefore, it was not possible
to add a codec with a priority of #1. This change removes the old capabilities
before the current ones are added.

ASTERISK-25182 #close
Reported by: Alexander Traud
patches:
 asterisk_13_allow_codec_reload.patch uploaded by Alexander Traud (License 6520)

Change-Id: I62a06bcf15e08e8c54a35612195f97179ebe5802

Due to the use of stasis_unsubscribe_and_join in the peer destructor
it is possible for a deadlock to occur when an event callback is
occurring at the same time.

This happens because the peer may be destroyed while holding the
peers container lock. If this occurs the event callback will never
be able to acquire the container lock and the unsubscribe will
never complete.

This change makes it so the peers that have been removed from the
peers container are not destroyed with the container lock held.

ASTERISK-25163 #close

Change-Id: Ic6bf1d9da4310142a4d196c45ddefb99317d9a33

This resolves two observed race conditions.

First, a bit of background on what the Stasis application does:

1a Creates a stasis_app_control structure. This structure is linked into
   a global container and can be looked up using a channel's unique ID.
2a Puts the channel in an event loop. The event loop can exit either
   because the stasis_app_control structure has been marked done, or
   because of some other factor, such as a hangup. In the event loop, the
   stasis_app_control determines if any specific ARI commands need to be
   run on the channel and will run them from this thread.
3a Checks if the channel is bridged. If the channel is bridged, then
   ast_bridge_depart() is called since channels that are added to Stasis
   bridges are always imparted as departable.
4a Unlink the stasis_app_control from the container.

When an ARI command is received by Asterisk, the following occurs
1b A thread is spawned to handle the HTTP request
2b The stasis_app_control(s) that corresponds to the channel(s) in the
   request is/are retrieved. If the stasis_app_control cannot be
   retrieved, then it is assumed that the channel in question has exited
   the Stasis app or perhaps was never in Stasis in the first place.
3b A command is queued onto the stasis_app_control, and the channel's
   event loop thread is signaled to run the command.
4b While most ARI commands do nothing further, some, such as adding or
   removing channels from a bridge, will block until the command they
   issued has been completed by the channel's event loop.

The first race condition that is solved by this patch involves a crash
that can occur due to faulty detection of the channel's bridged status
in step 3a. What can happen is that in step 2a, the event loop may run
the ast_bridge_impart() function to asynchronously place the channel
into a bridge, then immediately exit the event loop because the channel
has hung up. In step 3a, we would detect that the channel was not
bridged and would not call ast_bridge_depart(). The reason that the
channel did not appear to be bridged was that the depart_thread that is
spawned by ast_bridge_impart() had not yet started. That is the thread
where the channel is marked as being bridged. Since we did not call
ast_bridge_depart(), the Stasis application would exit, and then the
channel would be destroyed Then the depart_thread would start up and
try to manipulate the destroyed channel, causing a crash.

The fix for this is to switch from using ast_channel_is_bridged() to
checking the NULLity of ast_channel_internal_bridge_channel() to
determine if ast_bridge_depart() needs to be called. The channel's
internal bridge_channel is set when ast_bridge_impart() is called and
is NULLed by the call to ast_bridge_depart(). If the channel's internal
bridge_channel is non-NULL, then the channel must have been imparted
into the bridge and needs to be departed, even if the actual bridging
operation has not yet started. By departing the channel when necessary,
the thread that is running the Stasis application will block until the
bridge gives the okay that the depart_thread has exited.

The second race condition that is solved by this patch involves a leak
of HTTP handler threads. The problem was that step 2b would successfully
retrieve a stasis_app_control structure. Then step 2a would exit the
channel from the event loop due to a hangup. Steps 3a and 4a would
execute, and then finally steps 3b and 4b would. The problem is that at
step 4b, when attempting to add a channel to a bridge, the thread would
block forever since the channel would never execute the queued command
since it was finished with the event loop. This meant that the HTTP
handling thread would be leaked, along with any references that thread
may have owned (in my case, I was seeing bridges leaked).

The fix for this is to hone in better on when the channel has exited the
event loop. The stasis_app_control structure has an is_done field that
is now set at each point where the channel may exit the event loop. If
step 2b retrieves a valid stasis_app_control structure but the control
is marked as done, then the attempted operation exits immediately since
there will be nothing to service the attempted command.

ASTERISK-25091 #close
Reported by Ilya Trikoz

Change-Id: If66265b73b4c9f8f58599124d777fedc54576628

To prevent confusion I am removing the prefetch option until such
time as it is implemented. All other functionality, however, has
been implemented.

ASTERISK-25067

Change-Id: I9ce6aa3e5c6c5bc3c5baa8ff90fa036d73939895

This event was added some time ago in order to clarify when a channel
took the place of another channel in a parking lot. However, there was
no XML documentation added for the event. This patch adds the XML
documentation.

ASTERISK-24900 #close
Reported by Rusty Newton

Change-Id: I4cfe7777c4b94bbff91c9221c6096a7a02a92eac

ASTERISK-25162 #close

Change-Id: Id79aa3c6fe490016ee98efc97ac4c1d3f461f97e

Some phones send g.726 audio packed for AAL2, which differs from what is
recommended by RFC 3351. If Asterisk receives audio formatted as such when
negotiating g.726 then it sounds a bit distorted. Added an option to
res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
AAL2 packed.

ASTERISK-25158 #close
Reported by: Steve Pitts

Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615

The CDR_PROP function (as well as the NoCDR application) set the
'disable all' flag (AST_CDR_FLAG_DISABLE_ALL) on the current CDR. This
flag is supposed to be applied to all CDRs that are currently in the
chain, as well as all CDRs that may be created in the future. Currently,
however, the flag is only applied to the existing CDRs in the chain; new
CDRs do not receive the 'disable all' flag. In particular, this affects
parallel dials, which generate new CDRs for each pair of channels in
the dial attempt.

This patch carries over the 'disable all' flag when it is specified on a
CDR and a new CDR is generated for the chain.

ASTERISK-24344 #close

Change-Id: I91a0f0031e4d147bdf8a68ecd08304d506fb6a0e

When a parallel dial occurs, a new CDR will be created for each dial
attempt that is made. In most circumstances, the act of creating each
CDR in the chain will include a step that updates the Party A snapshot,
which causes the context/extension of the Party A to be copied onto the
CDR object.

However, when the Party A is in a subroutine, we explicitly do *not*
copy the context/extension onto the CDR. This prevents the Macro or
GoSub routine name from blowing away the context/extension that the
channel was originally executing in. For the original CDR, this is not a
problem: the original CDR already recorded the last known 'good' state
of the channel just prior to it going into the subroutine. However, for
newly generated CDRs in a chain, there is no context/extension set on
them. Since we are in a subroutine, we will never set the Party A's
context/extension on the CDR, and we end up with a CDR with no
destination recorded on it.

This patch updates the creation of a chained CDR such that it copies
over the original CDR's context/extension. This is the last known "good"
state of the CDR, and is a reasonable starting point for the newly
generated CDR. In the case where we are not in a subroutine, subsequent
code will update the location of the CDR from the Party A information;
in the case where we are in a subroutine, the context/extension on the
original CDR is the correct information.

ASTERISK-24443 #close

Change-Id: I6a3ef0d6e458d3b9b30572feaec70f2964f3bc2a

If a client sends and INVITE which is 401 rejected, then subsequently
sends a new INVITE with the auth info and uses a different fromtag
from the first INVITE, Asterisk will accept the new INVITE as part of
the original dialog - match_req_to_dialog() specifically ignores the
fromtag. However it does not update the stored dialog with the new
fromtag.

This results in Asterisk being unable to match future packets that are
part of this dialog (such as the ACK to the OK or the OK to the BYE),
and the call is dropped.

This problem was originally found when using an NEC-i SV8100-GE (NEC SIP
Card).

* After a successful match of a packet to the dialog, if the packet is
  not a SIP_RESPONSE, authentication is present and the fromtags are
  different, the stored fromtag is updated with the one from the recent
  INVITE.

ASTERISK-25154 #close
Reported by: Damian Ivereigh
Tested by: Damian Ivereigh

Change-Id: I5c16cf3b409e5ef9f2b2fe974b6bd2a45a6aa17e

Prior to this patch, chan_pjsip was failing to pass the endpoint's
context and the desired extension to the ast_channel_alloc_* routine.
This caused a new channel snapshot to be issued without a context and
extension, which can cause some reporting issues for users of AMI, CEL,
and other APIs. The channel driver would later set the context and
extension on the channel such that the channel would start in the
correct location in the dialplan, but the information reported in the
initial event would be incorrect.

This patch modifies the channel driver such that it now passes the
context and extension directly into the allocation routine. This
provides the information in the new channel snapshot published over
Stasis.

ASTERISK-25156 #close
Reported by: cloos

Change-Id: Ic6f8542836e596db8f662071d118e8f934fdf25e

When performing a blonde transfer the code uses the old masquerade
mechanism to move a channel around. As a result of this certain information,
such as connected line, is moved between the channels involved. Upon
completion of the move a frame is queued which is supposed to update the
connected line information on the channel. This does not occur as the
code considers it a redundant update since the masquerade operation
updated the channel (but did not inform it of the new connected line
information). The code also does not queue a connected line update
to be handled by the thread handling the channel. Without this any
other channel that may be loosely involved does not know it is
talking to a different caller.

This change does the following to resolve this:

1. The indicated connected line information is cleared upon
completion of the masquerade operation when doing a blonde transfer.
This prevents the connected line update from being considered
redundant.

2. A connected line update frame is now queued upon the completion
of the masquerade operation so any other channel loosely involved
knows that there is a different caller.

ASTERISK-25157 #close
Reported by: Joshua Colp

Change-Id: Ibb8798184a1dab3ecd35299faecc420034adbf20

The voicemail.conf mailbox key/value pair is defined as:
<mailbox>=[<password>[,<full-name>[,<email>[,<pager>[,<options>]]]]]
Where all fields in the value including the field values are optional.

Since the parsing code for the mailbox key/value pair is sloppy, this
patch tightens the parsing for the directory information.

* Renamed the 'pos' and 'bufptr' variables to 'name' and 'options'
respectively in search_directory_sub().  Those names make more sense.

* Made sure that search_directory_sub() is dealing with the voicemail.conf
mailbox options field if it even exists when looking for the 'hidefromdir'
and 'alias' options.

* Fix crash if a voicemail.conf mailbox is just
<mailbox>=<password>,<name> when the 'a' option is used.  If there were no
fields after the name then the 'options' pointer was not checked for NULL.

* Fix users.conf alias processing if the 'a' option is used.  The wrong
variable was used.

ASTERISK-25087 #close
Reported by: Chet Stevens

Change-Id: I86052ea77307beddddba5279824d39dc0d593374

All send/receive processing for a SIP transaction needs to be done under
the same threadpool serializer to prevent reentrancy problems inside
pjproject when using an external DNS resolver to process messages for the
transaction.

* Add threadpool API call to get the current serializer associated with
the worker thread.

* Pick a serializer from a pool of default serializers if the caller of
res_pjsip.c:ast_sip_push_task() does not provide one.

This is a simple way to ensure that all outgoing SIP request messages are
processed under a serializer.  Otherwise, any place where a pushed task is
done that would result in an outgoing out-of-dialog request would need to
be modified to supply a serializer.  Serializers from the default
serializer pool are picked in a round robin sequence for simplicity.

A side effect is that the default serializer pool will limit the growth of
the thread pool from random tasks.  This is not necessarily a bad thing.

* Made pjsip_resolver.c use the requesting thread's serializer to execute
the async callback.

* Made pjsip_distributor.c save the thread's serializer name on the
outgoing request tdata struct so the response can be processed under the
same serializer.

ASTERISK-25115 #close
Reported by: John Bigelow

Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a

* Fix query_set destruction before we are done kicking the queries off.

* Fixed no queries requested handling.

* Add empty queries request unit test.

* Added missing allocation check in ast_dns_query_set_add().

* Made initial pjsip resolving query vector slightly larger.

ASTERISK-25115
Reported by: John Bigelow

Change-Id: Ie8be8347d0992e93946d72b6e7b1299727b038f2

Those trailing newlines mess up test formatting.

Change-Id: I5e3f3a55b82c9d7acb9661201d4993d1958f1185

Change-Id: I1716c93d6e097ad28128ecb9e806aac7a4180c8a

Change-Id: Icafea3fb4ea64ac027561b23cbfe2b17997dc549

Change-Id: I4615771077c3c6a0a7273da6d7b5f77af7e8d976

Change-Id: Iee3bd8c8a528776056972066698fe735f0f6cf60

This patch fixes use-after-free bugs caught by AddressSanitizer.

1. PJSIP transport manager may decide to destroy transport on its own.
For example, when the contact registered via websocket has not renewed
its registration in time. The transport was destoyed, but the websocket
listener thread was still active until the socket closes, and then tried
to call transport_shutdown on transport that has been freed.

Also, the transport destructor accessed wstransport->rdata.tp_info.pool
right after freeing memory that contained wstransport itself.

This patch converts transport to an ao2 object, allowing it to be
refcounted, so that it is available until both websocket listener and
pjsip transport manager are finished with it.

2. The websocket listener deletes the last reference on websocket session
when the tcp connection is closed, and it gets destroyed, but
the transport manager may still use it, for example when disconnect
happens in the middle of a SIP transaction.

A new reference to websocket session has been added that is released
with the transport to prevent this.

ASTERISK-25096 #close
Reported by: Josh Kitchens

ASTERISK-24963 #close
Reported by: Badalian Vyacheslav

Change-Id: Idc0b63eb6e459c1ddfb2430127d34b3c4d8d373b

GCC 4.7 Manual:
http://gcc.gnu.org/onlinedocs/gcc-4.7.4/gcc/Function-Attributes.html

weakref ("target")

A weak reference is an alias that does not by itself require a definition
to be given for the target symbol.

ASTERISK-22559 #close
Reported by: Ibercom

Change-Id: I36a136cae947b65187a697533416f9ff9a0b8cdf