ASTERISK-25189 #close
Reported by: John Hardin

Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac

When performing some provider testing, the rewrite_contact option was
interfering with proper construction of a route set when sending an ACK
after receiving a 200 OK response to an INVITE.

The initial INVITE was sent to address sip:foo. The 200 OK had a Contact
header with URI sip:bar. In addition, the 200 OK had Record-Route
headers for sip:baz and sip:foo, in that order. Since the Record-Route
headers had the lr parameter, the result should have been:

* Set R-URI of the ACK to sip:bar.
* Add Route headers for sip:foo and sip:baz, in that order.

However, the rewrite_contact option resulted in our rewriting the
Contact header on the 200 OK to sip:foo. The result was:

* R-URI remained sip:foo.
* We added Route headers for sip:foo and sip:baz, in that order.

The result was that sip:bar was not indicated in the ACK at all, so the
far end never received our ACK. The call eventually dropped.

The intention of rewrite_contact is to rewrite the most immediate
destination of our SIP request to be the same address on which we
received a request or response. In the case of processing a SIP response
with Record-Route headers, this means that instead of rewriting the
Contact header, we should instead rewrite the bottom-most Record-Route
header. In the case of processing a SIP request with Record-Route
headers, this means we rewrite the top-most Record-route header.
Like when we rewrite the Contact header, we also ensure to update
the dialog's route set if it exists.

ASTERISK-25196 #close
Reported by Mark Michelson

Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f

It is best if the loading code creates and initializes the module's
infrastructure before letting the system know of its existence.  The
unloading code needs to reverse the actions of the loading code and in the
reverse order.

ASTERISK-24907
Reported by: Kevin Harwell

Change-Id: I5d151383e9787b5b60aa5e1627b10f040acdded4

Added checks when a unit test is registered to see that the summary and
description strings do not end with a new-line '\n' for consistency.

The check generates a warning message and will cause the
/main/test/registrations unit test to fail.

* Updated struct ast_test_info member doxygen comments.

Change-Id: I295909b6bc013ed9b6882e85c05287082497534d

Analyzing the code shows that the unit test summary and description
strings should not end with a new-line character.  Where these strings are
used in the code a new-line is provided for output.

Change-Id: I2f4f37988ec363c8d1c5077a2fc8ca841c5cd30c

Analyzing the code shows that the unit test summary and description
strings should not end with a new-line character.  Where these strings are
used in the code a new-line is provided for output.

Change-Id: I129284f5e7ca93d82532334076da4c462d3d9fba

Change-Id: Icf5f13c8e1c2c92a4473bb573ed2dd856ce1b64e

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

The res_pjsip_mwi previously required a reload to set up the proper
subscriptions to allow unsolicited MWI to work. This change
makes it so the act of registering will also cause this to occur.
This is particularly useful if realtime is involved as no reload
needs to occur within Asterisk to cause the MWI information
to get sent.

ASTERISK-25180 #close

Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252

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