ASTERISK-24482 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4142/
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In websocket_add_protocol_internal is used to add the "echo"
protocol, but ast_websocket_remove_protocol is used to remove
it.  This causes an extra call to ast_module_unref.

ASTERISK-24480 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4140/
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When merging from 12 to 13 there were conflicts,
I mistakenly had the loop run ast_closestream(others[0])
when it should be ast_closestream(others[x]).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427182 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Virtual line support establishes a relationship between messages
related to an outbound registration and a local endpoint. This is
accomplished by attaching a parameter to the Contact of the outbound
registration and looking for it on any received requests. If the
parameter exists and can be matched to an outbound registration
the configured endpoint is associated with the request.

Review: https://reviewboard.asterisk.org/r/2964/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427165 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a packet exceeds the MTU, pjproject will switch from UDP to TCP.  In
some circumstances (on some networks), this can cause some issues with
messages not getting sent to the correct destination - and can also cause
connections to get dropped due to quirks in pjproject deciding to
terminate TCP connections with no messages.

While fixing the routing/messaging issues is important, having a
configuration option in Asterisk that tells pjproject to not switch over
to TCP would be useful.  That way, if some glitch is discovered on some
other network/site, we can at least disable the behavior until a fix is
put into place.

AFS-197 #close

Review: https://reviewboard.asterisk.org/r/4137/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427113 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change adds an option, moh_passthrough, that when enabled will pass
hold and unhold requests through using a SIP re-invite. When placing on
hold a re-invite with sendonly will be sent and when taking off hold a
re-invite with sendrecv will be sent. This allows remote servers to handle
the musiconhold instead of the local Asterisk instance being responsible.

Review: https://reviewboard.asterisk.org/r/4103/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There is no procedure called ast_closeframe, fix code to use
ast_closestream.

Reported By: Matt Jordan
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Fix cleanup in __ast_play_and_record where others[x] may be leaked.
This was caught where prepend != NULL && outmsg != NULL, once
realfile[x] == NULL any further others[x] would be leaked. A cleanup
block was also added for prepend != NULL && outmsg == NULL.

11+: Fix leak of ast_writestream recording_fs in
app_voicemail:leave_voicemail.

ASTERISK-24476 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4138/
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Fix code paths where it is possible for frames to leak.
Fix uninitialized variable in jb_get_fixed and jb_get_adaptive.

ASTERISK-22409 #related
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4128/
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When the res_stasis module is unloaded, it will dispose of the apps_registry
container. This is a problem if an ARI operation is in flight that attempts
to use the registry, as the shutdown occurs in a separate thread. This patch
adds some sanity checks to the various routines that access the registry which
cause the operations to fail if the apps_registry does not exist.

Crash caught by the Asterisk Test Suite.
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Review: https://reviewboard.asterisk.org/r/4118/
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A question arose as to whether a .pem file
could be provided in place of the .crt and
.key files in a PJSIP TLS configuration. I
tested this and discovered that although a
cert will be read from the pem file, a key
will not, and thus the priv_key_file entry
is still required. This update to the fine
documentation clarifies the option usage.

AST-1448 #close
Review: https://reviewboard.asterisk.org/r/4129/
Reported by: John Bigelow
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This updates the status of the outbound registration
to reflect when it has been unregistered.  Since the
registration is unregistered but is not stopped, the
registration schedule remains active as before.  The
patch also updates the documentation of both the AMI
and CLI commands.

ASTERISK-24411 #close
Review: https://reviewboard.asterisk.org/r/4119/
Reported by: John Bigelow
patches:
  unregister-patch1.txt uploaded by John Bigelow (License 5091)
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When r426594 was made, it did not take into account a unit test that verified
that the function properly populated the unsupported buffer. The function
would previously memset the buffer if it detected it had any contents; since
this function can now be called iteratively on successive headers, the unit
tests would now fail. This patch updates the unit tests to reset the buffer
themselves between successive calls, and updates the documentation of the
function to note that this is now required.
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This change ensures refcounter.py is installed to a place where it
can be found by the Asterisk testsuite if REF_DEBUG is enabled.

ASTERISK-24432 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4094/
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set_member_value has a couple leaks to references in the variable q
found through testsuite tests/queues/set_penalty.  Also remove the
REF_DEBUG_ONLY_QUEUES compiler declaration, this is no longer possible
with the updated REF_DEBUG code.

ASTERISK-24466 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4125/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426808 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Cleanup references to in_translate[x].format and
out_translate[x].format in ast_audiohook_detach_list.

ASTERISK-24465 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4124/
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Currently, it is possible for some subscriptions to get into a NULL state. When
this occurs and the PJSIPShowSubscriptionsInbound ami action is issued and a
device is subscribed for extension state then the associated subscription state
object can't be located.  The code then attempts to dereference a NULL object.
Added a NULL check to avoid the problem.

Reported by: John Bigelow
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When removing the qualify_frequency from an AoR or a contact the statistics
shown when issuing "pjsip show aors" from the CLI are incorrect. This patch
deletes the contact's status object from sorcery, disassociating it from the
contact, if the qualify_freqency is removed from configuration.

ASTERISK-24462 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4116/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426761 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In update_messages_by_imapuser(), messages were appended to a finite
array which resulted in a crash when an IMAP mailbox contained more
than 256 entries. This memory is now dynamically increased as needed.

Observe that this patch adds a bunch of XXX's to questionable code. See
the review (url below) for more information.

ASTERISK-24190 #close
Reported by: Nick Adams
Tested by: Nick Adams

Review: https://reviewboard.asterisk.org/r/4126/
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Add additional checks for NULL pointers to fix several crashes reported.

ASTERISK-24304 #close
Reported by: dhanapathy sathya
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This patch adds support for 414, 493, 479, and a stray 400 response in REGISTER
response handling. This helps interoperability in a number of scenarios.

Review: https://reviewboard.asterisk.org/r/3437

patches:
  rb3437.patch uploaded by oej (License 5267)
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A SIP request may contain multiple Supported: and Required: headers. Currently,
chan_sip only parses the first Supported/Required header it finds. This patch
adds support for multiple Supported/Required headers for INVITE requests.

Review: https://reviewboard.asterisk.org/r/2478

ASTERISK-21721 #close
Reported by: Olle Johansson
patches:
  rb2478.patch uploaded by oej (License 5267)
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A left over from the formats conversion (Corey Farrell).

ASTERISK-24458 #close
Review: https://reviewboard.asterisk.org/r/4117/

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The feature_automonitor() and feature_automixmonitor() functions were not
locking the channel around ast_get_chan_features_general_config().
Accessing the channel datastore list without the channel locked is a good
way to corrupt the list or follow the pointer chain into oblivion.
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When frames are translated by a fax gateway they need to be freed.  The
existing call to ast_frfree was unreachable.  This change reorganizes
fax_gateway_framehook to ensure that ast_frfree is called when needed.

ASTERISK-24457 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4115/
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ASTERISK-24453 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4110/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a channel is imparted to a bridge, the invocation of the function may
provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart,
the caller must assume that ownership has passed to the function, as in all
paths the function destroys the struct prior to returning (as its purpose is
to configure the behavior of the channel while in the bridge). On one off
nominal path - where the channel already has a PBX thread - the struct was not
being destroyed.

This patch fixes that glitch.

ASTERISK-24437 #close
Reported by: Scott Griepentrog
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The parameter name is "Response", not "Resonse".

ASTERISK-24430 #close
Reported by: Dafi Ni
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426365 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426297 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Clean ao2_iterator, resolving reference leak to queue members.

ASTERISK-24454 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4111/
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Remove duplicate allocation of payload, preventing leak.

ASTERISK-24455 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4113/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426253 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Because opus transcoding support cannot be included in the standard Asterisk
distribution, a few codec_opus implementations have popped up.  To make it
easier for people to drop in opus support in their own installations, this
patch adds configure checks for libopus.

Review: https://reviewboard.asterisk.org/r/4106/
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When Moises committed the fixes for WSS (which was a great patch), wdoekes had
a few style nits that were on the review that got missed. This patch resolves
what I *think* were all of the ones that were still on the review.

Thanks to both moy for the patch, and wdoekes for the reviews.

Review: https://reviewboard.asterisk.org/r/3248/
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In res_phoneprov, unloading the module first destroys the http_routes
container, followed by the users. However, users may have a route in
the http_routes container; the validity of this container is not checked
in the users destructor. Hence, we hit an assert as the container has already
been set to NULL.

This patch does two things:
(1) It adds a sanity check in the user destructor (because why not)
(2) It switches the order of destruction, so that users are disposed of prior
    to the HTTP routes they may hold a reference to.

Note that this crash was caught by the Test Suite (go go testing!)
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In libsrtp 1.5.0, crypto_get_random is no longer resolved simply by including
srtp.h. Now, one must include crypto_kernel.h as well. As it turns out, this
header file has been provided by the library since 2006, so this is a
relatively benign change.

ASTERISK-24436 #close
Reported by: Patrick Laimbock
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Review: https://reviewboard.asterisk.org/r/4085/
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