ASTERISK-29244

Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5

refer_progress_notify wasn't always being called from the progress
serializer.  This could allow clearing notification->progress->sub
in one thread while another was trying to use it.

* Instances where refer_progress_notify was being called in-line,
  have been changed to use ast_sip_push_task().

Change-Id: Idcf1934c4e873f2c82e2d106f8d9f040caf9fa1e

When an endpoint requests to re-negotiate for fax and the incoming
re-invite is received prior to Asterisk sending out the 200 OK for
the initial invite the re-invite gets delayed. When Asterisk does
finally send the re-inivite the SDP includes streams for both audio
and T.38.

This happens because when the pending topology and active topologies
differ (pending stream is not in the active) in the delayed scenario
the pending stream is appended to the active topology. However, in
the fax case the pending stream should replace the active.

This patch makes it so when a delay occurs during fax negotiation,
to or from, the audio stream is replaced by the T.38 stream, or vice
versa instead of being appended.

Further when Asterisk sent the re-invite with both audio and T.38,
and the endpoint responded with a declined T.38 stream then Asterisk
would crash when attempting to change the T.38 state.

This patch also puts in a check that ensures the media state has a
valid fax session (associated udptl object) before changing the
T.38 state internally.

ASTERISK-29203 #close

Change-Id: I407f4fa58651255b6a9030d34fd6578cf65ccf09

Add option "srtpreplayprotection" rtp.conf to enable srtp
replay protection.

ASTERISK-29260
Reported by: Alexander Traud

Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458

New responses sent within a PJSIP sessions are based on those that were
sent before. Therefore, adding/modifying a header once causes it to be
sent on all responses that follow.

Sending 181 Call Is Being Forwarded many times first adds "histinfo"
duplicated more and more, and eventually overflows past the array
boundary.

This commit adds a check preventing adding "histinfo" more than once,
and skipping it if there is no more space in the header.

Similar overflow situations can also occur in res_pjsip_path and
res_pjsip_outbound_registration so those were also modified to
check the bounds and suppress duplicate Supported values.

ASTERISK-29227
Reported by: Ivan Poddubny

Change-Id: Id43704a1f1a0293e35cc7f844026f0b04f2ac322

ASTERISK-29205 #close

Change-Id: Ib7aa65644e8df76e2378d7613ee7cf751b9d0bea

If a remote side is broken and sends an SDP that can not be
negotiated the call will be torn down but there is a window
where a second 183 Session Progress or 200 OK that is forked
can be received that also attempts to negotiate SDP. Since
the code marked the SDP negotiation as being done and complete
prior to this it assumes that there is an active local and remote
SDP which it can modify, while in fact there is not as the SDP
did not successfully negotiate. Since there is no local or remote
SDP a crash occurs.

This patch changes the pjmedia_sdp_neg_modify_local_offer2
function to no longer assume that a previous SDP negotiation
was successful.

ASTERISK-29196

Change-Id: I22de45916d3b05fdc2a67da92b3a38271ee5949e

After some changes to streams and topologies, receiving fax through
local channels stopped working. This change adds a stream topology with
a stream of type IMAGE to the local channel pair and allows fax to be
received.

ASTERISK-29035 #close

Change-Id: Id103cc5c9295295d8e68d5628e76220f8f17e9fb

Provided a support of a MIME-type for wav16. Added new MIME-type
for classic wav.

ASTERISK-29275 #close

Change-Id: I749bda287ba1ab20c1e0af5e4c0153817d47873b

Two previous commits, 620d9f47 and 6d980de2, allow to set up a call
without audio, again. That was introduced originally with commit f04d5fb8
but changed and broke over time. The original commit missed one
scenario: A [peer] section in sip.conf, which does not allow audio at
all. In that case, chan_sip rejected the call, although even when the
requester offered no audio. Now, chan_sip does not check whether there
is no audio format but checks whether there is no format in general. In
other words, if there is at least one format to offer, the call succeeds.

However, to prevent calls with no-audio, chan_sip still rejects calls
when both call parties (caller = requester of the call *and* callee =
[peer] section in sip.conf) included audio. In such a case, it is
expected that the call should have audio.

ASTERISK-29280

Change-Id: I0fb74faf51ef22a60c10b467df6a4d1c1943b73e

If there's no secret specified for an iax2 peer and there's no secret
specified in the dial string, Asterisk will crash if the auth method
requested by the peer is MD5 or plaintext.  You also couldn't specify
a default auth method in the [general] section of iax.conf so if you
don't have static peers defined and just use the dial string, Asterisk
will still crash even if you have a secret specified in the dial string.

* Added logic to iax2_call() and authenticate_reply() to print
  a warning and hanhup the call if encryption is requested and
  there's no secret or auth method.  This prevents the crash.

* Added the ability to specify a default "auth" in the [general]
  section of iax.conf.

ASTERISK-29624
Reported by: N A

Change-Id: I5928e16137581f7d383fcc7fa04ad96c919e6254

Change-Id: I350939f2220f9e5d44ddf4c8d9a4c99fde4d169a

The previous commit 6d980de2 fixed this issue in the core of Asterisk.
With that, each channel technology can be used without audio
theoretically. Practically, the channel-technology driver chan_sip
turned out to have an invalid check preventing that. chan_sip tested
whether there is at least one audio format. However, chan_sip has to
test whether there is at least one format. More cannot be tested while
requesting chan_sip because only the [general] capabilities but not the
[peer] caps are known yet. And the [peer] caps might not be a subset or
show any intersection with the [general] caps. This change here fixes
this.

The original commit f04d5fb8, thirteen years ago, contained a software
bug as it passed ANY audio capability to the channel-technology driver.
Instead, it should have passed NO audio format. Therefore, this
addressed issue here was not noticed in Asterisk 1.6.x and Asterisk 1.8.
Then, Asterisk 10 changed that from ANY to NO, but nobody reported since
then.

ASTERISK-29265

Change-Id: Ic16a3bf13cd1b5c4fc4041ed74961177d96b600f

When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is
0 when no protocl specific error
SIP example of failure, 3xx-6xx for the SIP error code received

This allows applications to perform actions based on the failure
reason.

ASTERISK-29252 #close
Reported-by: Dan Cropp

Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4

if From number contain * or # asterisk will not add user=phone

Currently only number that uses AST_DIGIT_ANYNUM can have "user=phone" but the validation should use AST_DIGIT_ANY
this is a problem when you want to send call to ISUP
as they will disregard the From header and either replace From with anonymous or with p-asserted-identity

ASTERISK-29261
Reported by: Mark Petersen
Tested by: Mark Petersen

Change-Id: I3307bdbf757582740bfee4110e85f7b6c9291cc4

ASTERISK-29259

Change-Id: Ib6a6550e0e08355745d66da8e60ef49e81f9c6c5

This completes the fix for ASTERISK_24543. Only when the call is an
outgoing call, consult and append the configured format capabilities
(p->caps). When all audio formats got rejected the negotiated format
capabilities (p->jointcaps) contain no audio formats for incoming
calls. This is required when there are other accepted media streams.

ASTERISK-29258

Change-Id: I8bab31c7f3f3700dce204b429ad238a524efebb9

Log proper control frame names instead of "Unknown control '14'", etc.

Change-Id: I1724f2f4d1b064b25a5c93a7da0cb03be5143935

Provided a support of variuos URL-schemes for res_musiconhold,
registered by ast_bucket_scheme_register().

ASTERISK-29262 #close

Change-Id: If0ea8697587353dce358a70035d82649fd4632b6

function ast_sip_session_media_state_add.

Check ast_media_type matches when a ast_sip_session_media is found
otherwise when transitioning from say image to audio, the wrong
session is returned in the first if statement.

ASTERISK-29220 #close

Change-Id: I6f6efa9b821ebe8881bb4c8c957f8802ddcb4b5d

The last argument to ast_copy_string() is the buffer size, not the
number of characters, so we add 1 to avoid stamping out the final \n
in the persisted SUBSCRIBE message.

Change-Id: I019b78942836f57965299af15d173911fcead5b2

From https://www.mail-archive.com/bug-autoconf@gnu.org/msg04408.html



> ... the long-obsolete AC_HEADER_STDC, previously used internally by
> AC_INCLUDES_DEFAULT, used AC_EGREP_HEADER.  The AC_HEADER_STDC macro
> is now a no-op (and is not used at all within Autoconf anymore), so
> that change is likely what made the first use of AC_EGREP_HEADER the
> one inside the if condition, causing the observed results.

The implication is that the test does nothing anyway, and due to it
being a no-op from 2.70 onwards, results in the required not being set
to yes, resulting in ./configure to fail.

Change-Id: Ic1ff38d87f791fbf1f2a80512f81bb7110392460
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

Otherwise, Clang 10 warned because of logical-not-parentheses.

Change-Id: Ia8fb493f727b08070eb2dcf520c08df34ed11d79

ASTERISK-29248

Change-Id: I2b17bd5ffb246bc64c463402c9831413da78a556

There are a couple of parameters (datalen and data) that do not get set
in chan_pjsip_indicate which could cause an Invalid message to pop up
for things such as fax. This patch adds them to the frame.

Change-Id: Ia51be086a0708be905e73d1f433572c49c7e38f8

When both a tech subscription and an endpoint subscription exist for a given
endpoint, TextMessageReceived events are dispatched to the tech subscription
only.

ASTERISK-29229

Change-Id: I9eac4cba5f9e27285a282509395347abc58fc2b8

session->channel doesn't exist until chan_pjsip creates it, so intead of
setting a channel variable every new incoming call sets one and the same
global variable.

This patch moves the code to chan_pjsip so that SIPDOMAIN is set on
a newly created channel, it also removes a misleading reference to
channel->session used to fetch call pickup configuraion.

ASTERISK-29240

Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755

Previously, chan_sip parsed all known media streams in an SDP offer
like video (and text) even when videosupport=no (and textsupport=no).
This wasted processor power. Furthermore, chan_sip accepted SDP offers,
including no audio but just video (or text) streams although
videosupport=no (or textsupport=no). Finally, chan_sip denied the whole
offer instead of individual streams when they had encryption (SDES-sRTP)
unexpectedly enabled.

ASTERISK-29238
ASTERISK-29237
ASTERISK-29222

Change-Id: Ie49e4e2a11f0265f914b684738348ba8c0f89755

The fix for ASTERISK-27902 made chan_pjsip process SIP responses twice.
This resulted in extra noise in logs (for example, "is making progress"
and "is ringing" get logged twice by app_dial), as well as in noise in
signalling: one incoming 183 Session Progress results in 2 outgoing 183-s.

This change splits the response handler into 2 functions:
 - one for updating HANGUPCAUSE, which is still called twice,
 - another that does the rest, which is called only once as before.

ASTERISK-28016
Reported-by: Alex Hermann

ASTERISK-28549
Reported-by: Gant Liu

ASTERISK-28185
Reported-by: Julien

Change-Id: I0a1874be5bb5ed12d572d17c7f80de6e5e542940

With newer version of linux /var/run/ is a symlink to /run/ that has
been turned into tmpfs.

Added note that if asterisk has to bind to a specific IP that
systemd has to wait until the network is up.

Added note on how to make sure that the environment variable
HOSTNAME is included.

ASTERISK-29216
Reported by: Mark Petersen
Tested by: Mark Petersen

Change-Id: Ib3e560655befd3e99eec743687144f5569533379

This reverts commit 860e40dd.

Reason for revert: Too many issues reported.  Need to research and correct.

ASTERISK-29230
ASTERISK-29231
Reported by: Michael Maier

Change-Id: I9011e2eecda4e91e1cfeeda6d1a7f1a0453eab41

Under contention it becomes possible that multiple channels will be told
they successfully obtained the lock, which is a bug.  Please refer

ASTERISK-29217

This introduces a couple of changes.

1.  Replaces requesters ao2 container with simple counter (we don't
    really care who is waiting for the lock, only how many).  This is
    updated undex ->mutex to prevent memory access races.
2.  Correct semantics for ast_cond_timedwait() as described in
    pthread_cond_broadcast(3P) is used (multiple threads can be released
    on a single _signal()).
3.  Module unload races are taken care of and memory properly cleaned
    up.

Change-Id: I6f68b5ec82ff25b2909daf6e4d19ca864a463e29
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

On Gentoo it's possible to have multiple lua versions installed, all
with a path of /usr, so it's not possible to use the current --with-lua
option to determisticly pin to a specific version as is required by the
Gentoo PMS standards.

This environment variable allows to lock to specific versions,
unversioned check will be skipped if this variable is supplied.

Change-Id: I8c403eda05df25ee0193960262ce849c7d2fd088
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

Add channel reference count for PJSIP REFER. The call could be terminated
prior to the result of the transfer. In that scenario, when the SUBSCRIBE/NOTIFY
occurred several minutes later, it would attempt to access a session which was
no longer valid.  Terminate event subscription if pjsip_xfer_initiate() or
pjsip_xfer_send_request() fails in transfer_refer().

ASTERISK-29201 #close
Reported-by: Dan Cropp

Change-Id: I3fd92fd14b4e3844d3d7b0f60fe417a4df5f2435

launch_monitor_thread is responsible for creating and initializing
the mixmonitor, and dependent data structures. There was one off
nominal path after the datastore gets created that triggers when
the channel being monitored is hung up prior to monitor starting
itself.

If this happened the monitor thread would not "launch", and the
mixmonitor object and associated objects are freed, including the
underlying datastore data object. However, the datastore itself was
not removed from the channel, so when the channel eventually gets
destroyed it tries to access the previously freed datastore data
and crashes.

This patch removes and frees datastore object itself from the channel
before freeing the mixmonitor object thus ensuring the channel does
not call it when destroyed.

ASTERISK-28947 #close

Change-Id: Id4f9e958956d62473ed5ff06c98ae3436e839ff8

ASTERISK-28992 #close

Change-Id: Ia7d608924036139ee2520b840d077762d02668d0

A prior patch segmented channel snapshots, and changed the underlying
data object type associated with ast_channel_snapshot_type stasis
messages. Prior to Asterisk 18 it was a type ast_channel_snapshot, but
now it type ast_channel_snapshot_update.

When publishing ast_channel_snapshot_type in pbx_realtime the
ast_channel_snapshot was being passed in as the message data
object. When a handler, expecting a data object type of
ast_channel_snapshot_update, dereferenced this value a crash
would occur.

This patch makes it so pbx_realtime now uses the expected type, and
channel snapshot publish method when publishing.

ASTERISK-29168 #close

Change-Id: I9a2cfa0ec285169317f4b9146e4027da8a4fe896

Rename check_manager_enabled() and check_webmanager_enabled() to begin
with ast_ so that the symbols are automatically exported by the
linker.

ASTERISK~29184

Change-Id: I85762b9a5d14500c15f6bad6507138c8858644c9

Segfault occurs during outbound UDP registration when all
transport states are being iterated over. The transport object
in the transport is accessed, but flow transports have a NULL
transport object.

Modify to not iterate over any flow transport

ASTERISK-29210 #close

Change-Id: If28dc3a18bdcbd0a49598b09b7fe4404d45c996a