3 years agostream: Make ast_stream_topology_create_from_format_cap() allow NULL cap.
Richard Mudgett [Fri, 21 Apr 2017 00:25:10 +0000 (19:25 -0500)]
stream: Make ast_stream_topology_create_from_format_cap() allow NULL cap.

Change-Id: Ie29760c49c25d7022ba2124698283181a0dd5d08

3 years agoSDP: Make ast_sdp_state_set_remote_sdp() return error.
Richard Mudgett [Mon, 24 Apr 2017 21:55:23 +0000 (16:55 -0500)]
SDP: Make ast_sdp_state_set_remote_sdp() return error.

Change-Id: I7707c9d872c476d897ff459008652b35142a35e1

3 years agoSDP: Misc cleanups (Mostly memory leaks)
Richard Mudgett [Fri, 14 Apr 2017 16:52:33 +0000 (11:52 -0500)]
SDP: Misc cleanups (Mostly memory leaks)

Change-Id: I74431b385da333f2c5f5a6d7c55e70b69a4f05d2

3 years Fix compile error.
Richard Mudgett [Thu, 27 Apr 2017 23:15:48 +0000 (18:15 -0500)] Fix compile error.

Change-Id: I6d9edd34d8b2474222c86f44e379ead61e57a54f

3 years agoMerge "channel: Add ability to request an outgoing channel with stream topology."
Jenkins2 [Thu, 27 Apr 2017 22:53:53 +0000 (17:53 -0500)]
Merge "channel: Add ability to request an outgoing channel with stream topology."

3 years agoMerge "frame: Better handle interpolated frames."
Jenkins2 [Thu, 27 Apr 2017 22:29:02 +0000 (17:29 -0500)]
Merge "frame: Better handle interpolated frames."

3 years agoMerge "res_pjsip_session: Add cleanup to ast_sip_session_terminate"
Jenkins2 [Thu, 27 Apr 2017 22:14:48 +0000 (17:14 -0500)]
Merge "res_pjsip_session:  Add cleanup to ast_sip_session_terminate"

3 years agoMerge "res_pjsip/res_pjsip_callerid: NULL check on caller id name string"
Jenkins2 [Thu, 27 Apr 2017 21:47:34 +0000 (16:47 -0500)]
Merge "res_pjsip/res_pjsip_callerid: NULL check on caller id name string"

3 years agoMerge "vector: defaults and indexes"
Jenkins2 [Thu, 27 Apr 2017 20:44:45 +0000 (15:44 -0500)]
Merge "vector: defaults and indexes"

3 years agoMerge "cleanup: Fix fread() and fwrite() error handling"
Joshua Colp [Thu, 27 Apr 2017 18:59:02 +0000 (13:59 -0500)]
Merge "cleanup: Fix fread() and fwrite() error handling"

3 years agoMerge "pjproject_bundled: Add --disable-libwebrtc to configure"
Jenkins2 [Thu, 27 Apr 2017 16:57:33 +0000 (11:57 -0500)]
Merge "pjproject_bundled:  Add --disable-libwebrtc to configure"

3 years agores_pjsip_session: Add cleanup to ast_sip_session_terminate
George Joseph [Thu, 27 Apr 2017 13:02:12 +0000 (07:02 -0600)]
res_pjsip_session:  Add cleanup to ast_sip_session_terminate

If you use ast_request to create a PJSIP channel but then hang it
up without causing a transaction to be sent, the session will
never be destroyed.  This is due ot the fact that it's pjproject
that triggers the session cleanup when the transaction ends.
app_chanisavail was doing this to get more granular channel state
and it's also possible for this to happen via ARI.

* ast_sip_session_terminate was modified to explicitly call the
  cleanup tasks and unreference session if the invite state is NULL
  AND invite_tsx is NULL (meaning we never sent a transaction).

* chan_pjsip/hangup was modified to bump session before it calls
  ast_sip_session_terminate to insure that session stays valid
  while it does its own cleanup.

* Added test events to session_destructor for a future testsuite

ASTERISK-26908 #close
Reported-by: Richard Mudgett

Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9

3 years agoMerge "res_rtp_asterisk.c: Fix crash in RTCP DTLS operation."
Jenkins2 [Thu, 27 Apr 2017 15:05:16 +0000 (10:05 -0500)]
Merge "res_rtp_asterisk.c: Fix crash in RTCP DTLS operation."

3 years agochannel: Add ability to request an outgoing channel with stream topology.
Joshua Colp [Mon, 24 Apr 2017 15:59:44 +0000 (15:59 +0000)]
channel: Add ability to request an outgoing channel with stream topology.

This change extends the ast_request functionality by adding another
function and callback to create an outgoing channel with a requested
stream topology. Fallback is provided by either converting the
requested stream topology into a format capabilities structure if
the channel driver does not support streams or by converting the
requested format capabilities into a stream topology if the channel
driver does support streams.

The Dial application has also been updated to request an outgoing
channel with the stream topology of the calling channel.


Change-Id: Ifa9037a672ac21d42dd7125aa09816dc879a70e6

3 years agoMerge "sdp: Add support for T.38"
Joshua Colp [Thu, 27 Apr 2017 10:38:14 +0000 (05:38 -0500)]
Merge "sdp: Add support for T.38"

3 years agoMerge "SDP: Ensure SDPs "merge" properly."
Joshua Colp [Thu, 27 Apr 2017 10:38:07 +0000 (05:38 -0500)]
Merge "SDP: Ensure SDPs "merge" properly."

3 years agores_pjsip/res_pjsip_callerid: NULL check on caller id name string
Kevin Harwell [Wed, 26 Apr 2017 19:20:00 +0000 (14:20 -0500)]
res_pjsip/res_pjsip_callerid: NULL check on caller id name string

It's possible for a name in a party id structure to be marked as valid, but the
name string itself be NULL (for instance this is possible to do by using the
dialplan CALLERID function). There were a couple of places where the name was
validated, but the string itself was not checked before passing it to functions
like 'strlen'. This of course caused a crashed.

This patch adds in a NULL check before attempting to pass it into a function
that is not NULL tolerant.

ASTERISK-25823 #close

Change-Id: Iaa6ffe9d92f598fe9e3c8ae373fadbe3dfbf1d4a

3 years agovector: defaults and indexes
Kevin Harwell [Tue, 25 Apr 2017 16:43:26 +0000 (11:43 -0500)]
vector: defaults and indexes

Added an pre-defined integer vector declaration. This makes integer vectors
easier to declare and pass around. Also, added the ability to default a vector
up to a given size with a default value. Lastly, added functionality that
returns the "nth" index of a matching value.

Also, updated a unit test to test these changes.

Change-Id: Iaf4b51b2540eda57cb43f67aa59cf1d96cdbcaa5

3 years agoframe: Better handle interpolated frames.
Joshua Colp [Wed, 26 Apr 2017 10:38:31 +0000 (10:38 +0000)]
frame: Better handle interpolated frames.

Interpolated frames are frames which contain a number of
samples but have no actual data. Audiohooks did not
handle this case when translating an incoming frame into
signed linear. It assumed that a frame would always contain
media when it may not. If this occurs audiohooks will now
immediately return and not act on the frame.

As well for users of ast_trans_frameout the function has
been changed to be a bit more sane and ensure that the data
pointer on a frame is set to NULL if no data is actually
on the frame. This allows the various spots in Asterisk that
check for an interpolated frame based on the presence of a
data pointer to work as expected.


Change-Id: I7fa22f631fa28d540722ed789ce28e84c7f8662b

3 years agoMerge "res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP."
Jenkins2 [Wed, 26 Apr 2017 15:44:00 +0000 (10:44 -0500)]
Merge "res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP."

3 years agopjproject_bundled: Add --disable-libwebrtc to configure
George Joseph [Wed, 26 Apr 2017 13:45:31 +0000 (07:45 -0600)]
pjproject_bundled:  Add --disable-libwebrtc to configure

Without the disable, pjproject tries to build it's internal
webrtc implementation which requires sse2.  This fails on
platforms without sse2.

ASTERISK-26930 #close
Reported-by: abelbeck

Change-Id: I07231f9160c35cfa42b194d3aad4e7d51fd9a410

3 years agoMerge "alembic: Add table for 'resource_list' PJSIP RLS type."
George Joseph [Wed, 26 Apr 2017 14:02:17 +0000 (09:02 -0500)]
Merge "alembic: Add table for 'resource_list' PJSIP RLS type."

3 years agoMerge "res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions."
Jenkins2 [Tue, 25 Apr 2017 22:04:47 +0000 (17:04 -0500)]
Merge "res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions."

3 years agoMerge "res_hep: Add additional config initialization and validation"
George Joseph [Tue, 25 Apr 2017 21:39:03 +0000 (16:39 -0500)]
Merge "res_hep: Add additional config initialization and validation"

3 years agocleanup: Fix fread() and fwrite() error handling
Sean Bright [Fri, 21 Apr 2017 17:04:44 +0000 (13:04 -0400)]
cleanup: Fix fread() and fwrite() error handling

Cleaned up some of the incorrect uses of fread() and fwrite(), mostly in
the format modules. Neither of these functions will ever return a value
less than 0, which we were checking for in some cases.

I've introduced a fair amount of duplication in the format modules, but
I plan to change how format modules work internally in a subsequent
patch set, so this is simply a stop-gap.

Change-Id: I8ca1cd47c20b2c0b72088bd13b9046f6977aa872

3 years agoMerge "res_pjsip_session.c: Restructure ast_sip_session_alloc()"
George Joseph [Tue, 25 Apr 2017 20:37:15 +0000 (15:37 -0500)]
Merge "res_pjsip_session.c: Restructure ast_sip_session_alloc()"

3 years agoalembic: Add table for 'resource_list' PJSIP RLS type.
Joshua Colp [Tue, 25 Apr 2017 12:52:48 +0000 (12:52 +0000)]
alembic: Add table for 'resource_list' PJSIP RLS type.

This change adds an Alembic migration which adds a
ps_resource_list table that can contain resource_list
RLS configuration objects.


Change-Id: I7c888fafc67b3e87012de974f71ca7a5b8b1ec05

3 years agosdp: Add support for T.38
Joshua Colp [Fri, 14 Apr 2017 10:21:13 +0000 (10:21 +0000)]
sdp: Add support for T.38

This change adds a T.38 format which can be used in a stream
topology to specify that a UDPTL stream needs to be created.
The SDP API has been changed to understand T.38 and create
the UDPTL session, add the attributes, and parse the attributes.

This change does not change the boundary of the T.38 state
machine. It is still up to the channel driver to implement and
act on it (such as queueing control frames or reacting to them).


Change-Id: If28956762ccb8ead562ac6c03d162d3d6014f2c7

3 years agoSDP: Ensure SDPs "merge" properly.
Mark Michelson [Tue, 21 Mar 2017 20:44:44 +0000 (15:44 -0500)]
SDP: Ensure SDPs "merge" properly.

The gist of this work ensures that when a remote SDP is received, it is
merged properly with the local capabilities. The remote SDP is converted
into a stream topology. That topology is then merged with the current
local topology on the SDP state. That new merged topology is then used
to create an SDP. Finally, adjustments are made to RTP instances based
on knowledge gained from the remote SDP.

There are also a battery of tests in this commit that ensure that some
basic SDP merges work as expected.

While this may not sound like a big change, it has the property that it
caused lots of ancillary changes.

* The remote SDP is no longer stored on the SDP state. Biggest reason:
  there's no need for it. The remote SDP is used at the time it is being
  set and nowhere else.

* Some new SDP APIs were added in order to find attributes and convert
  generic SDP attributes into rtpmap structures.

* Writing tests made me realize that retrieving a value from an SDP
  options structure, the SDP options needs to be made const.

* The SDP state machine was essentially gutted by a previous commit.
  Initially, I attempted to reinstate it, but I found that as it had
  been defined, it was not all that useful. What was more useful was
  knowing the role we play in SDP negotiation, so the SDP state machine
  has been transformed into an indicator of role.

* Rather than storing separate local and joint stream state
  capabilities, it makes more sense to keep track of current stream
  state and update it as things change.

Change-Id: I5938c2be3c6f0a003aa88a39a59e0880f8b2df3d

3 years agores_hep: Add additional config initialization and validation
Sean Bright [Mon, 24 Apr 2017 18:16:45 +0000 (14:16 -0400)]
res_hep: Add additional config initialization and validation

* Initialize hepv3_runtime_data.sockfd to -1 so that our ao2 destructor
  does not close fd 0

* Add logging output when the required option - capture_address - is not

* Remove a no longer relevant #define and correct related documentation

* Pass appropriate flags to aco_option_register so that capture_address
  cannot be the empty string.

ASTERISK-26953 #close

Change-Id: Ief08441bc6596d6f1718fa810e54a5048124f076

3 years agocore: Use eventfd for alert pipes on Linux when possible
Sean Bright [Tue, 18 Apr 2017 00:06:10 +0000 (20:06 -0400)]
core: Use eventfd for alert pipes on Linux when possible

The primary win of switching to eventfd when possible is that it only
uses a single file descriptor while pipe() will use two. This means for
each bridge channel we're reducing the number of required file
descriptors by 1, and - if you're using timerfd - we also now have 1
less file descriptor per Asterisk channel.

The API is not ideal (passing int arrays), but this is the cleanest
approach I could come up with to maintain API/ABI.

I've also removed what I believe to be an erroneous code block that
checked the non-blocking flag on the pipe ends for each read. If the
file descriptor is 'losing' its non-blocking mode, it is because of a
bug somewhere else in our code.

In my testing I haven't seen any measurable difference in performance.

Change-Id: Iff0fb1573e7f7a187d5211ddc60aa8f3da3edb1d

3 years agoMerge "pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified"
George Joseph [Fri, 21 Apr 2017 20:47:36 +0000 (15:47 -0500)]
Merge "pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified"

3 years agoMerge "rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes."
George Joseph [Fri, 21 Apr 2017 20:46:21 +0000 (15:46 -0500)]
Merge "rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes."

3 years agores_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions.
Richard Mudgett [Fri, 21 Apr 2017 17:33:34 +0000 (12:33 -0500)]
res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions.

If ICE is enabled and a STUN server does not respond then we will block
until we give up on the STUN response.  This will take nine seconds.  In
the mean time the peer that sent the INVITE will send retransmissions.

* Restructure res_pjsip_session.c:new_invite() to send a 100 Trying out
earlier to prevent these retransmissions.


Change-Id: Ie3fc611e53a0eff6586ad55e4aacad81cf6319a8

3 years agores_pjsip_session.c: Restructure ast_sip_session_alloc()
Richard Mudgett [Fri, 21 Apr 2017 17:07:39 +0000 (12:07 -0500)]
res_pjsip_session.c: Restructure ast_sip_session_alloc()

* Restructure ast_sip_session_alloc() to need less cleanup on off nominal
error paths.

* Made ast_sip_session_alloc() and ast_sip_session_create_outgoing() avoid
unnecessary ref manipulation to return a session.  This is faster than
calling a function.  That function may do logging of the ref changes with
REF_DEBUG enabled.

Change-Id: I2a0affc4be51013d3f0485782c96b8fee3ddb00a

3 years agoMerge "build: Update config.guess and config.sub"
George Joseph [Thu, 20 Apr 2017 18:34:45 +0000 (13:34 -0500)]
Merge "build: Update config.guess and config.sub"

3 years agoMerge "res_stun_monitor: Don't fail to load if DNS resolution fails"
George Joseph [Thu, 20 Apr 2017 12:19:46 +0000 (07:19 -0500)]
Merge "res_stun_monitor: Don't fail to load if DNS resolution fails"

3 years agoMerge "make ari-stubs so doc periodic jobs can run"
George Joseph [Thu, 20 Apr 2017 12:17:28 +0000 (07:17 -0500)]
Merge "make ari-stubs so doc periodic jobs can run"

3 years agopbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified
Sean Bright [Wed, 19 Apr 2017 20:08:39 +0000 (16:08 -0400)]
pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified

Both ast_pbx_outgoing_app() and ast_pbx_outgoing_exten() cause the core
to spawn a new thread to perform the dial. When AST_OUTGOING_WAIT_COMPLETE
is passed to these functions, the calling thread will be blocked until
the newly created channel has been hung up.

After this patch, we run the dial on the current thread rather than
spawning a new one. The only in-tree code that passes
AST_OUTGOING_WAIT_COMPLETE is pbx_spool, so you should see reduced
thread usage if you are using .call files.

Change-Id: I512735d243f0a9da2bcc128f7a96dece71f2d913

3 years agores_rtp_asterisk.c: Fix crash in RTCP DTLS operation.
Richard Mudgett [Wed, 19 Apr 2017 18:23:55 +0000 (13:23 -0500)]
res_rtp_asterisk.c: Fix crash in RTCP DTLS operation.

Occasionally a crash happens when processing the RTCP DTLS timeout
handler.  The RTCP DTLS timeout timer could be left running if we have not
completed the DTLS handshake before we place the call on hold or we
attempt direct media.

* Made ast_rtp_prop_set() stop the RTCP DTLS timer when disabling RTCP.

* Made some sanity tweaks to ast_rtp_prop_set() when switching from
standard RTCP mode to RTCP multiplexed mode.

ASTERISK-26692 #close

Change-Id: If6c64c79129961acfa4b3d63a864e8f6b664acc0

3 years agortp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes.
Richard Mudgett [Wed, 22 Mar 2017 21:05:49 +0000 (16:05 -0500)]
rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes.

The struct ast_rtp_instance has historically been indirectly protected
from reentrancy issues by the channel lock because early channel drivers
held the lock for really long times.  Holding the channel lock for such a
long time has caused many deadlock problems in the past.  Along comes
chan_pjsip/res_pjsip which doesn't necessarily hold the channel lock
because sometimes there may not be an associated channel created yet or
the channel pointer isn't available.

In the case of ASTERISK-26835 a pjsip serializer thread was processing a
message's SDP body while another thread was reading a RTP packet from the
socket.  Both threads wound up changing the rtp->rtcp->local_addr_str
string and interfering with each other.  The classic reentrancy problem
resulted in a crash.

In the case of ASTERISK-26853 a pjsip serializer thread was processing a
message's SDP body while another thread was reading a RTP packet from the
socket.  Both threads wound up processing ICE candidates in PJPROJECT and
interfering with each other.  The classic reentrancy problem resulted in a

* rtp_engine.c: Make the ast_rtp_instance_xxx() calls lock the RTP
instance struct.

* rtp_engine.c: Make ICE and DTLS wrapper functions to lock the RTP
instance struct for the API call.

* res_rtp_asterisk.c: Lock the RTP instance to prevent a reentrancy
problem with rtp->rtcp->local_addr_str in the scheduler thread running

* res_rtp_asterisk.c: Avoid deadlock when local RTP bridging in
bridge_p2p_rtp_write() because there are two RTP instance structs

* res_rtp_asterisk.c: Avoid deadlock when trying to stop scheduler
callbacks.  We cannot hold the instance lock when trying to stop a
scheduler callback.

* res_rtp_asterisk.c: Remove the lock in struct dtls_details and use the
struct ast_rtp_instance ao2 object lock instead.  The lock was used to
synchronize two threads to prevent a race condition between starting and
stopping a timeout timer.  The race condition is no longer present between
dtls_perform_handshake() and __rtp_recvfrom() because the instance lock
prevents these functions from overlapping each other with regards to the
timeout timer.

* res_rtp_asterisk.c: Remove the lock in struct ast_rtp and use the struct
ast_rtp_instance ao2 object lock instead.  The lock was used to
synchronize two threads using a condition signal to know when TURN
negotiations complete.

* res_rtp_asterisk.c: Avoid deadlock when trying to stop the TURN
ioqueue_worker_thread().  We cannot hold the instance lock when trying to
create or shut down the worker thread without a risk of deadlock.

This patch exposed a race condition between a PJSIP serializer thread
setting up an ICE session in ice_create() and another thread reading RTP

* res_rtp_asterisk.c:ice_create(): Set the new rtp->ice pointer after we
have re-locked the RTP instance to prevent the other thread from trying to
process ICE packets on an incomplete ICE session setup.

A similar race condition is between a PJSIP serializer thread resetting up
an ICE session in ice_create() and the timer_worker_thread() processing
the completion of the previous ICE session.

* res_rtp_asterisk.c:ast_rtp_on_ice_complete(): Protect against an
uninitialized/null remote_address after calling

* res_rtp_asterisk.c: Eliminate the chance of ice_reset_session()
destroying and setting the rtp->ice pointer to NULL while other threads
are using it by adding an ao2 wrapper around the PJPROJECT ice pointer.
Now when we have to unlock the RTP instance object to call a PJPROJECT ICE
function we will hold a ref to the wrapper.  Also added some rtp->ice NULL
checks after we relock the RTP instance and have to do something with the
ICE structure.

ASTERISK-26835 #close
ASTERISK-26853 #close

Change-Id: I780b39ec935dcefcce880d50c1a7261744f1d1b4

3 years agobuild: Update config.guess and config.sub
Sean Bright [Wed, 19 Apr 2017 13:39:00 +0000 (09:39 -0400)]
build: Update config.guess and config.sub

Change-Id: Id078a1df07a771808775e1053cdfe1d99c8fb172

3 years agoMerge "format_wav: Read 16khz wav samples properly"
Joshua Colp [Wed, 19 Apr 2017 13:38:59 +0000 (08:38 -0500)]
Merge "format_wav: Read 16khz wav samples properly"

3 years agoMerge "format_ogg_vorbis: Clear ogg/vorbis data structures on close"
Joshua Colp [Wed, 19 Apr 2017 13:38:16 +0000 (08:38 -0500)]
Merge "format_ogg_vorbis: Clear ogg/vorbis data structures on close"

3 years agoMerge "Revert "bridging: Ensure successful T.38 negotation""
Joshua Colp [Wed, 19 Apr 2017 13:37:17 +0000 (08:37 -0500)]
Merge "Revert "bridging:  Ensure successful T.38 negotation""

3 years agoformat_wav: Read 16khz wav samples properly
Sean Bright [Fri, 14 Apr 2017 18:52:59 +0000 (14:52 -0400)]
format_wav: Read 16khz wav samples properly

When opening a PCM wave file for reading, we aren't tracking the
frequency of the opened file, so we treat 16khz files as 8khz and do
half reads.

This patch also cleans up some of the data types and an unnecessarily
complex `if` expression.

ASTERISK-26613 #close
Reported by: Vitaly K

Change-Id: I05f8b263058dc573ea8ffe0c62e7964506e11815

3 years agomake ari-stubs so doc periodic jobs can run
George Joseph [Mon, 17 Apr 2017 00:59:54 +0000 (18:59 -0600)]
make ari-stubs so doc periodic jobs can run

The periodic doc job does a make ari-stubs and checks that
there are no changes before generating the docs.  Since I changed
the mustache template (and the generated code directly) recently
and forgot to regenerate the stubs, the doc job thinks they're out
of date.

Change-Id: I94b97035311eccf52b0101b8590223265a7881d4

3 years agoformat_ogg_vorbis: Clear ogg/vorbis data structures on close
Sean Bright [Fri, 14 Apr 2017 17:51:31 +0000 (13:51 -0400)]
format_ogg_vorbis: Clear ogg/vorbis data structures on close

On filestream close, we need to clear out the ogg & vorbis data
structures to prevent a memory leak.

ASTERISK-26169 #close
Reported by: Ivan Myalkin

Change-Id: Iee94c5a5d5bdafbf8b181c5c064d15d90ace8274

3 years agoRevert "bridging: Ensure successful T.38 negotation"
Richard Mudgett [Fri, 14 Apr 2017 22:32:22 +0000 (17:32 -0500)]
Revert "bridging:  Ensure successful T.38 negotation"

This reverts commit 7819f95791fe0ca0e0cdc417e2687a5900444053.

Change-Id: Ib91a7e6c9856f5f41329e42f40ba2394fee861a4

3 years agores_stun_monitor: Don't fail to load if DNS resolution fails
Sean Bright [Fri, 14 Apr 2017 21:50:56 +0000 (17:50 -0400)]
res_stun_monitor: Don't fail to load if DNS resolution fails

res_stun_monitor will fail to load if DNS resolution of the STUN server
fails. Instead, we continue without the STUN server being resolved and
we will re-attempt the resolution on the STUN refresh interval.

ASTERISK-21856 #close
Reported by: Jeremy Kister

Change-Id: I6334c54a1cc798f8a836b4b47948e0bb4ef59254

3 years agoformat_pcm: Track actual header size of .au files
Sean Bright [Fri, 14 Apr 2017 19:36:57 +0000 (15:36 -0400)]
format_pcm: Track actual header size of .au files

Sun's Au file format has a minimum data offset 24 bytes, but this
offset is encoded in each .au file. Instead of assuming the minimum,
read the actual value and store it for later use.

ASTERISK-20984 #close
Reported by: Roman S.
asterisk- (license #6474) patch
uploaded by Roman S.

Change-Id: I524022fb19ff2fd5af2cc2d669d27a780ab2057c

3 years agomodules: change module LOAD_FAILUREs to LOAD_DECLINES (master)
George Joseph [Wed, 12 Apr 2017 12:50:49 +0000 (06:50 -0600)]
modules:  change module LOAD_FAILUREs to LOAD_DECLINES (master)

Change-Id: Iac40ecb20e10513d67bf0eaf61807f306067b258

3 years agoMerge "modules: change module LOAD_FAILUREs to LOAD_DECLINES (14)"
Joshua Colp [Thu, 13 Apr 2017 12:06:51 +0000 (07:06 -0500)]
Merge "modules:  change module LOAD_FAILUREs to LOAD_DECLINES (14)"

3 years agoMerge "modules: change module LOAD_FAILUREs to LOAD_DECLINES"
zuul [Thu, 13 Apr 2017 12:03:42 +0000 (07:03 -0500)]
Merge "modules:  change module LOAD_FAILUREs to LOAD_DECLINES"

3 years agores_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP.
Alexander Traud [Mon, 10 Apr 2017 10:13:39 +0000 (12:13 +0200)]
res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP.

This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in
SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over
UDP, if many codecs are allowed in Asterisk. This new feature is enabled
together with the optional feature compact_headers=yes via the file pjsip.conf.

ASTERISK-26932 #close

Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689

3 years agoMerge "strings.h: Avoid overflows in the string hash functions"
Joshua Colp [Thu, 13 Apr 2017 00:46:06 +0000 (19:46 -0500)]
Merge "strings.h:  Avoid overflows in the string hash functions"

3 years agomodules: change module LOAD_FAILUREs to LOAD_DECLINES (14)
George Joseph [Wed, 12 Apr 2017 12:47:59 +0000 (06:47 -0600)]
modules:  change module LOAD_FAILUREs to LOAD_DECLINES (14)

Change-Id: If99e3b4fc2d7e86fc3e61182aa6c835b407ed49e

3 years agomodules: change module LOAD_FAILUREs to LOAD_DECLINES
George Joseph [Tue, 11 Apr 2017 16:07:39 +0000 (10:07 -0600)]
modules:  change module LOAD_FAILUREs to LOAD_DECLINES

In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE.  This prevents asterisk from exiting
if a module can't be loaded.  If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.

A new API was added to logger: ast_is_logger_initialized().  This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout.  If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.

Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25

3 years agoMerge "bridging: Ensure successful T.38 negotation"
zuul [Wed, 12 Apr 2017 16:22:19 +0000 (11:22 -0500)]
Merge "bridging:  Ensure successful T.38 negotation"

3 years agoMerge "res_rtp_asterisk.c: Add stun_blacklist option"
zuul [Wed, 12 Apr 2017 14:55:42 +0000 (09:55 -0500)]
Merge "res_rtp_asterisk.c: Add stun_blacklist option"

3 years agobridging: Ensure successful T.38 negotation
Torrey Searle [Wed, 5 Apr 2017 11:41:29 +0000 (13:41 +0200)]
bridging:  Ensure successful T.38 negotation

When a T.38 happens immediatly after call establishment, the control
frame can be lost because the other leg is not yet in the bridge.

This patch detects this case an makes sure T.38 negotation happens
when the 2nd leg is being made compatible with the negotating
first leg

ASTERISK-26923 #close

Change-Id: If334125ee61ed63550d242fc9efe7987e37e1d94

3 years agoMerge "stun.c: Fix ast_stun_request() erratic timeout."
Joshua Colp [Wed, 12 Apr 2017 09:54:33 +0000 (04:54 -0500)]
Merge "stun.c: Fix ast_stun_request() erratic timeout."

3 years agoMerge "sorcery.c: Speed up ast_sorcery_retrieve_by_id()"
zuul [Wed, 12 Apr 2017 01:12:55 +0000 (20:12 -0500)]
Merge "sorcery.c: Speed up ast_sorcery_retrieve_by_id()"

3 years agoMerge "res_pjsip: Fix pointer use after unref."
zuul [Wed, 12 Apr 2017 01:12:53 +0000 (20:12 -0500)]
Merge "res_pjsip: Fix pointer use after unref."

3 years agoMerge "res_pjsip_sdp_rtp.c: Don't use deprecated transport struct member."
zuul [Wed, 12 Apr 2017 01:12:50 +0000 (20:12 -0500)]
Merge "res_pjsip_sdp_rtp.c: Don't use deprecated transport struct member."

3 years agostrings.h: Avoid overflows in the string hash functions
Torrey Searle [Fri, 7 Apr 2017 13:58:23 +0000 (15:58 +0200)]
strings.h:  Avoid overflows in the string hash functions

On 2's compliment machines abs(INT_MIN) behavior is undefined and
results in a negative value still being returnd.  This results in
negative hash codes that can result in crashes.

ASTERISK-26528 #close

Change-Id: Idff550145ca2133792a61a2e212b4a3e82c6517b

3 years agores_rtp_asterisk.c: Add stun_blacklist option
Richard Mudgett [Fri, 7 Apr 2017 21:14:16 +0000 (16:14 -0500)]
res_rtp_asterisk.c: Add stun_blacklist option

Added the stun_blacklist option to rtp.conf.  Some multihomed servers have
IP interfaces that cannot reach the STUN server specified by stunaddr.
Blacklist those interface subnets from trying to send a STUN packet to
find the external IP address.  Attempting to send the STUN packet
needlessly delays processing incoming and outgoing SIP INVITEs because we
will wait for a response that can never come until we give up on the
response.  Multiple subnets may be listed.

ASTERISK-26890 #close

Change-Id: I3ff4f729e787f00c3e6e670fe6435acce38be342

3 years agostun.c: Fix ast_stun_request() erratic timeout.
Richard Mudgett [Thu, 6 Apr 2017 22:31:14 +0000 (17:31 -0500)]
stun.c: Fix ast_stun_request() erratic timeout.

If ast_stun_request() receives packets other than a STUN response then we
could conceivably never exit if we continue to receive packets with less
than three seconds between them.

* Fix poll timeout to keep track of the time when we sent the STUN
request.  We will now send a STUN request every three seconds regardless
of how many other packets we receive while waiting for a response until we
have completed three STUN request transmission cycles.

Change-Id: Ib606cb08585e06eb50877f67b8d3bd385a85c266

3 years agosorcery.c: Speed up ast_sorcery_retrieve_by_id()
Richard Mudgett [Thu, 6 Apr 2017 23:30:11 +0000 (18:30 -0500)]
sorcery.c: Speed up ast_sorcery_retrieve_by_id()

Return early if ast_sorcery_retrieve_by_id() is not passed an id to find.
Also eliminated the RAII_VAR() usage in the function.

Change-Id: I871dbe162a301b5ced8b4393cec27180c7c6b218

3 years agores_pjsip: Fix pointer use after unref.
Richard Mudgett [Mon, 10 Apr 2017 16:30:35 +0000 (11:30 -0500)]
res_pjsip: Fix pointer use after unref.

Change-Id: I4b6e1b0070563eeaee223cb58326f1b962ed5bc1

3 years agores_pjsip_sdp_rtp.c: Don't use deprecated transport struct member.
Richard Mudgett [Thu, 6 Apr 2017 23:18:16 +0000 (18:18 -0500)]
res_pjsip_sdp_rtp.c: Don't use deprecated transport struct member.

* create_rtp(): Eliminate use of deprecated transport struct member.  That
member and several others in the transport structure were deprecated
because of an infinite loop created when using realtime configuration.
See 2451d4e4550336197ee2e482750cc53f30afa352


Change-Id: I0533aa13c9ce3c6cc394e0fd2b5bf1cd1b2ef3bc

3 years agotcptls.c: Cleanup TCP/TLS listener thread on abnormal exit.
Richard Mudgett [Mon, 10 Apr 2017 22:45:35 +0000 (17:45 -0500)]
tcptls.c: Cleanup TCP/TLS listener thread on abnormal exit.

Temporarily running out of file descriptors should not terminate the
listener thread.  Otherwise, when there becomes more file descriptors
available, nothing is listening.

* Added EMFILE exception to abnormal thread exit.

* Added an abnormal TCP/TLS listener exit error message.

* Closed the TCP/TLS listener socket on abnormal exit so Asterisk does not
appear dead if something tries to connect to the socket.

ASTERISK-26903 #close

Change-Id: I10f2f784065136277f271159f0925927194581b5

3 years agosamples: Undo removal of include from canonicalize-app-names commit.
Walter Doekes [Sat, 8 Apr 2017 08:05:03 +0000 (10:05 +0200)]
samples: Undo removal of include from canonicalize-app-names commit.

This include was accidentally removed in changeset
Ia79aea64de89531362e993e34230c2044a70aa93. My bad.

Change-Id: I1d716c7f9590b4e97909fb8bca1f2ed9bd0e4082

3 years agoMerge "pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete()."
Joshua Colp [Fri, 7 Apr 2017 22:33:12 +0000 (17:33 -0500)]
Merge "pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete()."

3 years agoMerge "pjsip: Add Alembic for PUBLISH support."
zuul [Fri, 7 Apr 2017 20:26:00 +0000 (15:26 -0500)]
Merge "pjsip: Add Alembic for PUBLISH support."

3 years agoMerge "samples: Canonicalize app names in extensions.conf.sample."
Joshua Colp [Fri, 7 Apr 2017 19:49:11 +0000 (14:49 -0500)]
Merge "samples: Canonicalize app names in extensions.conf.sample."

3 years agoMerge "core: Improve/simplify handling of required headers."
Joshua Colp [Fri, 7 Apr 2017 19:48:36 +0000 (14:48 -0500)]
Merge "core: Improve/simplify handling of required headers."

3 years agopjsip: Add Alembic for PUBLISH support.
Joshua Colp [Fri, 7 Apr 2017 13:35:33 +0000 (13:35 +0000)]
pjsip: Add Alembic for PUBLISH support.

This change adds database tables for the PUBLISH support so it
can be configured using realtime. A minor fix to the
res_pjsip_publish_asterisk module was done so that it read the
sorcery configuration from the correct section. Finally the
sample configuration files have been updated.


Change-Id: I81991ae5c75af98d247f7eacd1c0b0a763675952

3 years agopjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
Alexander Traud [Fri, 7 Apr 2017 13:06:11 +0000 (15:06 +0200)]
pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().

When the Asterisk channel driver res_pjsip offers SIP-over-TLS, sometimes, not
reproducible, Asterisk crashed in pj_ssl_sock_get_info() because a NULL pointer
was read. This change avoids this crash.

ASTERISK-26927 #close

Change-Id: I24a6011b44d1426d159742ff4421cf806a52938b

3 years agosamples: Canonicalize app names in extensions.conf.sample.
Walter Doekes [Wed, 5 Apr 2017 14:10:42 +0000 (16:10 +0200)]
samples: Canonicalize app names in extensions.conf.sample.

This takes care of warnings by ossobv/asterisklint.

Change-Id: Ia79aea64de89531362e993e34230c2044a70aa93

3 years agoMerge "pjproject_bundled: Add 3 upstream patches"
Joshua Colp [Thu, 6 Apr 2017 17:23:47 +0000 (12:23 -0500)]
Merge "pjproject_bundled: Add 3 upstream patches"

3 years agoMerge "chan_sip: Session Timers required but refused wrongly."
Joshua Colp [Thu, 6 Apr 2017 16:39:41 +0000 (11:39 -0500)]
Merge "chan_sip: Session Timers required but refused wrongly."

3 years agoMerge "app_queue: Log reason for PAUSEALL/UNPAUSEALL"
zuul [Thu, 6 Apr 2017 15:19:33 +0000 (10:19 -0500)]
Merge "app_queue: Log reason for PAUSEALL/UNPAUSEALL"

3 years agoMerge "sample_config: Add samples for pubsub to pjsip.conf.sample"
Joshua Colp [Thu, 6 Apr 2017 13:29:39 +0000 (08:29 -0500)]
Merge "sample_config:  Add samples for pubsub to pjsip.conf.sample"

3 years agoMerge "Unused realtime MOH classes not purged on 'moh reload'"
zuul [Thu, 6 Apr 2017 00:14:28 +0000 (19:14 -0500)]
Merge "Unused realtime MOH classes not purged on 'moh reload'"

3 years agoMerge "res_pjsip_session: Allow BYE to be sent on disconnected session."
zuul [Thu, 6 Apr 2017 00:04:25 +0000 (19:04 -0500)]
Merge "res_pjsip_session: Allow BYE to be sent on disconnected session."

3 years agopjproject_bundled: Add 3 upstream patches
George Joseph [Tue, 4 Apr 2017 21:20:22 +0000 (15:20 -0600)]
pjproject_bundled: Add 3 upstream patches


Also removed the progress bar from wget download to stdout.

ASTERISK-26905 #close
Reported-by: Ross Beer

Change-Id: I268fb3cf71a3bb24283ff0d24bd8b03239d81256

3 years agoapp_queue: Log reason for PAUSEALL/UNPAUSEALL
Troy Bowman [Tue, 4 Apr 2017 16:44:34 +0000 (10:44 -0600)]
app_queue: Log reason for PAUSEALL/UNPAUSEALL

We needed the reason for our reporting when agents pause/unpause all of
their queues at once.  This is a small, simple patch that adds a reason
for PAUSEALL and UNPAUSEALL.  I have been using it in production for years.

ASTERISK-26920 #close

Change-Id: Ifb3f0d1a0abd5194253d9794023546e1395baf3d

3 years agosample_config: Add samples for pubsub to pjsip.conf.sample
George Joseph [Wed, 5 Apr 2017 19:50:40 +0000 (13:50 -0600)]
sample_config:  Add samples for pubsub to pjsip.conf.sample

 * outbound-publish
 * resource_list
 * inbound-publication
 * asterisk-publication

Change-Id: I65043a896c35483f30a92d30b5b118359af7ba5a

3 years agores_pjsip_sdp_rtp.c: Don't alter global addr variable.
Richard Mudgett [Mon, 3 Apr 2017 20:38:06 +0000 (15:38 -0500)]
res_pjsip_sdp_rtp.c: Don't alter global addr variable.

* create_rtp(): Fix unexpected alteration of global address_rtp if a
transport is bound to an address.

* create_rtp(): Fix use of uninitialized memory if the endpoint RTP media
address is invalid or the transport has an invalid address.


Change-Id: Icde42e65164a88913cb5c2601b285eebcff397b7

3 years agoCDR: Protect from data overflow in ast_cdr_setuserfield.
Corey Farrell [Mon, 27 Mar 2017 14:03:49 +0000 (10:03 -0400)]
CDR: Protect from data overflow in ast_cdr_setuserfield.

ast_cdr_setuserfield wrote to a fixed length field using strcpy. This could
result in a buffer overrun when called from chan_sip or func_cdr. This patch
adds a maximum bytes written to the field by using ast_copy_string instead.

ASTERISK-26897 #close
  0001-CDR-Protect-from-data-overflow-in-ast_cdr_setuserfie.patch submitted
    by Corey Farrell (license #5909)

Change-Id: Ib23ca77e9b9e2803a450e1206af45df2d2fdf65c

3 years agoUnused realtime MOH classes not purged on 'moh reload'
Daniel Journo [Sun, 26 Mar 2017 00:01:06 +0000 (00:01 +0000)]
Unused realtime MOH classes not purged on 'moh reload'

Purge Realtime MOH classes on 'moh reload' even when musiconhold.conf
hasn't changed.

ASTERISK-25974 #close

Change-Id: I42c78ea76528473a656f204595956c9eedcf3246

3 years agocore: Improve/simplify handling of required headers.
Corey Farrell [Fri, 31 Mar 2017 17:09:38 +0000 (13:09 -0400)]
core: Improve/simplify handling of required headers.

* Report failures if configure finds a required header is missing.
* Deduplicate includes between asterisk.h, astmm.h and compat.h.
* Unconditionally include headers in compat.h if required elsewhere.

Change-Id: Ie67d0185ca71fbfb81c9bdfaebe46a49e3c56dc5

3 years agores_pjsip: Fix transport ref leak.
Richard Mudgett [Mon, 3 Apr 2017 18:56:43 +0000 (13:56 -0500)]
res_pjsip: Fix transport ref leak.

We were leaking a transport ref in multihomed_on_rx_message() which
resulted in the FRACK about excessive ref counts.

ASTERISK-26916 #close

Change-Id: I7a96658a9614a060565bb9ad51cb1c9c11ee145f

3 years agoMerge "sdp: Add support for setting connection address and clean up state."
Mark Michelson [Mon, 3 Apr 2017 14:32:08 +0000 (09:32 -0500)]
Merge "sdp: Add support for setting connection address and clean up state."

3 years agochan_sip: Session Timers required but refused wrongly.
Alexander Traud [Mon, 3 Apr 2017 07:30:43 +0000 (09:30 +0200)]
chan_sip: Session Timers required but refused wrongly.

SIP user-agents indicate which protocol extensions are allowed in headers
like Supported and Required. Such protocol extensions are Session Timers
(RFC 4028) for example. Session Timers are supported since Mantis-10665.
Since ASTERISK-21721, not only the first but multiple Supported/Required
headers in a message are parsed. In that change, an existing variable was
re-used within a newly added do-loop. Currently, at the end of that loop,
that variable is an empty string always. Previously, that variable was used
within log output. However, the log output was not changed.

ASTERISK-26915 #close

Change-Id: I09315f31b4d78fb214bb2a9fb6c0f5e143eae990

3 years agoMerge "Forward declare 'struct ast_json' in asterisk.h"
zuul [Sat, 1 Apr 2017 13:46:14 +0000 (08:46 -0500)]
Merge "Forward declare 'struct ast_json' in asterisk.h"

3 years agores_pjsip_session: Allow BYE to be sent on disconnected session.
Joshua Colp [Fri, 31 Mar 2017 21:31:24 +0000 (21:31 +0000)]
res_pjsip_session: Allow BYE to be sent on disconnected session.

It is perfectly acceptable for a BYE to be sent on a disconnected
session. This occurs when we respond to a challenge to the BYE
for authentication credentials.


Change-Id: I6ef0ddece812fea6665a1dd2549ef44fb9d90045

3 years Fix compiler error.
Richard Mudgett [Fri, 31 Mar 2017 18:14:05 +0000 (13:14 -0500)] Fix compiler error.

Added missing channel technology read/write stream callback

Change-Id: I829043a327d987e0d964485dd3d27964bebbd623

3 years agoMerge "cdr_pgsql: Fix buffer overflow calling libpq"
zuul [Fri, 31 Mar 2017 14:01:23 +0000 (09:01 -0500)]
Merge "cdr_pgsql: Fix buffer overflow calling libpq"