asterisk/asterisk.git
3 years agores/res_pjsip_session: Check for presence of an active negotiator
Matt Jordan [Sat, 25 Jun 2016 00:55:09 +0000 (19:55 -0500)]
res/res_pjsip_session: Check for presence of an active negotiator

It is possible in a hypothetical situation for a session refresh to be
invoked on a PJSIP when the negotiatior on the INVITE session has not
yet been established. While this shouldn't occur with existing uses of
ast_sip_session_refresh, the crashes that occur due to improperly
calling PJSIP functions that expect a non-NULL negotiatior are
avoidable. PJSIP will create the negotiator in pjsip_inv_reinvite; this
means that simply checking for the presence of the negotiator before
passing it to other PJSIP functions that use it is allowable. As such,
this patch adds checks for the presence of the negotiator before calling
PJSIP functions that assume it is non-NULL.

Change-Id: I1028323e7e01b0a531865e5412a71b6f6ec4276d

3 years agores/res_pjsip_pubsub: Add additional debug statements
Matt Jordan [Mon, 19 Oct 2015 23:55:58 +0000 (18:55 -0500)]
res/res_pjsip_pubsub: Add additional debug statements

When something very sad and wrong occurs, it's challenging sometimes to
figure out why. This patch adds some additional debug statements on
off-nominal paths to try and make debugging easier.

Change-Id: I7bffb73cc733b6f80193a23340881db4a102b640

3 years agores/res_corosync: Raise a Stasis message on node join/leave events
Matt Jordan [Mon, 19 Oct 2015 23:55:33 +0000 (18:55 -0500)]
res/res_corosync: Raise a Stasis message on node join/leave events

When res_corosync detects that a node leaves or joins, it currently is
informed of this via Corosync callbacks. However, there are a few
limitations with the information presented:
(1) While we have information that Corosync is aware of - such as the
    Corosync nodeid - that information is really only useful inside of
    Corosync or res_corosync. There's no way to translate a Corosync
    nodeid to some other internally useful unique identifier for the
    Asterisk instance that just joined or left the cluster.
(2) While res_corosync is notified of the instance joining or leaving
    the cluster, it has no mechanism to inform the Asterisk core or
    other modules of this event. This limits the usefulness of res_corosync
    as a heartbeat mechanism for other modules.

This patch addresses both issues.

First, it adds the notion of a cluster discovery message both within the
Stasis message bus, as well as the binary event messages that
res_corosync uses to transmit data back and forth within the cluster.
When Asterisk joins the cluster, it sends a discovery message to the other
nodes in the cluster, which correlates the Corosync nodeid along with
the Asterisk EID. res_corosync now maintains a hash of Corosync nodeids
to Asterisk EIDs, such that it can map changes in cluster state with the
Asterisk instance that has that nodeid. Likewise, when an Asterisk
instance receives a discovery message from a node in the cluster, it now
sends its own discovery message back to the originating node with the
local Asterisk EID. This lets Asterisk instances within the cluster
build a complete picture of the other Asterisk instances within the
cluster.

Second, it publishes the discovery messages onto the Stasis message bus.
Said messages are published whenever a node joins or leaves the cluster.
Interested modules can subscribe for the ast_cluster_discovery_type()
message under the ast_system_topic() and be notified when changes in
cluster state occur.

Change-Id: I9015f418d6ae7f47e4994e04e18948df4d49b465

3 years agoBuildSystem: Avoid obsolete warning with pthread.m4 on autoconf.
Alexander Traud [Wed, 13 Jul 2016 13:57:08 +0000 (15:57 +0200)]
BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf.

Updated the macro-set autoconf/ax_pthread.m4 to its latest upstream version.

ASTERISK-26046 #close

Change-Id: I11abc11d17acd2b6a8a5a5be8ae8e0949dab9cc7

3 years agoMerge "rest_api/channels: Fix multiple issues with create and dial"
zuul [Wed, 13 Jul 2016 13:08:41 +0000 (08:08 -0500)]
Merge "rest_api/channels:  Fix multiple issues with create and dial"

3 years agoMerge "res_pjsip: Fix statsd regression."
Joshua Colp [Wed, 13 Jul 2016 12:41:47 +0000 (07:41 -0500)]
Merge "res_pjsip: Fix statsd regression."

3 years agoMerge "BuildSystem: Allow own CFLAGS on ./configure."
Joshua Colp [Wed, 13 Jul 2016 11:42:57 +0000 (06:42 -0500)]
Merge "BuildSystem: Allow own CFLAGS on ./configure."

3 years agoMerge "install_prereq: Checkout of libSRTP 1.5.x."
Joshua Colp [Wed, 13 Jul 2016 00:30:38 +0000 (19:30 -0500)]
Merge "install_prereq: Checkout of libSRTP 1.5.x."

3 years agoMerge "chan_sip: Fix reference leaks in error paths."
Joshua Colp [Tue, 12 Jul 2016 23:49:13 +0000 (18:49 -0500)]
Merge "chan_sip: Fix reference leaks in error paths."

3 years agoMerge "res_sorcery_realtime: fix bug when successful UPDATE is treated as failed"
Joshua Colp [Tue, 12 Jul 2016 22:43:45 +0000 (17:43 -0500)]
Merge "res_sorcery_realtime: fix bug when successful UPDATE is treated as failed"

3 years agoMerge "res_pjsip: Added "subscribe_context" to endpoint"
Joshua Colp [Tue, 12 Jul 2016 22:14:23 +0000 (17:14 -0500)]
Merge "res_pjsip: Added "subscribe_context" to endpoint"

3 years agoMerge "BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf."
Joshua Colp [Tue, 12 Jul 2016 21:04:55 +0000 (16:04 -0500)]
Merge "BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf."

3 years agorest_api/channels: Fix multiple issues with create and dial
George Joseph [Tue, 12 Jul 2016 01:07:20 +0000 (19:07 -0600)]
rest_api/channels:  Fix multiple issues with create and dial

* We weren't properly subscribing to the channel and it's originator
  on create.
* We weren't doing a publish_dial after calling ast_call on dial.
* We weren't calling depart_bridge when a channel left the dial bridge.

The first 2 issues were causing events to not be generated and the third
was actually causing channels to not get properly destroyed when hung up.

Together these 3 issues were causing the new
rest_apichannels/create_dial_bridge tests to fail.

As a result of the fixes, the cdr state machine had to be slightly
tweaked to allow bridge leave events without asserting and the tests
themselves had to be updated to account for the channels now cleaning
themselves up.

Change-Id: Ibf23abf5a62de76e82afb4461af5099c961b97d8

3 years agores_pjsip: Fix statsd regression.
Richard Mudgett [Mon, 11 Jul 2016 15:25:04 +0000 (10:25 -0500)]
res_pjsip: Fix statsd regression.

The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f
patch introduced several regressions when the newly created "Updated"
state goes out for each endpoint registration refresh.

1) It restarted any OPTIONS RTT ping cycle.

2) It would interfere with a currently active ping and throw off that
ping's resulting RTT calculation.

3) It cleared the RTT time each time the endpoint was refreshed.

4) The cleared RTT time was sent out as a statsd update each time.

5) It created two AMI events for each update.

* Revert the original patch and reimplement it.  Now the current contact
status state is re-sent instead of the state being momentarily toggled
every time the endpoint refreshes its registration.  The statsd events are
not created for the re-sent refresh because they are sent after every
OPTIONS ping.

ASTERISK-26160 #close
Reported by: Matt Jordan

Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1

3 years agofunc_odbc: Fix connection deadlock.
Joshua Colp [Mon, 11 Jul 2016 00:08:28 +0000 (21:08 -0300)]
func_odbc: Fix connection deadlock.

The func_odbc module was modified to ensure that the
previous behavior of using a single database connection
was maintained. This was done by getting a single database
connection and holding on to it. With the new multiple
connection support in res_odbc this will actually starve
every other thread from getting access to the database as
it also maintains the previous behavior of having only
a single database connection.

This change disables the func_odbc specific behavior if
the res_odbc module is running with only a single database
connection active. The connection is only kept for the
duration of the request.

ASTERISK-26177 #close

Change-Id: I9bdbd8a300fb3233877735ad3fd07bce38115b7f

3 years agoBuildSystem: Allow own CFLAGS on ./configure.
Alexander Traud [Tue, 12 Jul 2016 08:50:22 +0000 (10:50 +0200)]
BuildSystem: Allow own CFLAGS on ./configure.

Before this change, make failed with the error
Unknown value '' found in build_tools/menuselect-deps for NATIVE_ARCH
when CFLAGS were supplied to the configure script. This was introduced with
<https://reviewboard.asterisk.org/r/1852/> which disabled BUILD_NATIVE when
CFLAGS were supplied. Those who need different -march= values, please, go for
./configure
make menuselect.makeopts or make menuselect
./menuselect/menuselect --disable BUILD_NATIVE

ASTERISK-25289 #close

Change-Id: Ic6365d5a97bb9b3556858f06432a8d1cfa83eebc

3 years agoast_expr2: Fix off-nominal memory leak.
Richard Mudgett [Mon, 11 Jul 2016 18:42:55 +0000 (13:42 -0500)]
ast_expr2: Fix off-nominal memory leak.

Thanks to ibercom for pointing out a memory leak that was missed
in the earlier patch for the issue.

ASTERISK-26119
Reported by: Alexei Gradinari

Change-Id: I9a151f5c4725d97fb82a9e938bc73dc659532b71

3 years agoinstall_prereq: Checkout of libSRTP 1.5.x.
Alexander Traud [Mon, 11 Jul 2016 15:17:47 +0000 (17:17 +0200)]
install_prereq: Checkout of libSRTP 1.5.x.

Since 5th November 2014, the master branch of libSRTP changed the prefix of
several member names and is not compatible with the source code in Asterisk
anymore. Therefore instead, this change checks out the latest version of the
libSRTP 1.5.x branch. Furthermore now, libSRTP is compiled with OpenSSL as
backend. This makes AES-GCM and AES-IN possible.

ASTERISK-22131 #close

Change-Id: I2e396cdc01da0ff610686e398ed210ca7408f7d6

3 years agochan_sip: Fix reference leaks in error paths.
Corey Farrell [Sat, 9 Jul 2016 18:32:27 +0000 (14:32 -0400)]
chan_sip: Fix reference leaks in error paths.

* get_sip_pvt_from_replaces leaks sip_pvt_ptr on any error.
* build_peer leaks peer on failure to allocate the endpoint.

This patch fixes get_sip_pvt by using an RAII_VAR, build_peer is fixed
with an unref in the appropriate place.

ASTERISK-26184 #close

Change-Id: I728b424648ad041409f7d90880f4c28b3ce2ca12

3 years agoMerge "chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled."
Joshua Colp [Fri, 8 Jul 2016 20:21:35 +0000 (15:21 -0500)]
Merge "chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled."

3 years agoMerge "REF_DEBUG: Prevent logging of container node objects."
Joshua Colp [Fri, 8 Jul 2016 12:09:25 +0000 (07:09 -0500)]
Merge "REF_DEBUG: Prevent logging of container node objects."

3 years agoREF_DEBUG: Prevent logging of container node objects.
Corey Farrell [Thu, 7 Jul 2016 17:44:39 +0000 (13:44 -0400)]
REF_DEBUG: Prevent logging of container node objects.

Using AO2_CONTAINER_ALLOC_OPT_DUPS_REPLACE can result in an unref being
recorded to the refs log for the node being replaced.  This prevents
logging of those unrefs since they would produce errors in
refcounter.py.

ASTERISK-26181 #close

Change-Id: Ie4fded84e8a1a58b3a59ce59dfd7eb0da3ddc5d4

3 years agores_sorcery_realtime: fix bug when successful UPDATE is treated as failed
Alexei Gradinari [Mon, 4 Jul 2016 21:38:57 +0000 (17:38 -0400)]
res_sorcery_realtime: fix bug when successful UPDATE is treated as failed

If the SQL UPDATE statement changes nothing then SQLRowCount returns 0.
This value should be treated as success.
But the function sorcery_realtime_update treats it as failed.

This bug was found using stress tests on PJSIP.
If there are 2 consecutive SIP REGISTER requests with the same contact data
during 1 second then res_pjsip_registrar adds contact location on 1st request
and tries to update contact location on 2nd.
The update fails and res_pjsip_registrar even removes correct contact location.

The test "object_update_uncreated" was removed from test_sorcery_realtime.c
because it's now a valid situation.

This patch also adds missing debug of extra SQL parameter.

ASTERISK-26172 #close

Change-Id: I05a7f3051455336c9dda29efc229decf86071303

3 years agochan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled.
Joshua Colp [Thu, 7 Jul 2016 15:38:45 +0000 (12:38 -0300)]
chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled.

Some T.38 implementations may send another re-invite after the initial
one which adds additional negotiation details (such as the max bitrate).
Currently this will fail when passthrough is being done in chan_sip as we
do nothing if T.38 is already active.

Other handlers of T.38 inside of Asterisk (such as res_fax) handle this
scenario so this change adds support for it to chan_sip and res_pjsip_t38.
If a request to negotiate is received while T.38 is already enabled a
new re-INVITE is sent and negotiation is done again.

ASTERISK-26179 #close

Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c

3 years agoPJSIP: provide valid tcp nodelay option for reuse
Scott Griepentrog [Thu, 7 Jul 2016 15:55:42 +0000 (10:55 -0500)]
PJSIP: provide valid tcp nodelay option for reuse

When using TCP transport with chan_pjsip, the TCP_NODELAY
option value was allocated on the stack, then passed as a
pointer to the tcp transport configuration structure, and
later re-used on subsequently created sockets when it was
no longer valid.  This patch changes the allocation to be
a static.

ASTERISK-26180 #close
Reported by: Scott Griepentrog

Change-Id: I3251164c7f710dbdab031282f00e30a9770626a0

3 years agores_pjsip: Added "subscribe_context" to endpoint
Alexei Gradinari [Wed, 6 Jul 2016 14:29:27 +0000 (10:29 -0400)]
res_pjsip: Added "subscribe_context" to endpoint

If specified, incoming SUBSCRIBE requests will be searched for the matching
extension in the indicated context. If no "subscribe_context" is specified,
then the "context" setting is used.

ASTERISK-25471 #close

Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514

3 years agoBuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf.
Alexander Traud [Mon, 4 Jul 2016 10:58:39 +0000 (12:58 +0200)]
BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf.

Updated the macro-set autoconf/libcurl.m4 to its latest upstream version. This
avoids a warning about an obsolete macro on AC_HELP_STRING, because Asterisk is
using AS_HELP_STRING everywhere else already.

ASTERISK-26046

Change-Id: I8299faf504ceaeee3e39930c59293809e116c631

3 years agoMerge "res_pjsip_session.c: Don't send extra BYE if SDP invalid."
Joshua Colp [Fri, 1 Jul 2016 16:37:03 +0000 (11:37 -0500)]
Merge "res_pjsip_session.c: Don't send extra BYE if SDP invalid."

3 years agoMerge "res_pjsip_session.c: End call on initial invalid SDP negotiation."
Joshua Colp [Fri, 1 Jul 2016 16:36:58 +0000 (11:36 -0500)]
Merge "res_pjsip_session.c: End call on initial invalid SDP negotiation."

3 years agoMerge "res_pjsip.c: Register PJMEDIA error code decoder."
Joshua Colp [Fri, 1 Jul 2016 16:36:53 +0000 (11:36 -0500)]
Merge "res_pjsip.c: Register PJMEDIA error code decoder."

3 years agoMerge "res_pjsip_session.c: Remove unused parameter from handle_incoming()."
Joshua Colp [Fri, 1 Jul 2016 16:36:48 +0000 (11:36 -0500)]
Merge "res_pjsip_session.c: Remove unused parameter from handle_incoming()."

3 years agoMerge "res_pjsip: Add missing NULL checks when using pjsip_inv_end_session()."
Joshua Colp [Fri, 1 Jul 2016 16:36:42 +0000 (11:36 -0500)]
Merge "res_pjsip: Add missing NULL checks when using pjsip_inv_end_session()."

3 years agoMerge "features: Fix channel datastore access."
zuul [Fri, 1 Jul 2016 16:12:48 +0000 (11:12 -0500)]
Merge "features: Fix channel datastore access."

3 years agoMerge "res_pjsip: improve realtime performance #2"
Joshua Colp [Thu, 30 Jun 2016 20:53:24 +0000 (15:53 -0500)]
Merge "res_pjsip: improve realtime performance #2"

3 years agores_pjsip_session.c: Don't send extra BYE if SDP invalid.
Richard Mudgett [Wed, 22 Jun 2016 22:26:38 +0000 (17:26 -0500)]
res_pjsip_session.c: Don't send extra BYE if SDP invalid.

When an answer SDP is invalid we were disconnecting the outgoing call and
sending two BYE requests.  The first BYE was sent by PJPROJECT because of
the invalid SDP answer.  The second BYE was sent by Asterisk because it
thought the canceled call was the result of the RFC5407 section 3.1.2 race
condition.

* Made not send the BYE on a canceled session if the SDP negotiation is
incomplete because PJPROJECT has already sent a BYE for the failed
negotiation.

ASTERISK-25772 #close
Reported by:  Dmitriy Serov

Change-Id: I44ad0bd0605e8eeb7035c890d6f97a1331f1a836

3 years agores_pjsip_session.c: End call on initial invalid SDP negotiation.
Richard Mudgett [Mon, 27 Jun 2016 22:19:08 +0000 (17:19 -0500)]
res_pjsip_session.c: End call on initial invalid SDP negotiation.

When an incoming call defers SDP negotiation and then sends us an invalid
SDP in the ACK, we need to send a BYE to disconnect the call.  In this
case SDP negotiation has failed and we don't have valid media streams
negotiated.

ASTERISK-25772

Change-Id: Ia358516b0fc1e6c4c139b78246f10b9da7a2dfb8

3 years agores_pjsip.c: Register PJMEDIA error code decoder.
Richard Mudgett [Thu, 23 Jun 2016 20:13:24 +0000 (15:13 -0500)]
res_pjsip.c: Register PJMEDIA error code decoder.

Registering the PJMEDIA error codes allows errors found when parsing an
incoming SDP to be easier to figure out.

"Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)"
is much easier to understand than "Unknown error 220030".

ASTERISK-25772

Change-Id: I44b2dcea656fedd7593171be9e845880a2c70ca0

3 years agores_pjsip_session.c: Remove unused parameter from handle_incoming().
Richard Mudgett [Mon, 27 Jun 2016 21:56:33 +0000 (16:56 -0500)]
res_pjsip_session.c: Remove unused parameter from handle_incoming().

Change-Id: Iedd182d189ec947c42edc2c66c4bda3c22060daa

3 years agores_pjsip: Add missing NULL checks when using pjsip_inv_end_session().
Richard Mudgett [Wed, 22 Jun 2016 23:02:59 +0000 (18:02 -0500)]
res_pjsip: Add missing NULL checks when using pjsip_inv_end_session().

pjsip_inv_end_session() is documented as being able to return the
passed in tdata parameter set to NULL on success.

Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047

3 years agofeatures: Fix channel datastore access.
Richard Mudgett [Thu, 30 Jun 2016 20:17:02 +0000 (15:17 -0500)]
features: Fix channel datastore access.

Found as a result of the testsuite tests/callparking test crashing.

Several calls to ast_get_chan_featuremap_config() and
ast_get_chan_features_xfer_config() did not lock the channel before
calling so the channel's datastore list was accessed without the lock's
protection.  Apparently another thread deleted a datastore on the
channel's list while the crashing thread was walking the list.  Crash at
0xdeaddead due to MALLOC_DEBUG's memory filler value as a result.

* Add missing channel locks to calls that were not already protected
as the doxygen for those calls indicates.

Change-Id: Id273b3d305cc616406c353cbc841b2b7655efaa1

3 years agoconfigure: Fix HAVE_PJSIP_EVSUB_GRP_LOCK not set with external pjproject
George Joseph [Thu, 30 Jun 2016 13:25:09 +0000 (07:25 -0600)]
configure:  Fix HAVE_PJSIP_EVSUB_GRP_LOCK not set with external pjproject

There was a typo in configure.ac preventing HAVE_PJSIP_EVSUB_GRP_LOCK
from getting set when using an external pjproject.

ASTERISK-26099 #close
Reported-by: Ross Beer

Change-Id: I709af70428e125fb5ccd44b171d25dd29141f0ae

3 years agoMerge "pjproject/patches/config_site: Increase the max number of ICE candidates"
Joshua Colp [Wed, 29 Jun 2016 23:49:38 +0000 (18:49 -0500)]
Merge "pjproject/patches/config_site: Increase the max number of ICE candidates"

3 years agohep.conf.sample: Default 'enabled' to 'no'
Matt Jordan [Wed, 29 Jun 2016 20:31:30 +0000 (15:31 -0500)]
hep.conf.sample: Default 'enabled' to 'no'

Following the principle of least surprise, we should not be sending
massive numbers of PJSIP and RTCP HEP packets out into the ether to some
only-slightly-random IP address. Having 'enabled' set to 'no' in the
sample configuration file should prevent this from happening for those
who run 'make samples'.

ASTERISK-26159 #close

Change-Id: I1753a64ca83a3442a6ebdc31061f8185c062d9b1

3 years agopjproject/patches/config_site: Increase the max number of ICE candidates
Matt Jordan [Wed, 29 Jun 2016 20:09:02 +0000 (15:09 -0500)]
pjproject/patches/config_site: Increase the max number of ICE candidates

When negotiating ICE candidates with WebRTC capable endpoints, many
networks will result in a browser offering ICE candidates that exceeds
the default number of max candidates, 16. This patch bumps the max
candidates to 32, with the max checks at twice the number of candidates.
In practice, this has shown to be sufficient for browser/WebRTC
negotiation.

Change-Id: Ifd8da8b315f5ae14814d4ce20e10d2e6355020e5

3 years agoMerge "codecs: Fix ABI incompatibility created by adding format_name to ast_codec"
zuul [Wed, 29 Jun 2016 17:24:14 +0000 (12:24 -0500)]
Merge "codecs:  Fix ABI incompatibility created by adding format_name to ast_codec"

3 years agoMerge "siren: Add format attribute modules for Siren7 and Siren14."
zuul [Wed, 29 Jun 2016 16:30:53 +0000 (11:30 -0500)]
Merge "siren: Add format attribute modules for Siren7 and Siren14."

3 years agoMerge "BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf."
zuul [Wed, 29 Jun 2016 16:16:05 +0000 (11:16 -0500)]
Merge "BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf."

3 years agocodecs: Fix ABI incompatibility created by adding format_name to ast_codec
George Joseph [Tue, 28 Jun 2016 14:00:32 +0000 (08:00 -0600)]
codecs:  Fix ABI incompatibility created by adding format_name to ast_codec

Adding format_name even to the end of ast_codec caused issued with
binary codec modules because the pointer would be garbage in asterisk
when they registered.  So, the ast_codec structure was reverted and an
internal_ast_codec structure was created just for use in codec.c.  A new
internal-only API was also added (__ast_codec_register_with_format) so
that codec_builtin could register codecs with the format_name in a
separate parameter rather than in the ast_codec structure.

ASTERISK-26144 #close
Reported-by: Alexei Gradinari

Change-Id: I6df1b08f6a6ae089db23adfe1ebc8636330265ba

3 years agoMerge "BuildSystem: Fix a few issues hightlighted by gcc 6.x"
Joshua Colp [Tue, 28 Jun 2016 19:57:06 +0000 (14:57 -0500)]
Merge "BuildSystem:  Fix a few issues hightlighted by gcc 6.x"

3 years agoBuildSystem: Fix a few issues hightlighted by gcc 6.x
George Joseph [Tue, 28 Jun 2016 13:22:24 +0000 (07:22 -0600)]
BuildSystem:  Fix a few issues hightlighted by gcc 6.x

gcc 6.1.1 caught a few more issues.
Made sure the unit tests still pass for the func_env and stdtime
issues.

ASTERISK-26157 #close

Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e

3 years agoconfigs/basic-pbx/modules.conf: Remove 'bad' modules
Matt Jordan [Tue, 28 Jun 2016 15:33:30 +0000 (10:33 -0500)]
configs/basic-pbx/modules.conf: Remove 'bad' modules

This patch removes the following modules:
 - pbx_functions: It never existed.
 - res_pjsip_log_forwarder: It no longer exists.
 - res_hep_pjsip: The base HEP module wasn't loaded, and most basic PBXs
                  aren't going to be installing HOMER
 - res_pjsip_phoneprov_provider: The basic res_phoneprov module isn't
                  loaded, and we aren't configured to make use of the
                  module

Change-Id: Id91f68cae7c9c8c3d370029fe1268cb51e4ff5a5

3 years agosiren: Add format attribute modules for Siren7 and Siren14.
Joshua Colp [Wed, 22 Jun 2016 16:19:32 +0000 (13:19 -0300)]
siren: Add format attribute modules for Siren7 and Siren14.

This change removes hardcoded SDP parsing and generation for
Siren7 and Siren14 from chan_sip and moves it to format attribute
modules so it can also be used by chan_pjsip.

With this the fmtp lines for both are added with the bitrate
information.

ASTERISK-26021

Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037

3 years agoBuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf.
Alexander Traud [Thu, 23 Jun 2016 09:33:06 +0000 (11:33 +0200)]
BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf.

Removed the obsolete macro AC_TYPE_SIGNAL because Asterisk does not use K&R C
but requires ANSI C anyway.

ASTERISK-26046

Change-Id: I914c014385e1862102d90fe7650621def78db02e

3 years agoMerge "res_fax: Fix reference leak in fax_v21_session_new."
zuul [Thu, 23 Jun 2016 02:50:22 +0000 (21:50 -0500)]
Merge "res_fax: Fix reference leak in fax_v21_session_new."

3 years agoMerge "res_rtp_asterisk: Fix a self-comparison identified by gcc 6"
Joshua Colp [Thu, 23 Jun 2016 01:16:03 +0000 (20:16 -0500)]
Merge "res_rtp_asterisk:  Fix a self-comparison identified by gcc 6"

3 years agoMerge "chan_unistim: Fix memcpy in get_to_address"
zuul [Wed, 22 Jun 2016 23:50:57 +0000 (18:50 -0500)]
Merge "chan_unistim:  Fix memcpy in get_to_address"

3 years agoMerge "BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf."
zuul [Wed, 22 Jun 2016 23:50:48 +0000 (18:50 -0500)]
Merge "BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf."

3 years agoMerge "Fix Alembic upgrades."
Joshua Colp [Wed, 22 Jun 2016 21:06:06 +0000 (16:06 -0500)]
Merge "Fix Alembic upgrades."

3 years agores_fax: Fix reference leak in fax_v21_session_new.
Corey Farrell [Wed, 22 Jun 2016 20:04:54 +0000 (16:04 -0400)]
res_fax: Fix reference leak in fax_v21_session_new.

fax_v21_session_new created a session details object but only released
the allocation reference during error conditions.  fax_session_new adds
it's own reference to details if needed so the caller is always
responsible for cleaning it's own reference.

ASTERISK-26141 #close

Change-Id: Ie7fc52a83b6596ce9ce2d5a2bd9f3e204f48fc88

3 years agoMerge "res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro."
zuul [Wed, 22 Jun 2016 19:36:46 +0000 (14:36 -0500)]
Merge "res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro."

3 years agores_pjsip: improve realtime performance #2
Alexei Gradinari [Wed, 22 Jun 2016 19:25:23 +0000 (15:25 -0400)]
res_pjsip: improve realtime performance #2

The patch removes updating all Endpoints' status on startup.
Instead, only non-qualified aors with static contact
and non-qualified non-expired contacts are retrieved from the realtime to
update the endpoint status to ONLINE.
The endpoint name was added to the contact object to simply find the endpoint
that created this contact.

The status of endpoints with qualified aors will be updated by 'qualify'
functions.

ASTERISK-26061 #close

Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df

3 years agores_rtp_asterisk: Fix a self-comparison identified by gcc 6
George Joseph [Wed, 22 Jun 2016 18:41:57 +0000 (12:41 -0600)]
res_rtp_asterisk:  Fix a self-comparison identified by gcc 6

gcc 6 caught a previously unidentified self-comparison in
ice_candidate_cmp.  Fixed it and re-ordered the predicates for better
short-circuiting.

ASTERISK-26140 #close

Change-Id: I3da713c568e24064430257b3502fbdafd35af7a7

3 years agochan_unistim: Fix memcpy in get_to_address
George Joseph [Wed, 22 Jun 2016 15:37:23 +0000 (09:37 -0600)]
chan_unistim:  Fix memcpy in get_to_address

A code block only enabled when HAVE_PKTINFO is not defined (FreeBSD)
was using a pointer to a pointer as the destination of a memcpy and a
'&' instead of '*' in the sizeof.

ASTERISK-26138 #close

Change-Id: Id4927ff256c0e470bdf7bcfc025146a2f656e708

3 years agoFix Alembic upgrades.
Mark Michelson [Mon, 20 Jun 2016 18:21:52 +0000 (13:21 -0500)]
Fix Alembic upgrades.

A non-existent constraint was being referenced in the upgrade script.
This patch corrects the problem by removing the reference.

In addition, the head of the alembic branch referred to a non-existent
revision. This has been fixed by referring to the proper revision.

This patch fixes another realtime problem as well. Our Alembic scripts
store booleans as yes or no values. However, Sorcery tries to insert
"true" or "false" instead. This patch introduces a new boolean type that
translates to "yes" or "no" instead.

ASTERISK-26128 #close

Change-Id: I51574736a881189de695a824883a18d66a52dcef

3 years agotest_res_pjsip_scheduler: Add 'depends' on pjproject in MODULEINFO
George Joseph [Wed, 22 Jun 2016 15:51:14 +0000 (09:51 -0600)]
test_res_pjsip_scheduler: Add 'depends' on pjproject in MODULEINFO

Since the file was missing the depends on pjproject, it wasn't
picking up the pjproject related include path.  If there was no
system installed pjproject and pjproject-bundled was used, a compile
would fail because pjsip.h wasn't found.

ASTERISK-26139 #close

Change-Id: I2ee64a999051452bc198c4e2c168c70769cd3757

3 years agoBuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf.
Alexander Traud [Wed, 22 Jun 2016 15:55:05 +0000 (17:55 +0200)]
BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf.

Removed the obsolete macro AC_FUNC_SETVBUF_REVERSED because Asterisk does not
support the platform SVR2 from the year 1987 anymore.

ASTERISK-26046

Change-Id: I28161b037feb2d29ab46ed20e785928460226c22

3 years agoMerge "res_rtp_asterisk: fix memory leak in dtls"
Joshua Colp [Wed, 22 Jun 2016 15:52:54 +0000 (10:52 -0500)]
Merge "res_rtp_asterisk: fix memory leak in dtls"

3 years agoMerge "res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription"
Joshua Colp [Wed, 22 Jun 2016 10:11:54 +0000 (05:11 -0500)]
Merge "res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription"

3 years agores_rtp_asterisk: fix memory leak in dtls
Torrey Searle [Tue, 21 Jun 2016 11:52:20 +0000 (13:52 +0200)]
res_rtp_asterisk: fix memory leak in dtls

ensure that cert bios get freed after creating the fingerprint

ASTERISK-26129 #close

Change-Id: I44d23aea07dce80176ca1ff877c5ace9452ef451

3 years agoMerge "res_rtp_asterisk: Use latest DTLS version available by underlying platform."
Joshua Colp [Wed, 22 Jun 2016 00:39:51 +0000 (19:39 -0500)]
Merge "res_rtp_asterisk: Use latest DTLS version available by underlying platform."

3 years agoMerge "res_pjsip_session: Handle race condition at shutdown with timer."
Joshua Colp [Tue, 21 Jun 2016 23:53:33 +0000 (18:53 -0500)]
Merge "res_pjsip_session: Handle race condition at shutdown with timer."

3 years agores_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro.
Richard Mudgett [Tue, 21 Jun 2016 22:42:28 +0000 (17:42 -0500)]
res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro.

Change-Id: I8799fb0a347ad76e747dafd0eacf1ea1086b9a8c

3 years agoMerge "PJSIP: provide transport type with received messages"
zuul [Tue, 21 Jun 2016 20:05:35 +0000 (15:05 -0500)]
Merge "PJSIP: provide transport type with received messages"

3 years agores_pjsip_pubsub: Address SEGV when attempting to terminate a subscription
George Joseph [Sun, 12 Jun 2016 16:19:27 +0000 (10:19 -0600)]
res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription

Occasionally under load we'll attempt to send a final NOTIFY on a
subscription that's already been terminated and a SEGV will occur
down in pjproject's evsub_destroy function.  This is a result of a
race condition between all the paths that can generate a notify
and/or destroy the underlying pjproject evsub object:

 * The client can send a SUBSCRIBE with Expires: 0.
 * The client can send a SUBSCRIBE/refresh.
 * The subscription timer can expire.
 * An extension state can change.
 * An MWI event can be generated.
 * The pjproject transaction timer (timer_b) can expire.

Normally when our pubsub_on_evsub_state is called with a terminate,
we push a task to the serializer and return at which point the dialog
is unlocked.  This is usually not a problem because the task runs
immediately and locks the dialog again.  When the system is heavily
loaded though, there may be a delay between the unlock and relock
during which another event may occur such as the subscription timer
or timer_b expiring, an extension state change, etc.  These may also
cause a terminate to be processed and if so, we could cause pjproject
to try to destroy the evsub structure twice.  There's no way for us to
tell that the evsub was already destroyed and the evsub's group lock
can't tolerate this and SEGVs.

The remedy is twofold.

 * A patch has been submitted to Teluu and added to the bundled
   pjproject which adds add/decrement operations on evsub's group lock.

 * In res_pjsip_pubsub:
   * configure.ac and pjproject-bundled's configure.m4 were updated
     to check for the new evsub group lock APIs.
   * We now add a reference to the evsub group lock when we create
     the subscription and remove the reference when we clean up the
     subscription.  This prevents evsub from being destroyed before
     we're done with it.
   * A state has been added to the subscription tree structure so
     termination progress can be tracked through the asyncronous tasks.
   * The pubsub_on_evsub_state callback has been split so it's not doing
     double duty.  It now only handles the final cleanup of the
     subscription tree.  pubsub_on_rx_refresh now handles both client
     refreshes and client terminates.  It was always being called for
     both anyway.
   * The serialized_on_server_timeout task was removed since
     serialized_pubsub_on_rx_refresh was almost identical.
   * Missing state checks and ao2_cleanups were added.
   * Some debug levels were adjusted to make seeing only off-nominal
     things at level 1 and nominal or progress things at level 2+.

ASTERISK-26099 #close
Reported-by: Ross Beer.

Change-Id: I779d11802cf672a51392e62a74a1216596075ba1

3 years agores_rtp_asterisk: Use latest DTLS version available by underlying platform.
Alexander Traud [Tue, 21 Jun 2016 12:05:30 +0000 (14:05 +0200)]
res_rtp_asterisk: Use latest DTLS version available by underlying platform.

Do not use DTLSv1_method() but DTLS_method() when available in OpenSSL of the
underlying platform. This change enables DTLS 1.2 since OpenSSL 1.0.2, for
WebRTC (DTLS-SRTP via SIP-over-WebSockets). This change enables AEAD-based
cipher-suites.

ASTERISK-26130 #close

Change-Id: I41f24448d6d2953e8bdb97c9f4a6bc8a8f055fd0

3 years agoPJSIP: provide transport type with received messages
Scott Griepentrog [Tue, 21 Jun 2016 15:53:05 +0000 (10:53 -0500)]
PJSIP: provide transport type with received messages

The receipt of a SIP MESSAGE may occur over any transport including TCP
and TLS. When the message is received, the original URI is added to the
message in the field PJSIP_RECVADDR, but this is insufficient to ensure
a reply message can reach the originating endpoint. This patch adds the
PJSIP_TRANSPORT field populated with the transport type.

ASTERISK-26132 #close

Change-Id: I28c4b1e40d573a056c81deb213ecf53e968f725e

3 years agoBuildSystem: Avoid obsolete warning with HELP_STRING on autoconf.
Alexander Traud [Tue, 21 Jun 2016 13:01:40 +0000 (15:01 +0200)]
BuildSystem: Avoid obsolete warning with HELP_STRING on autoconf.

Some configure scripts used both AC_HELP_STRING and its replacement
AS_HELP_STRING. For consistency and to avoid obsolete warnings, those were
changed to AS_HELP_STRING.

ASTERISK-26046

Change-Id: I8aad4fd2bdee40aa2a31ce3339a1eb33ff4f5b0f

3 years agoMerge "fix: memory leaks, resource leaks, out of bounds and bugs"
zuul [Tue, 21 Jun 2016 12:26:12 +0000 (07:26 -0500)]
Merge "fix: memory leaks, resource leaks, out of bounds and bugs"

3 years agoMerge "app_voicemail.c: Fix IMAP compile error."
zuul [Mon, 20 Jun 2016 19:45:16 +0000 (14:45 -0500)]
Merge "app_voicemail.c: Fix IMAP compile error."

3 years agores_pjsip_session: Handle race condition at shutdown with timer.
Joshua Colp [Mon, 20 Jun 2016 15:29:13 +0000 (12:29 -0300)]
res_pjsip_session: Handle race condition at shutdown with timer.

When shutting down res_pjsip_session will get unloaded before res_pjsip.
The act of unloading unregisters all the PJSIP services and sets
their module IDs to -1. In some cases it is possible for a timer to
occur after this happens which calls into res_pjsip_session. The
res_pjsip_session module can then try to get the session from the
INVITE session using the module ID. Since the module ID is now -1
this fails.

This change stores a copy of the module ID and uses it for the timer
callback scenario. If the module ID is -1 the callback immediately
returns but if the module ID is valid then it continues as normal.

This works as the original ID of the module is guaranteed to still
be valid when used with the INVITE session.

ASTERISK-26127 #close

Change-Id: I88df72525c4e9ef9f19c13aedddd3ac4a335c573

3 years agoMerge "http: leverage 'bindaddr' for TLS in http.conf"
zuul [Mon, 20 Jun 2016 18:28:12 +0000 (13:28 -0500)]
Merge "http: leverage 'bindaddr' for TLS in http.conf"

3 years agoapp_voicemail.c: Fix IMAP compile error.
Richard Mudgett [Mon, 20 Jun 2016 17:13:27 +0000 (12:13 -0500)]
app_voicemail.c: Fix IMAP compile error.

Fix compile error introduced by the patch for
ASTERISK-26045

Change-Id: I5b02876266f2824f4cec2b54d6ff4db5de5778d3

3 years agofix: memory leaks, resource leaks, out of bounds and bugs
Alexei Gradinari [Fri, 17 Jun 2016 18:51:57 +0000 (14:51 -0400)]
fix: memory leaks, resource leaks, out of bounds and bugs

ASTERISK-26119 #close

Change-Id: Iecbf7d0f360a021147344c4e83ab242fd1e7512c

3 years agoARI: Ensure announcer channels are destroyed.
Mark Michelson [Mon, 13 Jun 2016 22:40:07 +0000 (17:40 -0500)]
ARI: Ensure announcer channels are destroyed.

Announcer channels were not being destroyed because the
stasis_app_control structure that referenced them was not being
destroyed. The control structure was not being destroyed because it was
not being unlinked from its container. It was not being unlinked from
its container because the after bridge callback for the announcer
channel was not being run. The after bridge callback was not being run
because the after bridge datastore was not being removed from the
channel on destruction. The channel was not being destroyed because the
hangup that used to destroy the channel was now only reducing the
reference count to one. The reference count of the channel was only
being reduced to one because the stasis_app_control structure was
holding the final reference...

The control structure used to not keep a reference to the channel, so
that loop described above did not happen.

The solution is to manually remove the control structure from its
container when the playback on a bridge is complete.

ASTERISK-26083 #close
Reported by Joshua Colp

Change-Id: I0ddc0f64484ea0016245800b409b567dfe85cfb4

3 years agohttp: leverage 'bindaddr' for TLS in http.conf
Alexander Traud [Mon, 20 Jun 2016 13:05:09 +0000 (15:05 +0200)]
http: leverage 'bindaddr' for TLS in http.conf

The internal HTTP/WebSocket server supports both TCP and TLS, which can be
activated separately via the file http.conf. The source code intends to re-use
the TCP parameter 'bindaddr' for TLS, even if 'tlsbindaddr' is not specified
explicitly. This did not work because of a typo. This change resolves this typo.

ASTERISK-26126 #close

Change-Id: I5efb0409ae12044dfb3495b6b97b6d40a8c9c51f

3 years agoMerge "Add support for OGG/Speex file format"
Joshua Colp [Fri, 17 Jun 2016 19:03:57 +0000 (14:03 -0500)]
Merge "Add support for OGG/Speex file format"

3 years agoMerge "chan_sip: bigger buffers for headers, better failure mode"
zuul [Thu, 16 Jun 2016 22:59:32 +0000 (17:59 -0500)]
Merge "chan_sip: bigger buffers for headers, better failure mode"

3 years agores_pjsip_transport_management.c: Misc cleanups to survive shutdown.
Richard Mudgett [Wed, 18 May 2016 22:37:27 +0000 (17:37 -0500)]
res_pjsip_transport_management.c: Misc cleanups to survive shutdown.

* In unload_module(), reordered destroying things to minimize the window
that the global transports container could be used by other threads on
shutdown.  When shutting down you need to stop things in the opposite
order of creation.

* Put the global transports container into an AO2_GLOBAL_OBJ_STATIC to
eliminate the crash potential by other threads using the container on
shutdown.

* Made struct monitored_transport.sip_received not use
ast_atomic_fetchadd_int() since it is used as a boolean value that is only
set TRUE.  It was previously incremented for every received SIP message
and could theoretically overflow.

* In monitored_transport_state_callback(), allocated the monitored
transport object without a lock since the lock was unused.

* In keepalive_global_loaded(), removed releasing the transports container
if the keepalive_thread could not be started.  I set it up to be tried
again if the user reloads the configuration.

Change-Id: I8d12d16ef564290fa6d25a32334bb5ce8fdf87ff

3 years agores_pjsip.c: Add check that timer actually got scheduled.
Richard Mudgett [Wed, 6 Jan 2016 01:08:24 +0000 (19:08 -0600)]
res_pjsip.c: Add check that timer actually got scheduled.

Change-Id: Iabaa2e5dccf0762c258101ea0eb1487cf6959ad1

3 years agoMerge "res_pjsip_session.c: Reorganize ast_sip_session_terminate()."
zuul [Tue, 14 Jun 2016 18:36:41 +0000 (13:36 -0500)]
Merge "res_pjsip_session.c: Reorganize ast_sip_session_terminate()."

3 years agores_rtp_multicast.c: Fix warning message typo.
Richard Mudgett [Mon, 13 Jun 2016 18:33:53 +0000 (13:33 -0500)]
res_rtp_multicast.c: Fix warning message typo.

Change-Id: Ic9928208b9957e09866abe3d9649030942ec52b3

3 years agores_pjsip_session.c: Reorganize ast_sip_session_terminate().
Richard Mudgett [Fri, 12 Feb 2016 00:15:31 +0000 (18:15 -0600)]
res_pjsip_session.c: Reorganize ast_sip_session_terminate().

Change-Id: I68a2128bcba4830985d2d441e70dfd1ac5bd712b

3 years agoMerge "core: Not the configured but granted number of possible file descriptors."
zuul [Fri, 10 Jun 2016 20:50:35 +0000 (15:50 -0500)]
Merge "core: Not the configured but granted number of possible file descriptors."

3 years agocore: Not the configured but granted number of possible file descriptors.
Alexander Traud [Wed, 8 Jun 2016 11:15:15 +0000 (13:15 +0200)]
core: Not the configured but granted number of possible file descriptors.

With CLI "core show settings", simply the parameter maxfiles of the file
asterisk.conf was shown. If that parameter was not set, nothing was displayed
although the environment might have set a default number itself. Or if maxfiles
were not granted (completely), still maxfiles was shown. Now, the maximum number
of possible file descriptors in the environment is shown.

ASTERISK-26097

Change-Id: I2df5c58863b5007b34b77adbe28b885dfcdf7e0b

3 years agoMerge "astfd: With RLIMIT_NOFILE only the current value is sensible."
Joshua Colp [Fri, 10 Jun 2016 18:46:48 +0000 (13:46 -0500)]
Merge "astfd: With RLIMIT_NOFILE only the current value is sensible."

3 years agotranslate: Enables native Packet-Loss Concealment (PLC) for supporting codecs.
Joshua Colp [Fri, 10 Jun 2016 15:39:27 +0000 (12:39 -0300)]
translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs.

This reverts commit 5bfef2a8b4674382f959b21a3b8e14cf1d942bab as it
caused fax test failures.

ASTERISK-25629

Change-Id: I79de974dc4f63a1cafe0d2509169fd9a6b3cbaf4

3 years agoastfd: With RLIMIT_NOFILE only the current value is sensible.
Alexander Traud [Wed, 8 Jun 2016 11:05:22 +0000 (13:05 +0200)]
astfd: With RLIMIT_NOFILE only the current value is sensible.

With menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", both the maximum max
and current max of possible file descriptors were shown. Both show the same
value always. Not to confuse users, just the current maximum is shown now.

ASTERISK-26097

Change-Id: I49cf7952d73aec9e3f6a88942842c39be18380fa

3 years agoMerge "cel: Ensure only one dial status per channel exists."
zuul [Fri, 10 Jun 2016 03:38:52 +0000 (22:38 -0500)]
Merge "cel: Ensure only one dial status per channel exists."

3 years agoMerge "ARI: Ensure proper channel state on operations."
zuul [Fri, 10 Jun 2016 02:50:07 +0000 (21:50 -0500)]
Merge "ARI: Ensure proper channel state on operations."

3 years agoMerge "test_http_media_cache: Fix failing test."
zuul [Fri, 10 Jun 2016 02:50:05 +0000 (21:50 -0500)]
Merge "test_http_media_cache: Fix failing test."