asterisk/asterisk.git
2 years agoapp_voicemail: vm_authenticate accesses uninitialized memory
Sean Bright [Mon, 20 Feb 2017 12:28:23 +0000 (07:28 -0500)]
app_voicemail: vm_authenticate accesses uninitialized memory

vm_authenticate doesn't always set the passed ast_vm_user argument, so
we initialize to 0 before passing it in.

ASTERISK-25893 #close
Reported by: Filip Jenicek

Change-Id: Ia3cc0128f93d352ed9add8d5c2f0f7232c2cbe4a

2 years agoMerge "Revert "build: Execute ldconfig to build cache.""
zuul [Mon, 20 Feb 2017 20:09:27 +0000 (14:09 -0600)]
Merge "Revert "build: Execute ldconfig to build cache.""

2 years agoRevert "build: Execute ldconfig to build cache."
Joshua Colp [Mon, 20 Feb 2017 17:19:55 +0000 (11:19 -0600)]
Revert "build: Execute ldconfig to build cache."

This reverts commit 8851c3e0885cb704a5a6159a51768ea5297e9b10.

Change-Id: I124380be5e3bd57da978428a2a93604336ccd0db

2 years agoMerge "Binaural synthesis (confbridge): Adds utils/conf_bridge_binaural_hrir_importer"
Joshua Colp [Mon, 20 Feb 2017 16:24:55 +0000 (10:24 -0600)]
Merge "Binaural synthesis (confbridge): Adds utils/conf_bridge_binaural_hrir_importer"

2 years agoMerge "res_config_sqlite3: Fix crash when loading with invalid config"
zuul [Sun, 19 Feb 2017 19:24:44 +0000 (13:24 -0600)]
Merge "res_config_sqlite3: Fix crash when loading with invalid config"

2 years agoMerge "pjproject-bundled: Fix checksum verification when using cURL"
Joshua Colp [Sun, 19 Feb 2017 18:46:09 +0000 (12:46 -0600)]
Merge "pjproject-bundled: Fix checksum verification when using cURL"

2 years agores_config_sqlite3: Fix crash when loading with invalid config
Sean Bright [Wed, 15 Feb 2017 17:55:19 +0000 (12:55 -0500)]
res_config_sqlite3: Fix crash when loading with invalid config

When ast_config_load() fails with CONFIG_STATUS_FILEINVALID, it has
already destroyed the ast_config struct for us. Trying to do it again
results in a crash.

Change-Id: If6a5c0ca718ad428e01a1fb25beb209a9ac18bc6

2 years agoMerge "Remove extra ast_iostream_close() calls."
zuul [Fri, 17 Feb 2017 23:41:06 +0000 (17:41 -0600)]
Merge "Remove extra ast_iostream_close() calls."

2 years agopjproject-bundled: Fix checksum verification when using cURL
Sean Bright [Fri, 17 Feb 2017 23:06:47 +0000 (18:06 -0500)]
pjproject-bundled: Fix checksum verification when using cURL

ASTERISK-26802 #close
Reported by: Michael L. Young

Change-Id: Iad293080f55d4d69ab615717a15211d916eed613

2 years agoRemove extra ast_iostream_close() calls.
Mark Michelson [Fri, 17 Feb 2017 20:58:28 +0000 (14:58 -0600)]
Remove extra ast_iostream_close() calls.

When AMI encounters an error at the beginning of a session, it would
explicitly call ast_iostream_close() on its tcptls session's iostream.
It then would jump to a label where it would shut down the tcptls
session instance. The tcptls session instance would again attempt to
close the iostream.

Under normal circumstances, this might go by unnoticed. However, when
MALLOC_DEBUG is enabled, all fields on the iostream get set to
0xdeaddead when the iostream is freed. Thus a second call to
ast_iostream_close() after the iostream has been freed would reslt in an
attempt to call SSL_shutdown on 0xdeaddead, which would crash and burn
horribly.

The fix here is to not directly close the iostream from the dangerous
scenarios. The specific scenarios are:
* Exceeding the configured authlimit
* Failing to build a mansession on a new connection

Change-Id: I908f98d516afd5a263bd36b072221008a4731acd

2 years agobuild: Execute ldconfig to build cache.
Joshua Colp [Thu, 16 Feb 2017 16:30:00 +0000 (16:30 +0000)]
build: Execute ldconfig to build cache.

On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory.

This change does the minimally invasive thing and executes
ldconfig so that the libraries in the lib directory are found
and their location cached. By doing so Asterisk starts up fine.

ASTERISK-26705

Change-Id: I6d30b6427e9d5e69470e11327c7ff203fa7da519

2 years agoMerge "stream: Rename creates/destroys to allocs/frees"
zuul [Thu, 16 Feb 2017 19:24:30 +0000 (13:24 -0600)]
Merge "stream:  Rename creates/destroys to allocs/frees"

2 years agoMerge "stream: Add unit tests for channel stream usage."
Joshua Colp [Thu, 16 Feb 2017 17:34:40 +0000 (11:34 -0600)]
Merge "stream: Add unit tests for channel stream usage."

2 years agoMerge "chan_unistim: fix char type to have consistent behavior on ARM"
zuul [Thu, 16 Feb 2017 17:32:32 +0000 (11:32 -0600)]
Merge "chan_unistim: fix char type to have consistent behavior on ARM"

2 years agoMerge "http: Ensure capath is defined on all http creations"
Joshua Colp [Thu, 16 Feb 2017 16:40:10 +0000 (10:40 -0600)]
Merge "http: Ensure capath is defined on all http creations"

2 years agoMerge "res_pjsip_pubsub: Correctly implement persisted subscriptions"
Joshua Colp [Thu, 16 Feb 2017 15:48:52 +0000 (09:48 -0600)]
Merge "res_pjsip_pubsub:  Correctly implement persisted subscriptions"

2 years agostream: Rename creates/destroys to allocs/frees
George Joseph [Thu, 16 Feb 2017 14:28:33 +0000 (07:28 -0700)]
stream:  Rename creates/destroys to allocs/frees

To be consistent with sdp implementation.

Change-Id: I714e300939b4188f58ca66ce9d1e84b287009500

2 years agoMerge "pjsip_distributor.c: Fix off-nominal tdata ref leak."
zuul [Thu, 16 Feb 2017 13:08:29 +0000 (07:08 -0600)]
Merge "pjsip_distributor.c: Fix off-nominal tdata ref leak."

2 years agohttp: Ensure capath is defined on all http creations
Joshua Elson [Wed, 15 Feb 2017 20:44:32 +0000 (13:44 -0700)]
http: Ensure capath is defined on all http creations

ASTERISK-26794 #close

Change-Id: I9cbc3b6b6a8aab590f5ccde9c262a98e4d5253a1

2 years agochan_unistim: fix char type to have consistent behavior on ARM
Igor Goncharovsky [Thu, 16 Feb 2017 05:09:35 +0000 (08:09 +0300)]
chan_unistim: fix char type to have consistent behavior on ARM

There is difference exists in behaviour of char type on x86 and ARM.
On x86 by default char variable type means signed char, but in ARM
unsigned char used. This make binary calculations and negative values
works wrong on ARM.

This patch change type of char variables used for store negative
values and binary calculations to signed char.

ASTERISK-26714

Change-Id: Id78716dee9568a58419d4ef63c038affc3dfc7ab

2 years agoMerge "stream: Add stream topology to channel"
George Joseph [Thu, 16 Feb 2017 01:29:52 +0000 (19:29 -0600)]
Merge "stream:  Add stream topology to channel"

2 years agores_pjsip_pubsub: Correctly implement persisted subscriptions
George Joseph [Tue, 7 Feb 2017 19:17:12 +0000 (12:17 -0700)]
res_pjsip_pubsub:  Correctly implement persisted subscriptions

This patch fixes 2 original issues and more that those 2 exposed.

* When we send a NOTIFY, and the client either doesn't respond or
  responds with a non OK, pjproject only calls our
  pubsub_on_evsub_state callback, no others.  Since
  pubsub_on_evsub_state (which does the sub_tree cleanup) does not
  expect to be called back without the other callbacks being called
  first, it just returns leaving the sub_tree orphaned.  Now
  pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE
  which is what pjproject will set to tell us that it was the
  transaction that timed out or failed and not the subscription
  itself timing our or being terminated by the client. If is
  TSX_STATE, pubsub_on_evsub_state now does the proper cleanup
  regardless of the state of the subscription.

* When a client renews a subscription, we don't update the
  persisted subscription with the new expires timestamp.  This causes
  subscription_persistence_recreate to prune the subscription if/when
  asterisk restarts.  Now, pubsub_on_rx_refresh calls
  subscription_persistence_update to apply the new expires timestamp.
  This exposed other issues however...

* When creating a dialog from rdata (which sub_persistence_recreate
  does from the packet buffer) there must NOT be a tag on the To
  header (which there will be when a client refreshes a
  subscription).  If there is one, pjsip_dlg_create_uas will fail.
  To address this, subscription_persistence_update now accepts a flag
  that indicates that the original packet buffer must not be updated.
  New subscribes don't set the flag and renews do.  This makes sure
  that when the rdata is recreated on asterisk startup, it's done
  from the original subscribe packet which won't have the tag on To.

* When creating a dialog from rdata, we were setting the dialog's
  remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq.
  When the client tried to resubscribe after a restart with the
  correct cseq, we'd reject the request with an Invalid CSeq error.

* The acts of creating a dialog and evsub by themselves when
  recreating a subscription does NOT restart pjproject's subscription
  timer.  The result was that even if we did correctly recreate the
  subscription, we never removed it if the client happened to go away
  or send a non-OK response to a NOTIFY.  However, there is no
  pjproject function exposed to just set the timer on an evsub that
  wasn't created by an incoming subscribe request.  To address this,
  we create our own timer using ast_sip_schedule_task.  This timer is
  used only for re-establishing subscriptions after a restart.

  An earlier approach was to add support for setting pjproject's
  timer (via a pjproject patch) and while that patch is still included
  here, we don't use that call at the moment.

While addressing these issues, additional debugging was added and
some existing messages made more useful.  A few formatting changes
were also made to 'pjsip show scheduled tasks' to make displaying
the subscription timers a little more friendly.

ASTERISK-26696
ASTERISK-26756

Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e

2 years agores_rtp_asterisk: Use PJ_ICE_MAX_CAND instead of hard-coding 16
Sean Bright [Wed, 15 Feb 2017 17:03:00 +0000 (12:03 -0500)]
res_rtp_asterisk: Use PJ_ICE_MAX_CAND instead of hard-coding 16

pjsip limits the total number of ICE candidates to PJ_ICE_MAX_CAND,
which is a compile-time constant. Instead of hard-coding 16 when we
enumerate local interfaces, use PJ_ICE_MAX_CAND so that we can
potentially collect more interfaces if the compile time options are
changed.

Tangentially related to ASTERISK~24464

Change-Id: I1b85509e39e33b1fed63c86261fc229ba14bbabd

2 years agoBinaural synthesis (confbridge): Adds utils/conf_bridge_binaural_hrir_importer
Dennis Guse [Thu, 22 Dec 2016 15:42:46 +0000 (16:42 +0100)]
Binaural synthesis (confbridge): Adds utils/conf_bridge_binaural_hrir_importer

Adds the import tool for converting a HRIR database to hrirs.h

ASTERISK-26292

Change-Id: I51eb31b54c23ffd9b544bdc6a09d20c112c8a547

2 years agostream: Add unit tests for channel stream usage.
Joshua Colp [Tue, 14 Feb 2017 18:33:57 +0000 (18:33 +0000)]
stream: Add unit tests for channel stream usage.

This change adds unit tests cover the following:

1. That retrieving the first media stream of a specific media
type from a stream topology retrieves the expected media
stream.

2. That setting the native formats of a channel which does
not support streams results in the creation of streams on
its behalf according to the formats of the channel.

3. That setting a stream topology on a channel which supports
streams sets the topology to the provided one.

ASTERISK-26790

Change-Id: Ic53176dd3e4532e8c3e97d9e22f8a4b66a2bb755

2 years agoMerge "app_voicemail: Allow 'Comedian Mail' branding to be overriden"
zuul [Tue, 14 Feb 2017 23:42:07 +0000 (17:42 -0600)]
Merge "app_voicemail: Allow 'Comedian Mail' branding to be overriden"

2 years agoMerge "app_voicemail: VoiceMailPlayMsg did not play database stored messages"
zuul [Tue, 14 Feb 2017 23:18:08 +0000 (17:18 -0600)]
Merge "app_voicemail: VoiceMailPlayMsg did not play database stored messages"

2 years agoapp_voicemail: Allow 'Comedian Mail' branding to be overriden
Sean Bright [Mon, 13 Feb 2017 22:50:41 +0000 (17:50 -0500)]
app_voicemail: Allow 'Comedian Mail' branding to be overriden

Original patch by John Covert, slight modifications by me.

ASTERISK-17428 #close
Reported by: John Covert
Patches:
app_voicemail.c.patch (license #5512) patch uploaded by
        John Covert

Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6

2 years agostream: Add stream topology to channel
George Joseph [Mon, 13 Feb 2017 17:50:47 +0000 (10:50 -0700)]
stream:  Add stream topology to channel

Adds topology set and get to channel.

ASTERISK-26790

Change-Id: Ic379ea82a9486fc79dbd8c4d95c29fa3b46424f4

2 years agoMerge "app_record: Add option to prevent silence from being truncated"
zuul [Tue, 14 Feb 2017 21:04:40 +0000 (15:04 -0600)]
Merge "app_record: Add option to prevent silence from being truncated"

2 years agoMerge "cli: Fix various CLI documentation and completion issues"
zuul [Tue, 14 Feb 2017 20:34:03 +0000 (14:34 -0600)]
Merge "cli: Fix various CLI documentation and completion issues"

2 years agoMerge "channel: Protect flags in ast_waitfor_nandfds operation."
zuul [Tue, 14 Feb 2017 19:31:01 +0000 (13:31 -0600)]
Merge "channel: Protect flags in ast_waitfor_nandfds operation."

2 years agoMerge "stream: Add stream topology unit tests and fix uncovered bugs."
zuul [Tue, 14 Feb 2017 19:26:43 +0000 (13:26 -0600)]
Merge "stream: Add stream topology unit tests and fix uncovered bugs."

2 years agoapp_voicemail: VoiceMailPlayMsg did not play database stored messages
rrittgarn [Wed, 25 Jan 2017 22:25:21 +0000 (16:25 -0600)]
app_voicemail: VoiceMailPlayMsg did not play database stored messages

When attempting to use VoiceMailPlayMsg with a realtime data backend
the message is located, but never retrieved. This patch adds the
required RETRIEVE and DISPOSE calls that will fetch the message from
the database (and IMAP storage as well for that matter).

Also, removed extraneous make_file call.

ASTERISK-26723 #close

Change-Id: I1e122dd53c0f3d7faa10f3c2b7e7e76a47d51b8c

2 years agoMerge "libasteriskssl: do nothing with OpenSSL >= 1.1"
Joshua Colp [Tue, 14 Feb 2017 18:49:42 +0000 (12:49 -0600)]
Merge "libasteriskssl: do nothing with OpenSSL >= 1.1"

2 years agoMerge "tcptls: use TLS_client_method with OpenSSL 1.1"
zuul [Tue, 14 Feb 2017 18:41:06 +0000 (12:41 -0600)]
Merge "tcptls: use TLS_client_method with OpenSSL 1.1"

2 years agoMerge "openssl 1.1 support: use OPENSSL_VERSION_NUMBER"
zuul [Tue, 14 Feb 2017 18:33:01 +0000 (12:33 -0600)]
Merge "openssl 1.1 support: use OPENSSL_VERSION_NUMBER"

2 years agoapp_record: Add option to prevent silence from being truncated
Sean Bright [Tue, 14 Feb 2017 14:12:31 +0000 (09:12 -0500)]
app_record: Add option to prevent silence from being truncated

When using Record() with the silence detection feature, the stream is
written out to the given file. However, if only 'silence' is detected,
this file is then truncated to the first second of the recording.

This patch adds the 'u' option to Record() to override that behavior.

ASTERISK-18286 #close
Reported by: var
Patches:
app_record-1.8.7.1.diff (license #6184) patch uploaded by var

Change-Id: Ia1cd163483235efe2db05e52f39054288553b957

2 years agoMerge "core: Cleanup some channel snapshot staging anomalies."
Joshua Colp [Tue, 14 Feb 2017 13:14:50 +0000 (07:14 -0600)]
Merge "core: Cleanup some channel snapshot staging anomalies."

2 years agoMerge "app_queue: reset abandoned in sl for sl2 calculations"
zuul [Mon, 13 Feb 2017 22:28:38 +0000 (16:28 -0600)]
Merge "app_queue:  reset abandoned in sl for sl2 calculations"

2 years agoMerge "stream: Add media stream topology definition and API"
zuul [Mon, 13 Feb 2017 19:02:20 +0000 (13:02 -0600)]
Merge "stream:  Add media stream topology definition and API"

2 years agoapp_queue: reset abandoned in sl for sl2 calculations
Sebastian Gutierrez [Tue, 7 Feb 2017 17:13:09 +0000 (14:13 -0300)]
app_queue:  reset abandoned in sl for sl2 calculations

ASTERISK-26775 #close

Change-Id: I86de4b1a699d6edc77fea9b70d839440e4088284

2 years agoMerge "res_pjsip.c: Fix inconsistency between warning and action."
Joshua Colp [Mon, 13 Feb 2017 18:08:03 +0000 (12:08 -0600)]
Merge "res_pjsip.c: Fix inconsistency between warning and action."

2 years agostream: Add stream topology unit tests and fix uncovered bugs.
Joshua Colp [Mon, 13 Feb 2017 17:00:42 +0000 (17:00 +0000)]
stream: Add stream topology unit tests and fix uncovered bugs.

This change adds unit tests for the various API calls relating
to stream topologies. This includes creation, destruction,
inspection, and manipulation.

Through this a few bugs were uncovered in the implementation:

1. Creating a topology using a format capabilities would fail as
the code considered a return value of 0 from the append stream
function to indicate an error which is incorrect.

2. Not all functions which placed a stream into a topology
set the position on the stream itself.

3. Appending a stream would cause a frack if the position
provided was the last one. This occurred because the existing
stream was queried but the index was outside of what the
vector was currently at for size.

ASTERISK-26786

Change-Id: Id5590e87c8a605deea1a89e53169a9c011d66fa0

2 years agocli: Fix various CLI documentation and completion issues
Sean Bright [Sat, 11 Feb 2017 15:57:03 +0000 (10:57 -0500)]
cli: Fix various CLI documentation and completion issues

* app_minivm: Use built-in completion facilities to complete optional
arguments.

* app_voicemail: Use built-in completion facilities to complete
optional arguments.

* app_confbridge: Add missing colons after 'Usage' text.

* chan_alsa: Use built-in completion facilities to complete optional
arguments.

* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'

* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'

* func_odbc: Correct completions for 'odbc read' and 'odbc write'

* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.

* main/bridge: Correct completions for 'bridge kick.'

* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.

* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'

* main/pbx_app: Remove redundant completions for 'core show
applications.'

* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'

* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.

Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca

2 years agoMerge "chan_pjsip: Multidomain endpoint finding on call"
zuul [Mon, 13 Feb 2017 15:43:50 +0000 (09:43 -0600)]
Merge "chan_pjsip: Multidomain endpoint finding on call"

2 years agostream: Add media stream topology definition and API
George Joseph [Fri, 10 Feb 2017 21:45:43 +0000 (14:45 -0700)]
stream:  Add media stream topology definition and API

This change adds the media stream topology definition and API for
accessing and using it.

Some refactoring of the stream was also done.

ASTERISK-26786

Change-Id: Ic930232d24d5ad66dcabc14e9b359e0ff8e7f568

2 years agoMerge "manager: Restore Originate failure behavior from Asterisk 11"
zuul [Mon, 13 Feb 2017 13:11:16 +0000 (07:11 -0600)]
Merge "manager: Restore Originate failure behavior from Asterisk 11"

2 years agoMerge "stream: Add media stream definition and API with unit tests."
Joshua Colp [Mon, 13 Feb 2017 13:05:07 +0000 (07:05 -0600)]
Merge "stream: Add media stream definition and API with unit tests."

2 years agochan_pjsip: Multidomain endpoint finding on call
Norbert Varga [Fri, 13 Jan 2017 17:21:36 +0000 (18:21 +0100)]
chan_pjsip: Multidomain endpoint finding on call

When PJSIP tries to call an endpoint with a domain (e.g. 1000@test.com),
the user part is stripped down as it would be a trunk with a specified user,
and only the host part is called as a PJSIP endpoint and can't be found.
This is not correct in the case of a multidomain SIP account, so the stripping
after the @ sign is done only if the whole endpoint (in multidomain case
1000@test.com) can't be found.

ASTERISK-26248

Change-Id: I3a2dd6f57f3bd042df46b961eccd81d31ab202e6

2 years agochannel: Protect flags in ast_waitfor_nandfds operation.
Joshua Colp [Mon, 13 Feb 2017 11:05:51 +0000 (11:05 +0000)]
channel: Protect flags in ast_waitfor_nandfds operation.

The ast_waitfor_nandfds operation will manipulate the flags
of channels passed in. This was previously done without
the channel lock being held. This could result in incorrect
values existing for the flags if another thread manipulated
the flags at the same time.

This change locks the channel during flag manipulation.

ASTERISK-26788

Change-Id: I2c5c8edec17c9bdad4a93291576838cb552ca5ed

2 years agores_pjsip.c: Fix inconsistency between warning and action.
Richard Mudgett [Sat, 11 Feb 2017 17:25:30 +0000 (11:25 -0600)]
res_pjsip.c: Fix inconsistency between warning and action.

The original return value corresponded to AST_SIP_AUTHENTICATION_CHALLENGE
but we have no authenticator registered to create the challenge.

Change-Id: I62368180d774b497411b80fbaabd0c80841f8512

2 years agopjsip_distributor.c: Fix off-nominal tdata ref leak.
Richard Mudgett [Sat, 11 Feb 2017 17:26:58 +0000 (11:26 -0600)]
pjsip_distributor.c: Fix off-nominal tdata ref leak.

Change-Id: I571f371d0956a8039b197b4dbd8af6b18843598d

2 years agomanager: Restore Originate failure behavior from Asterisk 11
Sean Bright [Thu, 9 Feb 2017 16:01:22 +0000 (11:01 -0500)]
manager: Restore Originate failure behavior from Asterisk 11

In Asterisk 11, if the 'Originate' AMI command failed to connect the provided
Channel while in extension mode, a 'failed' extension would be looked up and
run. This was, I believe, unintentionally removed in 51b6c49. This patch
restores that behavior.

This also adds an enum for the various 'synchronous' modes in an attempt to
make them meaningful.

ASTERISK-26115 #close
Reported by: Nasir Iqbal

Change-Id: I8afbd06725e99610e02adb529137d4800c05345d

2 years agocore: Cleanup some channel snapshot staging anomalies.
Richard Mudgett [Wed, 8 Feb 2017 20:27:18 +0000 (14:27 -0600)]
core: Cleanup some channel snapshot staging anomalies.

We shouldn't unlock the channel after starting a snapshot staging because
another thread may interfere and do its own snapshot staging.

* app_dial.c:dial_exec_full() made hold the channel lock while setting up
the outgoing channel staging.  Made hold the channel lock after the called
party answers while updating the caller channel staging.

* chan_sip.c:sip_new() completed the channel staging on off-nominal exit.
Also we need to use ast_hangup() instead of ast_channel_unref() at that
location.

* channel.c:__ast_channel_alloc_ap() added a comment about not needing to
complete the channel snapshot staging on off-nominal exit paths.

* rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel
locks while staging the channels for the stats channel variables.

Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a

2 years agostream: Add media stream definition and API with unit tests.
Joshua Colp [Tue, 7 Feb 2017 12:56:41 +0000 (12:56 +0000)]
stream: Add media stream definition and API with unit tests.

This change adds the media stream definition and API for
accessing and using it. Unit tests have also been written
which exercise aspects of the API.

ASTERISK-26773

Change-Id: I3dbe54065b55aaa51f467e1a3bafd67fb48cac87

2 years agoconfigs/samples: Fix placement of 'identify' entry in sorcery.conf
George Joseph [Fri, 10 Feb 2017 15:35:51 +0000 (08:35 -0700)]
configs/samples: Fix placement of 'identify' entry in sorcery.conf

The entry for 'identify' was incorrectly placed in the
res_pjsip section when it should be in
res_pjsip_endpoint_identifier_ip.

ASTERISK-26785 #close

Change-Id: Ia1372b12a952bfe2df6b1b1e0e725ca306a5d41a

2 years agoRevert "Update qualifies when AOR configuration changes."
Mark Michelson [Wed, 8 Feb 2017 17:50:11 +0000 (11:50 -0600)]
Revert "Update qualifies when AOR configuration changes."

This reverts commit 6492e91392b8fd394193e411c6eb64b45486093f.

The change in question was intended to prevent the need to reload in
order to update qualifies on contacts when an AOR changes. However, this
ended up causing a deadlock instead.

Change-Id: I1a835c90a5bb65b6dc3a1e94cddc12a4afc3d71e

2 years agoMerge "srv: Fix crash when ast_srv_lookup is used and 0 records are returned."
zuul [Wed, 8 Feb 2017 16:00:05 +0000 (10:00 -0600)]
Merge "srv: Fix crash when ast_srv_lookup is used and 0 records are returned."

2 years agosrv: Fix crash when ast_srv_lookup is used and 0 records are returned.
Joshua Colp [Tue, 7 Feb 2017 18:01:03 +0000 (18:01 +0000)]
srv: Fix crash when ast_srv_lookup is used and 0 records are returned.

When performing an SRV lookup using the ast_srv_lookup function it
did not properly handle the situation where 0 records are returned.
If this happened it would wrongly assume that at least one record
was present.

This change fixes the code so it will exit early if an error occurs
or if 0 records are returned.

ASTERISK-26772
patches:
  srv_lookup.patch submitted by nappsoft (license 6822)

Change-Id: I09b19081c74e0ad11c12bf54a257243b1bcb2351

2 years agores_stasis_device_state: Protect the adding/removing of subscriptions.
Joshua Colp [Mon, 6 Feb 2017 17:40:45 +0000 (17:40 +0000)]
res_stasis_device_state: Protect the adding/removing of subscriptions.

The adding and removing of device state subscriptions did not protect
fully against simultaneous manipulation. In particular the subscribe
case allowed a small window where two subscriptions could be added for
the same device state instead of just one.

This change makes the code hold the subscriptions lock for the entirety
of each operation to ensure that two are not occurring at the same time.

ASTERISK-26770

Change-Id: I3e7f8eb9d09de440c9024d2dd52029f6f20e725b

2 years agores_pjsip: Fix some off nominal tdata leaks.
Richard Mudgett [Wed, 1 Feb 2017 23:56:13 +0000 (17:56 -0600)]
res_pjsip: Fix some off nominal tdata leaks.

Change-Id: I243a4be5e7fbfe604923764969c4ee04eee89b9d

2 years agores_ari: fix memory leak for channelvars
Sebastien Duthil [Fri, 3 Feb 2017 21:26:23 +0000 (16:26 -0500)]
res_ari: fix memory leak for channelvars

In ari.conf, when setting the option channelvars, every Stasis channel
snapshot would create a list of variable/value that would not be freed
when the snapshot is freed, resulting in a often-recurring memory
leak.

ASTERISK-26767 #close

Change-Id: Ia37dd9d68063d7f879193df02ede293e5ded716d

2 years agoMerge "Update qualifies when AOR configuration changes."
Joshua Colp [Fri, 3 Feb 2017 15:32:53 +0000 (09:32 -0600)]
Merge "Update qualifies when AOR configuration changes."

2 years agoMerge "channel.c: Fix unbalanced read queue deadlocking local channels."
Joshua Colp [Fri, 3 Feb 2017 11:32:26 +0000 (05:32 -0600)]
Merge "channel.c: Fix unbalanced read queue deadlocking local channels."

2 years agoMerge "res_agi: Prevent an AGI from eating frames it should not. (Re-do)"
Joshua Colp [Fri, 3 Feb 2017 11:32:12 +0000 (05:32 -0600)]
Merge "res_agi: Prevent an AGI from eating frames it should not. (Re-do)"

2 years agolibasteriskssl: do nothing with OpenSSL >= 1.1
Tzafrir Cohen [Fri, 3 Feb 2017 08:25:33 +0000 (10:25 +0200)]
libasteriskssl: do nothing with OpenSSL >= 1.1

OpenSSL 1.1 requires no explicit initialization. The hacks in the
library are not needed. They also happen to fail running Asterisk.

Change-Id: I3b3efd5d80234a4c45a8ee58dcfe25b15d9ad100

2 years agotcptls: use TLS_client_method with OpenSSL 1.1
Tzafrir Cohen [Sat, 21 Jan 2017 05:59:15 +0000 (07:59 +0200)]
tcptls: use TLS_client_method with OpenSSL 1.1

OpenSSL 1.1 introduced TLS_client_method() and deprecated the previous
version-specific methods (such as TLSv1_client_method(). Other than
being simpler to use and more correct (gain support for TLS newer that
TLS1, in our case), the older ones produce a deprecation warning that
fails the build in dev-mode.

Change-Id: I257b1c8afd09dcb0d96cda3a41cb9f7a15d0ba07

2 years agoopenssl 1.1 support: use OPENSSL_VERSION_NUMBER
Tzafrir Cohen [Sat, 21 Jan 2017 05:57:33 +0000 (07:57 +0200)]
openssl 1.1 support: use OPENSSL_VERSION_NUMBER

Use OPENSSL_VERSION_NUMBER instead of OPENSSL_API_COMPAT to detect
the openssl 1.1 API.

Change-Id: I4e448f55ef516aedf6ad154037c35577a421a458

2 years agoMerge "Add reload options to CLI/AMI stale object commands."
George Joseph [Fri, 3 Feb 2017 00:34:03 +0000 (18:34 -0600)]
Merge "Add reload options to CLI/AMI stale object commands."

2 years agoMerge "Frame deferral: Revert API refactoring."
Joshua Colp [Fri, 3 Feb 2017 00:08:43 +0000 (18:08 -0600)]
Merge "Frame deferral: Revert API refactoring."

2 years agoMerge "res_odbc: Remove deprecated settings from sample configuration file"
zuul [Thu, 2 Feb 2017 22:40:03 +0000 (16:40 -0600)]
Merge "res_odbc: Remove deprecated settings from sample configuration file"

2 years agochannel.c: Fix unbalanced read queue deadlocking local channels.
Richard Mudgett [Wed, 1 Feb 2017 00:28:15 +0000 (18:28 -0600)]
channel.c: Fix unbalanced read queue deadlocking local channels.

Using the timerfd timing module can cause channel freezing, lingering, or
deadlock issues.  The problem is because this is the only timing module
that uses an associated alert-pipe.  When the alert-pipe becomes
unbalanced with respect to the number of frames in the read queue bad
things can happen.  If the alert-pipe has fewer alerts queued than the
read queue then nothing might wake up the thread to handle received frames
from the channel driver.  For local channels this is the only way to wake
up the thread to handle received frames.  Being unbalanced in the other
direction is less of an issue as it will cause unnecessary reads into the
channel driver.

ASTERISK-26716 is an example of this deadlock which was indirectly fixed
by the change that found the need for this patch.

* In channel.c:__ast_queue_frame(): Adding frame lists to the read queue
did not add the same number of alerts to the alert-pipe.  Correspondingly,
when there is an exceptionally long queue event, any removed frames did
not also remove the corresponding number of alerts from the alert-pipe.

ASTERISK-26632 #close

Change-Id: Ia98137c5bf6e9d6d202ce0eb36441851875863f6

2 years agores_agi: Prevent an AGI from eating frames it should not. (Re-do)
Richard Mudgett [Tue, 31 Jan 2017 22:38:49 +0000 (16:38 -0600)]
res_agi: Prevent an AGI from eating frames it should not. (Re-do)

A dialplan intercept routine is equivalent to an interrupt routine.  As
such, the routine must be done quickly and you do not have access to the
media stream.  These restrictions are necessary because the media stream
is the responsibility of some other code and interfering with or delaying
that processing is bad.  A possible future dialplan processing
architecture change may allow the interception routine to run in a
different thread from the main thread handling the media and remove the
execution time restriction.

* Made res_agi.c:run_agi() running an AGI in an interception routine run
in DeadAGI mode.  No touchy channel frames.

ASTERISK-25951

ASTERISK-26343

ASTERISK-26716

Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43

2 years agoFrame deferral: Revert API refactoring.
Richard Mudgett [Tue, 31 Jan 2017 22:32:18 +0000 (16:32 -0600)]
Frame deferral: Revert API refactoring.

There are several issues with deferring frames that are caused by the
refactoring.

1) The code deferring frames mishandles adding a deferred frame to the
deferred queue.  As a result the deferred queue can only be one frame
long.

2) Deferrable frames can come directly from the channel driver as well as
the read queue.  These frames need to be added to the deferred queue.

3) Whoever is deferring frames is really only doing the __ast_read() to
collect deferred frames and doesn't care about the returned frames except
to detect a hangup event.  When frame deferral is completed we must make
the normal frame processing see the hangup as a frame anyway.  As such,
there is no need to have varying hangup frame deferral methods.  We also
need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real.
That fake hangup is to cause the PBX thread to break out of loops to go
execute a new dialplan location.

4) To properly deal with deferrable frames from the channel driver as
pointed out by (2) above, means that it is possible to process a dialplan
interception routine while frames are deferred because of the
AST_CONTROL_READ_ACTION control frame.  Deferring frames is not
implemented as a re-entrant operation so you could have the unsupported
case of two sections of code thinking they have control of the media
stream.

A worse problem is because of the bad implementation of the AMI PlayDTMF
action.  It can cause two threads to be deferring frames on the same
channel at the same time.  (ASTERISK_25940)

* Rather than fix all these problems simply revert the API refactoring as
there is going to be only autoservice and safe_sleep deferring frames
anyway.

ASTERISK-26343

ASTERISK-26716 #close

Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496

2 years agoMerge "audiohooks: Muting a hook can mute underlying frames"
zuul [Thu, 2 Feb 2017 17:51:21 +0000 (11:51 -0600)]
Merge "audiohooks:  Muting a hook can mute underlying frames"

2 years agores_odbc: Remove deprecated settings from sample configuration file
Sean Bright [Thu, 2 Feb 2017 17:26:12 +0000 (12:26 -0500)]
res_odbc: Remove deprecated settings from sample configuration file

ASTERISK-26704 #close
Reported by: Anthony Messina

Change-Id: I976a1f94cf79c5f31e76174c61f5c6a65fd6354f

2 years agoMerge "res_pjsip: Handle invocation of callback on outgoing request when error occurs."
zuul [Thu, 2 Feb 2017 16:44:58 +0000 (10:44 -0600)]
Merge "res_pjsip: Handle invocation of callback on outgoing request when error occurs."

2 years agores_resolver_unbound.c: Fix frequent ref leak caught by excessive ref trap.
Richard Mudgett [Wed, 1 Feb 2017 23:14:53 +0000 (17:14 -0600)]
res_resolver_unbound.c: Fix frequent ref leak caught by excessive ref trap.

ASTERISK-26765

Change-Id: I27eb97df7f8d7e624b0b9a61c0fcee4718c86d8d

2 years agoaudiohooks: Muting a hook can mute underlying frames
Sean Bright [Wed, 1 Feb 2017 21:56:50 +0000 (16:56 -0500)]
audiohooks:  Muting a hook can mute underlying frames

If an audiohook is placed on a channel that does not require transcoding,
muting that hook will cause the underlying frames to be muted as well.

The original patch is from David Woolley but I have modified slightly.

ASTERISK-21094 #close
Reported by: David Woolley
Patches:
      ASTERISK-21094-Patch-1.8-1.txt (license #5737) patch uploaded
      by David Woolley

Change-Id: Ib2b68c6283e227cbeb5fa478b2d0f625dae338ed

2 years agoMerge "res_rtp_asterisk: Swap byte-order when sending signed linear"
Joshua Colp [Wed, 1 Feb 2017 21:36:23 +0000 (15:36 -0600)]
Merge "res_rtp_asterisk:  Swap byte-order when sending signed linear"

2 years agoUpdate qualifies when AOR configuration changes.
Mark Michelson [Wed, 1 Feb 2017 19:54:50 +0000 (13:54 -0600)]
Update qualifies when AOR configuration changes.

Prior to this change, qualifies would only update in the following
cases:
* A reload of res_pjsip.so was issued.
* A dynamic contact was re-registered after its AOR's qualify_frequency
  had been changed
This does not work well if you are using realtime for your AORs. You can
update your database to have a new qualify_frequency, but the permanent
contacts on that AOR will not have their qualifies updated. And the
dynamic contacts on that AOR will not have their qualifies updated until
the next registration, which could be a long time.

This change seeks to fix this problem by making it so that whenever AOR
configuration is applied, the contacts pertaining to that AOR have their
qualifies updated.

Additions from this patch:
* AOR sorcery objects now have an apply handler that calls into a newly
  added function in the OPTIONS code. This causes all contacts
  associated with that AOR to re-schedule qualifies.
* When it is time to qualify a contact, the OPTIONS code checks to see
  if the AOR can still be retrieved. If not, then qualification is
  canceled on the contact.

Alterations from this patch:
* The registrar code no longer updates contact's qualify_frequence and
  qualify_timeout. There is no point to this since those values already
  get updated when the AOR changes.
* Reloading res_pjsip.so no longer calls the OPTIONS initialization
  function. Reloading res_pjsip.so results in re-loading AORs, which
  results in re-scheduling qualifies.

Change-Id: I2e7c3316da28f389c45954f24c4e9389abac1121

2 years agores_pjsip: Handle invocation of callback on outgoing request when error occurs.
Joshua Colp [Tue, 31 Jan 2017 17:17:50 +0000 (17:17 +0000)]
res_pjsip: Handle invocation of callback on outgoing request when error occurs.

There are some error cases in PJSIP when sending a request that will
result in the callback for the request being invoked.  The code did not
handle this case and assumed on every error case that the callback was not
invoked.

The code has been changed to check whether the callback has been invoked
and if so to absorb the error and treat it as a success.

ASTERISK-26679
ASTERISK-26699

Change-Id: I563982ba204da5aa1428989a11c06dd9087fea91

2 years agores_rtp_asterisk: Swap byte-order when sending signed linear
Sean Bright [Mon, 30 Jan 2017 15:02:14 +0000 (10:02 -0500)]
res_rtp_asterisk:  Swap byte-order when sending signed linear

Before Asterisk 13, signed linear was converted into network byte order by a
smoother before being sent over the network. We restore this behavior by
forcing the creation of a smoother when slinear is in use and setting the
appropriate flags so that the byte order conversion is always done.

ASTERISK-24858 #close
Reported-by: Frankie Chin

Change-Id: I868449617d1a7819578f218c8c6b2111ad84f5a9

2 years agodebug_utilities: Install ast_logescalator to /var/lib/asterisk/scripts
George Joseph [Tue, 31 Jan 2017 18:46:08 +0000 (11:46 -0700)]
debug_utilities: Install ast_logescalator to /var/lib/asterisk/scripts

Forgot to install it with the original patch

Change-Id: I8bdb540a6694971ae5fe21f48d532332c6482e4c

2 years agoMerge "make_build_h: handle backslashes in external strings"
zuul [Tue, 31 Jan 2017 15:43:51 +0000 (09:43 -0600)]
Merge "make_build_h: handle backslashes in external strings"

2 years agoMerge "app_queue: Fix queues randomly disappearing on reload"
zuul [Mon, 30 Jan 2017 17:39:26 +0000 (11:39 -0600)]
Merge "app_queue: Fix queues randomly disappearing on reload"

2 years agoMerge "debug_utilities: Add ast_logescalator"
zuul [Sun, 29 Jan 2017 17:22:05 +0000 (11:22 -0600)]
Merge "debug_utilities:  Add ast_logescalator"

2 years agoMerge "libastssl/pj: libastssl/pj should have an so_version"
zuul [Sat, 28 Jan 2017 01:18:13 +0000 (19:18 -0600)]
Merge "libastssl/pj: libastssl/pj should have an so_version"

2 years agodebug_utilities: Add ast_logescalator
George Joseph [Wed, 25 Jan 2017 12:50:43 +0000 (05:50 -0700)]
debug_utilities:  Add ast_logescalator

The escalator works by creating a set of startup commands in cli.conf
that set up logger channels and issue the debug commands for the
subsystems specified.  If asterisk is running when it is executed,
the same commands will be issued to the running instance.  The original
cli.conf is saved before any changes are made and can be restored by
executing '$prog --reset'.

The log output will be stored in...
$astlogdir/message.$uniqueid
$astlogdir/debug.$uniqueid
$astlogdir/dtmf.$uniqueid
$astlogdir/fax.$uniqueid
$astlogdir/security.$uniqueid
$astlogdir/pjsip_history.$uniqueid
$astlogdir/sip_history.$uniqueid

Some minor tweaks were made to chan_sip, and res_pjsip_history
so their history output could be send to a log channel as packets
are captured.

A minor tweak was also made to manager so events are output to verbose
when "manager set debug on" is issued.

Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543

2 years agoMerge "tests: use datadir for sound files"
zuul [Fri, 27 Jan 2017 18:44:05 +0000 (12:44 -0600)]
Merge "tests: use datadir for sound files"

2 years agolibastssl/pj: libastssl/pj should have an so_version
Torrey Searle [Mon, 23 Jan 2017 15:35:38 +0000 (16:35 +0100)]
libastssl/pj: libastssl/pj should have an so_version

Issue introduced in b59956a87.  In the non-darwin case libastssl/pj
should be versioned.  This causes the symbol file for this lib
to not be generated.

Change-Id: Ib07ae8c40252813c488e2c1ac6204fd42816dd4c
(cherry picked from commit 54b027916a71f2b83b2050cef5ef704ea5de39b2)

2 years agoMerge "media: Add experimental support for RTCP feedback."
George Joseph [Fri, 27 Jan 2017 13:04:52 +0000 (07:04 -0600)]
Merge "media: Add experimental support for RTCP feedback."

2 years agoMerge "res_pjsip_endpoint_identifier_ip: Fix memory leak of hosts when resolving."
zuul [Fri, 27 Jan 2017 04:11:37 +0000 (22:11 -0600)]
Merge "res_pjsip_endpoint_identifier_ip: Fix memory leak of hosts when resolving."

2 years agomake_build_h: handle backslashes in external strings
kkm [Wed, 25 Jan 2017 01:51:07 +0000 (17:51 -0800)]
make_build_h: handle backslashes in external strings

LikewiseOpen creates user names with a backslash in them. A gentle
massage with sed(1) allows such strings to be inserted into build.h
properly quoted. I am also adding the same for host name and other
strings used in the script that are more or less user-controlled.

ASTERISK-26754

Change-Id: Iac5ef2b67a68ee58f35ddbf86bb818ba6eabecae

2 years agoapp_queue: Fix queues randomly disappearing on reload
kkm [Wed, 25 Jan 2017 04:31:38 +0000 (20:31 -0800)]
app_queue: Fix queues randomly disappearing on reload

With 500+ queues and a reload every minute, a random queue disappears
upon reload. The cause is mususe of the 'dead' flag. Namely, all queues
were marked dead up front, and then "resurrected" by dropping this flag
for those found in the configuration. But a queue marked dead can be
removed also when control leaves the app entry point on a PBX thread.

With this change, the queue is marked only not found, and at the end of
reload only the queues that are still not found are actually marked as
dead, so the dead flag is never reset, and set only on positively dead
queues.

ASTERISK-26755

Change-Id: I3a4537aec9eb8d8aeeaa0193407e3523feb004bf

2 years agoMerge "PJPROJECT logging: Fix detection of max supported log level."
zuul [Fri, 27 Jan 2017 00:46:22 +0000 (18:46 -0600)]
Merge "PJPROJECT logging: Fix detection of max supported log level."

2 years agoMerge "ari: Implement 'debug all' and request/response logging"
George Joseph [Thu, 26 Jan 2017 23:06:40 +0000 (17:06 -0600)]
Merge "ari: Implement 'debug all' and request/response logging"

2 years agoMerge "res_musiconhold.c: Fix format ref leak when parsing MOH config class."
George Joseph [Thu, 26 Jan 2017 22:05:07 +0000 (16:05 -0600)]
Merge "res_musiconhold.c: Fix format ref leak when parsing MOH config class."

2 years agoMerge "frame.c: Fix off-nominal format ref leaks."
George Joseph [Thu, 26 Jan 2017 22:03:49 +0000 (16:03 -0600)]
Merge "frame.c: Fix off-nominal format ref leaks."