asterisk/asterisk.git
2 years agochan_iax2: Reload of iax peer results in loss of host address/port
Richard Begg [Tue, 14 Mar 2017 21:51:41 +0000 (08:51 +1100)]
chan_iax2: Reload of iax peer results in loss of host address/port

When using a non-dynamic peer address, build_peer() invalidates the
peer address structure by setting the address family to unspecified.
However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup()
will not amend the peer address if the cache is still valid, resulting
in peer connectivity failures.
To fix this, we call ast_dnsmgr_refresh() instead.

ASTERISK-26865

Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082

2 years agoconfigure: Don't use the progress bar with curl when downloading to stdout
Matt Jordan [Tue, 14 Mar 2017 20:12:28 +0000 (15:12 -0500)]
configure: Don't use the progress bar with curl when downloading to stdout

In some scenarios, such as when there may not be a terminal (such as
inside a Docker container), curl will apparently direct the progress bar
to stdout. This can cause extra data to be appended to a file curl'd
down to stdout, resulting in md5 verification failures.

This patch removes the progress bar, and tells curl to download the file
silently.

ASTERISK-26872 #close

Change-Id: Ie860b020f627d4372b3e7ce9453de5faafeebe6c

2 years agoMerge "chan_pjsip: Don't assume a session will have a channel."
zuul [Tue, 14 Mar 2017 19:07:51 +0000 (14:07 -0500)]
Merge "chan_pjsip: Don't assume a session will have a channel."

2 years agoRFC sdp: Initial SDP creation
George Joseph [Thu, 2 Mar 2017 23:11:06 +0000 (16:11 -0700)]
RFC sdp: Initial SDP creation

* Added additional fields to ast_sdp_options.
* Re-organized ast_sdp.
* Updated field names to correspond to RFC4566 terminology.
* Created allocs/frees for SDP children.
* Created getters/setters for SDP children where appropriate.
* Added ast_sdp_create_from_state.
* Refactored res_sdp_translator_pjmedia for changes.

Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48

2 years agoMerge "chan_sip: Call not cancelled after receiving a 422 response"
Joshua Colp [Tue, 14 Mar 2017 16:47:30 +0000 (11:47 -0500)]
Merge "chan_sip: Call not cancelled after receiving a 422 response"

2 years agoconfigs/samples/hep.conf.sample: Clarify how the HEP stack works
Matt Jordan [Tue, 14 Mar 2017 14:55:06 +0000 (09:55 -0500)]
configs/samples/hep.conf.sample: Clarify how the HEP stack works

This patch updates the documenation in hep.conf.sample to better specify
how the various HEP modules interact.

ASTERISK-26717 #close

Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124

2 years agofuncs/func_devstate: Remove new line in Device field of during module load
Matt Jordan [Tue, 14 Mar 2017 14:59:48 +0000 (09:59 -0500)]
funcs/func_devstate: Remove new line in Device field of during module load

During module loading of func_devstate, Asterisk emits the current
device state of all Custom device states currently stored in the AstDB.
This was erroneously including a new line character ('\n') to the end of
the device state, causing two new lines to be emitted in
DeviceStateChange AMI events.

Note that this only happened for those device state changes that
occurred during startup. Regular device state changes for Custom device
states are handled elsewhere, and did not have the newline.

ASTERISK-26643 #close
Reported by: Roman Bedros
Tested by: Matt Jordan
patches:
  ami_devstate.diff uploaded by Roman Bedros (License 6842)

Change-Id: I1f4c02fc79c448d43bf725f5039c83d9611d7d93

2 years agomain/stasis_cache: Demote the ERROR message when removing a nonexistent item
Matt Jordan [Tue, 14 Mar 2017 14:37:34 +0000 (09:37 -0500)]
main/stasis_cache: Demote the ERROR message when removing a nonexistent item

This patch demotes the ERROR message that is displayed when a
nonexistent item is removed from the Stasis cache. The genesis of this
demotion is due to chan_sip's realtime peers and their interaction with
Asterisk's core ast_endpoint code, but ostensibly it could happen from
other channel drivers as well.

Since Mark Michelson already did an excellent job of explaining on this
issue, it is quoted here for posterity:

"Internally, when a realtime peer is retrieved, Asterisk creates an
ast_endpoint structure. When that peer is destroyed, the ast_endpoint is
destroyed as well. Part of the destruction of the ast_endpoint involves
clearing the Stasis cache of all information about that endpoint. The
problem here is that the act of creating the ast_endpoint is not enough
to actually put any information in the Stasis cache. Instead, something
has to happen, such as a state change, in order for the Stasis cache to
have any information about that endpoint. When a device registers,
chan_sip creates an ast_endpoint structure, processes the REGISTER, and
then destroys the ast_endpoint. When the ast_endpoint is destroyed,
there is nothing to destroy in the Stasis cache, so an error message is
emitted. When you use rtcachefriends, ast_endpoint structures persist
for the lifetime of the module and so you do not see this error
message."

ASTERISK-25237 #close

Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70

2 years agores_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue
Matt Jordan [Wed, 8 Mar 2017 18:39:20 +0000 (12:39 -0600)]
res_pjsip_endpoint_identifier_ip: Clean up a spaces/tabs issue

Tabs > spaces. Always.

Change-Id: I899ff662361c7ab0327173bd7851a67b53dd65f1

2 years agochan_pjsip: Don't assume a session will have a channel.
Joshua Colp [Sun, 12 Mar 2017 14:21:16 +0000 (14:21 +0000)]
chan_pjsip: Don't assume a session will have a channel.

When querying for PJSIP specific information using the dialplan
function CHANNEL() it is possible that the underlying session
will no longer have a channel associated with it. This is
most likely to occur when the RTCP HEP module attempts to get
the channel name. If this happens then a crash will occur.

This change just adds a check that the channel exists on the
session before querying it.

ASTERISK-26857

Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01

2 years agopjproject_bundled: Reduce the need for rebuilds
George Joseph [Sat, 11 Mar 2017 02:29:04 +0000 (19:29 -0700)]
pjproject_bundled:  Reduce the need for rebuilds

Bundled pjproject should now only rebuild if one of the menuselect
"Compiler Flags" options changes.

Change-Id: If114a2e16b9e77af371a600d6a5e197bbf28fe43

2 years agoMerge "pjsip/cli_commands: pjsip show channelstats shows wrong codec"
Joshua Colp [Fri, 10 Mar 2017 22:02:08 +0000 (16:02 -0600)]
Merge "pjsip/cli_commands: pjsip show channelstats shows wrong codec"

2 years agoMerge "res_musiconhold: moh general section is a class and issues warning"
zuul [Fri, 10 Mar 2017 00:32:03 +0000 (18:32 -0600)]
Merge "res_musiconhold: moh general section is a class and issues warning"

2 years agoMerge "media_cache: Prefer ast_file_is_readable() over access()"
Joshua Colp [Thu, 9 Mar 2017 22:10:25 +0000 (16:10 -0600)]
Merge "media_cache: Prefer ast_file_is_readable() over access()"

2 years agopjsip/cli_commands: pjsip show channelstats shows wrong codec
Daniel Journo [Sun, 5 Mar 2017 21:26:07 +0000 (21:26 +0000)]
pjsip/cli_commands: pjsip show channelstats shows wrong codec

* cli_commands.c Fixed CLI output

ASTERISK-26822 #close

Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01

2 years agoMerge "pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel"
Joshua Colp [Thu, 9 Mar 2017 20:29:02 +0000 (14:29 -0600)]
Merge "pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel"

2 years agores_musiconhold: moh general section is a class and issues warning
Daniel Journo [Wed, 8 Mar 2017 20:29:16 +0000 (20:29 +0000)]
res_musiconhold: moh general section is a class and issues warning

* res_musiconhold.c: Ensure the general section is not treated as
a moh class.

ASTERISK-26353 #close

Change-Id: Ia3dbd11ea2b43ab3e6c820a9827811dd24bea82d

2 years agomedia_cache: Prefer ast_file_is_readable() over access()
Sean Bright [Wed, 8 Mar 2017 23:08:52 +0000 (18:08 -0500)]
media_cache: Prefer ast_file_is_readable() over access()

Change-Id: Icc0dc6e61b2e68d5cdcb74b016b2726a388c7def

2 years agopbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel
Sean Bright [Tue, 7 Mar 2017 12:25:25 +0000 (07:25 -0500)]
pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel

Set a variable on the channel that indicates which attempt number we
are currently performing to allow for attempt-specific behavior.

ASTERISK-26568 #close
Reported by: Roman Shubovich

Change-Id: Iacd7e8d43b0ed5b6cb021c62f41f1a1f5733dd89

2 years agores_pjsip_transport_websocket: Add support for IPv6.
Joshua Colp [Tue, 7 Mar 2017 13:37:52 +0000 (13:37 +0000)]
res_pjsip_transport_websocket: Add support for IPv6.

This change adds a PJSIP patch (which has been contributed upstream)
to allow the registration of IPv6 transport types.

Using this the res_pjsip_transport_websocket module now registers
an IPv6 Websocket transport and uses it for the corresponding
traffic.

ASTERISK-26685

Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647

2 years agoapp_voicemail: Cannot set fromstring on a per-mailbox basis
Daniel Journo [Wed, 8 Mar 2017 14:16:29 +0000 (14:16 +0000)]
app_voicemail: Cannot set fromstring on a per-mailbox basis

* apps/app_voicemail.c fromstring field added to mailbox which will
override the global fromstring if set.

ASTERISK-24562 #close

Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe

2 years agoMerge "res_http_websocket: Fix faulty read logic."
zuul [Wed, 8 Mar 2017 16:05:19 +0000 (10:05 -0600)]
Merge "res_http_websocket: Fix faulty read logic."

2 years agoMerge "pbx_spool: Gracefully handle long lines in call files"
zuul [Tue, 7 Mar 2017 23:54:29 +0000 (17:54 -0600)]
Merge "pbx_spool: Gracefully handle long lines in call files"

2 years agores_http_websocket: Fix faulty read logic.
Mark Michelson [Tue, 7 Mar 2017 19:38:17 +0000 (13:38 -0600)]
res_http_websocket: Fix faulty read logic.

When doing some WebRTC testing, I found that the websocket would
disconnect whenever I attempted to place a call into Asterisk. After
looking into it, I pinpointed the problem to be due to the iostreams
change being merged in.

Under certain circumstances, a call to ast_iostream_read() can return a
negative value. However, in this circumstance, the websocket code was
treating this negative return as if it were a partial read from the
websocket. The expected length would get adjusted by this negative
value, resulting in the expected length being too large.

This patch simply adds an if check to be sure that we are only updating
the expected length of a read when the return from a read is positive.

ASTERISK-26842 #close
Reported by Mark Michelson

Change-Id: Ib4423239828a013d27d7bc477d317d2f02db61ab

2 years agochan_sip: Call not cancelled after receiving a 422 response
Jean Aunis [Tue, 7 Mar 2017 14:12:48 +0000 (15:12 +0100)]
chan_sip: Call not cancelled after receiving a 422 response

When receiving a 422 response, the invitestate variable must be reset to
INV_CALLING.

ASTERISK-26841

Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099

2 years agocore: Add stream topology changing primitives with tests.
Joshua Colp [Tue, 7 Mar 2017 11:22:18 +0000 (11:22 +0000)]
core: Add stream topology changing primitives with tests.

This change adds a few things to facilitate stream topology changing:

1. Control frame types have been added for use by the channel driver
to notify the application that the channel wants to change the stream
topology or that a stream topology change has been accepted. They are
also used by the indicate interface to the channel that the application
uses to indicate it wants to do the same.

2. Legacy behavior has been adopted in ast_read() such that if a
channel requests a stream topology change it is denied automatically
and the current stream topology is preserved if the application is
not capable of handling streams.

Tests have also been written which confirm the multistream and
non-multistream behavior.

ASTERISK-26839

Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9

2 years agoSaynumber is trying to get "and" from "digits/" subfolder
Daniel Journo [Mon, 6 Mar 2017 21:54:23 +0000 (21:54 +0000)]
Saynumber is trying to get "and" from "digits/" subfolder

* say.c Changed 'digits/and' to 'vm-and' for en_GB

ASTERISK-26598 #close

Change-Id: If1b713e5daea6f952b339f139178d292a6c4fcfe

2 years agopbx_spool: Gracefully handle long lines in call files
Sean Bright [Mon, 6 Mar 2017 19:15:45 +0000 (14:15 -0500)]
pbx_spool: Gracefully handle long lines in call files

Per the linked issue, we aren't checking the buffer filled by fgets()
to determine if it contains a newline, so we will fail to correctly
parse the trailing portion of a long line.

This patch increases the buffer size from 256 to 1024, and skips any
line that exceeds that length, logging a warning in the process.

ASTERISK-17067 #close
Reported by: Dave Olszewski

Change-Id: I51bcf270c1b4347ba05b43f18dc2094c76f5d7b0

2 years agocore: Cleanup ast_get_hint() usage.
Richard Mudgett [Fri, 3 Mar 2017 03:27:00 +0000 (21:27 -0600)]
core: Cleanup ast_get_hint() usage.

* manager.c:manager_state_cb() Fix potential use of uninitialized hint[]
if a hint does not exist for the requested extension.  Ran into this when
developing a testsuite test.  The AMI event ExtensionStatus came out with
the hint header value containing garbage.  The AMI event PresenceStatus
also had the same issue.

* manager.c:action_extensionstate() no need to completely initialize the
hint[].  Only initialize the first element.

* pbx.c:ast_add_hint() Remove unnecessary assignment.

* chan_sip.c: Eliminate an unneeded hint[] local variable.  We only care
about the return value of ast_get_hint() there.

Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b

2 years agoMerge "res_pjsip WebRTC/websockets: Fix usage of WS vs WSS."
Joshua Colp [Wed, 1 Mar 2017 23:24:43 +0000 (17:24 -0600)]
Merge "res_pjsip WebRTC/websockets: Fix usage of WS vs WSS."

2 years agoMerge "stream: Unit tests for stream read and tweaks framework"
Joshua Colp [Wed, 1 Mar 2017 20:58:45 +0000 (14:58 -0600)]
Merge "stream: Unit tests for stream read and tweaks framework"

2 years agoMerge "res_config_pgsql: Make 'require' return consistent with other backends"
Joshua Colp [Wed, 1 Mar 2017 17:24:22 +0000 (11:24 -0600)]
Merge "res_config_pgsql: Make 'require' return consistent with other backends"

2 years agores_pjsip WebRTC/websockets: Fix usage of WS vs WSS.
Jørgen H [Thu, 16 Feb 2017 10:22:47 +0000 (10:22 +0000)]
res_pjsip WebRTC/websockets: Fix usage of WS vs WSS.

According to the RFC[1] WSS should only be used in the Via header
for secure Websockets.

* Use WSS in Via for secure transport.

* Only register one transport with the WS name because it would be
ambiguous.  Outgoing requests may try to find the transport by name and
pjproject only finds the first one registered.  This may mess up unsecure
websockets but the impact should be minimal.  Firefox and Chrome do not
support anything other than secure websockets anymore.

* Added and updated some debug messages concerning websockets.

* security_events.c: Relax case restriction when determining security
transport type.

* The res_pjsip_nat module has been updated to not touch the transport
on Websocket originating messages.

[1] https://tools.ietf.org/html/rfc7118

ASTERISK-26796 #close

Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12

2 years agostream: Unit tests for stream read and tweaks framework
George Joseph [Fri, 24 Feb 2017 21:30:33 +0000 (14:30 -0700)]
stream: Unit tests for stream read and tweaks framework

* Removed the AST_CHAN_TP_MULTISTREAM tech property.  We now rely
  on read_stream being set to indicate a multi stream channel.
* Added ast_channel_is_multistream convenience function.
* Fixed issue where stream and default_stream weren't being set on
  a frame retrieved from the queue.
* Now testing for NULL being returned from the driver's read or
  read_stream callback.
* Fixed issue where the dropnondefault code was crashing on a
  NULL f.
* Now enforcing that if either read_stream or write_stream are
  set when ast_channel_tech_set is called that BOTH are set.
* Added the unit tests.

ASTERISK-26816

Change-Id: If7792b20d782e71e823dabd3124572cf0a4caab2

2 years agores_config_pgsql: Make 'require' return consistent with other backends
Sean Bright [Wed, 1 Mar 2017 13:23:55 +0000 (08:23 -0500)]
res_config_pgsql: Make 'require' return consistent with other backends

res_config_pgsql should match the behavior of other realtime backend
drivers so that queue_log can disable adaptive logging.

ASTERISK-25628 #close
Reported by: Dmitry Wagin

Change-Id: Ic1fb1600c7ce10fdfb1bcdc43c5576b7e0014372

2 years agoSDP: Add initial SDP state machine.
Mark Michelson [Wed, 22 Feb 2017 21:11:29 +0000 (15:11 -0600)]
SDP: Add initial SDP state machine.

This introduces and documents the various states in the state machine.
This also introduces API functions that induce state changes, and places
TODO comments telling what needs to be done in addition to what is
already there. Those TODOs will be replaced with real code in upcoming
changes.

Change-Id: I871c0eb480b4c84d83e91ac5628e7a673e8b89ed

2 years agoMerge "media_cache: Mark cache entry stale if cache file is removed"
Joshua Colp [Wed, 1 Mar 2017 10:47:59 +0000 (04:47 -0600)]
Merge "media_cache: Mark cache entry stale if cache file is removed"

2 years agoMerge "res_config_pgsql: Release table locks where appropriate"
Joshua Colp [Wed, 1 Mar 2017 01:37:36 +0000 (19:37 -0600)]
Merge "res_config_pgsql: Release table locks where appropriate"

2 years agoMerge "res_pjsip_outbound_registration: Subscribe to network change events"
Joshua Colp [Wed, 1 Mar 2017 01:25:04 +0000 (19:25 -0600)]
Merge "res_pjsip_outbound_registration:  Subscribe to network change events"

2 years agoMerge "build: Warn if asterisk is installed in both 32 and 64 bit sys dirs"
Joshua Colp [Tue, 28 Feb 2017 23:47:46 +0000 (17:47 -0600)]
Merge "build:  Warn if asterisk is installed in both 32 and 64 bit sys dirs"

2 years agoMerge "res_pjsip_pubsub: Remove unneeded endpoint unref"
Joshua Colp [Tue, 28 Feb 2017 23:32:25 +0000 (17:32 -0600)]
Merge "res_pjsip_pubsub:  Remove unneeded endpoint unref"

2 years agoMerge "bridge_native_rtp: Handle case where channel joins already suspended."
zuul [Tue, 28 Feb 2017 23:14:25 +0000 (17:14 -0600)]
Merge "bridge_native_rtp: Handle case where channel joins already suspended."

2 years agomedia_cache: Mark cache entry stale if cache file is removed
Sean Bright [Tue, 28 Feb 2017 19:48:51 +0000 (14:48 -0500)]
media_cache: Mark cache entry stale if cache file is removed

In the event that a cache file is removed out from under us, we should
treat the cache entry as stale and force a refresh.

ASTERISK-26774 #close
Reported by: Igor Gamayunov

Change-Id: I3b1bd0c999d59d18664ef73a29823bc5b431dc52

2 years agoMerge "config: Improve documentation and behavior of outbound_proxy option."
Joshua Colp [Tue, 28 Feb 2017 20:44:29 +0000 (14:44 -0600)]
Merge "config: Improve documentation and behavior of outbound_proxy option."

2 years agoMerge "res_pjsip: Fix crash when contact has no status"
Joshua Colp [Tue, 28 Feb 2017 16:24:45 +0000 (10:24 -0600)]
Merge "res_pjsip: Fix crash when contact has no status"

2 years agores_config_pgsql: Release table locks where appropriate
Sean Bright [Tue, 28 Feb 2017 15:41:45 +0000 (10:41 -0500)]
res_config_pgsql: Release table locks where appropriate

The find_table() functions NULL or a locked table pointer. We are
not consistently calling release_table() in failure paths.

Change-Id: I6f665b455799c84b036e5b34904b82b05eab9544

2 years agopjsip.conf.sample: user_agent: not a specific version
Tzafrir Cohen [Tue, 28 Feb 2017 11:41:18 +0000 (13:41 +0200)]
pjsip.conf.sample: user_agent: not a specific version

Use the description of useragent from sip.conf here.

ASTERISK-26825 #close

Change-Id: I5b33a4aaa0ae1d793289d05e3bc09521affbf755

2 years agores_pjsip_pubsub: Remove unneeded endpoint unref
George Joseph [Tue, 28 Feb 2017 02:07:06 +0000 (19:07 -0700)]
res_pjsip_pubsub:  Remove unneeded endpoint unref

When a subscription was being recreated and the endpoint wasn't
found, we were trying to unref the endpoint.  This was causing
FRACKs.  Removed the unref.

ASTERISK-26823 #close

Change-Id: If86d2aecff8fe853c7f38a1bfde721fcef3cd164

2 years agores_pjsip: Fix crash when contact has no status
Jørgen H [Thu, 16 Feb 2017 10:16:47 +0000 (10:16 +0000)]
res_pjsip: Fix crash when contact has no status

This change fixes an assumption in res_pjsip that a contact will
always have a status. There is a race condition where this is
not true and would crash. The status will now be unknown when
this situation occurs.

ASTERISK-26623 #close

Change-Id: Id52d3ca4d788562d236da49990a319118f8d22b5

2 years agores_pjsip_outbound_registration: Subscribe to network change events
George Joseph [Wed, 22 Feb 2017 00:06:00 +0000 (17:06 -0700)]
res_pjsip_outbound_registration:  Subscribe to network change events

Outbound registration now subscribes to network change events
published by res_stun_monitor and refreshes all registrations
when an event happens.

The 'pjsip send (un)register' CLI commands were updated to accept
'*all' as an argument to operate on all registrations.

The 'PJSIP(Un)Register' AMI commands were also updated to
accept '*all'.

ASTERISK-26808 #close

Change-Id: Iad58a9e0aa5d340477fca200bf293187a6ca5a25

2 years agobuild: Warn if asterisk is installed in both 32 and 64 bit sys dirs
George Joseph [Mon, 27 Feb 2017 18:25:01 +0000 (11:25 -0700)]
build:  Warn if asterisk is installed in both 32 and 64 bit sys dirs

... and clean them both up on uninstall.

We've fixed the issue where 'make install' was installing to
/usr/lib on 64-bit systems that use /usr/lib64.  Now we need
to clean up the remnants in /usr/lib.

* 'make install' now prints a warning if DESTDIR/ASTLIBDIR
  contains 'lib64' and libasterisk* shared libraries or modules
  are also found in DESTDIR/ASTLIBDIR with 'lib64' transformed
  to 'lib'.

* 'make uninstall' ALWAYS cleans up both DESTDIR/ASTLIBDIR and
  DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'.

ASTERISK-26705

Change-Id: I6edddeb3c07a51e7c7ba7cac3c05e4bf3ec3f01f

2 years agobridge_native_rtp: Handle case where channel joins already suspended.
Joshua Colp [Mon, 27 Feb 2017 13:02:38 +0000 (13:02 +0000)]
bridge_native_rtp: Handle case where channel joins already suspended.

The bridge_native_rtp module did not properly handle the case where
a smart bridge operation occurs while a channel is suspended. In this
scenario the module would incorrectly set up local or remote RTP
bridging despite the media having to flow through Asterisk. The remote
endpoint would see two media streams and experience wonky audio.

The module has been changed so that it ensures both channels are
not suspended when performing the native RTP bridging and this
requirement has been documented in the bridge technology.

ASTERISK-26781

Change-Id: Id4022d73ace837d4a293106445e3ade10dbc7c7c

2 years agoMerge "channel: Add ast_read_stream function for reading frames from all streams."
George Joseph [Mon, 27 Feb 2017 14:51:26 +0000 (08:51 -0600)]
Merge "channel: Add ast_read_stream function for reading frames from all streams."

2 years agoMerge "Binaural synthesis (confbridge): DTMF conference management."
zuul [Fri, 24 Feb 2017 23:35:43 +0000 (17:35 -0600)]
Merge "Binaural synthesis (confbridge): DTMF conference management."

2 years agoBinaural synthesis (confbridge): DTMF conference management.
frahaase [Fri, 12 Aug 2016 16:23:40 +0000 (18:23 +0200)]
Binaural synthesis (confbridge): DTMF conference management.

DTMF configuration options for the binaural softmix bridge:
toggle binaural rendering (per channel).

ASTERISK-26292

Change-Id: Ibfe708b9fe26097c1798fcbfcc4dc461267d8af8

2 years agoconfig: Improve documentation and behavior of outbound_proxy option.
Joshua Colp [Fri, 24 Feb 2017 17:49:59 +0000 (17:49 +0000)]
config: Improve documentation and behavior of outbound_proxy option.

This change updates the documentation for the outbound_proxy option
to ensure it is consistently stated that a full SIP URI must be
provided for the option.

The res_pjsip_outbound_registration module has also been changed so
that the provided outbound_proxy value is checked to ensure it is a
URI and if not an error is output stating so.

ASTERISK-26782

Change-Id: I6c239a32274846fd44e65b44ad9bf6373479b593

2 years agoMerge "pjproject_bundled: Update for pjproject 2.6"
Joshua Colp [Fri, 24 Feb 2017 18:49:07 +0000 (12:49 -0600)]
Merge "pjproject_bundled: Update for pjproject 2.6"

2 years agoMerge "Binaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix."
Joshua Colp [Fri, 24 Feb 2017 18:49:00 +0000 (12:49 -0600)]
Merge "Binaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix."

2 years agoMerge "build: Execute ldconfig to build cache. (take two)"
zuul [Fri, 24 Feb 2017 18:12:13 +0000 (12:12 -0600)]
Merge "build: Execute ldconfig to build cache. (take two)"

2 years agoMerge "channel: Add support for writing to a specific stream."
zuul [Fri, 24 Feb 2017 17:16:13 +0000 (11:16 -0600)]
Merge "channel: Add support for writing to a specific stream."

2 years agochannel: Add ast_read_stream function for reading frames from all streams.
Joshua Colp [Thu, 23 Feb 2017 19:03:15 +0000 (19:03 +0000)]
channel: Add ast_read_stream function for reading frames from all streams.

This change introduces an ast_read_stream function and callback in
the channel technology which allows reading frames from all streams
and not just the default streams.

The stream number has also been added to frames. This is to allow the
case where frames are queued onto the channel instead of being read
directly from the driver.

This change does impose a restriction on reading though: a chain of
frames can only contain frames from the same stream.

ASTERISK-26816

Change-Id: I5d7dc35e86694df91fd025126f6cfe0453aa38ce

2 years agopjproject_bundled: Update for pjproject 2.6
George Joseph [Fri, 10 Feb 2017 00:05:01 +0000 (17:05 -0700)]
pjproject_bundled: Update for pjproject 2.6

 * Removed all 2.5.5 functional patches.
 * Updated usages of pj_release_pool to be "safe".
 * Updated configure options to disable webrtc.
 * Updated config_site.h to disable webrtc in pjmedia.
 * Added Richard Mudgett's recent resolver patches.

Change-Id: Ib400cc4dfca68b3d07ce14d314e829bfddc252c7

2 years agobuild: Execute ldconfig to build cache. (take two)
George Joseph [Thu, 23 Feb 2017 21:49:17 +0000 (14:49 -0700)]
build: Execute ldconfig to build cache. (take two)

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.  To make matters worse, options were being passed
to ldconfig on both Linux and FreeBSD that actually prevented
the rebuild of the cache.

 * Fedora has a /usr/share/config.site that automatically tells
   autoconf to use /usr/lib64 but CentOS does not. This logic was
   copied to configure.ac and modified so systems like Ubuntu,
   which still use /usr/lib for 64-bit systems, aren't affected.

Now that we have them in the correct directory...

In order for the system loader to find libasteriskssl and
libasteriskpj, one of 3 things has to happen...

  - The linker cache must be rebuilt including the directory
    where the libasterisk* libraries were installed.  Only root
    can rebuild the cache.  This was busted.
  - We have to link the asterisk binary with an rpath pointing
    to the directrory where the libasterisk* libraries were
    installed.  This makes things very complicated and will happen
    over the collective dead bodies of everyone who's had to
    package a distribution with an rpath.
  - Finally, you can start asterisk with LD_LIBRARY_PATH set to the
    directrory where the libasterisk* libraries were installed.

There are no other options. So...

 * The invokation of ldconfig has been moved from main/Makefile
   to ASTTOPDIR/Makefile, the options have been removed, and
   DESTDIR/ASTLIBDIR appended.  If you aren't root, you will be
   warned after the "Asterisk Installation Compete" banner that
   you must re-run 'make install' as root, manually run
   'ldconfig DESTDIR/ASTLIBDIR' as root, or run asterisk with
   LD_LIBRARY_PATH.

ASTERISK-26705

Change-Id: I2a64b7c33a7d3e9bde20f47e3d3ab771977af982

2 years agores_config_pgsql: Fix thread safety problems
Sean Bright [Thu, 23 Feb 2017 20:48:53 +0000 (15:48 -0500)]
res_config_pgsql: Fix thread safety problems

* A missing AST_LIST_UNLOCK() in find_table()

* The ESCAPE_STRING() macro uses pgsqlConn under the hood and we were
  not consistently locking before calling it.

* There were a handful of other places where pgsqlConn was accessed
  directly without appropriate locking.

Change-Id: Iea63f0728f76985a01e95b9912c3c5c6065836ed

2 years agoMerge "res_config_ldap: Various code improvements"
Joshua Colp [Thu, 23 Feb 2017 20:47:54 +0000 (14:47 -0600)]
Merge "res_config_ldap: Various code improvements"

2 years agochannel: Add support for writing to a specific stream.
Joshua Colp [Wed, 22 Feb 2017 11:00:57 +0000 (11:00 +0000)]
channel: Add support for writing to a specific stream.

This change adds an ast_write_stream function which allows
writing a frame to a specific media stream. It also moves
ast_write() to using this underneath by writing media
frames provided to it to the default streams of the channel.
Existing functionality (such as audiohooks, framehooks, etc)
are limited to being applied to the default stream only.

Unit tests have also been added which test the behavior of
both non-multistream and multistream channels to confirm that
the write() and write_stream() callbacks are invoked
appropriately.

ASTERISK-26793

Change-Id: I4df20d1b65bd4d787fce0b4b478e19d2dfea245c

2 years agoBinaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix.
frahaase [Fri, 12 Aug 2016 16:23:18 +0000 (18:23 +0200)]
Binaural synthesis (confbridge): Adds binaural synthesis to bridge_softmix.

Adds binaural synthesis to bridge_softmix (via convolution using libfftw3).
Binaural synthesis is conducted at 48kHz.
For a conference, only one spatial representation is rendered.
The default rendering is applied for mono-capable channels.

ASTERISK-26292

Change-Id: Iecdb381b6adc17c961049658678f6219adae1ddf

2 years agores_config_ldap: Various code improvements
Sean Bright [Wed, 22 Feb 2017 14:53:25 +0000 (09:53 -0500)]
res_config_ldap: Various code improvements

The initial motivation for this patch was to properly handle memory
allocation failures - we weren't checking the return values from the
various LDAP library allocation functions.

In the process, because update_ldap() and update2_ldap() were
substantially the same code, they've been consolidated.

Change-Id: Iebcfe404177cc6860ee5087976fe97812221b822

2 years agobuild_tools: Fix download_externals to allow the use of curl or wget
Michael L. Young [Wed, 22 Feb 2017 19:08:05 +0000 (14:08 -0500)]
build_tools:  Fix download_externals to allow the use of curl or wget

Not sure if this is really a bug versus an improvement. I can see it being
viewed as a bug though by some.

The current build_tools/download_externals file depends on wget in order to
download external modules.  The current build system is able to discover
which tool to use for fetching remote files - either wget or curl.

This patch takes advantage of this capability by modifying the two calls to
the wget binary to instead use what was discovered by the build system.

ASTERISK-26812 #close

Change-Id: If9411a2554f009274d377445613ae91192d948a1

2 years agoMerge "Revert "build: Execute ldconfig to build cache.""
zuul [Wed, 22 Feb 2017 19:56:48 +0000 (13:56 -0600)]
Merge "Revert "build: Execute ldconfig to build cache.""

2 years agoMerge "pbx_realtime: Prevent premature extension matching"
zuul [Wed, 22 Feb 2017 18:08:50 +0000 (12:08 -0600)]
Merge "pbx_realtime: Prevent premature extension matching"

2 years agoMerge "core: Show streams in "core show channel"."
zuul [Wed, 22 Feb 2017 17:40:02 +0000 (11:40 -0600)]
Merge "core: Show streams in "core show channel"."

2 years agoRevert "build: Execute ldconfig to build cache."
Joshua Colp [Wed, 22 Feb 2017 17:12:54 +0000 (11:12 -0600)]
Revert "build: Execute ldconfig to build cache."

This reverts commit 28c8e4f58f0f38792c7c79a05bd07788ebf15332.

Change-Id: Ie2e1aaf61fd49045994974a4581545ac8348fe4c

2 years agoMerge "Add initial SDP state code."
Joshua Colp [Wed, 22 Feb 2017 16:56:02 +0000 (10:56 -0600)]
Merge "Add initial SDP state code."

2 years agopbx_realtime: Prevent premature extension matching
Sean Bright [Tue, 21 Feb 2017 16:47:41 +0000 (11:47 -0500)]
pbx_realtime: Prevent premature extension matching

The patterns provided by pbx_realtime were checked in the order in
which they were returned from the realtime backend. If there was
overlap between multiple patterns, the first one to correctly match was
chosen even though it may not have been the best match.

We now sort the patterns descending by their length and compare in that
order. There may be cases where this still results in a sub-optimal
match, but this patch should improve the overall behavior.

ASTERISK-18271 #close
Reported by: Charlie Smurthwaite

Change-Id: I56d9ac15810eb1775966b669c3028e32cc7bd809

2 years agoMerge "pbx_dundi: DUNDi weight parameter not processed correctly"
zuul [Wed, 22 Feb 2017 16:19:45 +0000 (10:19 -0600)]
Merge "pbx_dundi: DUNDi weight parameter not processed correctly"

2 years agoMerge "realtime: Fix ast_load_realtime_multientry handling"
zuul [Wed, 22 Feb 2017 16:11:49 +0000 (10:11 -0600)]
Merge "realtime: Fix ast_load_realtime_multientry handling"

2 years agocore: Show streams in "core show channel".
Joshua Colp [Wed, 22 Feb 2017 14:32:23 +0000 (14:32 +0000)]
core: Show streams in "core show channel".

The "core show channel" CLI command will now output the streams
present on the channel with their details.

ASTERISK-26811

Change-Id: I9c95b57aa09415005f0677a1949a0feb07e4987a

2 years agoMerge "realtime: Centralize some common realtime backend code"
Joshua Colp [Wed, 22 Feb 2017 11:53:50 +0000 (05:53 -0600)]
Merge "realtime: Centralize some common realtime backend code"

2 years agoMerge "res_pjsip_authenticator_digest.c: Fix sorcery's immutable contract violation."
zuul [Wed, 22 Feb 2017 05:13:44 +0000 (23:13 -0600)]
Merge "res_pjsip_authenticator_digest.c: Fix sorcery's immutable contract violation."

2 years agoMerge "res_pjsip: Update artificial auth whenever default_realm changes."
zuul [Wed, 22 Feb 2017 04:55:13 +0000 (22:55 -0600)]
Merge "res_pjsip: Update artificial auth whenever default_realm changes."

2 years agoMerge "res_pjsip: Update authentication realm documentation."
zuul [Wed, 22 Feb 2017 04:39:09 +0000 (22:39 -0600)]
Merge "res_pjsip: Update authentication realm documentation."

2 years agoMerge "pjproject: Fixes to resolve DNS SRV crashes."
zuul [Wed, 22 Feb 2017 02:23:20 +0000 (20:23 -0600)]
Merge "pjproject: Fixes to resolve DNS SRV crashes."

2 years agoMerge "pjproject: Increase SENDER_WIDTH column size for 64-bit system logs."
zuul [Wed, 22 Feb 2017 02:14:26 +0000 (20:14 -0600)]
Merge "pjproject: Increase SENDER_WIDTH column size for 64-bit system logs."

2 years agoMerge "pjsip_distributor.c: Update some debug messages to get transaction name."
zuul [Wed, 22 Feb 2017 02:01:08 +0000 (20:01 -0600)]
Merge "pjsip_distributor.c: Update some debug messages to get transaction name."

2 years agoMerge "res_config_ldap: Don't try to delete non-existent attributes"
zuul [Wed, 22 Feb 2017 00:58:28 +0000 (18:58 -0600)]
Merge "res_config_ldap: Don't try to delete non-existent attributes"

2 years agoMerge "res_config_ldap: Remove extraneous line numbers from log messages"
zuul [Wed, 22 Feb 2017 00:58:27 +0000 (18:58 -0600)]
Merge "res_config_ldap: Remove extraneous line numbers from log messages"

2 years agoMerge "res_config_ldap: Make memory allocation more consistent"
zuul [Wed, 22 Feb 2017 00:58:26 +0000 (18:58 -0600)]
Merge "res_config_ldap: Make memory allocation more consistent"

2 years agoMerge "res_config_ldap: Fix configuration inheritance from _general"
zuul [Wed, 22 Feb 2017 00:24:05 +0000 (18:24 -0600)]
Merge "res_config_ldap: Fix configuration inheritance from _general"

2 years agoMerge "res_config_ldap: Fix erroneous LDAP_MOD_REPLACE in LDAP modify"
zuul [Tue, 21 Feb 2017 23:57:24 +0000 (17:57 -0600)]
Merge "res_config_ldap: Fix erroneous LDAP_MOD_REPLACE in LDAP modify"

2 years agopbx_dundi: DUNDi weight parameter not processed correctly
Sean Bright [Tue, 21 Feb 2017 21:09:47 +0000 (16:09 -0500)]
pbx_dundi: DUNDi weight parameter not processed correctly

The DUNDi weight field is not always converted from network byte order
to host byte order. This can result in incorrect weight values and
incorrect selection of DUNDi destinations.

ASTERISK-18731 #close
Reported by: Peter Racz
Patches:
dundi_weight.patch (license #6290) patch uploaded by Peter Racz

Change-Id: Iba3e1a700ff539db57211a7bbc26f7b22ea9a1be

2 years agoAdd initial SDP state code.
Mark Michelson [Wed, 15 Feb 2017 20:43:36 +0000 (14:43 -0600)]
Add initial SDP state code.

This establishes the basic allocation/destruction of an SDP state
object, plus some of the simpler getter methods involved. Subsequent
tasks will deal with adding a state machine, creating SDPs from
capabilities and options, and merging SDPs into a joint SDP.

Change-Id: Ie3757ce186f04b65e9d1883f5aace53f24e53709

2 years agoMerge changes from topic 'sdp_state_beginnings'
Joshua Colp [Tue, 21 Feb 2017 19:37:03 +0000 (13:37 -0600)]
Merge changes from topic 'sdp_state_beginnings'

* changes:
  Add SDP translator and PJMEDIA implementation.
  Add initial SDP options.

2 years agoMerge "build: Execute ldconfig to build cache."
Joshua Colp [Tue, 21 Feb 2017 19:36:38 +0000 (13:36 -0600)]
Merge "build: Execute ldconfig to build cache."

2 years agorealtime: Fix ast_load_realtime_multientry handling
Sean Bright [Tue, 21 Feb 2017 16:47:28 +0000 (11:47 -0500)]
realtime: Fix ast_load_realtime_multientry handling

ast_load_realtime_multientry() returns an ast_config structure whose
ast_categorys are keyed with the empty strings. Several modules were
giving semantic meaning to the category names causing problems at
runtime.

* app_directory: Treated the category name as the mailbox name, and
  would fail to direct calls to the appropriate extension after an
  entry was chosen.

* app_queue: Queues, queue members, and queue rules were all affected
  and needed to be updated.

* pbx_realtime: Pattern matching would never succeed because the
  extension entered by the user was always compared to the empty
  string.

Change-Id: Ie7e44986344b0b76ea8f6ddb5879f5040c6ca8a7

2 years agorealtime: Centralize some common realtime backend code
Sean Bright [Tue, 21 Feb 2017 14:56:54 +0000 (09:56 -0500)]
realtime: Centralize some common realtime backend code

All of the realtime backends create artificial ast_categorys to pass
back into the core as query results. These categories have no filename
or line number information associated with them and the backends differ
slightly on how they create them. So create a couple helper macros to
help make things more consistent.

Also updated the call sites to remove redundant error messages about
memory allocation failure.

Note that res_config_ldap sets the category filename to the 'table name'
but that is not read by anything in the core, so I've dropped it.

Change-Id: I3a1fd91e0c807dea1ce3b643b0a6fe5be9002897

2 years agoMerge "realtime: Fix LIKE escaping in SQL backends"
zuul [Tue, 21 Feb 2017 12:26:30 +0000 (06:26 -0600)]
Merge "realtime: Fix LIKE escaping in SQL backends"

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.

If DESTDIR is specified, however, the old logic is executed as
the install process may not have permission to alter the ldconfig
cache.

ASTERISK-26705

Change-Id: If4eca46ac510c6fea5568256280ffdb3888d7bb4

2 years agores_pjsip_authenticator_digest.c: Fix sorcery's immutable contract violation.
Richard Mudgett [Mon, 9 Jan 2017 02:32:17 +0000 (20:32 -0600)]
res_pjsip_authenticator_digest.c: Fix sorcery's immutable contract violation.

The inbound authentication object is supposed to be immutable when it is
stored in sorcery.  However, the immutable property is violated if the
authentication object does not have a realm set.

The immutable contract violation has a different effect depending upon
what sorcery back end is used.  If it is the config file back end you
would get the same object back until res_pjsip is reloaded.  If it is the
real-time or AstDB back end you would get a new object on each query.  If
it is cached you would get the same object back until it is refreshed from
the database.

Once an inbound authentication object has its realm set it may or may not
get updated again if the default_realm changes.

If the same authentication object is used for inbound and outbound
authentication then the immutable violation can make it very hard to
determine why the outbound authentication now fails.  The only diagnostic
message is a complaint about no realms matching when it had worked
earlier.  It fails because of the difference in behaviour for an empty
realm setting between inbound and outbound authentication objects.

* Fixed the sorcery object immutable violation by creating a new object
and setting the default_realm on it instead.  The new object is a shallow
copy for speed.

* The auth_store thread storage no longer holds an auth ref.  It
interferes with the shallow copy and never needed a ref anyway.

ASTERISK-26799 #close

Change-Id: I2328a52f61b78ed5fbba38180b7f183ee7e08956

2 years agores_pjsip: Update artificial auth whenever default_realm changes.
Richard Mudgett [Sun, 5 Feb 2017 02:17:36 +0000 (20:17 -0600)]
res_pjsip: Update artificial auth whenever default_realm changes.

There was code attempting to update the artificial authentication object
whenever the default_realm changed.  However, once the artificial
authentication object was created it would never get updated.  The
artificial authentication object would require a system restart for a
change to the default_realm to take effect.

ASTERISK-26799

Change-Id: Id59036e9529c2d3ed728af2ed904dc36e7094802