asterisk/asterisk.git
5 months agobundled-jansson: On OpenSuse Leap libjansson.a was placed in lib64
George Joseph [Mon, 4 Feb 2019 13:09:57 +0000 (06:09 -0700)]
bundled-jansson:  On OpenSuse Leap libjansson.a was placed in lib64

On OpenSuse Leap, libjansson.a is installed in
third-party/jansson/dest/lib64 instead of lib (which is where
the top-level makeopts looks).  This causes a link failure.

* Updated jansson/Makefile to add an explicit --libdir to force
  the installation to third-party/jansson/dest/lib.

ASTERISK-28271
Reported by: David Wilcox

Change-Id: Ibf8af75e5da13562105fcc39ed898c6ef0b5a5f3

5 months agoMerge "codecs.conf.sample: update codec opus docs"
George Joseph [Mon, 28 Jan 2019 13:46:51 +0000 (07:46 -0600)]
Merge "codecs.conf.sample: update codec opus docs"

5 months agoMerge "format_g726: add support for seeking"
George Joseph [Mon, 28 Jan 2019 13:46:17 +0000 (07:46 -0600)]
Merge "format_g726: add support for seeking"

5 months agoMerge "res_http_websocket: ensure control frames do not interfere with data"
George Joseph [Mon, 28 Jan 2019 13:22:01 +0000 (07:22 -0600)]
Merge "res_http_websocket: ensure control frames do not interfere with data"

5 months agocodecs.conf.sample: update codec opus docs
Kevin Harwell [Fri, 25 Jan 2019 18:27:41 +0000 (12:27 -0600)]
codecs.conf.sample: update codec opus docs

The option value "sdp" for some of the settings was removed a while back,
however the sample conf was not updated.

This patch removes any wording with regards to the old "sdp" option value,
and adjusts the defaults to what they are now.

ASTERISK-28263

Change-Id: I41bfa44e9f69446bcc5c8fd92e3675c676fdc445

5 months agoformat_g726: add support for seeking
eyalhasson [Tue, 22 Jan 2019 15:24:23 +0000 (17:24 +0200)]
format_g726: add support for seeking

Added support for the seek function in format_g726
so playback can start from anywhere.
Before the fix, playback of g726 files
always started from the beginning.

ASTERISK-28246

Change-Id: I626235bc4642df1479050d3d06828412603a9b40

5 months agoMerge "build : Fix cross-compilation errors"
Joshua C. Colp [Thu, 24 Jan 2019 14:23:53 +0000 (08:23 -0600)]
Merge "build : Fix cross-compilation errors"

5 months agoMerge "app_voicemail: Add Mailbox Aliases"
Joshua C. Colp [Thu, 24 Jan 2019 11:56:34 +0000 (05:56 -0600)]
Merge "app_voicemail:  Add Mailbox Aliases"

5 months agoMerge "res_pjsip_registrar: mitigate blocked threads on reliable transport shutdown"
Joshua C. Colp [Thu, 24 Jan 2019 11:52:57 +0000 (05:52 -0600)]
Merge "res_pjsip_registrar: mitigate blocked threads on reliable transport shutdown"

5 months agoMerge "Test_cel: Fails when DONT_OPTIMIZE is off"
Joshua C. Colp [Wed, 23 Jan 2019 17:26:34 +0000 (11:26 -0600)]
Merge "Test_cel: Fails when DONT_OPTIMIZE is off"

5 months agoMerge "manager_channels: Fix throwing of HangupHandler manager events"
Friendly Automation [Wed, 23 Jan 2019 15:46:00 +0000 (09:46 -0600)]
Merge "manager_channels: Fix throwing of HangupHandler manager events"

5 months agores_http_websocket: ensure control frames do not interfere with data
Jeremy Lainé [Wed, 23 Jan 2019 10:45:56 +0000 (11:45 +0100)]
res_http_websocket: ensure control frames do not interfere with data

Control frames (PING / PONG / CLOSE) can be received in the middle of a
fragmented message. In order to ensure they do not interfere with the
reassembly buffer, we exit early and do not return the payload to the
caller.

ASTERISK-28257 #close

Change-Id: Ia5367144fe08ac6141bba3309517a48ec7f013bc

5 months agobuild : Fix cross-compilation errors
Jean Aunis [Wed, 23 Jan 2019 13:59:00 +0000 (14:59 +0100)]
build : Fix cross-compilation errors

Bundled pjproject and jansson must be configured with the host and build
parameters provided to the configure script.
Autotools do not permit to check for the existence of local header files, so
the control of hrirs.h must not be done when cross-compiling.

ASTERISK-28250

Change-Id: If0a76e52a87d4ab82b7d4c72d27d8759ca931880

5 months agoMerge "stasis / manager / ari: Better filter messages."
Joshua C. Colp [Wed, 23 Jan 2019 00:58:48 +0000 (18:58 -0600)]
Merge "stasis / manager / ari: Better filter messages."

5 months agoMerge "bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix"
Joshua C. Colp [Wed, 23 Jan 2019 00:55:42 +0000 (18:55 -0600)]
Merge "bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix"

5 months agoMerge "pjsip_transport_management: Shutdown transport immediately on disconnect"
Joshua C. Colp [Wed, 23 Jan 2019 00:55:21 +0000 (18:55 -0600)]
Merge "pjsip_transport_management: Shutdown transport immediately on disconnect"

5 months agoMerge "res_http_websocket: respond to CLOSE opcode"
Joshua C. Colp [Wed, 23 Jan 2019 00:15:16 +0000 (18:15 -0600)]
Merge "res_http_websocket: respond to CLOSE opcode"

5 months agomanager_channels: Fix throwing of HangupHandler manager events
Gerald Schnabel [Tue, 22 Jan 2019 21:03:22 +0000 (22:03 +0100)]
manager_channels: Fix throwing of HangupHandler manager events

The type value extracted from stasis message data in channel_hangup_handler_cb
isn't compared against the valid values "run", "pop" and "push". Thus the
manager events HangupHandlerPush, HangupHandlerPop and HangupHandlerRun are
never thrown.

This regression was introduced by ASTERISK_21462.

ASTERISK-28252

Change-Id: I9956e35e18da1873113644df1ddc3c7cd37bf524

5 months agoTest_cel: Fails when DONT_OPTIMIZE is off
Chris-Savinovich [Sat, 19 Jan 2019 21:55:20 +0000 (15:55 -0600)]
Test_cel: Fails when DONT_OPTIMIZE is off

A bug in GCC causes TEST_CEL to return failure under the following
conditions:
1. TEST_FRAMEWORK on
2. DONT_OPTIMIZE off
3. Fedora and Ubuntu
4. GCC 8.2.1
5. Test name: test_cel_dial_pickup
6. There must exist a certain combination of multithreading.
The bug affects arithmetic calculations when the optimization level
is bigger than O1 and the -fpartial-inline flag is on. Provided these
conditions, function ast_str_to_lower() fails to convert to lower case
due to said function being of type force_inline.  The solution is to
remove the "force_inline" type declaration from function ast_str_to_lower()

Change-Id: Ied32e0071f12ed9d5f3b4cdd878b2532a1c769d7

5 months agoapp_voicemail: Add Mailbox Aliases
George Joseph [Mon, 10 Dec 2018 13:20:06 +0000 (06:20 -0700)]
app_voicemail:  Add Mailbox Aliases

You can now define an "aliases" context in voicemail.conf
whose entries point to actual mailboxes.  These can be used anywhere
the mailbox is specified.

Example:
[general]
aliasescontext = myaliases

[default]
1234 = yadayada

[myaliases]
4321@devices = 1234@default

Now you can use 4321@devices to refer to the 1234@default mailbox.

This can be useful to provide channel drivers with constant
mailbox specifications such as <extension>@devices leaving
app_voicemail to control exactly which mailbox the alias points to.
Now, only voicemail has to be reloaded to make changes instead of
individual channel drivers which are usually more expensive to
reload.

Change-Id: I395b9205c91523a334fe971be0d1de4522067b04

5 months agores_pjsip_registrar: mitigate blocked threads on reliable transport shutdown
Kevin Harwell [Tue, 22 Jan 2019 18:07:04 +0000 (12:07 -0600)]
res_pjsip_registrar: mitigate blocked threads on reliable transport shutdown

When a reliable transport is shutdown it's possible for the pjsip registrar
resource shutdown handler to get called multiple times. If this happens and one
of the threads is taking "too long" (slow database call for instance) then the
others get blocked waiting to delete.

Since it only takes one to delete the contact then the other threads should be
able to continue on if one of the threads is currently "deleting". This patch
makes it so now when a thread enters the shutdown handler it checks to see if a
thread is currently already "deleting". If so, then the thread does not attempt
to get the lock, and instead continues on thus avoiding the blockage.

ASTERISK-28213 #close

Change-Id: I7563ca596312b1dff4f3ab41483e89fe2862328a

5 months agopjproject_bundled: Add patch for double free issue in timer heap
George Joseph [Tue, 22 Jan 2019 15:02:37 +0000 (08:02 -0700)]
pjproject_bundled:  Add patch for double free issue in timer heap

Fixed #2172: Avoid double reference counter decrements in
timer in the scenario of race condition between
pj_timer_heap_cancel() and pj_timer_heap_poll().

Change-Id: If000e9438c83ac5084b678eb811e902c035bd2d8

5 months agobridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix
Xiemin Chen [Sun, 16 Dec 2018 12:43:42 +0000 (20:43 +0800)]
bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix

To avoid the stream name collide if there're more than one video track
in one client. If client has multi video tracks, the name of ast_stream
which represents each video track may be the same. Use the MSID:LABEL
here because it's identifiable.

ASTERISK-28196 #close
Reported-by: xiemchen

Change-Id: Ib62b2886e8d3a30e481d94616b0ceaeab68a870b

5 months agoMerge "channel.c: Fix segfault with Monitor(wav,file,i)"
Joshua C. Colp [Mon, 21 Jan 2019 19:18:20 +0000 (13:18 -0600)]
Merge "channel.c: Fix segfault with Monitor(wav,file,i)"

5 months agores_http_websocket: respond to CLOSE opcode
Jeremy Lainé [Tue, 8 Jan 2019 07:38:41 +0000 (08:38 +0100)]
res_http_websocket: respond to CLOSE opcode

This ensures that Asterisk responds properly to frames received from a
client with opcode 8 (CLOSE) by echoing back the status code in its own
CLOSE frame.

Handling of the CLOSE opcode is moved up with the rest of the opcodes so
that unmasking gets applied. The payload is no longer returned to the
caller, but neither ARI nor the chan_sip nor pjsip made use of the
payload, which is a good thing since it was masked.

ASTERISK-28231 #close

Change-Id: Icb1b60205fc77ee970ddc91d1f545671781344cf

5 months agopjsip_transport_management: Shutdown transport immediately on disconnect
Sean Bright [Fri, 18 Jan 2019 22:11:18 +0000 (17:11 -0500)]
pjsip_transport_management: Shutdown transport immediately on disconnect

The transport management code that checks for idle connections keeps a
reference to PJSIP's transport for IDLE_TIMEOUT milliseconds (32000 by
default). Because of this, if the transport is closed before this
timeout, the idle checking code will keep the transport from actually
being shutdown until the timeout expires.

Rather than passing the AO2 object to the scheduler task, we just pass
its key and look it up when it is time to potentially close the idle
connection. The other transport management code handles cleaning up
everything else for us.

Additionally, because we use the address of the transport when
generating its name, we concatenate an incrementing ID to the end of the
name to guarantee uniqueness.

Related to ASTERISK~28231

Change-Id: I02ee9f4073b6abca9169d30c47aa69b5e8ae9afb

5 months agochannel.c: Fix segfault with Monitor(wav,file,i)
Valentin Vidic [Sun, 20 Jan 2019 18:15:51 +0000 (19:15 +0100)]
channel.c: Fix segfault with Monitor(wav,file,i)

If the Monitor is started with the i option the read_stream will be
NULL. One code path in channel.c checks if write_stream is set but than
uses read_stream instead causing a segfault.

ASTERISK-28249

Change-Id: I1bae9126537be54895c7fea2d08dd9488d8cc525

6 months agostasis / manager / ari: Better filter messages.
Joshua C. Colp [Thu, 10 Jan 2019 19:34:32 +0000 (15:34 -0400)]
stasis / manager / ari: Better filter messages.

Previously both AMI and ARI used a default route on
their stasis message router to handle some of the
messages for publishing out their respective
connection. This caused messages to be given to
their subscription that could not be formatted
into AMI or JSON.

This change adds an API call to the stasis message
router which allows a default route to be set as well
as formatters that the default route is expecting.
This allows both AMI and ARI to specify that their
default route only wants messages of their given
formatter. By doing so stasis can more intelligently
filter at publishing time so that they do not receive
messages which will not be turned into AMI or JSON.

ASTERISK-28244

Change-Id: I65272819a53ce99f869181d1d370da559a7d1703

6 months agosched: Make sched_settime() return void because it cannot fail
Sean Bright [Thu, 17 Jan 2019 15:56:35 +0000 (10:56 -0500)]
sched: Make sched_settime() return void because it cannot fail

Change-Id: I66b8b2b2778f186919d73ae9bf592104b8fb1cd5

6 months agores_pjsip_transport_websocket: Don't assert on 0 length payloads
Sean Bright [Fri, 4 Jan 2019 23:14:45 +0000 (18:14 -0500)]
res_pjsip_transport_websocket: Don't assert on 0 length payloads

When --enable-dev-mode is used, pjsip_tpmgr_receive_packet() will assert
if passed a payload length of 0, so treat empty frames as if we didn't
receive them.

Change-Id: I9c5fdccd89cc8d2f3ed7e3ee405ef0fc78178f48

6 months agoMerge "res_pjsip: add option to enable ContactStatus event when contact is updated"
Joshua C. Colp [Mon, 14 Jan 2019 14:38:14 +0000 (08:38 -0600)]
Merge "res_pjsip: add option to enable ContactStatus event when contact is updated"

6 months agoMerge "stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure."
Joshua C. Colp [Mon, 14 Jan 2019 14:26:32 +0000 (08:26 -0600)]
Merge "stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure."

6 months agoMerge "res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled."
Joshua C. Colp [Mon, 14 Jan 2019 14:03:27 +0000 (08:03 -0600)]
Merge "res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled."

6 months agoMerge "RTP: reset DTMF last seqno/timestamp on RTP renegotiation"
Joshua C. Colp [Mon, 14 Jan 2019 14:03:03 +0000 (08:03 -0600)]
Merge "RTP: reset DTMF last seqno/timestamp on RTP renegotiation"

6 months agoMerge "app_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail"
Joshua C. Colp [Mon, 14 Jan 2019 12:19:45 +0000 (06:19 -0600)]
Merge "app_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail"

6 months agostasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure.
mohitdhiman [Sat, 12 Jan 2019 08:29:12 +0000 (13:59 +0530)]
stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure.

During Bridging of two channels if masquerade operation is performed on a
channel (clone channel) which was created with endpoint details
(ast_channel_alloc_with_endpoint()) and the original channel which is created
without endpoint details (ast_channel_alloc()) then both the channels must
exchange their endpoint details or else after masquerade when clone channel
is being destroyed the endpoint cleanup callbacks will be destroyed too and
after call completion unique_id of original channel will still be there in
ast_endpoint structure's channel_ids container.

ASTERISK-28197

Change-Id: I97ce73da390af20fd082fb09d722a6fe9cb2f39d

6 months agores_pjsip: add option to enable ContactStatus event when contact is updated
Alexei Gradinari [Fri, 11 Jan 2019 15:48:36 +0000 (10:48 -0500)]
res_pjsip: add option to enable ContactStatus event when contact is updated

The commit I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d removed sending out
the ContactStatus AMI event when a contact is updated.
Thist change broke things which rely on old behavior.

This patch adds a new PJSIP global configuration option
'send_contact_status_on_update_registration' to be able to preserve old
ContactStatus behavior.
By default new behavior, i.e. the ContactStatus event will not be sent when a
device refreshes its registration.

Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46

6 months agores_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled.
Joshua Colp [Mon, 7 Jan 2019 14:06:37 +0000 (14:06 +0000)]
res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled.

For video streams it was possible for the abs-send-time information
to be placed into RTP streams even if not negotiated. Depending on
the endpoint in use this could cause video to not flow.

We now only enable abs-send-time for negotiation if WebRTC is enabled.

ASTERISK-28230

Change-Id: I0eb682302f8da3a4ea3c42e839208d55f825ed0c

6 months agoRAII: Change order or variables in clang version
Diederik de Groot [Sat, 5 Jan 2019 17:14:26 +0000 (18:14 +0100)]
RAII: Change order or variables in clang version

This prevents use-after-scope issues when unwinding the stack,
which happens in reverse order. The varname variable needs to
remain alive for the destruction to be able to access it.
Issue was found using clang + address-sanitizer.

ASTERISK-28232 #close

Change-Id: I00811c34ae910836a5fb6d22304528aef92624db

6 months agoRTP: reset DTMF last seqno/timestamp on RTP renegotiation
Alexei Gradinari [Fri, 4 Jan 2019 15:57:06 +0000 (10:57 -0500)]
RTP: reset DTMF last seqno/timestamp on RTP renegotiation

The remote side may start a new stream when renegotiating RTP.
Need to reset the DTMF last sequence number and the timestamp
of the last END packet on RTP renegotiation.

If the new time stamp is lower then the timestamp of the last DTMF END packet
the asterisk drops all DTMF frames as out of order.

This bug was caught using Cisco ip-phone SPA5XX and codec g722.
On SIP session update the SPA50X resets stream and a new timestamp is twice
smaller then the previous.

ASTERISK-28162 #close

Change-Id: Ic72b4497e74d801b27a635559c1cf29c16c95254

6 months agoMerge "ast_coredumper: Refactor the pid determination process"
Joshua C. Colp [Fri, 4 Jan 2019 14:27:04 +0000 (08:27 -0600)]
Merge "ast_coredumper:  Refactor the pid determination process"

6 months agoMerge "stasis: Fix ABI between DEVMODE and non-DEVMODE."
Friendly Automation [Thu, 3 Jan 2019 23:39:22 +0000 (17:39 -0600)]
Merge "stasis: Fix ABI between DEVMODE and non-DEVMODE."

6 months agoMerge "stasic.c: Fix printf format type mismatches with arguments."
Friendly Automation [Thu, 3 Jan 2019 11:38:16 +0000 (05:38 -0600)]
Merge "stasic.c: Fix printf format type mismatches with arguments."

6 months agoapp_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail
Bryan Boatright [Wed, 2 Jan 2019 17:44:41 +0000 (11:44 -0600)]
app_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail

If a voicemail is marked "urgent" then the VM_MESSAGEFILE channel variable is
not updated correctly since urgent messages are in a different directory. The
fix is to update the channel variable when the path to the urgent message is
created.

ASTERISK-28225

Change-Id: I8efbace06e6122ea0793f7bdb073d4378e8274ca

6 months agoapp_queue: Fix crash when using 'b' option on non-ringall queue.
Joshua Colp [Wed, 2 Jan 2019 17:33:58 +0000 (17:33 +0000)]
app_queue: Fix crash when using 'b' option on non-ringall queue.

When using the 'b' option to Queue with a queue that was not configured
for ring all a crash would occur as the wrong pointer would be used.

ASTERISK-28218

Change-Id: If1390f64e321047dff24fd2410c95dde74904980

6 months agostasic.c: Fix printf format type mismatches with arguments.
Richard Mudgett [Wed, 19 Dec 2018 19:02:35 +0000 (13:02 -0600)]
stasic.c: Fix printf format type mismatches with arguments.

An int64_t is not likely the same size as a long.

* Changed the int64_t values in the statistics structs to longs so casting
is not necessary when generating the formatted CLI output.  The offending
members did not need to be int64_t anyway as they were only set by an int
type variable which was already truncating bits.

* Reordered the statistics structs to reduce potential padding bytes.

Change-Id: Ic090a070e9dc4ca650ebdb9c01ed50a581289962

6 months agoMerge "backtrace.c: Fix casting pointer to/from integral type."
George Joseph [Wed, 2 Jan 2019 15:51:36 +0000 (09:51 -0600)]
Merge "backtrace.c: Fix casting pointer to/from integral type."

6 months agostasis: Fix ABI between DEVMODE and non-DEVMODE.
Corey Farrell [Wed, 26 Dec 2018 17:49:57 +0000 (12:49 -0500)]
stasis: Fix ABI between DEVMODE and non-DEVMODE.

Eliminate differences with DEVMODE prototypes for public functions.

ASTERISK-28212 #close

Change-Id: I872c04842ab6b61e9dd6d37e4166bc619aa20626

6 months agoRevert "stasis_cache: Stop caching stasis subscription change messages"
George Joseph [Wed, 26 Dec 2018 16:26:36 +0000 (11:26 -0500)]
Revert "stasis_cache:  Stop caching stasis subscription change messages"

This reverts commit 5ec6d2c33e3b02755e0b2ea3fc94f048af5c741f.

This commit caused issues with polling when combined with
the revert commit "Revert "app_voicemail: Remove need to subscribe to stasis"

ASTERISK-28222
Reported by: abelbeck

Change-Id: I1e83a433e4202574181bc128dce876ef24936a52

6 months agoast_coredumper: Refactor the pid determination process
George Joseph [Mon, 24 Dec 2018 17:42:36 +0000 (10:42 -0700)]
ast_coredumper:  Refactor the pid determination process

In order to get a dump of the running process, we need to find the
pid of the main asterisk process.  This can be tricky if there are
also instances of "asterisk -r" running or if an alternate location
for asterisk.conf was specified on the command line with the -C
option that also specified an alternation location for the pid file.

So now...

1. We find the asterisk executable with "which" or the --asterisk-bin
   command line option.
2. If there's only 1 process with an executable path that matches,
   we use that pid.  If not...
3. We try "<asterisk-bin> -rx 'core show settings'" and parse the
   output to find the pidfile, then read that for the pid.  If that
   didn't work...
4. We get a list of all the pids matching <asterisk-bin> and look
   in /proc/<pid>/cmdline for a -C argument and retry the "core show
   settings" using the same -C option.  We can't parse the output
   of "ps" to get the -C path because it may contain spaces.  The
   contents of /proc/<pid>/cmdline are delimited by NULLs.  For BSDs
   we may have to mount /proc first. :(

ASTERISK-28221
Reported by: Andrew Nagy

Change-Id: I8aa1f3f912f949df2b5348908803c636bde1d57c

7 months agobacktrace.c: Fix casting pointer to/from integral type.
Richard Mudgett [Wed, 19 Dec 2018 18:39:08 +0000 (12:39 -0600)]
backtrace.c: Fix casting pointer to/from integral type.

The backtrace library bfd.h include file does not get the sizes of
pointers and ints right on some platforms.  On my old test box the size
of bfd_vma is 8 while the size of a pointer is 4.  gcc on the box
complains of the integer casting to/from pointers size mismatch.

* uintptr_t to the rescue by doing an appropriate two stage cast.

Change-Id: Icb2621583f50c8728de08a3c824d95fe53cc45d0

7 months agoMerge "res/res_ari: Add additional hangup reasons"
Friendly Automation [Wed, 19 Dec 2018 11:12:15 +0000 (05:12 -0600)]
Merge "res/res_ari: Add additional hangup reasons"

7 months agoMerge "app_voicemail: Don't delete mailbox state unless mailbox is deleted"
Friendly Automation [Wed, 19 Dec 2018 11:08:00 +0000 (05:08 -0600)]
Merge "app_voicemail:  Don't delete mailbox state unless mailbox is deleted"

7 months agoMerge "res_pjsip: Patch for res_pjsip_* module load/reload crash"
George Joseph [Tue, 18 Dec 2018 16:42:49 +0000 (10:42 -0600)]
Merge "res_pjsip: Patch for res_pjsip_* module load/reload crash"

7 months agoMerge "res_rtp_asterisk: Remove some unused structure fields."
George Joseph [Tue, 18 Dec 2018 16:42:26 +0000 (10:42 -0600)]
Merge "res_rtp_asterisk: Remove some unused structure fields."

7 months agoapp_voicemail: Don't delete mailbox state unless mailbox is deleted
George Joseph [Tue, 18 Dec 2018 16:33:50 +0000 (09:33 -0700)]
app_voicemail:  Don't delete mailbox state unless mailbox is deleted

The free_user function was automatically deleting the stasis mailbox
state but this only makes sense when the mailbox is actually
deleted, not just the structure freed.  This was causing issues
where leave_voicemail would publish the mwi message to stasis and
delete the state before the message could be processed by
res_pjsip_mwi.

* Removed the delete of state from free_user().

* Created a new free_user_final() function that both frees the data
  structure and deletes the state.  This function is only called
  during module load/unload where it's appropriate to delete the
  state.

ASTERISK-28215

Change-Id: I305e8b3c930e9ac41d901e5dc8a58fd7904d98dd

7 months agoMerge "res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set"
Joshua C. Colp [Mon, 17 Dec 2018 15:34:47 +0000 (09:34 -0600)]
Merge "res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set"

7 months agores_rtp_asterisk: Remove some unused structure fields.
Sean Bright [Fri, 14 Dec 2018 17:52:45 +0000 (12:52 -0500)]
res_rtp_asterisk: Remove some unused structure fields.

All of the fields that were removed were no longer referenced except for
'lastrxts' and 'rxseqno' which were only ever written to.

Change-Id: I5a5d31eb33e97663843698f58d0d97f22a76627c

7 months agoMerge "bridge_builtin_features.c: Set auto(mix)mon variables on both channels"
Joshua C. Colp [Fri, 14 Dec 2018 14:37:38 +0000 (08:37 -0600)]
Merge "bridge_builtin_features.c: Set auto(mix)mon variables on both channels"

7 months agores_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set
Sean Bright [Thu, 13 Dec 2018 21:56:50 +0000 (16:56 -0500)]
res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set

The profile-iop octet (the 2nd) of profile-level-id can be zero
according to RFC 6184 Section 8.1. So we ignore its value when deciding
to include profile-level-id in the outgoing SDP.

ASTERISK-27959 #close
Reported by: David Kuehling

Change-Id: Id28cd916a3d7748058fe9609b585d07d9e243f73

7 months agoMerge "confbridge: announce to the marked users when they join an empty conference"
Joshua C. Colp [Thu, 13 Dec 2018 14:00:06 +0000 (08:00 -0600)]
Merge "confbridge: announce to the marked users when they join an empty conference"

7 months agobridge_builtin_features.c: Set auto(mix)mon variables on both channels
Sean Bright [Tue, 11 Dec 2018 20:49:03 +0000 (15:49 -0500)]
bridge_builtin_features.c: Set auto(mix)mon variables on both channels

This is how features behaved up through Asterisk 11, but was changed
when the new bridging framework was implemented in Asterisk 12.

Reported by rrittgarn in #asterisk.

Change-Id: I72cf86223947a8118c75f46e2c603dbc11e3125b

7 months agoMerge "utils: Don't set or clear flags that don't need setting or clearing"
Joshua C. Colp [Wed, 12 Dec 2018 19:12:19 +0000 (13:12 -0600)]
Merge "utils: Don't set or clear flags that don't need setting or clearing"

7 months agoMerge "stasis: Add statistics gathering in developer mode."
Friendly Automation [Wed, 12 Dec 2018 19:08:23 +0000 (13:08 -0600)]
Merge "stasis: Add statistics gathering in developer mode."

7 months agoMerge "Use non-blocking socket() and pipe() wrappers"
Joshua C. Colp [Wed, 12 Dec 2018 17:31:00 +0000 (11:31 -0600)]
Merge "Use non-blocking socket() and pipe() wrappers"

7 months agoconfbridge: announce to the marked users when they join an empty conference
Alexei Gradinari [Fri, 7 Dec 2018 20:22:29 +0000 (15:22 -0500)]
confbridge: announce to the marked users when they join an empty conference

Currently the file sound_only_person is not played when a marked
user (with announce_only_user=yes) joins an empty conference.

This patch fixes it.

ASTERISK-28201 #close

Change-Id: I85b67687e6b220939c3af8091d83a70a7b174cf4

7 months agostasis: Add statistics gathering in developer mode.
Joshua C. Colp [Fri, 30 Nov 2018 11:40:40 +0000 (07:40 -0400)]
stasis: Add statistics gathering in developer mode.

This change adds statistics gathering to Stasis topics,
subscriptions, and message types. These can be viewed using
CLI commands and provide insight into how Stasis is used
and how long certain operations take to execute.

These are only available when Asterisk is compiled in
developer mode and do not have any impact under normal
operation.

ASTERISK-28117

Change-Id: I94411b53767f89ee01714daaecf0c2f1666e863f

7 months agoMerge "stasis: Allow filtering by formatter"
Friendly Automation [Wed, 12 Dec 2018 17:09:19 +0000 (11:09 -0600)]
Merge "stasis:  Allow filtering by formatter"

7 months agoMerge "build: Update config.guess and config.sub"
Joshua C. Colp [Wed, 12 Dec 2018 17:05:30 +0000 (11:05 -0600)]
Merge "build: Update config.guess and config.sub"

7 months agoMerge "pjproject_bundled: check whether UPDATE is supported on outgoing calls"
George Joseph [Wed, 12 Dec 2018 16:51:57 +0000 (10:51 -0600)]
Merge "pjproject_bundled: check whether UPDATE is supported on outgoing calls"

7 months agoMerge "Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit""
George Joseph [Tue, 11 Dec 2018 20:18:25 +0000 (14:18 -0600)]
Merge "Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit""

7 months agoUse non-blocking socket() and pipe() wrappers
Sean Bright [Tue, 11 Dec 2018 14:54:43 +0000 (09:54 -0500)]
Use non-blocking socket() and pipe() wrappers

Change-Id: I050ceffe5a133d5add2dab46687209813d58f597

7 months agoutils: Don't set or clear flags that don't need setting or clearing
Sean Bright [Tue, 11 Dec 2018 15:06:15 +0000 (10:06 -0500)]
utils: Don't set or clear flags that don't need setting or clearing

Change-Id: I0e7fb507ac09b15e45e1ff8501ecfca67afa5217

7 months agoMerge "CI: Various updates to buildAsterisk.sh"
George Joseph [Tue, 11 Dec 2018 15:07:59 +0000 (09:07 -0600)]
Merge "CI: Various updates to buildAsterisk.sh"

7 months agoMerge "utils: Wrap socket() and pipe() to reduce syscalls"
Joshua C. Colp [Tue, 11 Dec 2018 15:01:38 +0000 (09:01 -0600)]
Merge "utils: Wrap socket() and pipe() to reduce syscalls"

7 months agobuild: Update config.guess and config.sub
Sean Bright [Tue, 11 Dec 2018 12:55:16 +0000 (07:55 -0500)]
build: Update config.guess and config.sub

Pulled from the authoritative respository at:

  https://git.savannah.gnu.org/cgit/config.git/tree/

Change-Id: I748708ce24d4d47ff1f395075d0b08d3da3355e0

7 months agoRevert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit"
George Joseph [Tue, 11 Dec 2018 14:28:48 +0000 (09:28 -0500)]
Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit"

This reverts commit 3f53041267234b21aedd522c1197ec57cca90845.

Pending resolution of ASTERISK_28200

Change-Id: Iad4f3614cac95b00fdbb2b799aab8ae6285ec988

7 months agores/res_ari: Add additional hangup reasons
Sebastian Damm [Thu, 6 Dec 2018 17:23:50 +0000 (18:23 +0100)]
res/res_ari: Add additional hangup reasons

The ARI DELETE /channels command takes a "reason" parameter
Previously, there were only five reasons implemented
This patch adds more reasons to choose from for more
complex setups

ASTERISK-28198 #close

Change-Id: I85996f1076c9946d65c778413f040a845a90fecc

7 months agoMerge "chan_sip: Fix leak using contact ACL"
Joshua C. Colp [Mon, 10 Dec 2018 13:05:21 +0000 (07:05 -0600)]
Merge "chan_sip: Fix leak using contact ACL"

7 months agoutils: Wrap socket() and pipe() to reduce syscalls
Sean Bright [Fri, 7 Dec 2018 12:57:48 +0000 (07:57 -0500)]
utils: Wrap socket() and pipe() to reduce syscalls

Some platforms provide an implementation of socket() and pipe2() that allow the
caller to specify that the resulting file descriptors should be non-blocking.

Using these allows us to potentially elide 3 calls into 1 by avoiding extraneous
calls to fcntl() to set the O_NONBLOCK flag afterwards.

In passing, change ast_alertpipe_init() to use pipe2() directly instead of the
wrapper if it is available.

Change-Id: I3ebe654fb549587537161506c6c950f4ab298bb0

7 months agostasis: Allow filtering by formatter
George Joseph [Thu, 29 Nov 2018 15:53:51 +0000 (08:53 -0700)]
stasis:  Allow filtering by formatter

A subscriber can now indicate that it only wants messages
that have formatters of a specific type.  For instance,
manager can indicate that it only wants messages that have a
"to_ami" formatter.  You can combine this with the existing
filter for message type to get only messages with specific
formatters or messages of specific types.

ASTERISK-28186

Change-Id: Ifdb7a222a73b6b56c6bb9e4ee93dc8a394a5494c

7 months agoRemoving registrar_expire from basic-pbx config
David M. Lee [Wed, 5 Dec 2018 21:28:03 +0000 (15:28 -0600)]
Removing registrar_expire from basic-pbx config

The module has been removed, so it shouldn't be in the default config any more.

Change-Id: Ie7e09f00f9c9a885574e29478250de4c2cefd9f1

7 months agochan_sip: Fix leak using contact ACL
Giuseppe Sucameli [Wed, 5 Dec 2018 00:00:40 +0000 (01:00 +0100)]
chan_sip: Fix leak using contact ACL

Free old peer's contactacl before overwrite it within build_peer.

ASTERISK-28194

Change-Id: Ie580db6494e50cee0e2a44b38e568e34116ff54c

7 months agoCI: Various updates to buildAsterisk.sh
George Joseph [Wed, 5 Dec 2018 15:37:45 +0000 (08:37 -0700)]
CI: Various updates to buildAsterisk.sh

* Added ---no-configure, --no-menuselect, --no-make and --no-alembic
  options that prevent those actions from being performed.  Useful
  for testing and re-running portions of the build after fixing
  earlier failures.

* Added "set -e" to abort the script on command failure.
  Not sure why this wasn't there in the first place.

* Fixed a few echos that were redirecting to stderr when they shouldn't
  have been.

* Catch more alembic failures by actually trying to generate the SQL.

Change-Id: I9f395fa4e9254be7299e7c1014f1a13db78faffb

7 months agoMerge "test_websocket_client.c: Disable websocket_client_create_and_connect test."
George Joseph [Wed, 5 Dec 2018 14:18:36 +0000 (08:18 -0600)]
Merge "test_websocket_client.c: Disable websocket_client_create_and_connect test."

7 months agopjsip_add_use_callerid_contact: fixed alembic script
Kevin Harwell [Mon, 3 Dec 2018 23:45:57 +0000 (17:45 -0600)]
pjsip_add_use_callerid_contact: fixed alembic script

Change-Id: I413f1583c797fb79651786cd8d0b003599f8ed10

7 months agocore: Add some documentation to the malloc_trim code
Sean Bright [Mon, 3 Dec 2018 22:41:56 +0000 (17:41 -0500)]
core: Add some documentation to the malloc_trim code

This adds documentation to handle_cli_malloc_trim() indicating how it
can be useful when debugging OOM conditions.

Change-Id: I1936185e78035bf123cd5e097b793a55eeebdc78

7 months agoMerge "core: Merge malloc_trim patch"
George Joseph [Mon, 3 Dec 2018 22:26:51 +0000 (16:26 -0600)]
Merge "core: Merge malloc_trim patch"

7 months agocore: Merge malloc_trim patch
Chris-Savinovich [Mon, 3 Dec 2018 20:01:01 +0000 (14:01 -0600)]
core: Merge malloc_trim patch

We've had multiple opportunities where Richard Mudgett's
malloc_trim patch has been useful. Let's get it
pushed up to gerrit and merged.

Since malloc_trim is only available in libc, an entry is
added to configure.ac to create a definition for
HAVE_MALLOC_TRIM.

Change-Id: Ia38308c550149d9d6eae4ca414a649957de9700c

7 months agores_pjsip: Patch for res_pjsip_* module load/reload crash
Sungtae Kim [Sun, 11 Nov 2018 16:29:10 +0000 (17:29 +0100)]
res_pjsip: Patch for res_pjsip_* module load/reload crash

The session_supplements for the pjsip makes crashes when the module
load/unload.

ASTERISK-28157

Change-Id: I5b82be3a75d702cf1933d8d1417f44aa10ad1029

7 months agoapp_queue: Revert broken queue channel reference patch
lvl [Mon, 22 Oct 2018 12:47:56 +0000 (14:47 +0200)]
app_queue: Revert broken queue channel reference patch

Revert commit 6409e7b11a2310196a9978b30a6b79e2760be592, and add
NULL checks for all app_queue event handling code.

Related issues: ASTERISK~25185, ASTERISK~27006, ASTERISK~25844

ASTERISK-28125

Change-Id: I37334ea184ebb56e54471496b82937d4927815a0

7 months agotest_websocket_client.c: Disable websocket_client_create_and_connect test.
Chris-Savinovich [Fri, 30 Nov 2018 20:00:14 +0000 (14:00 -0600)]
test_websocket_client.c: Disable websocket_client_create_and_connect test.

This test was occasionally failing, with:

  WARNING[5812]: http.c:1939 httpd_helper_thread: Failed to set
      TCP_NODELAY on HTTP connection: Bad file descriptor
  ERROR[5812]: iostream.c:91 ast_iostream_nonblock: Failed to get
      fcntl() flags for file descriptor: Bad file descriptor
  ERROR[5812]: iostream.c:569 ast_iostream_close: close() failed: Bad
      file descriptor

Disabled for now by making the test explicit only.

Change-Id: I778f6cbb6104c6b4e89737a2eaf1a9540888d351

7 months agopjproject_bundled: check whether UPDATE is supported on outgoing calls
Pirmin Walthert [Wed, 28 Nov 2018 07:14:12 +0000 (08:14 +0100)]
pjproject_bundled: check whether UPDATE is supported on outgoing calls

In ASTERISK-27095 an issue had been fixed because of which chan_pjsip was not
trying to send UPDATE messages when connected_line_method was set to invite.
However this only solved the issue for incoming INVITES. For outgoing INVITES
(important when transferring calls) the options variable needs to be updated
at a different place.

ASTERISK-28182 #close
Reported-by: nappsoft

Change-Id: I76cc06da4ca76ddd6dce814a8b97cc66b98aaf29

7 months agoMerge "Revert "app_voicemail: Remove need to subscribe to stasis""
George Joseph [Fri, 30 Nov 2018 13:30:35 +0000 (07:30 -0600)]
Merge "Revert "app_voicemail: Remove need to subscribe to stasis""

7 months agoMerge "bridges: Remove reliance on stasis caching"
George Joseph [Thu, 29 Nov 2018 21:05:33 +0000 (15:05 -0600)]
Merge "bridges:  Remove reliance on stasis caching"

7 months agoRevert "app_voicemail: Remove need to subscribe to stasis"
George Joseph [Thu, 29 Nov 2018 19:26:16 +0000 (12:26 -0700)]
Revert "app_voicemail: Remove need to subscribe to stasis"

This reverts commit 29115e23848cceee0e2763bc70e87cb311919cdd.

That commit closed a long standing hole which allowed subscriptions
to mailboxes that weren't configured in voicemail.conf.  This
caused an issue with FreePBX which depdended on that behavior.
The commit is being reverted until FreePBX can handle the new
behavior.

ASTERISK-28151
Reported by: Ronald Raikes

Change-Id: I57b7b85e75d7dd97c742b5c69d718a0f61260c15

7 months agoMerge "jansson: Upgrade to 2.12."
Kevin Harwell [Thu, 29 Nov 2018 18:57:32 +0000 (12:57 -0600)]
Merge "jansson: Upgrade to 2.12."

7 months agotest_cel: Plug a few ref leaks
George Joseph [Mon, 26 Nov 2018 22:18:00 +0000 (15:18 -0700)]
test_cel:  Plug a few ref leaks

These are only a few of the leaks.  The large number of macros
and return paths in this file would make a weeks worth of work
to plug them all.

Change-Id: Ie2369fa944023d44767871c5c30974cb077ffb56

7 months agobridges: Remove reliance on stasis caching
George Joseph [Wed, 19 Sep 2018 19:34:41 +0000 (13:34 -0600)]
bridges:  Remove reliance on stasis caching

* The bridging core no longer uses the stasis cache for bridge
  snapshots.  The latest bridge snapshot is now stored on the
  ast_bridge structure itself.

* The following APIs are no longer available since the stasis cache
  is no longer used:
    ast_bridge_topic_cached()
    ast_bridge_topic_all_cached()

* A topic pool is now used for individual bridge topics.

* The ast_bridge_cache() function was removed since there's no
  longer a separate container of snapshots.

* A new function "ast_bridges()" was created to retrieve the
  container of all bridges.  Users formerly calling
  ast_bridge_cache() can use the new function to iterate over
  bridges and retrieve the latest snapshot directly from the
  bridge.

* The ast_bridge_snapshot_get_latest() function was renamed to
  ast_bridge_get_snapshot_by_uniqueid().

* A new function "ast_bridge_get_snapshot()" was created to retrieve
  the bridge snapshot directly from the bridge structure.

* The ast_bridge_topic_all() function now returns a normal topic
  not a cached one so you can't use stasis cache functions on it
  either.

* The ast_bridge_snapshot_type() stasis message now has the
  ast_bridge_snapshot_update structure as it's data.  It contains
  the last snapshot and the new one.

* cdr, cel, manager and ari have been updated to use the new
  arrangement.

Change-Id: I7049b80efa88676ce5c4666f818fa18ad1985369

7 months agoMerge "stasis: Segment channel snapshot to reduce creation cost."
Jenkins2 [Mon, 26 Nov 2018 20:07:47 +0000 (14:07 -0600)]
Merge "stasis: Segment channel snapshot to reduce creation cost."