Commit Graph

31881 Commits

Author SHA1 Message Date
George Joseph
3464093f85 Merge "res_pjsip: Patch for res_pjsip_* module load/reload crash" 2018-12-18 10:42:49 -06:00
George Joseph
2e21910ca1 Merge "res_rtp_asterisk: Remove some unused structure fields." 2018-12-18 10:42:26 -06:00
George Joseph
c23c8d92d5 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
2018-12-18 11:40:22 -05:00
Joshua C. Colp
768be60fbe Merge "res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set" 2018-12-17 09:34:47 -06:00
Sean Bright
357219dfb3 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
2018-12-14 12:57:06 -05:00
Joshua C. Colp
0eaa736541 Merge "bridge_builtin_features.c: Set auto(mix)mon variables on both channels" 2018-12-14 08:37:38 -06:00
Sean Bright
5b12dfa6dd 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
2018-12-13 17:03:59 -05:00
Joshua C. Colp
b701c8b8a0 Merge "confbridge: announce to the marked users when they join an empty conference" 2018-12-13 08:00:06 -06:00
Sean Bright
3db1df301e 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
2018-12-13 08:54:10 -05:00
Joshua C. Colp
e12ba0a600 Merge "utils: Don't set or clear flags that don't need setting or clearing" 2018-12-12 13:12:19 -06:00
Friendly Automation
4e8aa3b68c Merge "stasis: Add statistics gathering in developer mode." 2018-12-12 13:08:23 -06:00
Joshua C. Colp
a28f0382e8 Merge "Use non-blocking socket() and pipe() wrappers" 2018-12-12 11:31:00 -06:00
Alexei Gradinari
cb1a08bdcb 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
2018-12-12 12:15:49 -05:00
Joshua C. Colp
fe07093660 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
2018-12-12 12:14:53 -05:00
Friendly Automation
1f8062c6a6 Merge "stasis: Allow filtering by formatter" 2018-12-12 11:09:19 -06:00
Joshua C. Colp
3ee040e9cd Merge "build: Update config.guess and config.sub" 2018-12-12 11:05:30 -06:00
George Joseph
5a0a292843 Merge "pjproject_bundled: check whether UPDATE is supported on outgoing calls" 2018-12-12 10:51:57 -06:00
George Joseph
5d4d723844 Merge "Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit"" 2018-12-11 14:18:25 -06:00
Sean Bright
42ff856216 Use non-blocking socket() and pipe() wrappers
Change-Id: I050ceffe5a133d5add2dab46687209813d58f597
2018-12-11 12:29:09 -05:00
Sean Bright
bedf16b041 utils: Don't set or clear flags that don't need setting or clearing
Change-Id: I0e7fb507ac09b15e45e1ff8501ecfca67afa5217
2018-12-11 10:08:07 -05:00
George Joseph
82473227be Merge "CI: Various updates to buildAsterisk.sh" 2018-12-11 09:07:59 -06:00
Joshua C. Colp
6b017a51e0 Merge "utils: Wrap socket() and pipe() to reduce syscalls" 2018-12-11 09:01:38 -06:00
Sean Bright
00b36bb045 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
2018-12-11 09:34:31 -05:00
George Joseph
d1598dbc7d Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit"
This reverts commit 3f53041267.

Pending resolution of ASTERISK_28200

Change-Id: Iad4f3614cac95b00fdbb2b799aab8ae6285ec988
2018-12-11 09:28:48 -05:00
Sebastian Damm
a24bb1c4b6 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
2018-12-11 11:20:44 +01:00
Joshua C. Colp
fe49d04519 Merge "chan_sip: Fix leak using contact ACL" 2018-12-10 07:05:21 -06:00
Sean Bright
6d69fb3cc2 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
2018-12-07 09:06:08 -05:00
George Joseph
3f3dd992a2 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
2018-12-07 08:59:00 -05:00
David M. Lee
b899119a5d 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
2018-12-06 06:26:04 -05:00
Giuseppe Sucameli
0bde3751a0 chan_sip: Fix leak using contact ACL
Free old peer's contactacl before overwrite it within build_peer.

ASTERISK-28194

Change-Id: Ie580db6494e50cee0e2a44b38e568e34116ff54c
2018-12-05 17:17:57 -05:00
George Joseph
19c4e0f592 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
2018-12-05 12:03:08 -05:00
George Joseph
64b0bcacb5 Merge "test_websocket_client.c: Disable websocket_client_create_and_connect test." 2018-12-05 08:18:36 -06:00
Kevin Harwell
cbb7633ad3 pjsip_add_use_callerid_contact: fixed alembic script
Change-Id: I413f1583c797fb79651786cd8d0b003599f8ed10
2018-12-03 18:47:16 -05:00
Sean Bright
8f5df046f6 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
2018-12-03 17:47:26 -05:00
George Joseph
ff02c93d9e Merge "core: Merge malloc_trim patch" 2018-12-03 16:26:51 -06:00
Chris-Savinovich
58e50e56cb 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
2018-12-03 14:01:01 -06:00
Sungtae Kim
8644511cbf 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
2018-12-03 08:44:59 -06:00
lvl
140702ba2d app_queue: Revert broken queue channel reference patch
Revert commit 6409e7b11a, and add
NULL checks for all app_queue event handling code.

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

ASTERISK-28125

Change-Id: I37334ea184ebb56e54471496b82937d4927815a0
2018-12-03 11:12:20 +01:00
Chris-Savinovich
6c13b20803 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
2018-12-02 08:55:23 -05:00
Pirmin Walthert
ecb9ed0958 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
2018-11-30 09:34:09 -05:00
George Joseph
efeab21b52 Merge "Revert "app_voicemail: Remove need to subscribe to stasis"" 2018-11-30 07:30:35 -06:00
George Joseph
945451af90 Merge "bridges: Remove reliance on stasis caching" 2018-11-29 15:05:33 -06:00
George Joseph
4f0bf0270e Revert "app_voicemail: Remove need to subscribe to stasis"
This reverts commit 29115e2384.

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
2018-11-29 12:29:34 -07:00
Kevin Harwell
7d37967e7e Merge "jansson: Upgrade to 2.12." 2018-11-29 12:57:32 -06:00
George Joseph
f4924d40db 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
2018-11-26 15:18:00 -07:00
George Joseph
3667c5e1d2 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
2018-11-26 14:30:02 -07:00
Jenkins2
4ca709768d Merge "stasis: Segment channel snapshot to reduce creation cost." 2018-11-26 14:07:47 -06:00
Joshua Colp
a6b37e5c43 Merge "astobj2: Create function to copy weak proxied objects from container." 2018-11-26 13:48:00 -06:00
Joshua Colp
b80c9071e3 Merge "RTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit" 2018-11-26 13:47:32 -06:00
Joshua Colp
50ac85cb40 stasis: Segment channel snapshot to reduce creation cost.
When a channel snapshot was created it used to be done
from scratch, copying all data (many strings). This incurs
a cost when doing so.

This change segments the channel snapshot into different
components which can be reused if unchanged from the
previous snapshot creation, reducing the cost. In normal
cases this results in some pointers being copied with
reference count being bumped, some integers being set,
and a string or two copied. The other benefit is that it
is now possible to determine if a channel snapshot update
is redundant and thus stop it before a message is published
to stasis.

The specific segments in the channel snapshot were split up
based on whether they are changed together, how often they
are changed, and their general grouping. In practice only
1 (or 0) of the segments actually get changed in normal
operation.

Invalidation is done by setting a flag on the channel when
the segment source is changed, forcing creation of a new
segment when the channel snapshot is created.

ASTERISK-28119

Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423
2018-11-26 12:56:24 -06:00