Commit Graph

28583 Commits

Author SHA1 Message Date
zuul a1c62c930a Merge "app_queue: Added initialization for "context" parameter" 2016-10-17 15:08:58 -05:00
George Joseph 2a808b2fa6 pjproject_bundled: Add patch to address SSL crash
Addresses crashes when an attempt is made to operate on an SSL socket
after the socket has been closed.

ASTERISK-26477 #close

Change-Id: I421305b357558b4f9e690210dc0f4831ef4b3002
2016-10-17 11:55:52 -05:00
Leandro Dardini 973e57d5ce app_queue: Added initialization for "context" parameter
When using Asterisk Realtime Architecture, empty fields are skipped and the
default values are used. If the "context" parameter in queue was set and then
cleared from the database, the old value remains in memory and it continues
to be used. This change initialize the "context" parameter with an empty value,
allowing clearing the parameter.

ASTERISK-26462 #close

Change-Id: I64be73d5044ce38dd02408bd0e53de965ef65905
2016-10-17 08:15:26 -05:00
Michael Kuron e9315791b3 chan_sip: Only send video on outgoing channel if incoming channel supports it
Previously, the settings videosupport=always and videosupport=yes behaved
identically and unconditionally caused a video offer to be sent in the SDP on
an outgoing call. This was a regression introduced with commit
5a1d90e1fb in Asterisk 1.6.1.

This commit restores correct behavior: videosupport=always causes a video offer
to be sent unconditionally, while videosupport=yes will only offer video on an
outbound channel if the incoming channel it is bridged to also supports video.
That way, the device receiving the outgoing call can display the correct user
interface elements for audio or video and will not unnecessarily show a blank
video window on an audio-only call.

ASTERISK-17470 #close

Change-Id: I782f4409d436114dbc97061c3570c0cd24f7c3ae
2016-10-15 05:17:54 -05:00
zuul 5df575e16a Merge "Fix issues with bundled pjproject cached download." 2016-10-14 18:48:58 -05:00
zuul 6f49ca5927 Merge "Audit ast_json_pack() calls for needed UTF-8 checks." 2016-10-14 17:17:14 -05:00
zuul 9581407fd2 Merge "json: Check party id name, number, subaddresses for UTF-8." 2016-10-14 16:32:18 -05:00
zuul 5b70ae04d4 Merge "json: Add UTF-8 check call." 2016-10-14 14:40:34 -05:00
zuul 5acac79832 Merge "res_config_mysql: Fix several issues related to recent table changes" 2016-10-14 11:49:14 -05:00
zuul 01dd018aa4 Merge "aoc.c: Whitespace cleanup" 2016-10-14 11:13:50 -05:00
zuul 7947ee2894 Merge "app_queue.c: Fix clearing of pause reason string." 2016-10-14 10:08:23 -05:00
Corey Farrell aa39a87697 Fix issues with bundled pjproject cached download.
Previously when testing I had a preexisting makeopts in ASTTOPDIR.  The
ordering of configure.ac causes --with-externals-cache to be processed
after third-party configure.  In cases where the Asterisk clone is
cleaned it would cause pjproject to be downloaded to /tmp.  This
moves processing of the externals cache and sounds cache to happen
before third-party configure.

This also addresses a possible issue with the third-party Makefile.  If
TMPDIR is set by the environment it would override the path given to
--with-externals-cache.

ASTERISK-26416

Change-Id: Ifab7f35bfcd5a31a31a3a4353cc26a68c8c6592d
2016-10-14 07:48:59 -05:00
Richard Mudgett 9c49b96374 Audit ast_json_pack() calls for needed UTF-8 checks.
Added needed UTF-8 checks before constructing json objects in various
files for strings obtained outside the system.  In this case string values
from a channel driver's peer and not from the user setting channel
variables.

* aoc.c: Fixed type mismatch in s_to_json() for time and granularity json
object construction.

ASTERISK-26466
Reported by: Richard Mudgett

Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096
2016-10-13 18:13:00 -05:00
Richard Mudgett 774d5f7ef7 json: Check party id name, number, subaddresses for UTF-8.
* Updated unit test as ast_json_name_number() is now NULL tolerant.

ASTERISK-26466 #close
Reported by: Richard Mudgett

Change-Id: I7d4e14194f8f81f24a1dc34d1b8602c0950265a6
2016-10-13 18:12:59 -05:00
Richard Mudgett 1c4c6c082d json: Add UTF-8 check call.
Since the json library does not make the check function public we
recreate/copy the function in our interface module.

ASTERISK-26466
Reported by: Richard Mudgett

Change-Id: I36d3d750b6f5f1a110bc69ea92b435ecdeeb2a99
2016-10-13 18:12:59 -05:00
Richard Mudgett 6fe5202c2c aoc.c: Whitespace cleanup
* In s_to_json() removed unnecessary ast_json_ref() to ast_json_null()
when creating the type json object.  The ref is a noop.

Change-Id: I2be8b836876fc2e34a27c161f8b1c53b58a3889a
2016-10-13 15:59:54 -05:00
Richard Mudgett c3bf1632cd app_minivm.c: Fix malformed ast_json_pack() call.
Change-Id: I082b239022fac462666e52a14a44304748908dc0
2016-10-13 15:58:47 -05:00
Richard Mudgett 9c54964dc5 app_queue.c: Fix clearing of pause reason string.
The pause reason is not always cleared when it should be cleared.

* Made set_queue_member_pause() always clear pause reason if not pausing
with a reason string.

Change-Id: I993dad19626ec017478a230e980989438b778c53
2016-10-13 15:56:53 -05:00
George Joseph 3b3d06884c res_config_mysql: Fix several issues related to recent table changes
Unlike any of the other database drivers, res_config_mysql checks that
the table definition matches the requirements for every insert and
update statement.  Since all requirements are forced to 'char', any
column that isn't a char, like ps_contacts' expiration_time,
qualify_timeout, etc., will throw a warning.  It's kinda harmless but
very misleading.  Since no other driver does those checks on insert
or update, they've been removed from res_config_mysql.  Also, all
the logic that actually attempted to ALTER the table to fix the issue
has been removed.  With the move to alembic, the auto-alter
functionality is not only unnecessary, it's also dangerous.

The other issue is that res_config_mysql calls the mysql_insert_id
function inside store_mysql.  Presumably the intention was to return
the number of rows inserted DESPITE A NOTE IN THE CODE THAT THE VALUE
IS NON_PORTABLE AND MAY CHANGE.  That value is then returned to
config realtime as the number of rows inserted.  Guess what?  The value
changed.  It now only returns the number of rows inserted if there's an
auto increment column on the table, which ps_contacts doesn't have.
Otherwise it returns 0.  So now, the insert worked but we tell config
realtime and sorcery that no rows were inserted.  That call to
mysql_insert_id was removed and we now always return 1 if the insert
succeeded.  We're only inserting 1 row at a time anyway.  If the insert
fails, we still return -1.

ASTERISK-26362 #close
Reported-by: Carlos Chavez

Change-Id: I83ce633efdb477b03c8399946994ee16fefceaf4
2016-10-12 16:49:01 -05:00
frahaase dd6fc1bb7d Binaural synthesis (confbridge): Adds libfftw3 as dependency.
Adds libfftw3 to the build chain that is is going to be used for binaural
synthesis by bridge_softmix.

ASTERISK-26292

Change-Id: Iedc2f174e4ccb39ae5d9e698e339c6a17155867b
2016-10-12 11:43:38 -05:00
zuul 8b58db0962 Merge "Binaural synthesis (confbridge): interleaved two-channel audio." 2016-10-12 11:36:06 -05:00
zuul c44456db9d Merge "bundled_pjproject: Add tests for programs used by the Makefile, et al." 2016-10-12 11:04:53 -05:00
Torrey Searle 20c3dba39e res_fax: Fix a tight race condition causing fax to crash in audio fallback
When T.38 gets rejected and G711 failback occurs there is a period of
time where neither AST_FAX_TECH_T38 nor AST_FAX_TECH_AUDIO is set,
leading to a crash.

Change-Id: Icc3f457b2292d48a9d7843dac0028347420cc982
2016-10-12 06:54:13 -05:00
zuul 1ca148cae8 Merge "Add text of cdr directory into README.md for ast-db-manage" 2016-10-11 19:45:14 -05:00
zuul 3981bc2a54 Merge "res_calendar: Add support for fetching calendars when reloading" 2016-10-11 19:22:24 -05:00
zuul f866201598 Merge "audiohooks: Remove redundant codec translations when using audiohooks" 2016-10-11 17:45:56 -05:00
zuul 48bb93b4bb Merge "vector: After remove element recheck index" 2016-10-11 16:41:33 -05:00
zuul 33997ef6a7 Merge "app_dial: Add the "Q" option to set the cause on unanswered channels" 2016-10-11 15:15:56 -05:00
zuul c58eb12f19 Merge "logger: Prevent output of verbose messages initiated from rasterisk." 2016-10-11 13:57:56 -05:00
George Joseph 86e8716952 app_dial: Add the "Q" option to set the cause on unanswered channels
The "Q" option will set the cause on the unanswered channels when
another channel answers.  It overrides the default of
ANSWERED_ELSEWHERE.

NOTE:  chan_sip does not support setting the cause on a CANCEL to
anything other than ANSWERED_ELSEWHERE.

ASTERISK-26446 #close

Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47
2016-10-11 12:05:56 -05:00
Joshua Colp f22e36461f Merge "res_rtp_asterisk: Fix infinite DTMF issue when switching to P2P bridge" 2016-10-11 08:52:45 -05:00
Badalyan Vyacheslav 17031f12fe vector: After remove element recheck index
Small fix. It is necessary to double-check
the index that we just removed because there
is a new element.

ASTERISK-26453 #close

Change-Id: Ib947fa94dc91dcd9341f357f1084782c64434eb7
2016-10-11 07:12:45 -04:00
Joshua Colp 31a3e75025 Merge "cel_odbc: Fix memory leak on module unload" 2016-10-10 19:26:37 -05:00
Joshua Colp 28c0e7ab18 Merge "pjproject_bundled: Add MALLOC_DEBUG capability" 2016-10-10 18:06:30 -05:00
Torrey Searle cc269766b8 res_rtp_asterisk: Fix infinite DTMF issue when switching to P2P bridge
If a bridge switched to P2P when a DTMF was in progress it
was possible for the DTMF to continue being sent indefinitely.

Change-Id: I7e2a3efe0d59d4b214ed50cd0b5d0317e2d92e29
2016-10-10 17:00:23 -05:00
Corey Farrell fafdde322c logger: Prevent output of verbose messages initiated from rasterisk.
Remote asterisk consoles should only display verbose log messages
created by the daemon.  The first patch for ASTERISK-26410 caused
a couple verbose messages to be printed when the rasterisk process
ended.

ASTERISK-26410

Change-Id: Ie2a1bb3753ad2724c0349ec1a336f52f7117b52a
2016-10-10 16:54:51 -05:00
Michael Walton 7af7490e42 audiohooks: Remove redundant codec translations when using audiohooks
The main frame read and write handlers in main/channel.c don't use the
optimum placement in the processing flow for calling audiohooks
callbacks, as far as codec translation is concerned. This change places
the audiohooks callback code:
 * After the channel read translation if the frame is not linear before
the translation, thereby increasing the chance that the frame is linear
as required by audiohooks
 * Before the channel write translation if the frame is linear at this
point
This prevents the audiohooks code from instantiating additional
translation paths to/from linear where a linear frame format is already
available, saving valuable CPU cycles

ASTERISK-26419

Change-Id: I6edd5771f0740e758e7eb42558b953f046c01f8f
2016-10-10 11:41:42 -05:00
Badalyan Vyacheslav 3ab7fae96b res_pjsip_config_wizard: Memory leak in module_unload
Fixed a memory leak. It removes only the first element.
Added a useful feature in vector.h to remove all items
under the CMP through a callback function / macro.

ASTERISK-26453 #close

Change-Id: I84508353463456d2495678f125738e20052da950
2016-10-10 12:01:20 -04:00
Ludovic Gasc (GMLudo) 9f62feca60 res_calendar: Add support for fetching calendars when reloading
We use a lot res_calendar, we are very happy with that, especially
because you use libical, the almost alone opensource library that
supports really ical format with all types of recurrency.

Nevertheless, some features are missed for our business use cases.

This first patch adds a new option in calendar.conf:
fetch_again_at_reload. Be my guest for a better name.

If it's true, when you'll launch "module reload res_calendar.so",
Asterisk will download again the calendar.

The business use case is that we have a WebUI with a scheduler planner,
we know when the calendars are modified.

For now, we need to define 1 minute of timeout to have a chance that
our user doesn't wait too long between the modification and the real
test.  But it generates a lot of useless HTTP traffic.


ASTERISK-26422 #close

Change-Id: I384b02ebfa42b142bbbd5b7221458c7f4dee7077
2016-10-10 10:43:53 -05:00
Joshua Colp 7126194187 Merge "Revert "Packet-Loss Concealment (PLC) for supporting codecs."" 2016-10-10 06:01:07 -05:00
Badalyan Vyacheslav ca2f3e5b99 cel_odbc: Fix memory leak on module unload
Change-Id: Ic7a1236eba2408090fdabb5f717b5fa455ead715
2016-10-09 23:30:07 -04:00
George Joseph 5fb848eebd bundled_pjproject: Add tests for programs used by the Makefile, et al.
Added tests for bzip2, tar, patch, sed and nm to configure.ac.

Set DOWNLOAD_TO_STDOUT to a working command line regardless of
whether the download program is wget, curl or fetch.

Added a 'configure.m4' file to the third-party directory which takes
care of calling any third-party project setup.  Had to move some
pjproject_bundled stuff up in configure.ac so it was called before
the third-party configure macro.

The pjproject tarball is now downloaded to the externals_cache_dir if
it was specified on the ./configure command line

Removed regeneration of the pjproject aconfigure file.  It was only
needed for an old patch that no longer applies.

Converted the tests for symbols to explicit tests since we know that
they're now available in the bundled version.  Saves a little time
during configure.

ASTERISK-26416 #close
Reported-by: Corey Farrell

Change-Id: Id1d94251c0155f8dd41b7de7067f35cfbaafbb9b
(cherry picked from commit e6b0053d75)
(cherry picked from commit a0d02f3832)
2016-10-09 21:25:20 -06:00
Joshua Colp 73f75c246b Revert "Packet-Loss Concealment (PLC) for supporting codecs."
This change introduced some fax test failures
that have not yet been addressed. So this is
not forgotten I'm submitting a change which
reverts it.

This reverts:
d56fc3b36b.

ASTERISK-25629

Change-Id: Ibc2f23c38643f5a2c89cf8915ae2d805b81bc3d5
2016-10-09 23:55:28 +00:00
George Joseph c5e8f50169 pjproject_bundled: Add MALLOC_DEBUG capability
pjproject_bundled will now use the asterisk memory debugging APIs
if MALLOC_DEBUG is turned on in menuselect.

Because this required stubs for the executable programs and the python
bindings, some Makefile reorganization was needed to properly handle
the dependencies.  As a result, the makefile now individually makes
each of the pjproject libraries separately instead of making them all
in 1 shot.  The only visible change is that there are separate status
lines printed for each library instead oif 1 for all libs.  Also, the
making of the pjproject dependency files was eliminated.  They're not
needed for building unless you're actively modifying pjproject source
files and it makes the build process faster.  Finally, any issues with
parallel builds should be resolved again making the build faster.

Change-Id: Icc5e3d658fbfb00e0a46b44c66dcc2522d5171b0
2016-10-09 18:15:12 -05:00
George Joseph 442b597929 alembic: Allow cdr, config and voicemail to exist in the same schema
cdr, config and voicemail are all separate alembic trees.  Because
alembic's default is to use a table named 'alembic_version' to store
the current tree revision, the 3 trees can't exist in the same schema
without stepping on each other.

Now each tree uses 'alembic_version_<tree_name>' as the version table.
Each tree's env.py script now first checks for 'alembic_version'.  If
it finds it AND its revision is in the tree's history, the script
renames it to 'alembic_version_<tree_name>'.  Regardless, the script
then continues with the migration using 'alembic_version_<tree_name>'
and creates that table if it's not found.  The result is that if an
existing 'alembic_version' table was found but it didn't belong to this
tree, it's left alone and 'alembic_version_<tree_name>' is used or
created.

WARNING:  If multiple trees are using the same schema, they MUST NOT
CRU or D any objects with names that might exist in the other trees.
An example would be 'yesno_values' type.  If two trees perform
operations on it, one tree could pull it out from under the other.
Thankfully we currently don't share any names among cdr, config and
voicemail.

NOTE:  Since the env.py scripts in each tree were identical, a common
env.py has been placed in the ast-db-manage directory and a symlink
to it has been placed in each tree directory.

ASTERISK-24311 #close
Reported-by: Dafi Ni

Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898
2016-10-07 07:49:42 -05:00
Joshua Colp 55e8c9eff2 Merge "chan_sip: Honor support of Symmetric Response (rport) for SIP requests." 2016-10-07 05:58:56 -05:00
Alexander Traud c4268ec734 chan_sip: Honor support of Symmetric Response (rport) for SIP requests.
In the SIP channel driver chan_sip, the default is "auto_force_rport". When no
NAT was detected, for example in case of IPv6, Asterisk uses the IP address
from the headers within the SIP-REGISTER for subsequent SIP signaling. When
the remote party specifies support for Symmetric Response (RFC 3581) via the
parameter "rport", Asterisk should not extract the port from the SIP headers
but reuse the port of the transport. This did not happen because of a typo.

ASTERISK-26438 #close

Change-Id: If6e7891848aaf96666dee5305695f7c6667cd5a6
2016-10-05 11:25:11 +02:00
Joshua Colp ac3c51f284 Merge "app_queue: Update dynamic members ringinuse on reload." 2016-10-04 12:45:54 -05:00
frahaase c455823657 Binaural synthesis (confbridge): interleaved two-channel audio.
Asterisk only supports mono audio at the moment.
This patch adds interleaved two-channel audio to Asterisk's channels.

ASTERISK-26292

Change-Id: I7a547cea0fd3c6d1e502709d9e7e39605035757a
2016-10-03 03:12:50 -05:00
Corey Farrell 2a03575c30 astobj2: Add backtrace to log_bad_ao2.
* Compile __ast_assert_failed unconditionally.
* Use __ast_assert_failed to log messages from log_bad_ao2
* Remove calls to ast_assert(0) that happen after log_bad_ao2 was run.

Change-Id: I48f1af44b2718ad74a421ff75cb6397b924a9751
2016-09-30 19:25:40 -04:00