Age | Commit message (Collapse) | Author |
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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
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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
5a1d90e1fbfc4b48927aad55311f3b38efbf1f54 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
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changes" into 13
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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
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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
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* Updated unit test as ast_json_name_number() is now NULL tolerant.
ASTERISK-26466 #close
Reported by: Richard Mudgett
Change-Id: I7d4e14194f8f81f24a1dc34d1b8602c0950265a6
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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
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* 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
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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
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Change-Id: I082b239022fac462666e52a14a44304748908dc0
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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
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al." into 13
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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
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audiohooks" into 13
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Change-Id: I68321c4bea50730c39fdb486e5f23aeadd1ad636
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channels" into 13
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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
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bridge" into 13
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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
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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
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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
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Change-Id: Ic7a1236eba2408090fdabb5f717b5fa455ead715
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schema" into 13
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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
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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
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* 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
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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
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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
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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
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This reverts commit 40aa28131bc30b4516da2b20eb1a1e043920169c.
ASTERISK-26426 #close
Change-Id: I81e55c3c512f1dd6f49896f0c6b97a07d74fd8f5
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Needed to ignore an xmlstarlet return code for optional element.
Change-Id: I6a96f709b4b38c9a3f3dda4e8b07903787e16873
Reported-by: Dan Jenkins
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codec_opus" into 13
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Updated codecs/codecs.xml to add codec_opus to the external
download list.
ASTERISK-26409
Change-Id: Ia07b36539f30e852125fb2b94147dc9774df31a4
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