The readdir_r function has been deprecated and should no longer be used. This
patch removes the readdir_r dependency (replaced it with readdir) and also moves
the directory search code to a more centralized spot (file.c)
Also removed a strict dependency on the dirent structure's d_type field as it
is not portable. The code now checks to see if the value is available. If so,
it tries to use it, but defaults back to using the stats function if necessary.
Lastly, for most implementations of readdir it *should* be thread-safe to make
concurrent calls to it as long as different directory streams are specified.
glibc falls into this category. However, since it is possible that there exist
some implementations that are not safe, locking has been added for those other
than glibc.
ASTERISK-26412
ASTERISK-26509 #close
Change-Id: Id8f54689b1e2873e82a09d0d0d2faf41964e80ba
Since adding all remaining rates of Signed Linear (ASTERISK-24274), SILK
(Gerrit 3136) and Codec 2 (ASTERISK-26217), no RTP Payload Type is left in the
dynamic range (96-127). RFC 3551 section 3 allows to reassign other ranges.
Consequently, when the dynamic range is exhausted, this change utilizes payload
types in the range between 35 and 63 giving room for another 29 payload types.
ASTERISK-26311 #close
Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964
This patch adds three new CLI commands:
- ari show apps: list the registered ARI applications
- ari show app: show detailed information about an ARI application
- ari set debug: dump events being sent to an ARI application
Note that while these CLI commands live in the res_stasis module, we use
the 'ari' family for these commands. This was done as most users of
Asterisk aren't aware of the semantic differences between ARI and
res_stasis, and some 'ari' CLI commands already exist.
ASTERISK-26488 #close
Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5
PATH_MAX is not guaranteed to be defined. In parctice, all but the HURD
define it to a constant. It is indeed not safe to assume there won't be
longer paths and Asterisk generally does err safely on such cases.
So even for HURD we'll just pretend PATH_MAX is 4096.
ASTERISK-25070 #close
Change-Id: I53d10ba18c34c132bcb640a5fd8e0da1d9b22db3
Headers declare that memcpy does not accept NULL argument for the first
two parameters. Add a conditional block to prevent memcpy and ast_free
from running on vectors with NULL element array.
ASTERISK-26526 #close
Change-Id: I988a476bb5fcfcbd3f6d6c6b3e7769e4f9629b71
It is only safe to run ast_register_cleanup callbacks when all modules
have been unloaded. Previously these callbacks were run during graceful
shutdown, making it possible to crash during shutdown.
ASTERISK-26513 #close
Change-Id: Ibfa635bb688d1227ec54aa211d90d6bd45052e21
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.
Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename
This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled. This variable was only used in lock.c so it
is now initialized in that file only.
ASTERISK-26480 #close
Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
When channel format changes occurred as a result of an RTP
re-negotiation the bridge was not informed this had happened.
As a result the bridge technology was not re-evaluated and the
channel may have been in a bridge technology that was incompatible
with its formats. The bridge is now unbridged and the technology
re-evaluated when this occurs.
The chan_pjsip module also allowed asymmetric codecs for sending
and receiving. This did not work with all devices and caused one
way audio problems. The default has been changed to NOT do this
but to match the sending codec to the receiving codec. For users
who want asymmetric codecs an option has been added, asymmetric_rtp_codec,
which will return chan_pjsip to the previous behavior.
The codecs returned by the chan_pjsip module when queried by
the bridge_native_rtp module were also not reflective of the
actual negotiated codecs. The nativeformats are now returned as
they reflect the actual negotiated codecs.
ASTERISK-26423 #close
Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc
ARI and AMI allow for an explicit channel ID to be specified
when originating channels. Unfortunately, there is nothing in
place to prevent someone from using the same ID for multiple
channels. Further complicating things, adding ID validation to channel
allocation makes it impossible for ARI to discern why channel allocation
failed, resulting in a vague error code being returned.
The fix for this is to institute a new method for channel errors to be
discerned. The method mirrors errno, in that when an error occurs, the
caller can consult the channel errno value to determine what the error
was. This initial iteration of the feature only introduces "unknown" and
"channel ID exists" errors. However, it's possible to add more errors as
needed.
ARI uses this feature to determine why channel allocation failed and can
return a 409 error during origination to show that a channel with the
given ID already exists.
ASTERISK-26421
Change-Id: Ibba7ae68842dab6df0c2e9c45559208bc89d3d06
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
Adds libfftw3 to the build chain that is is going to be used for binaural
synthesis by bridge_softmix.
ASTERISK-26292
Change-Id: Iedc2f174e4ccb39ae5d9e698e339c6a17155867b
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
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
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
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)
Asterisk only supports mono audio at the moment.
This patch adds interleaved two-channel audio to Asterisk's channels.
ASTERISK-26292
Change-Id: I7a547cea0fd3c6d1e502709d9e7e39605035757a
* 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
This allows asterisk to compiled with LOW_MEMORY to load modules built
without LOW_MEMORY.
ASTERISK-26398 #close
Change-Id: I24b78ac9493ab933b11087a8b6794f3c96d4872d
Add Ogg/Opus playback support.
This uses libopusfile in order to be able to read .opus files and play
them back.
Writing/recording support is not present at this time.
ASTERISK-26409
Change-Id: I8815d23345108d8ca7c0bd640f6a1ce6b4f56955
(cherry picked from commit daee8bbd5209b4158bc1785eede845a26e6cbeaa)
Users upgrading from asterisk 13.5 to a later version and who use
realtime with peers that have mailboxes were experiencing runaway
situations that manifested as a continuous stream of taskprocessor
congestion errors, memory leaks and an unresponsive chan_sip.
A related issue was that setting rtcachefriends=no NEVER worked in
asterisk 13 (since the move to stasis). In 13.5 and earlier, when a
peer tried to register, all of the stasis threads would block and
chan_sip would again become unresponsive. After 13.5, the runaway
would happen.
There were a number of causes...
* mwi_event_cb was (indirectly) calling build_peer even though calls to
mwi_event_cb are often caused by build_peer.
* In an effort to prevent chan_sip from being unloaded while messages
were still in flight, destroy_mailboxes was calling
stasis_unsubscribe_and_join but in some cases waited forever for the
final message.
* add_peer_mailboxes wasn't properly marking the existing mailboxes
on a peer as "keep" so build_peer would always delete them all.
* add_peer_mwi_subs was unsubscribing existing mailbox subscriptions
then just creating them again.
All of this was causing a flood of subscribes and unsubscribes on
multiple threads all for the same peer and mailbox.
Fixes...
* add_peer_mailboxes now marks mailboxes correctly and build_peer only
deletes the ones that really are no longer needed by the peer.
* add_peer_mwi_subs now only adds subscriptions marked as "new" instead
of unsubscribing and resubscribing everything. It also adds the peer
object's address to the mailbox instead of its name to the subscription
userdata so mwi_event_cb doesn't have to call build_peer.
With these changes, with rtcachefriends=yes (the most common setting),
there are no leaks, locks, loops or crashes at shutdown.
rtcachefriends=no still causes leaks but at least it doesn't lock, loop
or crash. Since making rtcachefriends=no work wasnt in scope for this
issue, further work will have to be deferred to a separate patch.
Side fixes...
* The ast_lock_track structure had a member named "thread" which gdb
doesn't like since it conflicts with it's "thread" command. That
member was renamed to "thread_id".
ASTERISK-25468 #close
Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0
sd_notify() is used to notify systemd of changes to the status of the
process. This allows the systemd daemon to know when the process
finished loading (and thus only start another program after Asterisk has
finished loading).
To use this, use a systemd unit with 'Type=notify' for Asterisk.
This commit also adds the function ast_sd_notify(), a wrapper around
sd_notify that does nothing if not built with systemd support.
Also adds support for libsystemd detection in the configure script.
Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811
This implements the chan_sip legacy_useroption_parsing option but with a
better name.
* Made the caller-id number and redirecting number strings obtained from
incoming SIP URI user fields always truncated at the first semicolon.
People don't care about anything after the semicolon showing up on their
displays even though the RFC allows the semicolon.
ASTERISK-26316 #close
Reported by: Kevin Harwell
Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
This patch add config to pjsip by endpoint.
;preferred_codec_only=yes
; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.
ASTERISK-26317 #close
Reported by: AaronAn
Tested by: AaronAn
Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762
On heavy loaded system the TCP/TLS incoming calls could be
disconnected by pjproject while these calls are being
processed by asterisk which could use the session's memory pools.
If the session in the disconnected state then the session memory
pools were already freed, so we get segfault.
This patch adds a lifetime control on an INVITE session to pjproject.
The lifetime of the session is manipulated by calling
pjsip_inv_add_ref/pjsip_inv_dec_ref.
This patch uses these functions to inform pjproject that the
session is in use.
This patch adds check if the session state is not disconnected
and also checks if the memory pool is not NULL.
This patch also places tasks 'session_end' and 'session_end_completion'
into session's serializer to avoid race condition.
ASTERISK-26291 #close
Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7
Create an alternative to ast_sorcery_generic_alloc which uses astobj2
shared locking. Use this new method for the 'struct ast_sip_aor' allocator.
Change-Id: I3f62f2ada64b622571950278fbb6ad57395b5d6f
This allows standard ao2 functions to be used to release references to
an ast_named_lock. This change can cause less frequent locking of the
global named_locks container. The container is no longer locked when a
named_lock reference is being release except when this causes the
named_lock to be destroyed.
Change-Id: I644e39c6d83a153d71b3fae77ec05599d725e7e6
ast_channel_get_t38_state() calls ast_channel_queryoption() with
AST_OPTION_T38_STATE. If the passed in channel is a local channel then a
deadlock can happen if a channel lock is held when called.
* Made ast_channel_get_t38_state() callers not hold a channel lock before
calling.
* Update ast_channel_get_t38_state() doxygen to note that no channel locks
can be held when calling the function.
ASTERISK-26203 #close
Reported by: Etienne Lessard
ASTERISK-24822 #close
Reported by: David Brillert
ASTERISK-22732 #close
Reported by: Richard Mudgett
Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214
ASTERISK-25980 added the FAXMODE channel variable to res_fax.c.
Unfortunately, it also introduced a deadlock potential because
set_channel_variables() which sets FAXMODE can be called during a
masquerade. The ast_channel_get_t38_state() which gets the value used to
set FAXMODE cannot be called with the channel locked. As a result, local
channels can deadlock because of how they must acquire the locks necessary
to operate.
The intent of FAXMODE is for dialplan to know how a fax was transferred
after the fax completes. However, the previous patch sets FAXMODE to the
channel's current T.38 state AFTER the fax has completed and where T.38
may have already disconnected.
* Set FAXMODE based upon T.38 negotiations exchanged either with the fax
applications or the fax framehooks.
ASTERISK-26203
Reported by: Etienne Lessard
ASTERISK-24822
Reported by: David Brillert
ASTERISK-22732
Reported by: Richard Mudgett
Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1
MALLOC_DEBUG should not be used to check if debugging is actually
enabled, __AST_DEBUG_MALLOC should be used instead. MALLOC_DEBUG only
indicates that debugging is requested, __AST_DEBUG_MALLOC indicates it
is active.
Change-Id: I3ce9cdb6ec91b74ee1302941328462231be1ea53
Allocator functions that take file/line/func parameters are prefixed
with single-underscore when MALLOC_DEBUG is not defined,
double-underscore when it is defined. This change updates all
allocators that accept file/line/func to have the same prototype in
either ABI mode. The parameter order of __ast_vasprintf and
__ast_asprintf in utils.h have been changed to match that of astmm.h.
End-use allocator macro's have been removed from astmm.h and moved to an
unconditional part of utils.h.
Change-Id: I823bb6ce2b5675b3a4735948f10a3b420e9a023a
