We joked about splitting pbx.c into multiple files but this first step was
fairly easy. All of the pbx_builtin dialplan applications have been moved
into pbx_builtins.c and a new pbx_private.h file was added. load_pbx_builtins()
is called by asterisk.c just after load_pbx().
A few functions were renamed and are cross-exposed between the 2 source files.
Change-Id: I87066be3dbf7f5822942ac1449d98cc43fc7561a
The test_timezone_watch unit test is written to expect a
condition to be signaled when the inotify daemon thread runs.
There exists a small window where the test_timezone_watch
thread can signal the inotify daemon thread while it is not
reading on the underlying file descriptor. If this occurs
the test_timezone_watch thread will wait indefinitely for a
signal that will never arrive.
This change adds a timeout to the condition so it will return
regardless after a period of time.
Change-Id: Ifed981879df6de3d93acd3ee0a70f92546517390
When an endpoint is created, its messages are forwarded to both the tech
endpoint topic and the all endpoints topic. This is done so that various
parties interested in endpoint messages can subscribe to just the tech
endpoint and receive all messages associated with that particular technology,
as opposed to subscribing to the all endpoints topic. Unfortunately, when the
tech endpoint is created, it also forwards all of its messages to the all
topic. This results in duplicate messages whenever an endpoint publishes its
messages.
This patch resolves the duplicate message issue by creating a new function
for Stasis caching topics, stasis_cp_sink_create. In most respects, this acts
as a normal caching topic, save that it no longer forwards messages it receives
to the all endpoints topic. This allows it to act as an aggregation "sink",
while preserving the necessary caching behaviour.
ASTERISK-25137 #close
Reported-by: Vitezslav Novy
ASTERISK-25116 #close
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
Change-Id: Ie47784adfb973ab0063e59fc18f390d7dd26d17b
Use of GOTO_ON_BLINDXFR would not work at all. The target location would
never be executed by the transferring channel.
* Made feature_blind_transfer() call ast_bridge_set_after_go_on() with
valid context, exten, and priority parameters from the transferring
channel.
* Renamed some feature_blind_transfer() local variables for clarity.
ASTERISK-25641 #close
Reported by Dmitry Melekhov
Change-Id: I19bead9ffdc4aee8d58c654ca05a198da1e4b7ac
This change introduces the configuration option 'full_backend_cache'
which changes the cache to be a full mirror of the backend instead
of a per-object cache. This allows all sorcery retrieval operations
to be carried out against it and is useful for object types which
are used in a "retrieve all" or "retrieve some" pattern.
ASTERISK-25625 #close
Change-Id: Ie2993487e9c19de563413ad5561c7403b48caab5
When applying an empty DTLS configuration the filenames in the
configuration will be empty. This is actually valid to do and
each filename should simply be ignored.
Change-Id: Ib761dc235638a3fb701df337952f831fc3e69539
The JSON library Asterisk uses, jansson, is not thread
safe for us in a few ways. To help with this wrappers for JSON
object reference count increasing and decreasing were added
which use a global lock to ensure they don't clobber over
each other. This does not extend to reference count manipulation
within the jansson library itself. This means you can't safely
use the object borrowing specifier (O) in ast_json_pack and
you can't share JSON instances between objects.
This change removes uses of the O specifier and replaces them
with the o specifier and an explicit ast_json_ref. Some cases
of instance sharing have also been removed.
ASTERISK-25601 #close
Change-Id: I06550d8b0cc1bfeb56cab580a4e608ae4f1ec7d1
When the asterisk sending OriginateResponse message,
it doesn't set the "Uniqueid".
And it didn't support correct response message for
Application originate.
ASTERISK-25624 #close
Change-Id: I26f54f677ccfb0b7cfd4967a844a1657fd69b74d
An ERROR or WARNING message should generally indicate that something has gone
wrong in Asterisk. In the case of writing to a file descriptor, Asterisk is not
in control of when the far end closes its reading on a file descriptor. If the
far end does close the file descriptor in an unclean fashion, this isn't a bug
or error in Asterisk, particularly when the situation can be gracefully
handled in Asterisk.
Currently, when this happens, a user would see the following somewhat cryptic
ERROR message:
"utils.c: write() returned error: Broken pipe"
There's a few problems with this:
(1) It doesn't provide any context, other than 'something broke a pipe'
(2) As noted, it isn't actually an error in Asterisk
(3) It can get rather spammy if the thing breaking the pipe occurs often, such
as a FastAGI server
(4) Spammy ERROR messages make Asterisk appear to be having issues, or can even
mask legitimate issues
This patch changes ast_carefulwrite to only log an ERROR if we actually had one
that was reasonably under our control. For debugging purposes, we still emit
a debug message if we detect that the far side has stopped reading.
Change-Id: Ia503bb1efcec685fa6f3017bedf98061f8e1b566
pjproject < 2.5.0 will segfault on a tls transport if async_operations
is greater than 1. A runtime version check has been added to throw
an error if the version is < 2.5.0 and async_operations > 1.
To assist in the check, a new api "ast_compare_versions" was added
to utils which compares 2 major.minor.patch.extra version strings.
ASTERISK-25615 #close
Change-Id: I8e88bb49cbcfbca88d9de705496d6f6a8c938a98
Reported-by: George Joseph
Tested-by: George Joseph
Both transport and endpoint now check for the existence and readability
of tls certificate and key files before passing them on to pjproject.
This will cause the object to not load rather than waiting for pjproject
to discover that there's a problem when a session is attempted.
NOTE: chan_sip also uses ast_rtp_dtls_cfg_parse but it's located
in build_peer which is gigantic and I didn't want to disturb it.
Error messages will emit but it won't interrupt chan_sip loading.
ASTERISK-25618 #close
Change-Id: Ie43f2c1d653ac1fda6a6f6faecb7c2ebadaf47c9
Reported-by: George Joseph
Tested-by: George Joseph
This patch fixes a crash which would occur when an audiohook was
applied to a channel using an audio codec that could not be translated
to signed linear (such as when using pass-through codecs like OPUS or
when the codec translator module for the format in use is not loaded).
ASTERISK-25498 #close
Reported by: Ben Langfeld
Change-Id: Ib6ea7373fcc22e537cad373996136636201f4384
Currently if a channel is transferred out of a bridge, the BRIDGEPEER
variable (also BRIDGEPVTCALLID) remain set even once the channel is
out of the bridge. This patch removes these variables when leaving
the bridge.
ASTERISK-25600 #close
Reported by: Mark Michelson
Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da
According to the API doxygen a sched ID of 0 is valid. Unfortunately, 0
was never returned historically and several users incorrectly coded usage
of the returned sched ID assuming that 0 was invalid.
ASTERISK-25476
Change-Id: Ib19c7ebb44ec9fd393ef6646dea806d4f34e3a20
channels/chan_iax2.c:
* Initialize struct chan_iax2_pvt scheduler ids earlier because of
iax2_destroy_helper().
channels/chan_sip.c:
channels/sip/config_parser.c:
* Fix initialization of scheduler id struct members. Some off nominal
paths had 0 as a scheduler id to be destroyed when it was never started.
chan_skinny.c:
* Fix some scheduler id comparisons that excluded the valid 0 id.
channel.c:
* Fix channel initialization of the video stream scheduler id.
pbx_dundi.c:
* Fix channel initialization of the packet retransmission scheduler id.
ASTERISK-25476
Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8
dns.c and dns_system_resolver.c were spitting out errors for lookup
failures for things like not finding a SRV record even though
there was an A record. Those have been changed to debug messages.
Logging not finding ANY record is left to the higher level caller.
Also, dns_system_resolver was using Windows line endings so I
converted them to Unix style. The actual log changes are on lines
156 and 159.
Change-Id: I65be16ea15304b96f9dcb4d289dbd3e2286fc094
Several issues are addressed here:
- main() is large, and half of it is only used if we're not rasterisk;
fixed by spliting up the daemon part into a separate function.
- Call ast_term_init from rasterisk as well.
- Remove duplicate code reading/writing asterisk history file.
- Attempt to tackle background color issues and color changes that
occur. Tested by starting asterisk -c until the colors stopped
changing at odd locations.
- Remove unused term_prep() and term_prompt() functions.
ASTERISK-25585 #close
Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f
Because the context, extension, and application are stored in stringfields,
checking for them being NULL doesn't work so well. This patch uses the
appropriate string library call, ast_strlen_zero, to see if there is a value
in the context/exten/app values.
Change-Id: Ie09623bfdf35f5a8d3b23dd596647fe3c97b9a23
Previously, a trancoding module did not have access to the joint but cached
format. Therefore, the module did not have access to the attributes negotiated
via SDP (line fmtp). Now, a translation module receives the joint format.
ASTERISK-25545 #close
Change-Id: Id6878a989b50573298dab115d3371ea369e1a718
When dns_parse_answer_ex was iterating over the answers it
wasn't incrementing the answer pointer correctly after the first
answer. The result was that no answers after the first
were being returned. For results where multiple records should
have been sorted by priority, weight, etc., there was nothing
to sort so the only the first record was returned even if it
wouldn't have been the correct record based on the sort.
ASTERISK-25565 #close
Reported-by: Daniel Tryba
Tested-by George Joseph
Change-Id: I8622604fefdcd3c11e2c5609a6382e53b1467b0b
The hashtab API is pretty NULL tolerant which has resulted
in remaining callers not doing much checks themselves.
Unfortunately the function to destroy an iterator does not
do a NULL check and will result in a crash if passed NULL.
This change fixes that.
ASTERISK-25552 #close
Change-Id: Ic1bf8eec3639e5a440f1c941d3ae3893ac6ed619
In practical tests, we have seen certain taskprocessors, specifically
Stasis subscription taskprocessors, cross the recently-added high-water
mark and emit a warning. This high-water mark warning is only intended
to be emitted when things have tanked on the system and things are
heading south quickly. In the practical tests, the Stasis taskprocessors
sometimes had a max depth of 180 tasks in them, and Asterisk wasn't in
any danger at all.
As such, this ups the high-water mark to 500 tasks instead. It also
redefines the SIP threadpool request denial number to be a multiple of
the taskprocessor high-water mark.
Change-Id: Ic8d3e9497452fecd768ac427bb6f58aa616eebce
In Asterisk 13, cached formats are created before their corresponding format-
attribute module is registered. Cached formats are involved when a local
extension is called. Therefore, ast_format_generate_sdp_fmtp did not work
on local extensions. This change affects the Opus Codec, H.263 (Plus), H.264,
and format-attribute modules provided externally.
ASTERISK-25160 #close
Change-Id: I1ea1f0483e5261e2a050112e4ebdfc22057d1354
We have observed situations where the SIP threadpool may become
deadlocked. However, because incoming traffic is still arriving, the SIP
threadpool's queue can continue to grow, eventually running the system
out of memory.
This change makes it so that incoming traffic gets rejected with a 503
response if the queue is backed up too much.
Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816
When appending all formats of a type all the codecs are iterated
and added. This operation was incorrectly adding the ast_format_none
format which is special in that it is supposed to be used when no
format is present. It shouldn't be appended.
ASTERISK-25535
Change-Id: I7b00f3bdf4a5f3022e483d6ece602b1e8b12827c
This change adds handling of dead worker threads when moving them
to be active. When this happens the worker thread is removed from
both the active and idle threads container. If no threads are able
to be moved to active then the pool grows as configured.
A unit test has also been added which thrashes the idle timeout
and thread activation to exploit any race conditions between the
two.
ASTERISK-25546 #close
Change-Id: I6c455f9a40de60d9e86458d447b548fb52ba1143
Previously, format-attribute modules relied on an existing fmtp line in SDP
negotiation. However, fmtp is optional for several formats like the Opus Codec.
Now, the format-attribute module is called with an empty fmtp, which allows the
module to initialise itself to RFC defaults. Furthermore now, Asterisk is able
to differentiate between internally and externally created formats.
ASTERISK-25537 #close
Change-Id: I28f680cef7fdf51c0969ff8da71548edad72ec52
ABI compatibility stubs existed for ast_app_separate_args and ast_verbose,
this is not needed in master.
Change-Id: I07b4d2c16079da3c2c6efa55df4a74368e0bd453
Since Asterisk 13, formats are immutable and cached. However while loading a
module like chan_sip, some formats were created instead using cached ones.
ASTERISK-25535 #close
Change-Id: I479cdc220d5617c840a98f3389b3bd91e91fbd9b
