Add a function to check if there is an exact match a one string between
delimiters in another string.
Add a function that will create an ast_json object out of a list of
Asterisk variables. An excludes string can also optionally be passed
in.
Also, add a macro to make it easier to get object integers.
Change-Id: I5f34f18e102126aef3997f19a553a266d70d6226
Some ast_stun_request users do not provide a destination address when
sending to a connection-mode socket.
ASTERISK-29691
Change-Id: Idd9114c3380216ba48abfc3c68619e79ad37defc
The current versions do not support future dates in all say application when using the 'Q' or 'q' format parameter and says "today" for everything that is greater than today
ASTERISK-29637
Change-Id: I1fb1cef0ce3c18d87b1fc94ea309d13bc344af02
The MessageSend AMI action has been updated to allow the Destination
and the To addresses to be provided separately. This brings the
MessageSend manager command in line with the capabilities of the
MessageSend dialplan application.
ASTERISK-29663 #close
Change-Id: I8513168d3e189a9fed88aaab6f5547ccb50d332c
Adds the ability for users to log to custom log levels
by providing custom log level names in logger.conf. Also
adds a logger show levels CLI command.
ASTERISK-29529
Change-Id: If082703cf81a436ae5a565c75225fa8c0554b702
Up until now, all of the logic used to translate
arguments to the Say applications has been
directly coupled to playback, preventing other
modules from using this logic.
This refactors code in say.c and adds a SAYFILES
function that can be used to retrieve the file
names that would be played. These can then be
used in other applications or for other purposes.
Additionally, a SayMoney application and a SayOrdinal
application are added. Both SayOrdinal and SayNumber
are also expanded to support integers greater than
one billion.
ASTERISK-29531
Change-Id: If9718c89353b8e153d84add3cc4637b79585db19
dsp.c contains arbitrary tone detection functionality
which is currently only used for fax tone recognition.
This change makes this functionality publicly
accessible so that other modules can take advantage
of this.
Additionally, a WaitForTone and TONE_DETECT app and
function are included to allow users to do their
own tone detection operations in the dialplan.
ASTERISK-29546
Change-Id: Ie38c395000f4fd4d04e942e8658e177f8f499b26
ncurses 6.1 introduced an extended number format for terminfo files
which the terminfo parsing in Asterisk is not able to parse. This
results in some TERM values that do support color (screen-256color on
Ubuntu 20.04 for example) to not get a color console.
ASTERISK-29630 #close
Change-Id: I27a4fcfab502219924af2d6b1c46feba92903cb3
IPv6 nameserver addresses are stored in different part of the
__res_state structure, so look there if we appear to have support for
it.
ASTERISK-28004 #close
Change-Id: I67067077d8a406ee996664518d9c8fbf11f6977d
There are 3 separate changes here but they are all closely related:
* Only try to set matchfield attributes on 'field' nodes
* We need to adjust how we treat the category pointer based on the
value of the category_match, to avoid memory corruption. We now
generate a regex-like string when match types other than
ACO_WHITELIST and ACO_BLACKLIST are used.
* Switch app_agent_pool from ACO_BLACKLIST_ARRAY to
ACO_BLACKLIST_EXACT since we only have one category we need to
ignore, not two.
ASTERISK-29614 #close
Change-Id: I7be7bdb1bb9814f942bc6bb4fdd0a55a7b7efe1e
Allows for the digit # to be read as a digit,
just like any other DTMF digit, as opposed to
forcing it to be used as an end of input
indicator. The default behavior remains
unchanged.
ASTERISK-18454 #close
Change-Id: I3033432adb9d296ad227e76b540b8b4a2417665b
This allows the STUN server to change its IP address without having to
reload the res_rtp_asterisk module.
The refresh of the name resolution occurs first when the module is
loaded, then recurringly, slightly after the previous DNS answer TTL
expires.
ASTERISK-29508 #close
Change-Id: I7955a046293f913ba121bbd82153b04439e3465f
The attended transfer feature will emit a warning if the user
cancels the transfer or the attended transfer doesn't complete
for any reason. Changes the warning to a verbose message,
since nothing is actually wrong here.
ASTERISK-29612 #close
Change-Id: I64c93cdb21360a0a8d45e9cb6db3af8168f66e6d
When playing a remote sound file, which is not in cache, first we need
to download it with ast_bucket_file_retrieve.
This can take a while if the remote host is slow. The current CURL
timeout is 180secs, so in extreme situations, it can take 3 minutes to
return.
Because ast_media_cache_retrieve has a lock on all function, while we
are waiting for the delayed download, Asterisk is not able to play any
more files, even the files already cached locally.
ASTERISK-29544 #close
Change-Id: I8d4142b463ae4a1d4c41bff2bf63324821567408
With Asterisk 1.6.0, in the main parser for the configuration file
extensions.conf, the separator was changed from vertical bar to comma.
However, the first separator was not changed in aelparse; it still had
to be a vertical bar, and no comma was allowed.
Additionally, this change allows the vertical bar for the first and
last parameter again, even in the main parser, because the vertical bar
was still accepted for the other parameters.
ASTERISK-29540
Change-Id: I882e17c73adf4bf2f20f9046390860d04a9f8d81
If Asterisk gets G.729 6-byte VAD frames inbound, then at outbound Asterisk sends this G.729 stream with non-continuous timestamps.
This makes the audio stream not-playable at the receiver side.
Linphone isn't able to play such an audio - lots of disruptions are heard.
Also I had complains of bad audio from users which use other types of phones.
After debugging, I found this is a regression connected with RTP Smoother (main/smoother.c).
Smoother has a special code to handle G.729 VAD frames (search for AST_SMOOTHER_FLAG_G729 in smoother.c).
However, this flag is never set in Asterisk-12 and newer.
Previously it has been set (see Asterisk-11).
ASTERISK-29526 #close
Change-Id: I6f51ecb1a3ecd9c6d59ec5a6811a27446e17065d
Use cURL's URL parsing API, falling back to the urlparser library, to
parse playback URLs in order to find their file extensions.
For backwards compatibility, we first look at the full URL, then at
any Content-Type header, and finally at just the path portion of the
URL.
ASTERISK-27871 #close
Change-Id: I16d0682f6d794be96539261b3e48f237909139cb
This appears to just have been a copy/paste error from 6258bbe7. Fix
suggested by Ross Beer in ASTERISK~29166.
Change-Id: I51e0de92042e53f37597c6f83a75621ef0d1ae37
If the system time has stepped backwards because of a time
adjustment between the time a frame is timestamped and the
time we check the timestamps in abstract_jb:hook_event_cb(),
we get a negative interval, but we don't check for that there.
abstract_jb:hook_event_cb() then calls
fixedjitterbuffer:fixed_jb_get() (via abstract_jb:jb_get_fixed)
and the first thing that does is assert(interval >= 0).
There are several issues with this...
* abstract_jb:hook_event_cb() saves the interval in a variable
named "now" which is confusing in itself.
* "now" is defined as an unsigned int which converts the negative
value returned from ast_tvdiff_ms() to a large positive value.
* fixed_jb_get()'s parameter is defined as a signed int so the
interval gets converted back to a negative value.
* fixed_jb_get()'s assert is NOT an ast_assert but a direct define
that points to the system assert() so it triggers even in
production mode.
So...
* hook_event_cb()'s "now" was renamed to "relative_frame_start" and
changed to an int64_t.
* hook_event_cb() now checks for a negative value right after
retrieving both the current and framedata timestamps and just
returns the frame if the difference is negative.
* fixed_jb_get()'s local define of ASSERT() was changed to call
ast_assert() instead of the system assert().
ASTERISK-29480
Reported by: Dan Cropp
Change-Id: Ic469dec73c2edc3ba134cda6721a999a9714f3c9
When using the Busy() and Congestion() applications the
function ast_safe_sleep is used by wait_for_hangup to safely
wait on the channel. This function may send silence if Asterisk
is configured to do so using the transmit_silence option.
In a scenario where an answered channel dials a Local channel
either directly or through call forwarding and the Busy()
or Congestion() dialplan applications were executed with the
transmit_silence option enabled the busy or congestion
tone would not be heard.
This is because inband generation of tones (such as busy
and congestion) is stopped when other audio is sent to
the channel they are being played to. In the given
scenario the transmit_silence option would result in
silence being sent to the channel, thus stopping the
inband generation.
This change adds a variant of ast_safe_sleep which can be
used when silence should not be played to the channel. The
wait_for_hangup function has been updated to use this
resulting in the tones being generated as expected.
ASTERISK-29485
Change-Id: I066bfc987a3ad6f0ccc88e0af4cd63f6a4729133
Previously, SayNumber always emitted a warning if the caller hung up
during execution. Usually this isn't correct, so check if the channel
hung up and, if so, don't emit a warning.
ASTERISK-29475
Change-Id: Ieea4a67301c6ea83bbc7690c1d4808d79a704594
Although Asterisk can receive and propogate flash events, it currently
provides no mechanism for doing anything with them itself.
This AMI event allows flash events to be processed by Asterisk.
Additionally, AST_CONTROL_FLASH is included in a switch statement
in channel.c to avoid throwing a warning when we shouldn't.
ASTERISK-29380
Change-Id: Ie17ffe65086e0282c88542e38eed6a461ec79e81
AST_CONTROL_FLASH isn't accounted for in a switch statement in file.c
where it should be ignored. Adding this to the switch ensures a
warning isn't thrown on RFC2833 flash events, since nothing's amiss.
ASTERISK-29372
Change-Id: I4fa549bfb7ba1894a4044de999ea124877422fbc
STIR/SHAKEN encodes using base64 URL format. Currently, we just use
base64. New functions have been added that convert to and from base64
encoding.
The origid field should also be an UUID. This means there's no reason to
have it as an option in stir_shaken.conf, as we can simply generate one
when creating the Identity header.
https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021
Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c
Enhancements:
* The MessageSend dialplan application now takes an optional
third argument that can set the message's "To" field on
outgoing messages. It's an alternative to using the
MESSAGE(to) dialplan function.
NOTE: No channel driver currently implements this field. A
follow-on commit for res_pjsip_messaging will implement it for
the chan_pjsip channel driver.
* To prevent confusion with the first argument, currently named
"to", it's been renamed to "destination". Its function,
creating the request URI, hasn't changed.
* The documentation for MessageSend was updated to be
more clear about the parameters and how they interact
the MESSAGE() dialplan function.
* With the rename of MessageSend's first parameter, and the fact
that message.c references <info> elements in chan_sip.c,
res_pjsip_messaging.c and res_xmpp, they each needed
documentation updates to use MessageDestinationInfo instead of
MessageToInfo.
* appdocsxml.dtd was updated to include a missing element
declaration for "dataType". This was showing up as an error
in Eclipse's dtd editor.
* Despite the changes in this commit, there should be
no impact to current users of MessageSend.
Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a
When a stream topology is provided to chan_local when dialing
it filters the audio formats down. This operation did not skip
streams which were removed (that have no formats) resulting in
calling being aborted.
This change causes such streams to be skipped.
ASTERISK-29407
Change-Id: I1de8b98727cb2d10f4bc287da0b5fdcb381addd6
Up/down sampling changes the number of samples produced by a translation.
This must be taken into account when checking the codec's buffer size.
ASTERISK-29328
Change-Id: I9aebe2f8788e00321a7f5c47aa97c617f39e9055
There is a possibility, when bridge_channel_write_frame() is
called, that the bridge_channel->chan will be NULL. The first
thing bridge_channel_write_frame() does though is call
ast_channel_is_multistream() which had no check for a NULL
channel and therefore caused a segfault. Since it's still
possible for bridge_channel_write_frame() to write the frame to
the other channels in the bridge, we don't want to bail before we
call ast_channel_is_multistream() but we can just skip the
multi-channel stuff. So...
bridge_channel_write_frame() only calls ast_channel_is_multistream()
if bridge_channel->chan is not NULL.
As a safety measure, ast_channel_is_multistream() now returns
false if the supplied channel is NULL.
ASTERISK-29379
Reported-by: Vyrva Igor
Reported-by: Ross Beer
Change-Id: Idfe62dbea8c69813ecfd58e113a6620dc42352ce
Using the information from the MODULEINFO XML we can
now output useful information at the end of module
loading for deprecated modules. This includes the
version it was deprecated in, the version it will be
removed in, and the replacement if available.
ASTERISK-29339
Change-Id: I2080dab97d2186be94c421b41dabf6d79a11611a
Added a TIME_UNIT enumeration, and a function that converts a
string to one of the enumerated values. Also, added functions
that create and initialize a timeval object using a specified
value, and unit type.
Change-Id: Ic31a1c3262a44f77a5ef78bfc85dcf69a8d47392
The 'core' console (ie: asterisk -c) does read logger.conf and does
use the dateformat= option.
Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf
and uses a hard coded dateformat option for printing received verbose messages:
main/logger.c: static char dateformat[256] = "%b %e %T"
This change will load logger.conf for each remote console session and
use the dateformat= option to set the per-line timestamp for verbose messages
Change-Id: I3ea10990dbd920e9f7ce8ff771bc65aa7f4ea8c1
ASTERISK-25358: #close
Reported-by: Igor Liferenko
When using the ast_unreal_lock_all function no channel
locks can be held before calling it.
This change unlocks the channel that indicate was
called on before doing so and then relocks it afterwards.
ASTERISK-29035
Change-Id: Id65016201b5f9c9519a216e250f9101c629e19e9
There exists an inconsistency with framehook usage
such that it is only on reads that the frame should
be freed, not on writes as well.
ASTERISK-29071
Change-Id: I5ef918ebe4debac8a469e8d43bf9d6b673e8e472
Some sorcery objects actually contain dynamic content
that can change despite the underlying configuration
itself not changing. A good example of this is the
res_pjsip_endpoint_identifier_ip module which allows
specifying hostnames. While the configuration may not
change between reloads the DNS information of the
hostnames can.
This change adds the ability for a sorcery object to be
marked as having dynamic contents which is then taken
into account when reloading by the sorcery file based
config module. If there is an object with dynamic content
then a reload will be forced while if there are none
then the existing behavior of not reloading occurs.
ASTERISK-29321
Change-Id: I9342dc55be46cc00204533c266a68d972760a0b1
A frame suppression API exists as part of channels
which allows audio frames to or from a channel to
be dropped. The MuteAudio AMI action uses this
API to perform its job.
This API uses a framehook to intercept flowing
audio and drop it when appropriate. It is the
responsibility of the framehook to free the
frame it is given if it changes the frame. The
suppression API failed to do this resulting in
a leak of audio frames.
This change adds the freeing of these frames.
ASTERISK-29071
Change-Id: Ie50acd454d672d36af914050c327d2e120d8ba7b
Implemented the english way of saying the year in ast_say_date_with_format_nl.
Currently the numbers are spoken correctly until 2020 and stopped working
this year.
ASTERISK-29297 #close
Reported-by: Jacek Konieczny
Change-Id: If5918eed5ab05df31df4dd23f08a909a60f6aba4
Instead of looking for pass-through formats in the list of transcodable
formats (which is going to find nothing), go through the result which
is going to be the jointcaps of the tech_pvt of the channel. Finally,
only with that list, ast_format_cap_remove(.) is going to succeed.
This restores the behaviour of Asterisk 1.8. However, it does not fix
ASTERISK_29282 because that issue report is about chan_sip and PJSIP.
Here, only chan_sip is fixed because PJSIP does not even call
ast_rtp_instance_available_formats -> ast_translate_available_format.
Change-Id: Icade2366ac2b82935b95a9981678c987da2e8c34
After some changes to streams and topologies, receiving fax through
local channels stopped working. This change adds a stream topology with
a stream of type IMAGE to the local channel pair and allows fax to be
received.
ASTERISK-29035 #close
Change-Id: Id103cc5c9295295d8e68d5628e76220f8f17e9fb
When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is
0 when no protocl specific error
SIP example of failure, 3xx-6xx for the SIP error code received
This allows applications to perform actions based on the failure
reason.
ASTERISK-29252 #close
Reported-by: Dan Cropp
Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4
Rename check_manager_enabled() and check_webmanager_enabled() to begin
with ast_ so that the symbols are automatically exported by the
linker.
ASTERISK~29184
Change-Id: I85762b9a5d14500c15f6bad6507138c8858644c9
This was dead code, test code introduced with Asterisk 13. This was
found while analyzing ASTERISK_28416 and ASTERISK_29185. This change
partly fixes, not closes those two issues.
Change-Id: I42d0daa37f6f334c7d86672f06f085858a3f3940
The documentation in the wiki says there should be spyee-channel
information elements in the ChanSpyStop AMI event.
https://wiki.asterisk.org/wiki/x/Xc5uAg
However, this is not the case in Asterisk <= 16.10.0 Version. We're
using these Spyee* arguments since Asterisk 11.x, so these arguments
vanished in Asterisk 12 or higher.
For maximum compatibility, we still send the ChanSpyStop event even if
we are not able to find any 'Spyee' information.
ASTERISK-28883 #close
Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f
