When executing dial, the topology of the incoming channel is cloned and
used for the outgoing channel. This creates issues when an incoming
stream is sendonly or recvonly as the stream state of the outgoing
channel will be the same as the stream state of the incoming channel.
Now the stream state is flipped for the outgoing stream in
dial_exec_full if the incoming stream topology is recvonly or sendonly.
ASTERISK-29655
Reported by: Michael Auracher
ASTERISK-29638
Reported by: Michael Auracher
Change-Id: I294dc834ac9a5f048b101b691669959e9df630e1
The ReceiveMF and ReceiveSF applications currently always
return 0, even if a channel has hung up. The call will still
end but generally applications are expected to return -1 if
the channel has hung up.
We now return -1 if a hangup occured to bring this behavior
in line with this norm. This has no functional impact, but
merely increases conformity with how these modules interact
with the PBX core.
ASTERISK-29951 #close
Change-Id: I234d755050ab8ed58f197c6925b968ba26b14033
Adds the m option to the Queue application, which allows a
music on hold class to be specified at runtime which will
override the class configured in queues.conf.
This option functions like the m option to Dial.
ASTERISK-29876 #close
Change-Id: Ie25a48569cf8755c305c9438b1ed292c3adcf8d7
Currently, if a user tries to access a non-dynamic
MeetMe conference and the conference is not found,
the call simply silent hangs up. There is no indication
to the user that anything went wrong at all.
This changes the relevant debug message to a warning
so that the user is notified of this invalidity.
ASTERISK-29954 #close
Change-Id: Iebcfae3755d00f2150d676ee211c57bc59530048
Adds documentation for all of the possible return values
for the DIALSTATUS variable in the Dial application.
ASTERISK-25716
Change-Id: Id22593f1f1f7ea86e5734cee49516ec50848e8c0
This adds a new AMI action called QueueWithdrawCaller.
This AMI action makes it possible to withdraw a caller from a queue,
in a safe and a generic manner.
This can be useful for retrieving a specific call and
dispatching it to a specific extension.
It works by signaling the caller to exit the queue application
whenever it can. Therefore, it is not guaranteed
that the call will leave the queue.
ASTERISK-29909 #close
Change-Id: Ic15aa238e23b2884abdcaadff2fda7679e29b7ec
Adds the since tag to the documentation DTD so
that individual applications, functions, etc.
can now specify when they were added to Asterisk.
This tag is added at the individual application,
function, etc. level as opposed to at the module
level because modules can expand over time as new
functionality is added, and granularity only
to the module level would generally not be useful.
This enables the ability to more easily determine
when new functionality was added to Asterisk, down
to minor version as opposed to just by major version.
This makes it easier for users to write more portable
dialplan if desired to not use functionality that may
not be widely available yet.
ASTERISK-29896 #close
Change-Id: Ibbb35c702d8038bdc3fd0a944fbfa69384cc15d5
Currently, if VoiceMailMain is called with a mailbox, if that
mailbox doesn't exist, then the application silently falls back
to prompting the user for the mailbox, as if no arguments were
provided.
However, if a specific mailbox is requested and it doesn't exist,
then no warning at all is emitted.
This fixes this behavior to now warn if a specifically
requested mailbox could not be accessed, before falling back to
prompting the user for the correct mailbox.
ASTERISK-29920 #close
Change-Id: Ib4093b88cd661a2cabc5d685777d4e2f0ebd20a4
mpg123 doesn't support HTTPS, but the MP3Player application
doesn't document this or warn the user about this. HTTPS
streams have become more common nowadays and users could
reasonably try to play them without being aware they should
use the HTTP stream instead.
This adds documentation to note this limitation. It also
throws a warning if users try to use the HTTPS stream to
tell them to use the HTTP stream instead.
ASTERISK-29900 #close
Change-Id: Ie3b029be5258c5a701f71ed3b1a7a80d1e03b827
Adds an option to the ReceiveMF application to allow specifying a
maximum number of digits.
Originally, this capability was not added to ReceiveMF as it was
with ReceiveSF because typically a ST digit is used to denote that
sending of digits is complete. However, there are certain signaling
protocols which simply transmit a digit (such as Expanded In-Band
Signaling) and for these, it's necessary to be able to read a
certain number of digits, as opposed to until receiving a ST digit.
This capability is added as an option, as opposed to as a parameter,
to remain compatible with existing usage (and not shift the
parameters).
ASTERISK-29877 #close
Change-Id: I4229167c9aa69b87402c3c2a9065bd8dfa973a0b
There are a lot of Queue AMI actions and Queue applications
which do not load queue and queue members from Realtime.
AMI actions
QueuePause - if queue not in memory - response "Interface not found".
QueueStatus/QueueSummary - if queue not in memory - empty response.
Applications:
PauseQueueMember - if queue not in memory
Attempt to pause interface %s, not found
UnpauseQueueMember - if queue not in memory
Attempt to unpause interface xxxxx, not found
This patch adds a new function load_realtime_queues
which loads queue and queue members for desired queue
or all queues and all members if param 'queuename' is NULL or empty.
Calls the function load_realtime_queues when needed.
Also this patch fixes leak of ast_config in function set_member_value.
Also this patch fixes incorrect LOG_WARNING when pausing/unpausing
already paused/unpaused member.
The function ast_update_realtime returns 0 when no record modified.
So 0 is not an error to warn about.
ASTERISK-29873 #close
ASTERISK-18416 #close
ASTERISK-27597 #close
Change-Id: I554ee0eebde93bd8f49df7f84b74acb21edcb99c
if holdtime is (0 min, 0 sec) there is no hold time announcements
we should then also not playing queue-thankyou
ASTERISK-29831
Change-Id: Ic7e51dcde526b23f1cd8d24e1d1e2d81e10f9d2c
gethostbyname() and gethostbyname_r() are deprecated in favor of
getaddrinfo() which we use in the ast_sockaddr family of functions.
ASTERISK-29819 #close
Change-Id: Ie277c0ef768d753b169c121ef570a71665692ab7
adding support for playing the correct en/et for nordic languages
by adding 'n' for neuter gender in the relevant ast_say_number
ASTERISK-29827
Change-Id: I03ebc827d2f0dc95132ab2f42799893c70edc5b1
Currently, the MP3Player application doesn't
emit a warning if attempting to play a stream
which no longer exists. This can be a common
scenario as many mp3 streams are valid at some
point but can disappear at any time.
Now a warning is thrown if attempting to play
a nonexistent MP3 stream, instead of silently
exiting.
ASTERISK-29829 #close
Change-Id: I53a0bf1ed1740166655eb66fe7675f6f808bf535
Adds missing documentation for some channel,
bridge, and queue events.
ASTERISK-24427
ASTERISK-29515
Change-Id: I92b06b88c8cadc0155f95ebe3e870b3e795a8c64
Adds tech-agnostic support for SF signaling
by adding SF sender and receiver applications
as well as Dial integration.
ASTERISK-29802 #close
Change-Id: I7ec50752e9a661af639425e5d1e339f17411bcad
A previous patch for ASTERISK_29578 caused a 'leak' of
extension state information across queues, causing the
state of the first member of unrelated queues to be
updated in addition to the correct member. Which queues
and members depended on the order of queues in the
iterator.
Additionally, it is possible to use the same 'hint:' on
multiple queue members, so the update cannot break out
of the update loop early when a match is found.
ASTERISK-29806 #close
Change-Id: If2c1d1cc2a752afd9286d79710fc818596e7a7ad
added that we set DIALEDPEERNUMBER on the outgoing channels
so it is avalible in b(content^extension^line)
this add the same behaviour as Dial
ASTERISK-29795
Change-Id: Icbc589ea2066f0c401a892bf478f6b2fd44e62f6
added support for playing the correct plural sound file
dependen on where you have 1 or multipe messages
based on the existing SE/NO code
ASTERISK-29797
Change-Id: I88aa814d02f3772bb80b474204b1ffb26fe438c2
Adds a ReceiveText application that can be used in
conjunction with SendText. Currently, there is no
way in Asterisk to receive text in the dialplan
(or anywhere else, really). This allows for Asterisk
to be the recipient of text instead of just the sender.
ASTERISK-29759 #close
Change-Id: Ica2c354a42bff69f323a0493d3a7cd0fb129d52d
Adds tech-agnostic support for MF signaling by adding
MF sender and receiver applications as well as Dial
integration.
ASTERISK-29496-mf #do-not-close
Change-Id: I61962b359b8ec4cfd05df877ddf9f5b8f71927a4
changed that when we recive a CANCEL that we set HANGUPCAUSE to AST_CAUSE_NORMAL_CLEARING
ASTERISK-28053
Reported by: roadkill
Change-Id: Ib653aec2282f55b59d87484391cc07c8e6612b89
Most examples in the XML documentation use the
example tag to demonstrate examples, which gets
parsed specially in the Wiki to make it easier
to follow for users.
This fixes a few modules to use the example
tag instead of vanilla para tags to bring them
in line with the standard syntax.
ASTERISK-29777 #close
Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b
Refactors generic functions used for email generation
into utils.c so that they can be used by multiple
modules, including app_voicemail and app_minivm,
to avoid code duplication.
ASTERISK-29715 #close
Change-Id: I1de0ed3483623e9599711129edc817c45ad237ee
Fixes a deadlock in app_morsecode caused by locking
the channel twice when reading variables from the
channel. The duplicate lock is simply removed.
ASTERISK-29744 #close
Change-Id: I204000701f123361d7f85e0498fedc90243c75e4
Currently, when the t option is specified with no arguments,
the # character is still treated as a terminator, even though
no character should be treated as a terminator.
This is because a previous regression fix was modified to
remove the use of NULL as a default altogether. However,
NULL and an empty string actually refer to different
arrangements and should be treated differently. NULL is the
default terminator (#), while an empty string removes the
terminator altogether. This is the behavior being used by
the rest of the core.
Additionally, since S_OR catches empty strings as well as
NULL (not intended), this is changed to a ternary operator
instead, which fixes the behavior.
ASTERISK-29705 #close
Change-Id: I9b6b72196dd04f5b1e0ab5aa1b0adf627725e086
If users are able to press # for options while leaving
a message and then press 3 to rerecord the message, if
the caller hangs up during the rerecord prompt but before
Asterisk starts recording a message, then an "empty"
voicemail gets processed whereby an email gets sent out
notifying the user of a 0:00 duration message. The file
doesn't actually exist, so playback will fail since there
was no message to begin with.
This adds a check after the streaming of the rerecord
announcement to see if the caller has hung up. If so,
we bail out early so that we can clean up properly.
ASTERISK-29391 #close
Change-Id: Id965d72759a2fd3b39afb76fec08aaebebe75c31
Add a time_t logintime to storage a time when a member is added into a
queue.
Also, includes show this time (in seconds) using a 'queue show' command
and the field LoginTime for response for AMI events.
ASTERISK-18069 #close
Change-Id: Ied6c3a300f78d78eebedeb3e16a1520fc3fff190
I am adding a mix option that will play by filename and say.conf unlike
say option that will only play with say.conf. It
will look on the format of the name, if it is like say it play with
say.conf if not it will play the file name.
ASTERISK-29662
Change-Id: I815816916a308f0fa8f165140dc15772dcbd547a
If the terminator character is not explicitly specified
and an indications tone is used for reading a digit,
there is no null pointer check so Asterisk crashes.
This prevents null usage from occuring.
ASTERISK-29673 #close
Change-Id: Ie941833e123c3dbfb88371b5de5edbbe065514ac
Previously, if custom hints were used with the hint:
format in app_queue, when device state changes occured,
app_queue would only do a literal string comparison of
the context used for the hint in app_queue and the context
of the hint which just changed state. This caused hints
to not update and become stale if the context associated
with the agent included the context which actually changes
state, essentially completely breaking device state for
any such agents defined in this manner.
This fix adds an additional check to ensure that included
contexts are also compared against the context which changed
state, so that the behavior is correct no matter whether the
context is specified to app_queue directly or indirectly.
ASTERISK-29578 #close
Change-Id: I8caf2f8da8157ef3d9ea71a8568c1eec95592b78
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
In new mpg123 versions (since 1.26) the default output is 32 bits
Asterisk expects the output in 16 bits, so we force the output to be on 16 bits.
It will work wit new and old versions of mpg123.
Thanks Thomas Orgis <thomas-forum@orgis.org> for giving the key!
ASTERISK-29635 #close
Change-Id: I88c7740118b5af4e895bd8b765b68ed5c11fc816
Adds a SendMF application and PlayMF manager
event to send arbitrary R1 MF tones on the
current or specified channel.
ASTERISK-29496
Change-Id: I5d89afdbccee3f86cc702ed96d882f3d351327a4
Previously, the error emitted when app_stack tries
to branch to a dialplan location that doesn't exist
has included only the information about the attempted
branch in the error log. This adds the current location
as well so users can see where the branch failed in
the logs.
ASTERISK-29626
Change-Id: Ia23502ab2ad21485a1ac74295063a8f25a6df5ce
There is an option to silence voicemail instructions but it does not
take into consideration if a recorded greeting exists or not. Add a
new 'S' option that does that.
ASTERISK-29632 #close
Change-Id: I03f2f043a9beb9d99deab302247e2a8686066fb4
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
Prevents reloads of app_queue from also resetting
queue statistics.
Also preserves individual queue agent statistics
if we're just reloading members.
ASTERISK-28701
Change-Id: Ib5d4cdec175e44de38ef0f6ede4a7701751766f1
The Milliwatt application uses incorrect tone timings
that cause it to play the 1004 Hz tone constantly.
This adds an option to enable the correct timing
behavior, so that the Milliwatt application can
be used for milliwatt test lines. The default behavior
remains unchanged for compatability reasons, even
though it is incorrect.
ASTERISK-29575 #close
Change-Id: I73ccc6c6fcaa31931c6fff3b85ad1805b2ce9d8c
Previously, the Morsecode application only supported international
Morse code. This adds support for American Morse code and adds an
option to configure the frequency used in off intervals.
Additionally, the application checks for hangup between tones
to prevent application execution from continuing after hangup.
ASTERISK-29541
Change-Id: I172431a2e18e6527d577e74adfb05b154cba7bd4
A list of codecs to use for dialplan-originated calls can
now be specified in Originate, similar to the ability
in call files and the manager action.
Additionally, we now default to just using the slin codec
for originated calls, rather than all the slin* codecs up
through slin192, which has been known to cause issues
and inconsistencies from AMI and call file behavior.
ASTERISK-29543
Change-Id: I96a1aeb83d54b635b7a51e1b4680f03791622883
app_meetme is deprecated in 19, to be removed in 21.
app_osplookup is deprecated in 19, to be removed in 21.
chan_alsa is deprecated in 19, to be removed in 21.
chan_mgcp is deprecated in 19, to be removed in 21.
chan_skinny is deprecated in 19, to be removed in 21.
res_pktccops is deprecated in 19, to be removed in 21.
app_macro was deprecated in 16, to be removed in 21.
chan_sip was deprecated in 17, to be removed in 21.
res_monitor was deprecated in 16, to be removed in 21.
ASTERISK-29548
ASTERISK-29549
ASTERISK-29550
ASTERISK-29551
ASTERISK-29552
ASTERISK-29553
ASTERISK-29558
ASTERISK-29567
ASTERISK-29572
Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131
Allows multiple files comprising an agent announcement
to be played by separating on the ampersand, similar
to the multi-file support in other Asterisk applications.
ASTERISK-29528
Change-Id: Iec600d8cd5ba14aa1e4e37f906accb356cd7891a
Adds application to asynchronously collect digits
dialed on a channel in the TX or RX direction
using a framehook and stores them in a specified
variable, up to a configurable number of digits.
ASTERISK-29477
Change-Id: I51aa93fc9507f7636ac44806c4420ce690423e6f
While several applications exist to wait for
a certain event to occur, none allow waiting
for any generic expression to become true.
This application allows for waiting for a condition
to become true, with configurable timeout and
checking interval.
ASTERISK-29444
Change-Id: I08adf2824b8bc63405778cf355963b5005612f41
Hitherto, the A option has made it possible to play
audio upon answer to the called party only. This option
is expanded to allow for playback of an audio file to
the caller instead of or in addition to the audio
played to the answerer.
ASTERISK-29442
Change-Id: If6eed3ff5c341dc8c588c8210987f2571e891e5e
Caller ID can now be set on the called channel and
Variables can now be set on the destination
using the Originate application, just as
they can be currently using call files
or the Manager Action.
ASTERISK-29450
Change-Id: Ia64cfe97d2792bcbf4775b3126cad662922a8b66
Adds a new ConfKick() application, which may
be used to kick a specific channel, all channels,
or all non-admin channels from a specified
conference bridge, similar to existing CLI and
AMI commands.
ASTERISK-29446
Change-Id: I5d96b683880bfdd27b2ab1c3f2e897c5046ded9b
A new user option, answer_channel, adds the capability to
prevent answering the channel if it hasn't already been
answered yet.
ASTERISK-29440
Change-Id: I26642729d0345f178c7b8045506605c8402de54b
Hitherto, VoiceMail() played a non-customizable beep tone to indicate
the caller could leave a message. In some cases, the beep may not
be desired, or a different tone may be desired.
To increase flexibility, a new option allows customization of the tone.
If the t option is specified, the default beep will be overridden.
Supplying an argument will cause it to use the specified file for the tone,
and omitting it will cause it to skip the beep altogether. If the option
is not used, the default behavior persists.
ASTERISK-29349
Change-Id: I1c439c0011497e28a28067fc1cf1e654c8843280
Also removed the sample documentation, and some oddly-placed
documentation about the timeout argument to the Queue() application
itself. There is a large section on the timeout behavior below.
ASTERISK-26614 #close
Change-Id: I8f84e8304b50305b7c4cba2d9787a5d77c3a6217
If a queue member was updated with the same status multiple
times each time a QueueMemberStatus event would be sent
which would be a duplicate of the previous.
This change makes it so that the QueueMemberStatus event is
only sent if the status actually changes.
ASTERISK-29355
Change-Id: I580c60d992a0a8f2bea8b91c868771b3b490d116
This change embeds the MODULEINFO block of modules
into the core XML documentation. This provides a shared
mechanism for use by both menuselect and Asterisk for
information and a definitive source of truth.
ASTERISK-29335
Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90
Some modules have a different support level documented in their
MODULEINFO XML and Asterisk module definition. This change
brings the two in sync for the modules which were not matching.
ASTERISK-29336
Change-Id: If2f819103d4a271e2e0624ef4db365e897fa3d35
Queue members using dialplan hints as a state interface must handle
INUSE+RINGING hint as RINGINUSE devstate, and INUSE + ONHOLD as INUSE.
ASTERISK-28369
Change-Id: I127e06943d4b4f1afc518f9e396de77449992b9f
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
launch_monitor_thread is responsible for creating and initializing
the mixmonitor, and dependent data structures. There was one off
nominal path after the datastore gets created that triggers when
the channel being monitored is hung up prior to monitor starting
itself.
If this happened the monitor thread would not "launch", and the
mixmonitor object and associated objects are freed, including the
underlying datastore data object. However, the datastore itself was
not removed from the channel, so when the channel eventually gets
destroyed it tries to access the previously freed datastore data
and crashes.
This patch removes and frees datastore object itself from the channel
before freeing the mixmonitor object thus ensuring the channel does
not call it when destroyed.
ASTERISK-28947 #close
Change-Id: Id4f9e958956d62473ed5ff06c98ae3436e839ff8
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
When using this option, answering the channel is deferred until
all prompts/greetings have been played and the caller is about
to leave their message.
ASTERISK-29118 #close
Change-Id: I41b9f0428783c0bd697c8c994f906d1e75ce9ddb
Operations that update queues when shared_lastcall is set lock the
queue in question, then have to lock the queues container to find the
other queues with the same member. On the other hand, __queues_show
(which is called by both the CLI and AMI) does the reverse. It locks
the queues container, then iterates over the queues locking each in
turn to display them. This creates a deadlock.
* Moved queue print logic from __queues_show to a separate function
that can be called for a single queue.
* Updated __queues_show so it doesn't need to lock or traverse
the queues container to show a single queue.
* Updated __queues_show to snap a copy of the queues container and iterate
over that instead of locking the queues container and iterating over
it while locked. This prevents us from having to hold both the
container lock and the queue locks at the same time. This also
allows us to sort the queue entries.
ASTERISK-29155
Change-Id: I78d4dc36728c2d7bc187b97d82673fc77f2bcf41
app_confbridge now has the ability to set the estimated bitrate on an
SFU bridge. To use it, set a bridge profile's remb_behavior to "force"
and set remb_estimated_bitrate to a rate in bits per second. The
remb_estimated_bitrate parameter is ignored if remb_behavior is something
other than "force".
Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a
This fixes a bug introduced mistakenly in ASTERISK-25665:
If leave-empty is enabled, a call may sometimes be removed from
a queue without recording it as abandoned.
This causes Asterisk to not generate an abandon event for that
call, and for the queue abandoned counter to be incorrect.
ASTERISK-29043 #close
Change-Id: I1a71b81df78adff59af587f1d8483cf57df430c7
Rather than putting messages into INBOX and then moving them to Urgent
later, put them directly in to the Urgent folder. This prevents
mailcmd from being skipped.
ASTERISK-27273 #close
Change-Id: I49934e093290d308506ab8d45a40ef705c5ae4f5
This fixes the reseting members lastpause problem when realtime members is being used,
the function rt_handle_member_record was forcing the reset members lastpause because it
does not exist in realtime
ASTERISK-29034 #close
Change-Id: Ic9107e4456732a1f78412a32adb2ef87f5da40b5
The SCOPE_ENTER and SCOPE_EXIT* macros now print debug messages
at the same level as the scope level. This allows the same
messages to be printed to the debug log when AST_DEVMODE
isn't enabled.
Also added a few variants of the SCOPE_EXIT macros that will
also call ast_log instead of ast_debug to make it easier to
use scope tracing and still print error messages.
Change-Id: I7fe55f7ec28069919a0fc0b11a82235ce904cc21
Allow passing a topology from the called channel back to the
calling channel.
* Added a new function ast_queue_answer() that accepts a stream
topology and queues an ANSWER CONTROL frame with it as the
data. This allows the called channel to indicate its resolved
topology.
* Added a new virtual function to the channel tech structure
answer_with_stream_topology() that allows the calling channel
to receive the called channel's topology. Added
ast_raw_answer_with_stream_topology() that invokes that virtual
function.
* Modified app_dial.c and features.c to grab the topology from the
ANSWER frame queued by the answering channel and send it to
the calling channel with ast_raw_answer_with_stream_topology().
* Modified frame.c to automatically cleanup the reference
to the topology on ANSWER frames.
Added a few debugging messages to stream.c.
Change-Id: I0115d2ed68d6bae0f87e85abcf16c771bdaf992c
If your queues.conf had _no_ [general] section, they would default to
'yes'. Now, they always default to 'no'.
(Actually, commit ed615afb7e already
partially fixed it for shared_lastcall.)
ASTERISK-28951
Change-Id: Ic39d8a0202906bc454194368bbfbae62990fe5f6
Prior to making any modifications to the pjsip infrastructure
for ACN, I've added the tracing functions to the existing code.
This should make the final commit easier to review, but we can also
now run a "before and after" trace.
No functional changes were made with this commit.
Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c
When stream support was added to Asterisk the stream state
was used inconsistently, resulting in odd behavior. This
was then standardized to be the state of a stream from the
perspective of Asterisk.
This change updates the StreamEcho dialplan application
to use the correct state, send only, since we are only
sending to the endpoint and not expecting them to send us
multiple video streams.
ASTERISK-28954
Change-Id: I35bfd533ef1184ffe62586b22bbd253c82872a56
Because ring_entry() is not called, outgoing->chan is not touched here
either.
ASTERISK-28950
ASTERISK-28644
Change-Id: I564613715dfaf45af868251eb75a451f512af90f
Before this changeset, it was possible that a queue member (agent) was
called even though they just got out of a call, and wrapuptime seconds
hadn't passed yet.
This could happen if a member ended a call _between_ a new call attempt
and asterisk trying that particular member for a new call.
In that case, Asterisk would check the hangup time of the
call-before-the-last-call instead of the hangup time of the-last-call.
ASTERISK-28952
Change-Id: Ie0cab8f0e8d639c01cba633d4968ba19873d80b3
When send_events is enabled for a user, we were leaking a reference
to the bridge channel in confbridge_manager.c:send_message(). This
also caused the bridge snapshot to not be destroyed.
Change-Id: I87a7ae9175e3cd29f6d6a8750e0ec5427bd98e97
This patch fixes a few compile warnings/errors that now occur when using gcc
10+.
Also, the Makefile.rules check to turn off partial inlining in gcc versions
greater or equal to 8.2.1 had a bug where it only it only checked against
versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures
any version above the specified version is correctly compared.
Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9
Ensure that output buffers for the osp_convert_inout
function have sufficient space for additional data
such as brackets and ports.
ASTERISK-28804
Change-Id: Ie54c8241ff0cc653910539c2db00ff2a4869750b
Add a new "masquarade" channel event, and use it in app_queue to track unique id's.
Testcase is submitted as https://gerrit.asterisk.org/c/testsuite/+/14210
ASTERISK-28829 #close
ASTERISK-25844 #close
Change-Id: Ifc5f9f9fd70903f3c6e49738d3bc632b085d2df6
The gcc 10 -Wrestrict option was causing "overlap" errors when
snprintf was copying one char[256] structure member to another
char[256] member in the same structure.
Using ast_alloca instead of declaring the structure inline
solves the issue.
Here's a link to the "enhancement":
https://gcc.gnu.org/legacy-ml/gcc-patches/2019-10/msg00570.html
We may follow up with a gcc bug report.
Change-Id: Ie0099adcb0a9727bd9aa99e024dd912a67eaf534
Since Asterisk 14, app_fax did not compile at all because Asterisk
requires that not malloc but ast_malloc is used everywhere. However,
the system headers of SpanDSP use malloc. Because we cannot (and do
not need to) change system headers, let us ignore this.
ASTERISK-28848
Change-Id: I31f7a6b92a07032c5cef1c16b8901b107fe35546
When in a conference bridge it may be necessary to have
text messages disabled for specific participants or for
all. This change adds a configuration option, "text_messaging",
which can be used to enable or disable this on the
user profile. By default existing behavior is preserved
as it defaults to "yes".
ASTERISK-28841
Change-Id: I30b5d9ae6f4803881d1ed9300590d405e392bc13
Given a scenario where MixMonitor was initiated over AMI it
was possible for the channel and MixMonitor thread to remain
alive past hang up of the channel. This scenario required
the AMI initiated MixMonitor to retrieve the channel, a
hangup to occur on the channel in another thread, and then
for MixMonitor to actually start. If this occurred the
MixMonitor thread would remain alive indefinitely and
the channel reference would remain.
This change ensures that audiohooks are never able to
be attached to channels that have been hung up. An
additional fix has also been done in app_mixmonitor to
properly release the channel reference if this occurs.
ASTERISK-28780
Change-Id: I8044c06daa06f0f16607788c596f55623be26f58
The optional synchronization behavior created in
64906c4c9b is now the default for
MixMonitor.
* Add a new flag 'n' that allows for this behavior to be turned off
* Add a notice when the 'S' option is used indicating that it is no
longer necessary
Change-Id: I158987c475cda4e1ff1256dd0daccdd99df568b4
When opening a file for writing, Asterisk silently converts filenames
ending with 'wav49' to 'WAV.' We aren't taking that in to account when
setting the MIXMONITOR_FILENAME variable in MixMonitor.
* If the user wants to write to a wav49 file, make sure that it is
reflected properly in MIXMONITOR_FILENAME.
* Add a note to the documentation describing this behavior.
* Add a note in main/file.c indicating that app_mixmonitor needs to be
changed if the logic in build_filename was changed.
ASTERISK-24798 #close
Reported by: xrobau
Change-Id: I384691ce624eb55c80a125b9ca206d2d691c574c
* The MailboxExists dialplan application was deprecated on 2006-09-26
in Asterisk 1.6.0 (commit ec83b11183)
* The MAILBOX_EXISTS dialplan function was deprecated on 2011-12-06 in
Asterisk 11.0.0 (commit fd64bb66f9)
Change-Id: I71cfc9d7b9217a37b802f4cc6ef2d57900b7398f
In af90afd90c, Japanese language support
was added to app_voicemail and main/say.c, but the leading whitespace
is not consistent with Asterisk coding guidelines. This patch fixes
that.
Whitespace only, no functional change.
ASTERISK~23324
Reported by: Kevin McCoy
Change-Id: I72c725f5930084673749bd7c9cc426a987f08e87
ast_store_realtime() is not NULL tolerant, so we need to initialize
the field values we pass to it to the empty string to avoid a crash.
ASTERISK-23739 #close
Reported by: Stas Kobzar
Change-Id: I756c5dd0299c77f4274368f7c99eb0464367466c
If voicemail.conf exists but is empty, the config parsing process will
default a number of global variables to non-zero values. On the other
hand, if voicemail.conf is missing (arguably semantically equivalent
to an empty file), this process is skipped and the globals are
defaulted to 0.
Set the globals to the same values they would be set to if a
configuration were present. This allows voicemail configuration to be
done completely by Realtime without the need to create an empty
voicemail.conf file.
ASTERISK-27622 #close
Reported by: Jim Van Meggelen
Change-Id: Id907d280f310f12e542ca527e6a025432b9fb409
This commit adds support for
[AudioSocket](
https://wiki.asterisk.org/wiki/display/AST/AudioSocket),
a very simple bidirectional audio streaming protocol. There are both
channel and application interfaces.
A description of the protocol can be found on the above referenced
GitHub page. A short talk about the reasons and implementation can be
found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from
CommCon 2019.
ARI support has also been added via the existing "externalMedia" ARI
functionality. The UUID is specified using the arbitrary "data" field.
ASTERISK-28484 #close
Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5
The QueueMemberPause AMI event includes two fields that return the
reason a member was paused.
* In release branches, deprecate Reason in favor of PausedReason.
* In master, remove the Reason field entirely.
ASTERISK-28349 #close
Reported by: Niksa Baldun
Change-Id: I01da58f2b0ab927baeee754870f62b51b7b3d296
Additionally alter the warning to mention that it was "beep" which could
not be streamed to give admins a better clue about what the warning
means.
ASTERISK-28682
Change-Id: If5aed21226a173117ed17589f44826dd1ba6576e
This patch fixes some wrongly formatted documentation for the AgentLogin
application. A couple of "see also" links should contain only the function
name, and no parameters.
Change-Id: I3f788b47dce3292e311f8a9856938d59a0bd0661
Dialplan has to be careful about passing an empty device list or empty
positions in the list. As a result, dialplan has to check for these
conditions before using ChanIsAvail. Simplify dialplan by making
ChanIsAvail handle these conditions gracefully.
* Made tolerate empty positions in the device list.
* Simplified the code and eliminated some unnecessary indention.
ASTERISK-28638
Change-Id: I9e4b67e2cbf26b2417c2d03485b8568e898931d3
* Made BridgeAdd not hangup the call if there is a problem.
* Reduced message level from warning to verbose for normal exception
cases.
* Added a loop safety check to BridgeAdd.
* Made BridgeAdd set BRIDGERESULT with the status when dialplan is
resumed.
Change-Id: I374d39b8a3edcc794eeb5c6b9f31a01424cdc426
Dialplan has to be careful about passing an empty destination list or
empty positions in the list. As a result, dialplan has to check for
these conditions before using Dial. Simplify dialplan by making Dial
handle these conditions gracefully.
* Made tolerate empty positions in the dialed device list.
* Reduced some message log levels from notice to verbose.
ASTERISK-28638
Change-Id: I6edc731aff451f8bdfaee5498078dd18c3a11ab9
Dialplan has to be careful about passing an empty destination list or
empty positions in the list. As a result, dialplan has to check for
these conditions before using Page. Simplify dialplan by making Page
handle these conditions gracefully.
* Made tolerate empty positions in the paged device list.
* Reduced some warnings associated with the 's' option to verbose
messages. The warning level for those messages really serves no purpose
as that is why the 's' option exists.
ASTERISK-28638
Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3
Temporary channel lifespan is very short and CDR deactivation request
through ast_cdr_set_property() may happen when CDR is not available
yet. Use CDR_PROP() dialplan function instead, it will first wait
for pending CDR insertion requests to be processed.
ASTERISK-28636
Change-Id: I1cbe09e8d2169c0962c1195133ff260d291f2074
ConfBridge has the ability to move between different sample
rates for mixing the conference bridge. Up until now there has
only been the ability to set the conference bridge to mix at
a specific sample rate, or to let it move between sample rates
as necessary. This change adds the ability to configure a
conference bridge with a maximum sample rate so it can move
between sample rates but only up to the configured maximum.
ASTERISK-28658
Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee
ASTERISK_28143 attempted to fix an issue where calls with no audio would never
timeout. It did so by adding AST_FRAME_NULL as a frame type to process in its
calculations. Unfortunately these frames seem to show up at irregular time
intervals. This resulted in app_amd returning prematurely most of the time.
* Removed AST_FRAME_NULL from the calculations
* Added a check to see how much time has actually passed since app_amd began
ASTERISK-28608
Change-Id: I642a21b02d389b17e40ccd5357754b034c3daa42
This patch fixes several issues reported by the lgtm code analysis tool:
https://lgtm.com/projects/g/asterisk/asterisk
Not all reported issues were addressed in this patch. This patch mostly fixes
confirmed reported errors, potential problematic code points, and a few other
"low hanging" warnings or recommendations found in core supported modules.
These include, but are not limited to the following:
* innapropriate stack allocation in loops
* buffer overflows
* variable declaration "hiding" another variable declaration
* comparisons results that are always the same
* ambiguously signed bit-field members
* missing header guards
Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25
The following modules needed tweaks for API changes.
addons/cdr_mysql.c
addons/chan_ooh323.c
apps/app_meetme.c
ASTERISK-28604
Change-Id: Ib40e513ae55b5114be035cdc929abb6a8ce2d06d
If you specify multiple formats in voicemail.conf, eg. "format = gsm|wav"
and are using realtime ODBC backend, only the first format gets stored
in the database. So when you forward a message later on, there is a bug
generating the email, related to the stored format (GSM) being different
than the desired email format (WAV) specified for the user. Sox can
handle this, but Asterisk needs to tell sox exactly what to do.
ASTERISK-22192
Change-Id: I7321e7f7e7c58adbf41dd4fd7191c887b9b2eafd
This reverts commit fd2e8d0da7.
Reason for revert: Problematic for users who store their voicemail
on network storage devices, or share voicemail storage between
multiple Asterisk instances.
ASTERISK-28567 #close
Change-Id: I3ff4ca983d8e753fe2971f3439bd154705693c41
ChanIsAvail() creates a temporary channel with ast_request() to test
resource availability. It should not generate a CDR when it hangs up
this temporary channel.
This patch disables CDR generation for the temporary channel with
ast_cdr_set_property().
ASTERISK-28527
Change-Id: I7b0555c6909c7d322e452dde97c9ea5b111552d1
There are 4 scenarios to consider when capturing audio from a channel
with an audiohook:
1. There is no rx and no tx audio, so return nothing.
2. There is rx but no tx audio, so return rx.
3. There is tx but no rx audio, so return tx.
4. There is rx and tx audio, so mix them and return.
The file passed as the primary argument to MixMonitor will be written to
in scenarios 2, 3, and 4. However, if you pass the r() and t() options
to MixMonitor, a frame will only be written to the r() file if there was
rx audio and a frame will only be written to the t() file if there was
tx audio.
If you subsequently take the r() and t() files and try to mix them, the
sides of the conversation will 'drift' and be non-representative of the
user experience.
This patch adds a new 'S' option to MixMonitor that injects a frame of
silence on either the r() side or the t() side of the channel so that
when later mixed, there is no such drift.
Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e
The function leave_voicemail checks if expungeonhangup is set,
but does not check if IMAP stream is closed,
so it could call imap function with NULL stream.
This leads to segfault.
ASTERISK-28505 #close
Change-Id: Ib66c57c1f1ba97774e447b36349198e2626a8d7c
Maximilian Fridrich
Naveen Albert
Kfir Itzhak
Alexei Gradinari
Mark Petersen
Sean Bright
Steve Davies
Alexander Traud
Josh Soref
Rodrigo Ramírez Norambuena
Shloime Rosenblum
Carlos Oliva
Joshua C. Colp
Naveen Albert
Ivan Poddubnyi
Sebastien Duthil
Dan Cropp
Kevin Harwell
George Joseph
Evandro César Arruda
Walter Doekes
Kevin Harwell
Nathan Bruning
Jaco Kroon
Friendly Automation
Seán C McCord
Corey Farrell
Richard Mudgett
Frederic LE FOLL
Michael Cargile
lvl
cmaj