The Dial, Queue, and FollowMe applications need to inhibit the bridging
initial connected line exchange in order to support the 'I' option.
* Replaced the pass_reference flag on ast_bridge_join() with a flags
parameter to pass other flags defined by enum ast_bridge_join_flags.
* Replaced the independent flag on ast_bridge_impart() with a flags
parameter to pass other flags defined by enum ast_bridge_impart_flags.
* Since the Dial, Queue, and FollowMe applications are now the only
callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the
calling contract to require the initial COLP exchange to already have been
done by the caller.
* Made all callers of ast_bridge_impart() check the return value. It is
important. As a precaution, I also made the compiler complain now if it
is not checked.
* Did some cleanup in parking_tests.c as a result of checking the
ast_bridge_impart() return value.
An independent, but associated change is:
* Reduce stack usage in ast_indicate_data() and add a dropping redundant
connected line verbose message.
(closes issue ASTERISK-22072)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2845/
........
Merged revisions 399136 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
DTMF start/end and hold/unhold events have state because a DTMF begin
event and hold event must be ended by something.
The following cases need to be handled when a channel is moved around in
the system.
* When a channel leaves a bridge it may owe a DTMF end event to the
bridge.
* When a channel leaves a bridge it may owe an UNHOLD event to the bridge.
(This case is explicitly ignored because things like transfers need
explicit control over this.)
* When a channel leaves the bridging system it may need to simulate a DTMF
end event to the channel.
* When a channel leaves the bridging system it may need to simulate an
UNHOLD event to the channel.
The patch also fixes the following:
* Fixes playing a file and restarting MOH using the latest MOH class used.
(closes issue ASTERISK-22043)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2791/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds pass through support for Opus and VP8. That includes:
* Format attribute negotiation for Opus. Note that unlike some other codecs,
the draft RFC specifies having spaces delimiting the attributes in addition
to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in
chan_sip, so a small tweak was also included in this patch for that.
* A format attribute negotiation module for Opus, res_format_attr_opus
* Fast picture update for VP8. Since VP8 uses a different RTCP packet number
than FIR, this really is specific to VP8 at this time.
Note that the format attribute negotiation in res_pjsip_sdp_rtp was written
by mjordan. The rest of this patch was written completely by Lorenzo Miniero.
Review: https://reviewboard.asterisk.org/r/2723/
(closes issue ASTERISK-21981)
Reported by: Tzafrir Cohen
patches:
asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a new dialplan application, SayAlphaCase, that performs much
the same function as SayAlpha except that it takes additional options
which allow the user to specify whether the case of each letter should
be announced for uppercase, lowercase, or all letters. Similar
functionality has been added to the SAY ALPHA AGI command via an
optional parameter.
Original Patch by: Kevin Scott Adams
Reported by: Kevin Scott Adams
Review: https://reviewboard.asterisk.org/r/2725/
(closes issue ASTERISK-20782)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does three things:
1. It provides a Surrogate channel technology with a consolidated
"implementation detail flag" on the channel technology. This tells
consumers of Stasis that the creation of this channel is an implementation
detail in Asterisk and can be ignored (if they so choose). This
consolidates the conference recorder/announcer flags as well - these flags
had no additional meaning beyond "ignore this channel please".
2. It modifies allocation of a channel in two ways:
(a) If a channel technology can be determined from the name, we set it
directly in the allocation routine. This prevents the initial
publication of the message from going out with a NULL channel technology
where possible. This lets Stasis consumers get the right channel
technology on the first publication.
(b) It reorganizes allocation to make use of the 'finalized' property on the
channel. This was already used to know that a channel had completely
finished its construction in the masquerade routine; now we also use it
to know whether or not the setting of certain channel properties is
occurring during or post construction. The various set routines were
modified accordingly as well.
3. The masquerade event is now dead, Jim. It no longer served any purpose
whatsoever - if you perform a call pickup you'll get a Pickup event;
if you perform an attended transfer you will still get those events; if you
steal a channel to put it elsewhere you'll get the corresponding NewExten or
BridgeEnter events.
Review: https://reviewboard.asterisk.org/r/2740
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch implements the controls from ARI recordings. The controls
are:
* DELETE /recordings/live/{recordingName} - stop recording and
discard it
* POST /recordings/live/{recordingName}/stop - stop recording
* POST /recordings/live/{recordingName}/pause - pause recording
* POST /recordings/live/{recordingName}/unpause - resume recording
* POST /recordings/live/{recordingName}/mute - mute recording (record
silence to the file)
* POST /recordings/live/{recordingName}/unmute - unmute recording.
Since this underlying functionality did not already exist, is was
added to app.c by a set of control frames, similar to how playback
control works. The pause/mute control frames are toggles, even though
the ARI controls are idempotent, to be consistent with the playback
control frames.
(closes issue ASTERISK-22181)
Review: https://reviewboard.asterisk.org/r/2697/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new res_ari_asterisk.so module presents several config options
from asterisk main. Unfortunately, they aren't exported, so the module
won't load on Linux.
This patch renames the variables, adding the ast_ prefix so they will
be exported.
Review: https://reviewboard.asterisk.org/r/2737
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is smaller than the initial review placed on review board. This is because
a change to allow for channel drivers to access parking functionality externally was
committed and invalidated quite a few of the changes initially made.
(closes issue ASTERISK-22039)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2717
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk naming conventions:
* channel is not "channeling"
* monitor is not "monitoring"
etc.
A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is
the act of using a bridge on a set of channels - and the API that fulfills that
role is more than just the action.
(closes issue ASTERISK-22130)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It pulls out bridge_channel and puts it into its own translation unit
* It adds public and protected headers for bridging_channel. Protected
functions are appropriate only for the Bridging API and sub-classes of a
bridge.
(issue ASTERISK-22130)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In previous versions of Asterisk, the zombies roamed freely,
unchecked and uncontrolled. They ravaged Asterisk systems with
their biting and their nashing and their pointy teeth.
Sometimes, you couldn't even hang them up.
Now, zombies are rare. They still *technically* exist in certain
places, but they are controlled. Kind of like a zombie zoo: you can
see them, but you can't touch them, and they can't touch you.
Bring your kids!
Because zombies are now population controlled with a very short lifespan,
there's no reason to rename the channels to '%s<ZOMBIE>'. The channels
are guaranteed to die off quickly; the rename really is just confusing
at this point.
This patch finally removes the renaming. On the plus side: this made
my life easier in CDRs during call pickup and attended transfers to
an Asterisk application. It will make other folks lives easier as well!
Review: https://reviewboard.astierks.org/r/2690/
(closes issue ASTERISK-21699)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds CEL support for blind and attended transfers and call pickup.
During the course of adding this functionality I noticed that
CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly
useless without a bridge identifier, so I added that as well.
This adds tests for blind transfers, several types of attended
transfers, and call pickup.
The extra field in CEL records now consists of a JSON blob whose fields
are defined on a per-event basis.
Review: https://reviewboard.asterisk.org/r/2658/
(closes issue ASTERISK-21565)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made ast_audiohook_detach_list() and ast_audiohook_write_list_empty()
NULL tolerant.
* Made ast_audiohook_detach_list() return void since it is a destructor.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since ast_hangup() is effectively a channel destructor, it should be a
void function.
* Make the few silly callers checking the return value no longer do so.
Only the CDR and CEL unit tests checked the return value.
* Make all callers take advantage of the NULL safe change and remove the
NULL check before the call.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch reorders certain actions that may raise Stasis messages in the
channel destructor such that they occur before the Stasis cache is cleared.
Once the Stasis cache is cleared, its rather a bad idea to be trying to
publish information about a channel.
(closes issue ASTERISK-22001)
Reported by: Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* It adds a new soft hangup flag AST_SOFTHANGUP_HANGUP_EXEC that is set when a
channel is executing dialplan hangup logic, i.e., the 'h' extension or a
hangup handler. Stasis messages now also convey the soft hangup flag so
consumers of the messages can know when a channel is executing said
hangup logic.
* It adds a new channel flag, AST_FLAG_DEAD, which is set when a channel is
well and truly dead. Not just a zombie, but dead, Jim. Manager, CEL, CDRs,
and other consumers of Stasis have been updated to look for this flag to
know when the channel should by lying six feet under.
* The CDR engine has been updated to better handle a channel entering and
leaving a bridge. Previously, a new CDR was automatically created when a
channel left a bridge and put into the 'Pending' state; however, this
way of handling CDRs made it difficult for the 'endbeforehexten' logic to
work correctly - there was always a new CDR waiting in the hangup logic
and, even if 'ended', wouldn't be the CDR people wanted to inspect in the
hangup routine. This patch completely removes the Pending state and instead
defers creation of the new CDR until it gets a new message that requires
a new CDR.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This process also involved a large amount of rework regarding how to redial
the Parker when a channel leaves a parking lot due to timeout. An attended
transfer channel variable has been added to attended transfers to extensions
that will eventually park (but haven't at the time of transfer) as well.
This resolves one of the two BUGBUG comments remaining in res_parking.
(issues ASTERISK-21877)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2638/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is the first step in adding recording support to the
Asterisk REST Interface.
Recordings are stored in /var/spool/recording. Since recordings may be
destructive (overwriting existing files), the API rejects attempts to
escape the recording directory (avoiding issues if someone attempts to
record to ../../lib/sounds/greeting, for example).
(closes issue ASTERISK-21594)
(closes issue ASTERISK-21581)
Review: https://reviewboard.asterisk.org/r/2612/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is the initial push to update Asterisk's CDR engine for the new
bridging framework. This patch guts the existing CDR engine and builds the new
on top of messages coming across Stasis. As changes in channel state and bridge
state are detected, CDRs are built and dispatched accordingly. This
fundamentally changes CDRs in a few ways.
(1) CDRs are now *very* reflective of the actual state of channels and bridges.
This means CDRs track well with what an actual channel is doing - which
is useful in transfer scenarios (which were previously difficult to pin
down). It does, however, mean that CDRs cannot be 'fooled'. Previous
behavior in Asterisk allowed for CDR applications, channels, and other
properties to be spoofed in parts of the code - this no longer works.
(2) CDRs have defined behavior in multi-party scenarios. This behavior will not
be what everyone wants, but it is a defined behavior and as such, it is
predictable.
(3) The CDR manipulation functions and applications have been overhauled. Major
changes have been made to ResetCDR and ForkCDR in particular. Many of the
options for these two applications no longer made any sense with the new
framework and the (slightly) more immutable nature of CDRs.
There are a plethora of other changes. For a full description of CDR behavior,
see the CDR specification on the Asterisk wiki.
(closes issue ASTERISK-21196)
Review: https://reviewboard.asterisk.org/r/2486/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This uses the channel state change events from Stasis-Core to determine
when channel-related CEL events should be raised. Those refactored in
this patch are:
* AST_CEL_CHANNEL_START
* AST_CEL_ANSWER
* AST_CEL_APP_START
* AST_CEL_APP_END
* AST_CEL_HANGUP
* AST_CEL_CHANNEL_END
Retirement of Linked IDs is also refactored.
CEL configuration has been refactored to use the config framework.
Note: Some HANGUP events are not generated correctly because the bridge
layer does not propagate hangupcause/hangupsource information yet.
Review: https://reviewboard.asterisk.org/r/2544/
(closes issue ASTERISK-21563)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Stasis cache clear message payloads now consist of a stasis_message
representative of the message to be cleared from the cache. This allows
multiple parallel caches to coexist and be cleared properly by the same
cache clear message even when keyed on different fields.
This change fixes a bug where multiple cache clears could be posted for
channels. The cache clear is now produced in the destructor instead of
ast_hangup.
Additionally, dummy channels are no longer capable of producing channel
snapshots.
Review: https://reviewboard.asterisk.org/r/2596
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* DTMF attended and blind transfers have hold/unhold behavior restored.
* External attended and blind transfers unhold the transfered party when
the transfer is initiated.
* Made prohibit blind transferring a bridge marked as masquerade only.
(ConfBridge bridges)
* Made running an application or playing a file inside a bridge post the
hold/unhold messages if MOH is requested.
Review: https://reviewboard.asterisk.org/r/2574/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses issues during immediate shutdowns, where modules
are not unloaded, but Asterisk atexit handlers are run.
In the typical case, this usually isn't a big deal. But the
introduction of the Stasis message bus makes it much more likely for
asynchronous activity to be happening off in some thread during
shutdown.
During an immediate shutdown, Asterisk skips unloading modules. But
while it is processing the atexit handlers, there is a window of time
where some of the core message types have been cleaned up, but the
message bus is still running. Specifically, it's still running
module subscriptions that might be using the core message types. If a
message is received by that subscription in that window, it will
attempt to use a message type that has been cleaned up.
To solve this problem, this patch introduces ast_register_cleanup().
This function operates identically to ast_register_atexit(), except
that cleanup calls are not invoked on an immediate shutdown. All of
the core message type and topic cleanup was moved from atexit handlers
to cleanup handlers.
This ensures that core type and topic cleanup only happens if the
modules that used them are first unloaded.
This patch also changes the ast_assert() when accessing a cleaned up
or uninitialized message type to an error log message. Message type
functions are actually NULL safe across the board, so the assert was a
bit heavy handed. Especially for anyone with DO_CRASH enabled.
Review: https://reviewboard.asterisk.org/r/2562/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Check the returned bridged pointer for NULL to avoid a crash. It looks
like chan_agent is returning a NULL pointer when it probably should be
returning a pointer to the channel the Agent channel is pretending to be.
(closes issue ASTERISK-21793)
Reported by: Rodrigo P. Telles
Patches:
jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Rodrigo P. Telles
........
Merged revisions 390044 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 390047 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The attended transfer API call can complete the attended transfer in a number of ways
depending on the current bridged states of the channels involved.
The hiding of masquerades is done in some bridging-related functions, such as the manager
Bridge action and the Bridge dialplan application. In addition, call pickup was edited
to "move" a channel rather than masquerade it.
Review: https://reviewboard.asterisk.org/r/2511
(closes issue ASTERISK-21334)
Reported by Matt Jordan
(closes issue Asterisk-21336)
Reported by Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Breaks many things until they can be reworked. A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does two things:
* It fixes a bug where the outbound channel's application/data set by the
dialing API/app_dial is not communicated until the channel is hung up.
If that happens, AMI would incorrectly send a NewExten event immediately
after a Hangup. This isn't really AMI's fault, as the dialing APIs never
communicated the 'helpful' app/data on the outbound channel until it was
hungup.
* It makes public sending a stasis message about a change in channel state.
This is useful enough that - for now at least - it should be public. If
operations on a channel go to being more coarse-grained, this function
could be made private again.
Review: https://reviewboard.asterisk.org/r/2548
Note that this problem was found and reported by Matt DiMeo.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When we first introduced the channel blob types, the JSON blobs were
self identifying by a required "type" field in the JSON object
itself. This, as it turns out, was a bad idea.
When we introduced the message router, it was useless for routing based
on the JSON type. And messages had two type fields to check: the
stasis_message_type() of the message itself, plus the type field in the
JSON blob (but only if it was a blob message).
This patch corrects that mistake by removing the required type field
from JSON blobs, and introducing first class stasis_message_type objects
for the actual message type.
Since we now will have a proliferation of message types, I introduced a
few macros to help reduce the amount of boilerplate necessary to set
them up.
Review: https://reviewboard.asterisk.org/r/2509
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The two party bridging loops were changing the bridge peer pointers
without the channel locks held. Thus when ast_channel_massquerade()
tested and used the pointer there is a small window of opportunity for the
pointers to become NULL even though the masquerade code has the channels
locked.
(closes issue ASTERISK-21356)
Reported by: William luke
Patches:
jira_asterisk_21356_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: William luke
........
Merged revisions 386256 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 386286 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
refactored to use these events rather than producing the events directly
in channel.c. Finally, the code was added to app_stasis to produce
DTMF events on the WebSocket.
The AMI events are completely backward compatible, including sending
events on transmitted DTMF, and sending DTMF start events.
The Stasis-HTTP events are somewhat simplified. Since DTMF start and
DTMF send events are generally less useful, Stasis-HTTP will only send
events on received DTMF end.
(closes issue ASTERISK-21282)
(closes issue ASTERISK-21359)
Review: https://reviewboard.asterisk.org/r/2439
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch does the following:
* A new Stasis payload has been defined for multi-channel messages. This
payload can store multiple ast_channel_snapshot objects along with a single
JSON blob. The payload object itself is opaque; the snapshots are stored
in a container keyed by roles. APIs have been provided to query for and
retrieve the snapshots from the payload object.
* The Dial AMI events have been refactored onto Stasis. This includes dial
messages in app_dial, as well as the core dialing framework. The AMI events
have been modified to send out a DialBegin/DialEnd events, as opposed to
the subevent type that was previously used.
* Stasis messages, types, and other objects related to channels have been
placed in their own file, stasis_channels. Unit tests for some of these
objects/messages have also been written.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
HangupRequest and SoftHangupRequest are now ast_channel_blob Stasis
messages, with the cause code as an optional field in the blob.
NewCallerid now simply watches for changes in the callerid information
in channel snapshots, and creates the AMI event appropriately.
Since the original NewCallerid event honored the channelvars setting
in manager.conf, the channel variables configured there had to become
a part of the channel snapshot. These are now a part of every snapshot
based event, making the configuration description "every time a
channel-oriented event is emitted" less of a lie.
There a a few other changes wrapped up in here as well.
* When ast_channel_topic() is given NULL for a channel, it returns
the ast_channel_topic_all() topic instead of NULL. This can clean
up a lot of NULL checking we're doing currently.
* The fields Cause and Cause-txt were removed from the base channel
information and put only on the Hangup events, since those fields
are meaningless outside of a Hangup event.
* Removed the pipe-delimiter processing of the channelvars field,
since that's been deprecated forever.
(closes issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2405/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch started out simply as fixing the bouncing tests introduced
in r382685, but required some other changes to give it a decent
implementation.
To fix the bouncing tests, the UserEvent and Newexten AMI events
needed to be refactored to dispatch via Stasis. Dispatching directly
to AMI resulted in those events sometimes getting ahead of the
associated Newchannel events, which would understandably confuse anyone.
I found that instead of creating a zillion different message types and
structures associated with them, it would be preferable to define a
message type that has a channel snapshot and a blob of structured data
with a small bit of additional information. The JSON object model
provides a very nice way of representing structured data, so I went
with that.
* Move JSON support from res_json.c to main/json.c
* Made libjansson-dev a required dependency
* Added an ast_channel_blob message type, which has a channel
snapshot and JSON blob of data.
* Changed UserEvent and Newexten events so that they are dispatched
via ast_channel_blob messages on the channel's topic.
* Got rid of the ast_channel_varset message; used ast_channel_blob
instead.
* Extracted the manager functions converting Stasis channel events to
AMI events into manager_channel.c.
(issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2381/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Remove MWI's dependency on the event system by moving it to
Stasis-core. This also introduces forwarding topic pools in Stasis-core
which aggregate many dynamically allocated topics into a single primary
topic.
Review: https://reviewboard.asterisk.org/r/2368/
(closes issue ASTERISK-21097)
Patch-by: Kinsey Moore
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For the initial use of this bus, I took some work kmoore did creating
channel snapshots. So rather than create AMI events directly in the
channel code, this patch generates Stasis events, which manager.c uses
to then publish the AMI event.
This message bus provides a generic publish/subscribe mechanism within
Asterisk. This message bus is:
- Loosely coupled; new message types can be added in seperate modules.
- Easy to use; publishing and subscribing are straightforward
operations.
In addition to basic publish/subscribe, the patch also provides
mechanisms for message forwarding, and for message caching.
(issue ASTERISK-20887)
(closes issue ASTERISK-20959)
Review: https://reviewboard.asterisk.org/r/2339/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the capability for asynchronous manipulation of audio being
played back to a channel though a new AMI action "ControlPlayback". The
ControlPlayback action supports a number of operations, the availability of
which depend on the application being used to send audio to the channel.
When the audio playback was initiated using the ControlPlayback application
or CONTROL STREAM FILE AGI command, the audio can be paused, stopped,
restarted, reversed, or skipped forward. When initiated by other mechanisms
(such as the Playback application), the audio can be stopped, reversed, or
skipped forward.
Review: https://reviewboard.asterisk.org/r/2265/
(closes issue ASTERISK-20882)
Reported by: mjordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Masquerades are an insane implementation detail within Asterisk. It generates
a number of useless and confusing events, and manipulates channels in a way
that semantically doesn't make sense. I've given a fairly thorough review of
masquerade code and its usage on the wiki at
https://wiki.asterisk.org/wiki/x/IwBRAQ.
While ultimately it makes the most sense to abandon masquerades altogether,
it will take some time to completely irradicate. Even then, there may always
be code that's not worth rewriting to get rid of the masquerade.
This patch does two things to make masquerades slightly less insane:
* When swapping the names of the original and clone channel, only emit a
single rename event of original -> original<ZOMBIE>. The original code
issued three rename events to accomplish the same end.
* In addition to swapping the names of the channels, also swap their
uniqueid's. This allows the 'Uniqueid' field to be used as a stable
identifier for a channel from and external interface, such as AMI.
Review: https://reviewboard.asterisk.org/r/2266/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix an unbalanced manager_bridge_event(unlink) call if
AST_SOFTHANGUP_UNBRIDGE is set in ast_channel_bridge().
* Make ast_channel_bridge() use common cleanup code when leaving the
bridge.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Squeezed some redundancy out of update_bridge_vars().
* Wrapped long line in __ast_change_name_nolink().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.
This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
event-cachability-3.diff uploaded by jcolp (license 5000)
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* Add red-black tree container type.
* Add CLI command "astobj2 container dump <name>"
* Added ao2_container_dump() so the container could be dumped by other
modules for debugging purposes.
* Changed ao2_container_stats() so it can be used by other modules like
ao2_container_check() for debugging purposes.
* Updated the unit tests to check red-black tree containers.
(closes issue ASTERISK-19970)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2110/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generator.
This patch introduces an internal helper function to safely check whether the
current generator is the one that is expected before deactivating it. The
current externally accessible ast_channel_stop_generator() function has been
modified to be implemented in terms of the new function.
(closes issue ASTERISK-19918)
Reported by: Eduardo Abad
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r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines
Fix misuses of timeouts throughout the code.
Prior to this change, a common method for determining if a timeout
was reached was to call a function such as ast_waitfor_n() and inspect
the out parameter that told how many milliseconds were left, then use
that as the input to ast_waitfor_n() on the next go-around.
The problem with this is that in some cases, submillisecond timeouts
can occur, resulting in the out parameter not decreasing any. When this
happens thousands of times, the result is that the timeout takes much
longer than intended to be reached. As an example, I had a situation where
a 3 second timeout took multiple days to finally end since most wakeups
from ast_waitfor_n() were under a millisecond.
This patch seeks to fix this pattern throughout the code. Now we log the
time when an operation began and find the difference in wall clock time
between now and when the event started. This means that sub-millisecond timeouts
now cannot play havoc when trying to determine if something has timed out.
Part of this fix also includes changing the function ast_waitfor() so that it
is possible for it to return less than zero when a negative timeout is given
to it. This makes it actually possible to detect errors in ast_waitfor() when
there is no timeout.
(closes issue ASTERISK-20414)
reported by David M. Lee
Review: https://reviewboard.asterisk.org/r/2135/
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r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines
Remove some debugging that accidentally made it in the last commit.
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Currently, if an acknowledgement of a timer fails Asterisk will not realize
that a serious error occurred and will continue attempting to use the timer's
file descriptor. This can lead to situations where errors stream to the
CLI/log file. This consumes significant resources, masks the actual problem
that occurred (whatever caused the timer to fail in the first place), and
can leave channels in odd states.
This patch propagates the errors in the timing resource modules up through
the timer core, and makes users of these timers handle acknowledgement
failures. It also adds some defensive coding around the use of timers
to prevent using bad file descriptors in off nominal code paths.
Note that the patch created by the issue reporter was modified slightly for
this commit and backported to 1.8, as it was originally written for
Asterisk 10.
Review: https://reviewboard.asterisk.org/r/2178/
(issue ASTERISK-20032)
Reported by: Jeremiah Gowdy
patches:
jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358)
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This allows for the REDIRECTING dialplan function to be
used to set the reason to any string.
The SIP channel driver has been modified to set the redirecting
reason string to the value received in a Diversion header. In
addition, SIP 480 response reason text will set the redirecting
reason as well.
(closes issue AST-942)
reported by Malcolm Davenport
(closes issue AST-943)
reported by Malcolm Davenport
Review: https://reviewboard.asterisk.org/r/2101
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* ASTERISK-20383
Missing named call pickup group features:
CHANNEL(callgroup) - Need CHANNEL(namedcallgroup)
CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup)
Pickup() - Needs to also select from named pickup groups.
* ASTERISK-20384
Using the pickupexten, the pickup channel selection could fail even though
there was a call it could have picked up. In a call pickup race when
there are multiple calls to pickup and two extensions try to pickup a
call, it is conceivable that the loser will not pick up any call even
though it could have picked up the next oldest matching call.
Regression because of the named call pickup group feature.
* See ASTERISK-20386 for the implementation improvements. These are the
changes in channel.c and channel.h.
* Fixed some locking issues in CHANNEL().
(closes issue ASTERISK-20383)
Reported by: rmudgett
(closes issue ASTERISK-20384)
Reported by: rmudgett
(closes issue ASTERISK-20386)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2112/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new API allows for sorted containers, insertion options, duplicate
handling options, and traversal order options.
* Adds the ability for containers to be sorted when they are created.
* Adds container creation options to handle duplicates when they are
inserted.
* Adds container creation option to insert objects at the beginning or end
of the container traversal order.
* Adds OBJ_PARTIAL_KEY to allow searching with a partial key. The partial
key works similarly to the OBJ_KEY flag. (The real search speed
improvement with this flag will come when red-black trees are added.)
* Adds container traversal and iteration order options: Ascending and
Descending.
* Adds an AST_DEVMODE compile feature to check the stats and integrity of
registered containers using the CLI "astobj2 container stats <name>" and
"astobj2 container check <name>". The channels container is normally
registered since it is one of the most important containers in the system.
* Adds ao2_iterator_restart() to allow iteration to be restarted from the
beginning.
* Changes the generic container object to have a v_method table pointer to
support other types of containers.
* Changes the container nodes holding objects to be ref counted.
The ref counted nodes and v_method table pointer changes pave the way to
allow other types of containers.
* Includes a large astobj2 unit test enhancement that tests the new
features.
(closes issue ASTERISK-19969)
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/2078/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes numerous doxygen warnings across Asterisk. It also updates
the makefile to regenerate the doxygen configuration on the local system
before running doxygen to help prevent warnings/errors on the local system.
Much thanks to Andrew for tackling one of the Asterisk janitor projects!
(issue ASTERISK-20259)
Reported by: Andrew Latham
Patches:
doxygen_partial.diff uploaded by Andrew Latham (license 5985)
make_progdocs.diff uploaded by Andrew Latham (license 5985)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the feature "Private representation of caller, connected
and redirecting party ids", as previously discussed with us (DATUS) and
Digium.
1. Feature motivation
Until now it is quite difficult to modify a party number or name which can
only be seen by exactly one particular instantiated technology channel
subscriber. One example where a modified party number or name on one
channel is spread over several channels are supplementary services like
call transfer or pickup. To implement these features Asterisk internally
copies caller and connected ids from one channel to another. Another
example are extension subscriptions. The monitoring entities (watchers)
are notified of state changes and - if desired - of party numbers or names
which represent the involving call parties. One major feature where a
private representation of party names is essentially needed, i.e. where a
party name shall be exclusively signaled to only one particular user, is a
private user-specific name resolution for party numbers. A lookup in a
private destination-dependent telephone book shall provide party names
which cannot be seen by any other user at any time.
2. Feature Description
This feature comes along with the implementation of additional private
party id elements for caller id, connected id and redirecting ids inside
Asterisk channels.
The private party id elements can be read or set by the user using
Asterisk dialplan functions.
When a technology channel is initiating a call, receives an internal
connected-line update event, or receives an internal redirecting update
event, it merges the corresponding public id with the private id to create
an effective party id. The effective party id is then used for protocol
signaling.
The channel technologies which initially support the private id
representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and
PRI (chan_dahdi).
Once a private name or number on a channel is set and (implicitly) made
valid, it is generally used for any further protocol signaling until it is
rewritten or invalidated.
To simplify the invalidation of private ids all internally generated
connected/redirecting update events and also all connected/redirecting
update events which are generated by technology channels -- receiving
regarding protocol information - automatically trigger the invalidation of
private ids.
If not using the private party id representation feature at all, i.e. if
using only the 'regular' caller-id, connected and redirecting related
functions, the current characteristic of Asterisk is not affected by the
new extended functionality.
3. User interface Description
To grant access to the private name and number representation from the
Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan
functions are extended by the following data types. The formats of these
data types are equal to the corresponding regular 'non-private' already
existing data types:
CALLERID:
priv-all
priv-name priv-name-valid priv-name-charset priv-name-pres
priv-num priv-num-valid priv-num-plan priv-num-pres
priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd
priv-tag
CONNECTEDLINE:
priv-name priv-name-valid priv-name-pres priv-name-charset
priv-num priv-num-valid priv-num-pres priv-num-plan
priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd
priv-tag
REDIRECTING:
priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset
priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan
priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd
priv-orig-tag
priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset
priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan
priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd
priv-from-tag
priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset
priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan
priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd
priv-to-tag
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2030/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is based on the work done by Olle Johansson on review board.
The idea is that the channel specified in an AMI originate or call
file is typically not connected to the outgoing extension until the
channel has been answered. With this change, an EarlyMedia header can
be specified for AMI originates and an early_media option can
be specified in call files. With this option set, once early media is
received on a channel, it will be connected with the outgoing extension.
(closes issue ASTERISK-18644)
Reported by Olle Johansson
Review: https://reviewboard.asterisk.org/r/1472
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are
implemented in parallel to the existing numbered callgroup/pickupgroup
implementation. However, unlike the existing implementation, which is limited
to a maximum of 64 defined groups, the number of defined groups allowed for
named callgroups/pickupgroups is effectively unlimited.
Named groups are configured with the keywords "namedcallgroup" and
"namedpickupgroup". This corresponds to the numbered group definitions of
"callgroup" and "pickupgroup". Note that as the implementation of named groups
coexists with the existing numbered implementation, a defined named group of
"4" does not equate to numbered group 4.
Support for the named groups has been added to the SIP, DAHDI, and mISDN channel
drivers.
Review: https://reviewboard.asterisk.org/r/2043
Uploaded by:
Guenther Kelleter(license #6372)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now
been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan
functions to better facilitate access to the AST_CAUSE translations
for technology-specific cause codes. The HangupCauseClear application
has also been added to remove this data from the channel.
(closes issue SWP-4738)
Review: https://reviewboard.asterisk.org/r/2025/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It's harder to correlate the Newchannel and Hangup AMI events without
specifying "AccountCode" in both.
(closes issue ASTERISK-19963)
Reported by: Oleg A. Arkhangelsky
Patches:
hangup_acctcode.diff (license #6397) patch uploaded by Oleg A. Arkhangelsky
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The awk script parses out the first instance of the DOCUMENTATION tag that it
finds within a file. If a file did not previously have a DOCUMENTATION tag
but received one due to it having an AMI event, then the XML fragment
associated with the AMI event was erroneously placed in the resulting XML
file. Without the python scripts, these XML fragments will not validate.
This patch adds DOCUMENTATION tags at the top of those files that did
not previously have them to prevent the awk script from pulling AMI event
documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds some basic documentation for a number of modules. This
includes core source files in Asterisk (those in main), as well as
chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri. The DTD
has also been updated to allow referencing of AMI commands.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Hangup handlers are an alternative to the h extension. They can be used
in addition to the h extension. The idea is to attach a Gosub routine to
a channel that will execute when the call hangs up. Whereas which h
extension gets executed depends on the location of dialplan execution when
the call hangs up, hangup handlers are attached to the call channel. You
can attach multiple handlers that will execute in the order of most
recently added first.
(closes issue ASTERISK-19549)
Reported by: Mark Murawski
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2002/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Channel drivers that allow native bridging need to handle
AST_CONTROL_PVT_CAUSE_CODE frames and previously did not handle them
properly, usually breaking out of the native bridge. This change
corrects that behavior and exposes the available cause code information
to the dialplan while native bridges are in place. This required
exposing the HANGUPCAUSE hash setter outside of channel.c, so
additional documentation has been added.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines
Add support-level indications to many more source files.
Since we now have tools that scan through the source tree looking for files
with specific support levels, we need to ensure that every file that is
a component of a 'core' or 'extended' module (or the main Asterisk binary)
is explicitly marked with its support level. This patch adds support-level
indications to many more source files in tree, but avoids adding them to
third-party libraries that are included in the tree and to source files
that don't end up involved in Asterisk itself.
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r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
Add a script to enable finding source files without support-levels defined.
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* Make non-normal dialplan execution routines be able to run on a hung up
channel. This is preparation work for hangup handler routines.
* Fixed ability to support relative non-normal dialplan execution
routines. (i.e., The context and exten are optional for the specified
dialplan location.) Predial routines are the only non-normal routines that
it makes sense to optionally omit the context and exten. Setting a hangup
handler also needs this ability.
* Fix Return application being able to restore a dialplan location
exactly. Channels without a PBX may not have context or exten set.
* Fixes non-normal execution routines like connected line interception and
predial leaving the dialplan execution stack unbalanced. Errors like
missing Return statements, popping too many stack frames using StackPop,
or an application returning non-zero could leave the dialplan stack
unbalanced.
* Fixed the AGI gosub application so it cleans up the dialplan execution
stack and handles the autoloop priority increments correctly.
* Eliminated the need for the gosub_virtual_context return location.
Review: https://reviewboard.asterisk.org/r/1984/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Presence support has been added. This is accomplished by
allowing for presence hints in addition to device state
hints. A dialplan function called PRESENCE_STATE has been
added to allow for setting and reading presence. Presence
can be transmitted to Digium phones using custom XML
elements in a PIDF presence document.
Voicemail has new APIs that allow for moving, removing,
forwarding, and playing messages. Messages have had a new
unique message ID added to them so that the APIs will work
reliably. The state of a voicemail mailbox can be obtained
using an API that allows one to get a snapshot of the mailbox.
A voicemail Dialplan App called VoiceMailPlayMsg has been
added to be able to play back a specific message.
Configuration hooks have been added. Configuration hooks
allow for a piece of code to be executed when a specific
configuration file is loaded by a specific module. This is
useful for modules that are dependent on the configuration
of other modules.
chan_sip now has a public method that allows for a custom
SIP INFO request to be sent mid-dialog. Digium phones use
this in order to display progress bars when files are played.
Messaging support has been expanded a bit. The main
visible difference is the addition of an AMI action
MessageSend.
Finally, a ParkingLots manager action has been added in order
to get a list of parking lots.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Restructure ast_do_masquerade() to not hold channel locks while it calls
ast_indicate().
* Simplify many calls to ast_do_masquerade() since it will never return a
failure now. If it does fail internally because a channel driver callback
operation failed, the only thing ast_do_masquerade() can do is generate a
warning message about strange things may happen and press on.
* Fixed the call to ast_bridged_channel() in ast_do_masquerade(). This
change fixes half of the deadlock reported in ASTERISK-19801 between
masquerades and chan_iax.
(closes issue ASTERISK-19537)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1915/
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This patch fixes to situations that could cause the CEL LINKEDID_END event to
be missed.
1) During a core stop gracefully, modules are unloaded when ast_active_channels
== 0. The LINKDEDID_END event fires during the channel destructor. This means
that occasionally, the cel_* module will be unloaded before the channel is
destroyed. It seemed generally useful to wait until the refcount of all
channels == 0 before unloading, so I added a channel counter and used it in the
shutdown code.
2) During a masquerade, ast_channel_change_linkedid is called. It calls
ast_cel_check_retire_linkedid which unrefs the linkedid in the linkedids
container in cel.c. It didn't ref the new linkedid. Now it does.
Review: https://reviewboard.asterisk.org/r/1900/
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There was a slight discrepancy in the behaviors of the old SIP_CAUSE and the
new SIP_CAUSE/HANGUPCAUSE when a channel had been originated and had not yet
been answered. This caused the noload_res_srtp_attempt_srtp test to fail since
the SIP_CAUSE variable was never actually set. This behavior has been restored.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch improves the handling of call id logging significantly with regard
to transfers and adding APIs to better handle specific aspects of logging.
Also, changes have been made to chan_sip in order to better handle the creation
of callids and to enable the monitor thread to bind itself to a particular
call id when a dialog is determined to be related to a callid. It then unbinds
itself before returning to normal monitoring.
review: https://reviewboard.asterisk.org/r/1886/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is the starting point for the Asterisk 11: Who Hung Up work and provides
a framework which will allow channel drivers to report the types of hangup
cause information available in SIP_CAUSE without incurring the overhead of the
MASTER_CHANNEL dialplan function. The initial implementation only includes
cause generation for chan_sip and does not include cause code translation
utilities.
This change deprecates SIP_CAUSE and replaces its method of reporting cause
codes with the new framework. This change also deprecates the 'storesipcause'
option in sip.conf.
Review: https://reviewboard.asterisk.org/r/1822/
(Closes issue SWP-4221)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Before this patch, the predial routine executes on the ;1 channel of a
local channel pair. Executing predial on the ;1 channel of a local
channel pair is of limited utility. Any channel variables set by the
predial routine executing on the ;1 channel will not be available when the
local channel executes dialplan on the ;2 channel.
* Create ast_pre_call() and an associated pre_call() technology callback
to handle running the predial routine. If a channel technology does not
provide the callback, the predial routine is simply run on the channel.
Review: https://reviewboard.asterisk.org/r/1903/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20,
22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111,
and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped
were either extended/deprecated or in areas of code that shouldn't be
disturbed.
(Closes issue ASTERISK-19650)
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Asterisk has a setting for the minimum allowed DTMF. If we get shorter
DTMF tones, these will be changed to the minimum on the outbound call
leg.
(closes issue ASTERISK-19772)
Review: https://reviewboard.asterisk.org/r/1882/
Reported by: oej
Tested by: oej
Patches by: oej
Thanks to the reviewers.
1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Found a small amount of curly brackets in my hotel room here in Denmark.
I hereby donate them to the Asterisk project.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who
was the original redirecting party of a call.
* Added support for the original redirecting party and reason to the
REDIRECTING function and the system core as well as to the stubbed
locations in sig_pri.c.
Review: https://reviewboard.asterisk.org/r/1829/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Hangup now can take a regular expression as the Channel option. If you want
to hangup multiple channels, use /regex/ as the Channel option. Existing
behavior to hanging up a single channel is unchanged, but if you pass a regex,
the manager will send you a list of channels back that were hung up.
(closes issue ASTERISK-19575)
Reported by: Mark Murawski
Tested by: Mark Murawski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Q.951 indicates that when the presentation indicator is "Number not
available due to interworking" for a number then the screening indicator
field should be "Network provided".
* Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
when the presentation is "Number not available due to interworking". This
fix makes Asterisk consistent and it also makes it consistent with earlier
branches as far as this presentation value is concerned.
* Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions
handle the "Number not available due to interworking" case better in
sig_pri.c. This change is possible because the minimum required libpri
version (v1.4.11) has the necessary defines in libpri.h.
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Everything still compiled after making these changes, so I assume these
whitespace-only changes didn't break anything (and shouldn't have).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When connected line support was added, the wait_for_answer() variable
single changed its meaning slightly. Unfortunately, the places where
single was used did not necessarily get updated to reflect that change.
Also audio/video frames were sent to all forked calls when the endpoints
were never made compatible.
* Don't pass audio/video media frames when the channels have not been made
compatible.
* Added handling of AST_CONTROL_SRCCHANGE to app_dial.c.
* Fixed app_dial.c passing on AST_CONTROL_HOLD because that frame can also
pass a requested MOH class.
(closes issue ASTERISK-16901)
Reported by: Chris Gentle
(closes issue ASTERISK-17541)
Reported by: clint
Review: https://reviewboard.asterisk.org/r/1805/
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The set_format() function was more subtle in how it modified the
struct ast_channel readtrans/writetrans values.
* Fixed ast_activate_generator() conversion correctly.
(closes issue ASTERISK-19434)
Reported by: Birger Harzenetter
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The check if an ISDN call is bridged before it could be placed on hold is
not necessary and is overly restrictive. The check was originally done to
prevent problems with call transfers in case a user tried to transfer a
call connected to an application to another call connected to an
application. The ISDN transfer code has not required this restriction for
quite some time because ECT could transfer any two active calls to each
other.
* Remove ISDN hold restriction for calls connected to applications.
* Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD instead of generating a warning message.
(closes issue ASTERISK-19388)
Reported by: Birger Harzenetter
Tested by: rmudgett
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This takes two actions.
1. Move the reading of the alertpipe in __ast_read() to immediately before the
removal of frames from the readq. This means we won't do something silly like
read from the alertpipe, then ignore the fact that there's a frame to get from
the readq since channel's fdno is the AST_TIMING_FD.
2. When ast_settimeout() sets the rate to 0 and the timingfunc to NULL, if the
channel's fdno is the AST_TIMING_FD, then set the fdno to -1. This is because
if the rate is 0 and the timingfunc is NULL, it means that the channel's timing
fd is being invalidated, so any pending reads should not occur.
This may actually solve more issues than the referenced one below, but it's not
known at this time for sure.
(closes issue ASTERISK-19223)
reported by Frank-Michael Wittig
Review: https://reviewboard.asterisk.org/r/1779
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* Fix opaquification conversion error.
(closes issue ASTERISK-19424)
Reported by: Jeremy Pepper
Patches:
asterisk-19424-initialize_priority_regression.diff (license #5026) patch uploaded by Michael L. Young
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit adds GoSub alternatives to connected line, redirecting, and CCSS
macro hooks so that macro can finally be deprecated. This also adds
deprecation warnings for those features when used and in documentation.
Review: https://reviewboard.asterisk.org/r/1760/
(closes issue SWP-4256)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The accessors names for the "emulate_dtmf_digit" field on the ast_channel
are misleading. Change them to ast_channel_dtmf_digit_to_emulate*.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch removes some unnecessary locking of the channels container in
ast_hangup(). The reason this came up is that this lock can very quickly block
the entire system. If any of the channel cleanup code decides to block, it
causes a problem for the whole system. For example, when audiohooks get
destroyed, if that blocks for a while waiting on the mixmonitor thread to exit
because it's busy blocking on some I/O, it causes a problem for many other
threads in the meantime.
Review: https://reviewboard.asterisk.org/r/1712/
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When ast_channel name was opaquified, the channel search functions did not
get converted correctly. As a result ExtenSpy which uses a channel
iterator search by exten@context could never find anything.
* Updated the doxygen documentation for the search functions in channel.h.
Review: https://reviewboard.asterisk.org/r/1702/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.
Review: https://reviewboard.asterisk.org/r/1661/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Initially, support was left in for the old style of searching, even
though it wasn't actually used. In the case of name_len != 0, the
OBJ_KEY flag isn't passed because we aren't matching on a full key
and therefor can't use the hash function to optimize. The code left
in to support the old way of searching unfortunately treated a prefix
search like this as though an ast_channel struct was passed as an arg
and caused a crash.
This patch also adds needed parentheses around some matching conditions.
(closes issue ASTERISK-19182)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.
This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.
The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.
The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).
Review: https://reviewboard.asterisk.org/r/1655/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Add locking when a channel inherits variables and datastores in
__ast_request_and_dial() and ast_call_forward(). Note: The involved
channels are not active so there was minimal potential for problems.
* Remove calls to ast_set_callerid() in __ast_request_and_dial() and
ast_call_forward() because the set information is for the wrong direction.
* Don't use C++ keywords for variable names in ast_call_forward().
* Run the redirecting interception macro if defined when forwarding a call
in ast_call_forward(). Note: Currently will never execute because the
only callers that supply a calling channel supply a hungup or zombie
channel.
* Make feature_request_and_dial() put the transferee into autoservice when
it calls ast_call_forward() in case a redirection interception macro is
run. Note: Currently will never happen because the caller channel (Party
B) is always hungup at this time.
* Make feature_request_and_dial() ignore the AST_CONTROL_PROCEEDING frame
to silence a log message.
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int blah = 1;
...
write(chan->alertpipe[1], &blah, new_frames * sizeof(blah)) !=
(new_frames * sizeof(blah)))
is only valid when new_frames == 1. Otherwise we start reading into adjacent
variables declared on the stack. The read end discards what is read, so the
values don't matter but it's not a good idea to read past where we want even
though new_frames is almost always 1 and should never be large. This patch is
basically taken out of kpfleming's eventfd branch, as he mentioned that he
remembered fixing it there when I talked to him about this issue.
Review: https://reviewboard.asterisk.org/r/1583/
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https://origsvn.digium.com/svn/asterisk/branches/10
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r337431 | irroot | 2011-09-22 08:29:09 +0200 (Thu, 22 Sep 2011) | 25 lines
Merged revisions 337430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) | 19 lines
Its possible to loose audio on ast_write when the channel is not transcoded correctly.
in the case of DAHDI the channel is hungup.
This patch tries to "fix" the problem and make the channel compatiable and warn the user of
this problem.
Please note there is a underlying problem with codec negotion this does not fix the problem
it does try to rectify it and prevent loss of service.
Review: https://reviewboard.asterisk.org/r/1442/
(closes issue ASTERISK-17541)
(closes issue ASTERISK-18063)
(issue ASTERISK-14384)
(issue ASTERISK-17502)
(issue ASTERISK-18325)
(issue ASTERISK-18422)
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r336307 | jrose | 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) | 20 lines
Merged revisions 336294 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines
Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would
break when starting a call with directmedia. This patch queues a new type of control frame
so that our RTP bridge loop can properly detect when these situations occur and check to see
if peers need to be updated in order to send their media to the proper location.
(Closes issue ASTERISK-18340)
Reported by: Thomas Arimont
(Closes issue ASTERISK-17725)
Reported by: kwk
Tested by: twilson, jrose
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r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
Merged revisions 335064 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then
the 484 Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete
application dialplan logic was automatically triggered; now, explicit
dialplan usage of the application is required.
Additionally, this patch adds a new AST_CONTOL_FRAME type called
AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame,
it is an indication that the dialplan expects more digits back from the
device. If the device supports overlap dialing it should attempt to
notify the device that the dialplan is waiting for more digits; otherwise,
it can handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288)
Reported by: Mikael Carlsson
Tested by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/1416/
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r334010 | rmudgett | 2011-08-31 10:23:11 -0500 (Wed, 31 Aug 2011) | 50 lines
Merged revisions 334009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011) | 43 lines
Call pickup race leaves orphaned channels or crashes.
Multiple users attempting to pickup a call that has been forked to
multiple extensions either crashes or fails a masquerade with a "bad
things may happen" message.
This is the scenario that is causing all the grief:
1) Pickup target is selected
2) target is marked as being picked up in ast_do_pickup()
3) target is unlocked by ast_do_pickup()
4) app dial or queue gets a chance to hang up losing calls and calls
ast_hangup() on target
5) SINCE A MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with
ast_channel_masquerade(), ast_hangup() completes successfully and the
channel is no longer in the channels container.
6) ast_do_pickup() then calls ast_channel_masquerade() to schedule the
masquerade on the dead channel.
7) ast_do_pickup() then calls ast_do_masquerade() on the dead channel
8) bad things happen while doing the masquerade and in the process
ast_do_masquerade() puts the dead channel back into the channels container
9) The "orphaned" channel is visible in the channels list if a crash does
not happen.
This patch does the following:
* Made ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up channel
and not release the channel lock until that has happened.
* Made __ast_channel_masquerade() not setup a masquerade if either channel
has AST_FLAG_ZOMBIE set.
* Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer work.
(closes issue ASTERISK-18222)
Reported by: Alec Davis
Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer
(closes issue ASTERISK-18273)
Reported by: Karsten Wemheuer
Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer
Review: https://reviewboard.asterisk.org/r/1400/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Richard Mudgett
Matthew Jordan
Kinsey Moore
David M. Lee
Mark Michelson
Jason Parker
Joshua Colp
Jonathan Rose
Brent Eagles
Kevin P. Fleming
Terry Wilson
Olle Johansson
Mark Murawki
Russell Bryant
Tzafrir Cohen
Gregory Nietsky
Matthew Nicholson