Kinsey Moore [Fri, 19 Jul 2013 18:09:43 +0000 (18:09 +0000)]
Fix crash when using temporary peers
Temporary peers do not have an associated Stasis endpoint and quite a
bit of code in chan_sip assumes that all peers have a Stasis endpoint.
All endpoint accesses in chan_sip are now wrapped in an endpoint
NULL-check.
Jonathan Rose [Thu, 18 Jul 2013 16:49:44 +0000 (16:49 +0000)]
bridge_holding/app_bridgewait: Add new entertainment options
This patch adds more entertainment options to holding bridges and the
bridge_wait application. Also, holding bridges will now use music on
hold as the default entertainment option instead of none. The
parameters for app_bridgewait have changed to (role, options) from
the previous (options) and the options themselves have changed as
well (entertainment options are now contained in an enumerator, role
specification is handled by the role parameter, etc)
(closes issue ASTERISK-21923)
Reported by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/2679/
Matthew Jordan [Thu, 18 Jul 2013 14:50:56 +0000 (14:50 +0000)]
Tweak debug statements
This patch does two things:
1. It moves the debug statement that shows the HTTP sub-protocols being
compared after the string length calculation such that it shows the correct
string length in the output
2. It adds some additional debug that displays when it matches on a
sub-protocol and when it fails
David M. Lee [Thu, 18 Jul 2013 14:08:21 +0000 (14:08 +0000)]
Fix caching topic shutdown assertions
The recent changes to update stasis_cache_topics directly from the
publisher thread uncovered a race condition, which was causing asserts
in the /stasis/core tests.
If the caching topic's subscription is the last reference to the
caching topic, it will destroy the caching topic after the final
message has been processed. When dispatching to a different thread,
this usually gave the unsubscribe enough time to finish before
destruction happened. Now, however, it consistently destroys before
unsubscription is complete.
This patch adds an extra reference to the caching topic, to hold it
for the duration of the unsubscription.
This patch also removes an extra unref that was happening when the
final message was received by the caching topic. It was put there
because of an extra ref that was put into the caching topic's
constructor. Both have been removed, which makes the destructor a bit
less confusing.
Michael L. Young [Thu, 18 Jul 2013 12:54:50 +0000 (12:54 +0000)]
Properly indicate failure to open an audio stream in res_agi
If there is an error streaming an audio file, the current return status makes it
difficult for an AGI script to determine that there was an error with the audio
file.
This patches changes the result to return -1 and the function returns
RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other parts of
res_agi, this would appear to be the proper way to handle an error.
(closes issue ASTERISK-21903)
Reported by: Ariel Wainer
Tested by: Ariel Wainer
Patches:
asterisk-21903-return-stream-res_1.8.diff
by Michael L. Young (license 5026)
When a DAHDI device is removed at run-time it sends the event
DAHDI_EVENT_REMOVED on each channel. This is intended to signal the
userspace program to close the respective file handle, as the driver of
the device will need all of them closed to properly clean-up.
This event has long since been handled in chan_dahdi (chan_zap at the
time). However the event that is sent on a D-Channel of a "PRI" (ISDN)
span simply gets ignored.
This commit adds handling for closing the file descriptor (and shutting
down the span, while we're at it).
It also adds a CLI command 'pri destroy span <N>' to destroy the span
and its DAHDI channels.
Matthew Jordan [Tue, 16 Jul 2013 22:33:27 +0000 (22:33 +0000)]
Add 'kick all' capability to ConfBridge CLI command
This patch adds the ability to kick all users out of a conference from the
ConfBridge kick CLI command. It is invoked by passing 'all' as the channel
parameter to the CLI command, i.e., "confbridge kick <conf> all".
Note that this patch was modified slightly to conform to trunk.
Matthew Jordan [Tue, 16 Jul 2013 22:25:33 +0000 (22:25 +0000)]
Re-order handlers in CEL to ensure that HANGUP events happen after APP_END
When a channel is hungup, both an APP_END event and a HANGUP event can be
fired. To ensure that HANGUP events occur after APP_END events, the method
callbacks for the APP_END event should be processed prior to the callbacks
for the HANGUP event.
Matthew Jordan [Tue, 16 Jul 2013 18:22:07 +0000 (18:22 +0000)]
Re-order cleanup
This patch attempts to fix some possible race conditions in shutdown of the
CDR engine. It:
* Adds a cleanup handler to only unsubscribe and join on stasis messages during
graceful shutdown. The cleanup handler should execute before the regular atexit
handler, as we want to unsubscribe for any further messages before dispatching
the CDRs.
* The CDRs are now locked when we dispatch them on shutdown.
Richard Mudgett [Mon, 15 Jul 2013 23:20:55 +0000 (23:20 +0000)]
Replace chan_agent with app_agent_pool.
The ill conceived chan_agent is no more. It is now replaced by
app_agent_pool.
Agents login using the AgentLogin() application as before. The
AgentLogin() application no longer does any authentication.
Authentication is now the responsibility of the dialplan. (Besides, the
authentication done by chan_agent did not match what the voice prompts
asked for.)
Sample extensions.conf
[login]
; Sample agent 1001 login
; Set COLP for in between calls so the agent does not see the last caller COLP.
exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>)
; Give the agent DTMF transfer and disconnect features when connected to a caller.
same => n,Set(CHANNEL(dtmf-features)=TX)
same => n,AgentLogin(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
[caller]
; Sample caller direct connect to agent 1001
exten => 800,1,AgentRequest(1001)
same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same => n,Hangup()
; Sample caller going through a Queue to agent 1001
exten => 900,1,Queue(agent_q)
same => n,Hangup()
Sample queues.conf
[agent_q]
member => Local/800@caller,,SuperAgent,Agent:1001
Under the hood operation overview:
1) Logged in agents wait for callers in an agents holding bridge.
2) Caller requests an agent using AgentRequest()
3) A basic bridge is created, the agent is notified, and caller joins the
basic bridge to wait for the agent.
4) The agent is either automatically connected to the caller or must ack
the call to connect.
5) The agent is moved from the agents holding bridge to the basic bridge.
6) The agent and caller talk.
7) The connection is ended by either party.
8) The agent goes back to the agents holding bridge.
To avoid some locking issues with the agent holding bridge, I needed to
make some changes to the after bridge callback support. The after bridge
callback is now a list of requested callbacks with the last to be added
the only active callback. The after bridge callback for failed callbacks
will always happen in the channel thread when the channel leaves the
bridging system or is destroyed.
(closes issue ASTERISK-21554)
Reported by: Matt Jordan
Matthew Jordan [Sun, 14 Jul 2013 02:41:43 +0000 (02:41 +0000)]
Provide error message for QUEUE_MEMBER when member is not in queue
When QUEUE_MEMBER is used and the member specified is not in the queue,
Asterisk provides an ERROR message that indicates that the option specified
is not valid. This patch now properly displays an ERROR message that the
member is not in the queue if an interface is specified.
(closes issue ASTERISK-21980)
Reported by: Avraam David
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Merged revisions 394345 from http://svn.asterisk.org/svn/asterisk/branches/11
Matthew Jordan [Sun, 14 Jul 2013 02:05:04 +0000 (02:05 +0000)]
Remove redundant code in dns.c
Peter J Philipp pointed out that there are two checks that ensure that len is
not less than 0. If len is less than 0, the function returns. Having both of
them is clearly redundant.
This removes the second and attempts to clarify (slightly) the error condition.
(closes issue ASTERISK-21772)
Reported by: Peter J Philipp
Matthew Jordan [Sun, 14 Jul 2013 02:00:02 +0000 (02:00 +0000)]
Clarify documentation for function PASSTHRU
It is not apparent to the average user that the PASSTHRU function should not
be passed as ${PASSTHRU(string)} but just as PASSTHRU(string) to functions
which take a variable name and not its contents.
This patch clarifies the behavior in the documentation and provides an example.
(closes issue ASTERISK-21717)
Reported by: Richard Miller
patches:
func_strings.diff uploaded by Richard Miller (license 5685)
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Merged revisions 394302 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 394303 from http://svn.asterisk.org/svn/asterisk/branches/11
Matthew Jordan [Sat, 13 Jul 2013 23:28:23 +0000 (23:28 +0000)]
Fix FRACK message from external redirects; handle outbound channels better
This patch does the following:
* It simplifies the Dial handling in CDRs. As a rule, the caller in a dial
relationship is always the Party A. There was some logic present in the
handling of the dial message that could, conceivably, pick the caller
as Party A for the beginning of the dial and the peer as Party A for the
end of the dial. This shouldn't have happened if the code in the bridging
framework was doing its job; however, that was broken and it led to the
FRACK. As it is, this code was overly ocmplex and not needed: the caller,
if present, should always be Party A. Period.
* It properly checks to see if a channel will continue on in the dialplan.
ast_check_hangup - much like cake at the end - is a lie. It will tell
you that you are hungup when you are not. Do not believe it.
I would make this function tell the truth, but I'm nervous that we've been
depending on it sitting on its throne of lies for far too long, and it would
probably break lots of things. So I'm just checking the "internal" soft
hangup flags, like everyone else.
(closes issue ASTERISK-22060)
Reported by: Mark Michelson
Fix a longstanding issue with MFC-R2 configuration that prevented users
from mixing different variants or general MFC-R2 settings within the same E1 line.
Most users do not have a problem with this since MFC-R2 lines are usually fractional E1s, or
the whole E1 has the same country variant and R2 settings.
In Venezuela however is common to have inbound MFC-R2 and outbound DTMF-R2 within the same E1.
This fix now properly parses the chan_dahdi.conf file to generate a new openr2 context every
time a new channel => section is found and the configuration was changed.
(closes issue ASTERISK-21117)
Reported by: Rafael Angulo
Related Elastix issue: http://bugs.elastix.org/view.php?id=1612
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Merged revisions 394106 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 394173 from http://svn.asterisk.org/svn/asterisk/branches/11
David M. Lee [Fri, 12 Jul 2013 18:23:39 +0000 (18:23 +0000)]
Fixed intermittent crash when loading test_json.so
The JSON test attempted an overly clever use of RAII_VAR to run code
at the beginning and end of each test, in order to validate that no
JSON objects were leaked during the test.
The problem is that the validation code would run during the initial
load, when the tests were initialized. This happens during startup,
when other parts of the system might actively be allocating and
freeing JSON objects.
This patch changes the RAII_VAR to use the new
ast_test_register_{init,cleanup} functions to run the validations
properly.
Major changes are to pull all packet reading functions into skinny_session
and move timeout handling to scheduling arrangements. Thread cancelling is
now undertaken directly rather than waiting for the read to timeout
(cleanup is popped on thread cancel). Also added some keepalive timings in
debugging messages.
Keepalive timeout has been increased from 1.1 by keepalive to 3 times
keepalive. This seems to align (after keepalives stabilise) with when
devices reset after not receiving keepalives. Probably needs more work,
especially around the first and/or second keepalives that vary
significantly by device and firmware version.
David M. Lee [Thu, 11 Jul 2013 15:37:51 +0000 (15:37 +0000)]
Correct test_cel cleanup.
When I corrected the CEL test crash in r394037, I didn't quite pay attention
to how the globals and locals were being shuffled around in the cleanup
callback. I removed the nulling of the global variables, which caused them
to be double cleaned.
This patch puts the global nulling code back (since the vars are cleaned up
by RAII_VARs), and removes the explicit ao2_cleanup() (since they were no-ops,
because the variables had just been nulled).
David M. Lee [Thu, 11 Jul 2013 14:39:55 +0000 (14:39 +0000)]
Change ARI user config to use a type field
When I initially wrote the configuration support for ARI users, I
determined the section type by a category prefix (i.e., [user-admin]).
This is neither idiomatic Asterisk configuration, nor is it really
that user friendly. This patch replaces the category prefix with a
type field in the section, which is much cleaner.
David M. Lee [Thu, 11 Jul 2013 04:34:49 +0000 (04:34 +0000)]
test_voicemail_api: fix warning found by gcc-4.8
The voicemail_api test had code like strncmp(a, b, sizeof(a)), but a was a
char pointer, instead of a literal or char array. This meant that sizeof was
the size of the pointer, not the length of the string.
Since the string is in a stringfield and should be null terminated, I just
changed it to a plain strcmp.
Kevin Harwell [Wed, 10 Jul 2013 22:26:13 +0000 (22:26 +0000)]
PSJIP - sip.conf to res_sip.conf script
** This script is in no way finished.
Started the initial "cut" at converting a sip.conf file to a res_sip.conf file.
Hopefully the bulk of the framework is in place and only a few minor adjustments
need to be made when an option mapping is added that "doesn't fit". This script
and supporting files should be executable against python version 2.5.
An OrderedDict class (backported from a newer version of python) is included.
A MultiOrderedDict class is implemented so options, when added, should be able
to be added in order and allowed to have multiple values.
Currently the scripts supports the majority of endpoint options found in
res_sip.conf. Support has also been added for Aor(s) and the ACL/security
sections. Inside the sip_to_res_sip.py file one can see a list of options
that still need to be mapped.
Also items that still need to be done: templates, includes, parsing '=>'
delimiter. Note that some code is hopefully in place already to support
templates (e.g. lookup/retrieving defaults from them). However, the
parsing of and adding of the section needs to be done.
Russell Bryant [Wed, 10 Jul 2013 01:56:15 +0000 (01:56 +0000)]
astobj2-ify the SLA code
The SLA code within app_meetme was written before asotbj2 had been
merged into Asterisk. Worse, support for reloads did not exist at first
and was added later as a bolt-on feature. I knew at the time that
reloading was not safe at all while SLA was in use, so the reload would
be queued up to execute when the system was idle. Unfortunately, this
approach was still prone to errors beyond the fact that this was the
only place in Asterisk where configuration was not reloaded
instantly when requested.
This patch converts various SLA objects to be reference counted objects
using astobj2. This allows reloads to be processed while the system is
in use. The code ensures that the objects will not disappear while one
of the other threads is using them. However, they will be immediately
removed from the global trunk and station containers so no new calls
will use them if removed from configuration.
Matthew Jordan [Mon, 8 Jul 2013 14:26:40 +0000 (14:26 +0000)]
Create Local channel messages on the Stasis message bus and produce AMI events
This patch does the following:
* It adds a virtual table of callbacks to core_unreal. These callbacks can be
supplied by concrete implementations of "unreal" channel drivers, which lets
the unreal channel driver call specific functionality when it performs some
action. Currently, this is done to notify implementations when an
optimization operation has begun, and when an optimization operation has
succeeded.
* It adds Stasis-Core messages for Local channel bridging and Local channel
optimization. Local channel optimization is now two events: a Begin and an
End. Some consumers of Stasis-Core may want to know when an operation is
beginning so that they can 'prepare' their information; others will be more
concerned about when the operation has completed, so that they can 'fix up'
information. Stasis-Core allows for both, as does AMI.
Mark Michelson [Mon, 8 Jul 2013 13:57:28 +0000 (13:57 +0000)]
Fix some broken logic in sending outbound caller ID.
* trust_id_outbound was required even when the caller ID was not marked
private. This is against intentions and documentation.
* We now check both name and number privacy instead of checking name privacy
twice.
Matthew Jordan [Sun, 7 Jul 2013 21:29:40 +0000 (21:29 +0000)]
In a channel destructor dispose of items that raise Stasis message properly
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
Matthew Jordan [Sun, 7 Jul 2013 20:34:38 +0000 (20:34 +0000)]
Handle hangup logic in the Stasis message bus and consumers of Stasis messages
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.
Matthew Jordan [Fri, 5 Jul 2013 17:33:33 +0000 (17:33 +0000)]
Refactor RTCP events over to Stasis; associate with channels
This patch does the following:
* It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel
information in the RTCP events. Because Stasis provides a cache, Jaco's
patch was modified to pass the channel uniqueid to the RTP layer as
opposed to a pointer to the channel. This has the following benefits:
(1) It keeps the RTP engine 'clean' of references back to channels
(2) It prevents circular dependencies and other potential ref counting issues
* The RTP engine now allows any RTP implementation to raise RTCP messages.
Potentially, other implementations (such as res_rtp_multicast) could also
raise RTCP information. The engine provides structs to represent RTCP headers
and RTCP SR/RR reports.
* Some general refactoring in res_rtp_asterisk was done to try and tame the
RTCP code. It isn't perfect - that's *way* beyond the scope of this work -
but it does feel marginally better.
* A few random bugs were fixed in the RTCP statistics. (Example: performing an
assignment of a = a is probably not correct)
* We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't
raise an event when we sent a RR report.
Note that this work will be of use to others who want to monitor call quality
or build modules that report call quality statistics. Since the events are now
moving across the Stasis message bus, this is far easier to accomplish. It is
also a first step (though by no means the last step) towards getting Olle's
pinefrog work incorporated.
Again: note that the patch by Jaco Kroon was modified slightly for this work;
however, he did all of the hard work in finding the right places to set the
channel in the RTP engine across the channel drivers. Much thanks goes to Jaco
for his hard work here.
Matthew Jordan [Fri, 5 Jul 2013 03:08:58 +0000 (03:08 +0000)]
Remove parkinglot from the channel snapshot
Legacy channel drivers often include the ability to set a default parking lot
on an endpoint basis; when channels are created for that endpoint, they inherit
the parkinglot option. Parking used to use this option more frequently; while
it is still supported, other options (such as using channel variables or
creation of a custom parkinglot) are supported. More importantly, conveying the
parkinglot information through a channel snapshot isn't terribly useful - it
is rarely (if ever) changed on a channel and some consumers of channel
snapshots, such as ARI, will never use the information.
(closes issue ASTERISK-21968)
Reported by: Matt Jordan
Jonathan Rose [Thu, 4 Jul 2013 18:46:56 +0000 (18:46 +0000)]
res_parking: Replace Parker snapshots with ParkerDialString
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/
Richard Mudgett [Wed, 3 Jul 2013 22:36:38 +0000 (22:36 +0000)]
OneTouchRecord: Make so Monitor/MixMonitor can be toggled/started/stopped.
The OneTouchRecord feature has historically been a toggle. This patch
adds the ability to make the OneTouchRecord hook optionally start/stop
recording only. If OneTouchRecord is already doing what is requested then
only the invoker hears the courtesy tone and/or start/stop recording
message.
The new feature is written so we could easily add explicit start/stop
recording DTMF hooks for Monitor and MixMonitor.
The majority of the changes in bridge_builtin_features.c is a refactoring
of the OneTouchRecord code (Monitor and MixMonitor versions) so it is easy
to direct the toggle/start/stop functionality.
Matthew Jordan [Wed, 3 Jul 2013 22:04:08 +0000 (22:04 +0000)]
Fix some bugs in CDRs; add some CLI commands to help debugging
This patch fixes a few minor bugs and one major one: the CDR by bridge
container was less than helpful. The mechanism previously used to try
and find all of the CDRs in a particular bridge ended up missing CDRs,
resulting in incorrect records.
When looking up CDRs in a bridge, we now just bite the bullet and do
a selection across all existing CDRs.
Mark Michelson [Wed, 3 Jul 2013 20:41:00 +0000 (20:41 +0000)]
Publish a bridge enter before pulling on a push-and-swap operation.
Prior to this patch, the order of procedures on a bridge push was
* Add new bridge channel to bridge's array.
* Pull the swap channel out of the bridge
* Publish a bridge enter event.
The problem is that when the swap channel was pulled from the bridge,
a bridge leave event would be published. The bridge snapshot
published during the bridge leave showed the new channel that had
been added to the bridge, but there had been no bridge enter event
for that channel.
The fix provided here was to change the order a bit
* Add new bridge channel to bridge's array.
* Publish bridge enter event.
* Pull the swap channel out of the bridge.
This makes it so that the bridge snapshots during the stasis
events are accurate.
David M. Lee [Wed, 3 Jul 2013 19:46:50 +0000 (19:46 +0000)]
Fix load errors related to the new ari_model_validators.
The Asterisk strategy of loading modules with RTLD_LAZY to extract metadata
from the module works well enough, until you try to take the address of a
function.
If a module takes the address of a function, that function needs to be
resolved at load time. That kinda defeats RTLD_LAZY.
This patch adds some ari_validator_{id}_fn() wrapper functions for safely
getting the function pointer from a different module.
David M. Lee [Wed, 3 Jul 2013 17:58:45 +0000 (17:58 +0000)]
ARI - channel recording support
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).
David M. Lee [Wed, 3 Jul 2013 16:33:13 +0000 (16:33 +0000)]
ARI authentication.
This patch adds authentication support to ARI.
Two authentication methods are supported. The first is HTTP Basic
authentication, as specified in RFC 2617[1]. The second is by simply
passing the username and password as an ?api_key query parameter
(which allows swagger-ui[2] to authenticate more easily).
ARI usernames and passwords are configured in the ari.conf file
(formerly known as stasis_http.conf). The user may be set to
`read_only`, which will prohibit the user from issuing POST, DELETE,
etc. Also, the user's password may be specified in either plaintext,
or encrypted using the crypt() function.
Several other notes about the patch.
* A few command line commands for seeing ARI config and status were
also added.
* The configuration parsing grew big enough that I extracted it to
its own file.
The event discriminator was the field name instead of being a value in
the object, due to limitations in how Swagger 1.1 could model objects.
While technically sufficient in communicating event information, it was
really difficult to deal with in terms of client side JSON handling.
This patch takes advantage of a proposed extension[1] to Swagger which
allows type variance through the use of a discriminator field. This had
a domino effect that made this a surprisingly large patch.
In changing the models, I also had to change the swagger_model.py
processor so it can handle the type discriminator and subtyping. I took
that a big step forward, and using that information to generate an
ari_model module, which can validate a JSON object against the Swagger
model.
The REST and WebSocket generators were changed to take advantage of the
validators. If compiled with AST_DEVMODE enabled, JSON objects that
don't match their corresponding models will not be sent out. For REST
API calls, a 500 Internal Server response is sent. For WebSockets, the
invalid JSON message is replaced with an error message.
Since this took over about half of the job of the existing JSON
generators, and the .to_json virtual function on messages took over the
other half, I reluctantly removed the generators.
The validators turned up all sorts of errors and inconsistencies in our
data models, and the code. These were cleaned up, with checks in the
code generator avoid some of the consistency problems in the future.
* The model for a channel snapshot was trimmed down to match the
information sent via AMI. Many of the field being sent were not
useful in the general case.
* The model for a bridge snapshot was updated to be more consistent
with the other ARI models.
Another impact of introducing subtyping was that the swagger-codegen
documentation generator was insufficient (at least until it catches up
with Swagger 1.2). I wanted it to be easier to generate docs for the API
anyways, so I ported the wiki pages to use the Asterisk Swagger
generator. In the process, I was able to clean up many of the model
links, which would occasionally give inconsistent results on the wiki. I
also added error responses to the wiki docs, making the wiki
documentation more complete.
Finally, since Stasis-HTTP will now be named Asterisk REST Interface
(ARI), any new functions and files I created carry the ari_ prefix. I
changed a few stasis_http references to ari where it was non-intrusive
and made sense.
projects / thirdparty / asterisk.git / log
