Increasing the non-breaking AMI and ARI version numbers due to changes and
additions in those API's. Note, some changes may be forthcoming (will be added
between now and the next release of Asterisk), thus not listed here. As well
a few changes listed below may have been released in a previous release of
Asterisk, but the API version numbers were not increased at that time, so
including here.
AMI:
* app_queue: set the wrapuptime from AddQueueMember application - e806990
* res_pjsip: option for ContactStatus event updates - 7f22c9f
ARI:
* bridging: Add creation timestamps - 0d70120
* res_stasis: Add ability to switch applications - 50a4b61
* ARI event type filtering - 1c5def4
* Added ARI resource /ari/asterisk/ping - 19fc99a
ASTERISK-28314
Change-Id: Iebc813840f8230afa6b20579772e15549064b787
The download_externals script wasn't getting the PJPROJECT_BUNDLED
environment variable passed down to it so it wasn't downloading
the appropriate variant of res_digium_phone. This could cause
crashes in the DPMA.
Change-Id: I5daa9369c7af1fd556d892e89a85f279a2533425
Apply flag -fno-partial-inlining on default optimization if and only if
gcc version >= 8.2.1 (this is the current ver on Fedora and Ubuntu).
This is done to avoid a bug that causes arithmetic calculations to fail
if the following conditions are met:
1. TEST_FRAMEWORK on
2. DONT_OPTIMIZE off
3. Fedora and Ubuntu
4. GCC 8.2.1
5. There must exist a certain combination of multithreading.
6. Optimization level -O2 and -O3
7. Flag -fpartial-inline activated (default when optimization level>=2)
The following link points to a similar gcc bug reported in 2015. This leads me
to believe the bug has regressed. Note I am not able to replicate this bug
in an environment other than Asterisk + Test Framework + Test_cel because the
multithreading combination that causes it seems to be unique. Therefore I
am temporarily abandoning any thoughts of reporting the new occurrence of this
bug to gcc.gnu.org. https://gcc.gnu.org/bugzilla/show_bug.cgi?id=65307
Change-Id: Ibd1afe60e0a38b88e85fdcd9b051004601c2f102
Topic names now follow: <subsystem>:<functionality>[/<object>]
This ensures that they are all unique, and also provides better
insight in to what each topic is for.
Subscriber ids now also use the main topic name they are
subscribed to and an incrementing integer as their identifier to
make it easier to understand what the subscription is primarily
responsible for.
Both the CLI commands for listing topic and subscription statistics
now sort to make it a bit easier to see what is going on.
Subscriptions will now show all topics that they are receiving messages
from, not just the main topic they were subscribed to.
ASTERISK-28335
Change-Id: I484e971a38c3640f2bd156282e532eed84bf220d
Currently, when the Asterisk calculates rtp statistics, it uses
sample_count as a unsigned integer parameter. This would be fine
for most of cases, but in case of large enough number of sample_count,
this might be causing the divide by zero error.
ASTERISK-28321
Change-Id: If7e0629abaceddd2166eb012456c53033ea26249
Add logical group at DAHDIChannel event
and create "dahdi_group" at CHANNEL function.
ASTERISK-28317
Change-Id: Ic1f834cd53982a9707a9748395ee746d6575086a
chan_sip will always ignore 183 responses that do not contain SDP
however, chan_pjsip will currently always translate it into a
183 with SDP. This new flag allows chan_pjsip to have the same
behavior as chan_sip.
ASTERISK-28322 #close
Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a
Add a json_pack at startup that will fail if runtime links against a
library older than jansson-2.11.
Change-Id: I101aebafe0f9407650206f7c552dad3d69377b5a
Added the ability to move between Stasis applications within Stasis.
This can be done by calling 'move' in an application, providing (at
minimum) the channel's id and the application to switch to. If the
application is not registered or active, nothing will happen and the
channel will remain in the current application, and an event will be
triggered to let the application know that the move failed. The event
name is "ApplicationMoveFailed", and provides the "destination" that the
channel was attempting to move to, as well as the usual channel
information. Optionally, a list of arguments can be passed to the
function call for the receiving application. A full example of a 'move'
call would look like this:
client.channels.move(channelId, app, appArgs)
The control object used to control the channel in Stasis can now switch
which application it belongs to, rather than belonging to one Stasis
application for its lifetime. This allows us to use the same control
object instead of having to tear down the current one and create
another.
ASTERISK-28267 #close
Change-Id: I43d12b10045a98a8d42541889b85695be26f288a
Fixes an intermittent segmentation fault which occured when accessing
nativeformats of a channel which entered into a queue.
ASTERISK-27964
Reported by: Francisco Seratti
Change-Id: Ic87fa7a363f3b487c24ce07032f4b2201c22db9e
While the 'interface' column is a NOT NULL, the empty string is still
allowed. res_config_odbc treats the empty string as a NULL and we crash
when trying to dereference.
Also cleaned up an adjacent error message for consistency.
ASTERISK-28168 #close
Change-Id: I55e012b540fbcda99bb40bede3099b7ae5db8202
In Asterisk configuration, a multiline comment starts with ;-- as long as it is
not followed by another dash (i.e. ;--- is not a multiline comment).
ASTERISK-28323 #close
Change-Id: I32dc38e0fac01d3c0805d27d35d2365a7c37ca72
The res_pjsip_websocket module requires the res_http_websocket
module so ensure it is loaded. As well the res_pjsip_notify
module needs the pjsip_notify.conf configuration file so
ensure it is installed.
ASTERISK-28272
Change-Id: I261659b84e7a6ac4cb49990d9badb4b2ad01bacd
This small feature will help to checking the bridge's status to
figure out which bridge is in old/zombie or not. Also added
detail items for the 'bridge show *' cli to provide more detail
info. And added creation item to the ARI as well.
ASTERISK-28279
Change-Id: I460238c488eca4d216b9176576211cb03286e040
PJSIP assumes that these header names are not allocated, and does not
clone the name strings when reusing headers.
Block unload of res_pjsip_diversion until shutdown to ensure static
memory stays valid.
ASTERISK-28312 #close
Change-Id: Ibd6ea55ec4a604bbd43ac07f8d0b54da2c39b8b9
This file was added to the Subversion repository on 2007-03-15 by
Russell Bryant, a Digium employee at the time.
ASTERISK-24173 #close
Change-Id: Ie866fa9d31d550467613d362b35b03c031ee594d
apply_negotiated_sdp_stream was returning a "1" when no joint
capabilities were found on an outgoing call instead of a "-1".
This indicated to res_pjsip_session that the handler DID handle
the sdp when in fact it didn't. Without the appropriate setup,
a subsequent media frame coming in would have an invalid stream_num
and cause a seg fault when the stream was attempted to be retrieved.
apply_negotiated_sdp_stream now returns the correct "-1" and any
media is now discarded before it reaches the core stream processing.
ASTERISK-28260
Reported by: Sotiris Ganouris
Change-Id: Ia095cb16b4862f2f6ad6d2d2a77453fa2542371f
This reverts commit d524ad523d.
Reason for revert: This causes Contact and Via headers to have the wrong
transport address.
ASTERISK-28309 #close
Change-Id: Ibba4d6176f68e39279fcd9a545f81d56e747bed8
If both send_registrations and send_auth are both set to yes,
outbound_auth/username must be set or we crash.
ASTERISK-27992 #close
Change-Id: I6418d56de1ae53f80393b314c2584048fbf7f11d
Rather than calling ast_odbc_find_table() in the prepare callback, call
it beforehand and pass it in to the callback to avoid the need for a
second connection.
ASTERISK-28166 #close
Change-Id: I6f8a0b9990d636fd6bc1a92ed70f7050d2436202
When a contact was removed by the registrar it did not always check to see if
the circumstances involved a monitored reliable transport. For instance, if the
'remove_existing' option was set to 'true' then when existing contacts were
removed due to 'max_contacts' being reached, those existing contacts being
removed did not unregister the transport monitor.
Also, it was possible to add more than one monitor on a reliable transport for
a given aor and contact.
This patch makes it so all contact removals done by the registrar also remove
any associated transport monitors if necessary. It also makes it so duplicate
monitors cannot be added for a given transport.
ASTERISK-28213
Change-Id: I94b06f9026ed177d6adfd538317c784a42c1b17a
The recent upgrade of Gerrit to 2.16 elimiated referencing a
repository in a way the jenkinsfiles were relying on so
the URL references were changed to a more consistent and supported
format.
Change-Id: I2e8e3f213b9a96bb1b27665eca4a9a24bc49820e
(cherry picked from commit 5ce084579f)
This module allows presence subscriptions to voicemail boxes. This
allows common BLF keys to act as voicemail waiting indicators.
ASTERISK-28301
Change-Id: I62a246c24f3d7d432e33e22d7a4a57c15c292fdd
