Because StasisEnd's timestamp created it's own timestamp, it makes
wrong timestamp, compare to other channel event(ChannelDestroyed).
Fixed to getting a timestamp from the Channel's timestamp.
ASTERISK-28333
Change-Id: I5eb8380fc472f1100535a6bc4983c64767026116
This might be useful in situations where you are loading an undetermined number
of items into a vector and don't want to keep (potentially) 2x the necessary
memory around indefinitely.
Change-Id: I9711daa0fe01783fc6f04c5710eba84f2676d7b9
A size_t is not always an unsigned long.
* Use the %zu format specifier in the ast_cli() printf format string since
AST_VECTOR_SIZE() returns a size_t value.
Change-Id: Ib102dd36bbe6c2a7a4ce6870ae9110d978dd7e98
As part of an earlier voicemail refactor, ast_delete_mwi_state_full
was modified to remove the pool topic for a mailbox when the state
was deleted. This was an attempt to prevent stale topics from
accumulating when app_voicemail was reloaded and a mailbox went
away. Unfortunately because of the fact that when app_voicemail
reloads, ALL mailboxes are deleted then only current ones recreated,
topics were being removed from the pool that still had subscribers
on them, then recreated as new topics of the same name. So now
modules like res_pjsip_mwi are listening on a topic that will
never receive any messages because app_voicemail is publishing on
a different topic that happens to have the same name. The solutiuon
to this is not easy and given that accumulating topics for
deleted mailboxes is less evil that not sending NOTIFYs...
* Removed the call to stasis_topic_pool_delete_topic in
ast_delete_mwi_state_full.
Also:
* Fixed a topic reference leak in res_pjsip_mwi
mwi_stasis_subscription_alloc.
* Added some debugging to mwi_stasis_subscription_alloc,
stasis_topic_create, and topic_dtor.
* Fixed a topic reference leak in an error path in
internal_stasis_subscribe.
ASTERISK-28306
Reported-by: Jared Hull
Change-Id: Id7da0990b3ac4be4b58491536b35f41291247b27
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
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:
* res_pjsip: option for ContactStatus event updates - 4a8564c
ARI:
* bridging: Add creation timestamps - 4dd4dbd
* res_stasis: Add ability to switch applications - 65170ba
* ARI event type filtering - da93d17
* Added ARI resource /ari/asterisk/ping - 67d587f
ASTERISK-28314
Change-Id: I96951b19c27c196e410b09fe82b00c8ca328cccc
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
When variable ALTCONF is defined, the command start prints the message
"Unable to open specified master config file '"/etc/asterisk/asteris..."
and use default configurations.
ASTERISK-28332
Change-Id: I7595e582a0ee2c1051ea35435e247e27906957ef
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
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
Add logical group at DAHDIChannel event
and create "dahdi_group" at CHANNEL function.
ASTERISK-28317
Change-Id: Ic1f834cd53982a9707a9748395ee746d6575086a
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
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
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