agupta [Thu, 6 Jun 2019 12:48:18 +0000 (18:18 +0530)]
chan_pjsip.c: Check for channel and session to not be NULL in hangup
We have seen some rare case of segmentation fault in hangup function
and we could notice that channel pointer was NULL. Debug log shows
that there is a 200 OK answer and SIP timeout at the same time. It
looks that while the SIP session was being destroyed due to timeout
call hangup due to answer event lead to race condition and channel
is being destroyed from two different places. The check ensures we
check it not to be NULL before freeing it.
[custom_blindxfer]
exten => s,1,
same => n,Playback(pbx-transfer)
same => n,Read(dest,dial,10,i,3,3)
same => n,BlindTransfer(${dest},default)
same => n,Return()
;;;
Fixes an error occurring in function pgsql_reconnect() caused when value of
hostname is blank. Which in turn will cause the connection string to look
like this: "host= port=xx", which creates a sintax error. This fix now checks
if the corresponding values for host, port, dbname, and user are blank. Note
that since this is a reconnect function the database library will replace any
missing value pairs with default ones.
Alexei Gradinari [Tue, 28 May 2019 20:35:17 +0000 (16:35 -0400)]
res_fax: fix segfault on inactive "reserved" fax session
The change #10017 "Handle fax gateway being started more than once"
introdiced a bug which leads to segfault in res_fax_spandsp.
The res_fax_spandsp module does not support reserving sessions, so
fax_session_reserve returns a fax session with state AST_FAX_STATE_INACTIVE.
The fax_gateway_start does not create a real fax session if the fax session
is already present and the state is not AST_FAX_STATE_RESERVED.
But the "reserved" session created for res_fax_spandsp has state
AST_FAX_STATE_INACTIVE, so fax_gateway_start not starting.
Then when fax_gateway_framehook is called and gateway T.38 state is
NEGOTIATED the call of gateway->s->tech->write(gateway->s, f) leads to
segfault, because session tech_pvt is not set, i.e. the tech session
was not initialized/started.
This patch adds check also on AST_FAX_STATE_INACTIVE to the "reserved"
session created for res_fax_spandsp will start.
This patch also adds extra check and log ERROR if tech_pvt is not set
before call tech->write.
Alexei Gradinari [Tue, 28 May 2019 22:15:40 +0000 (18:15 -0400)]
res_fax: add channel name to CLI 'fax show session'
This patch adds a channel name to output of CLI 'fax show session'
and also expands the channel name field up to 30 characters on
CLI 'fax show sessions'
George Joseph [Fri, 17 May 2019 23:44:37 +0000 (17:44 -0600)]
res_rtp_asterisk: Add ability to propose local address in ICE
You can now add the "include_local_address" flag to an entry in
rtp.conf "[ice_host_candidates]" to include both the advertized
address and the local address in ICE negotiation:
Alexei Gradinari [Mon, 13 May 2019 20:37:50 +0000 (16:37 -0400)]
pjsip: replace 180 by 183 if SDP negotiation has completed
The caller endpoint hears dead silence if a callee replies 180 (without SDP)
and the caller already received 183 (with SDP).
It happens because Asterisk sends 180 (WITH SDP) to the caller,
there are not incoming RTP packets from the callee
and Asterisk does not generate inband ringing,
so there are not any outgoing RTP packets to the caller.
This patch replaces 180 by 183 if SDP negotiation has completed,
as if the caller endpoint is configured with "inband_progress=yes".
In this case Asterisk will generate inband ringing untill Asterisk receive
incoming RTP packets from the callee.
Ben Ford [Tue, 7 May 2019 16:08:33 +0000 (11:08 -0500)]
pjsip_options.c: Allow immediate qualifies for new contacts.
When multiple endpoints try to register close together using the same
AOR with qualify_frequency set, one contact would qualify immediately
while the other contacts would have to wait out the duration of the
timer before being able to qualify. Changing the conditional to check
the contact container count for a non-zero value allows all contacts to
qualify immediately.
George Joseph [Fri, 3 May 2019 18:31:06 +0000 (12:31 -0600)]
build: Pass --fno-partial-inlining to third-party when appropriate
When the gcc version is >= 8.2.1, we were already setting the
--fno-partial-inlining flag for Asterisk source files to get around
a gcc bug but we weren't passing the flag down to the bundled
builds of pjproject and jansson.
George Joseph [Thu, 2 May 2019 18:29:49 +0000 (12:29 -0600)]
res_pjsip: Check return from pjsip_parse_uri calls
Updated ast_sip_create_rdata_with_contact and registrar_find_contact
to check the return from pjsip_parse_uri before attempting to
use the uri returned.
ASTERISK-28402 Reported-by: Ross Beer
Change-Id: I9810b3b163c45ed5a56ec743586e5ce107f13ba7
Ben Ford [Tue, 23 Apr 2019 14:47:45 +0000 (09:47 -0500)]
stasis: Fix crash at shutdown.
When compiling in dev mode, stasis statistics are enabled and can cause
a crash at shutdown due to the following:
- Containers are freed
- Topics and subscriptions remain
- When those topics and subscriptions are deallocated, they go to do
things with the container
This changes the containers to global ao2 objects, and whenever needed
in the code, a reference must be obtained and checked before any
operations can be done.
Antoni Goldstein [Fri, 29 Mar 2019 14:04:46 +0000 (14:04 +0000)]
app_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings
Added RINGTIME, RINGTIME_MS, PROGRESSTIME, PROGRESSTIME_MS variables filled
at the earliest received PROGRESS or RINGING.
Added millisecond versions of DIALEDTIME and ANSWEREDTIME.
Added millisecond versions of ast_channel_get_up_time and
ast_channel_get_duration in channel.c.
Kevin Harwell [Tue, 9 Apr 2019 19:09:44 +0000 (14:09 -0500)]
mwi core: Move core MWI functionality into its own files
There is enough MWI functionality to warrant it having its own 'c' and header
files. This patch moves all current core MWI data structures, and functions
into the following files:
main/mwi.h
main/mwi.c
Note, code was simply moved, and not modified. However, this patch is also in
preparation for core MWI changes, and additions to come.
George Joseph [Mon, 22 Apr 2019 16:09:51 +0000 (10:09 -0600)]
ARI: Bump non-breaking version number to 1.10.2
main/json.c: Added app_name, app_data to channel type
res/res_ari: Added ARI resource /ari/channels/{channelId}/rtp_statistics
res/res_ari: Added timestamp as a requirement for all ARI events
core/buildsystem: check the actual compiler being version
Make compiler check use the output of the actual compiler being
used as reported by the CC variable, instead of unconditionally
running the "gcc" binary. Also only run the check if the compiler
is gcc or a cross-compile gcc.
We changed the validation of autocomplete parameter in the "indications
remove" command to avoid continue the execution of the command after
asking for autocomplete out of range parameters.
Checks the PJSIP global setting value.
If it is true (default) it adds the norefersub capability to PJSIP.
If it is false (disabled) it does not add the norefersub capability
to PJSIP.
This is useful for Cisco switches that do not follow RFC4488.
ASTERISK-28375 #close Reported-by: Dan Cropp
Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9
George Joseph [Fri, 12 Apr 2019 16:32:44 +0000 (10:32 -0600)]
CI: Move test group config files to Jenkins
One of the downaides of having things like test configuration
in the git repo is that it can't be changed at runtime. You have
to create a review for the changes and merge it mefore it will
take effect.
This review moves the data currently held in
tests/CI/periodic-dailyTestGroups.json and
tests/CI/gateTestGroups.json into a Jenkins Config File attached
to the job definitions. This allows us to alter it from the
Jenkins UI at runtime. The original files stay in the repo
as documentation.
Sean Bright [Tue, 9 Apr 2019 15:10:12 +0000 (11:10 -0400)]
app_voicemail: Cleanup stale lock files on module load
If Asterisk crashes while a VM directory is locked, lock files in the VM
spool directory will not get properly cleaned up. We now clear them on
module load.
Sean Bright [Thu, 11 Apr 2019 20:48:49 +0000 (16:48 -0400)]
res_ael: Create consistent label names across reloads
Reset the internal counter that the AEL2 compiler uses for unique label
names before compiling. This keeps dialplan labels consistent across
reloads assuming the AEL2 has not changed.
Sean Bright [Thu, 11 Apr 2019 16:03:07 +0000 (12:03 -0400)]
res_ael: Fix pattern matching against literal '+'
When generating the regular expression that matches against existing
extensions, we need to escape literal characters that can also be
regular expression metacharacters. This was already being done for '*'
but we need to do the same for '+'.
In passing, remove some unreachable code - strcmp() is already run
immediately when entering extension_matches().
Ben Ford [Mon, 1 Apr 2019 19:42:55 +0000 (14:42 -0500)]
build: Revise CHANGES and UPGRADE.txt handling.
This changes the way that we handle adding changes to CHANGES and
UPGRADE.txt. The reason for this is because whenever someone needed to
make a change to one of these files and someone else had already done
so, you would run into merge conflicts. With this new setup, there will
never be merge conflicts since all changes will be documented in the
doc/<file>-staging directory. The release script is now responsible for
merging all of these changes into the appropriate files.
There is a special format that these files have to follow in order to be
parsed. The files do not need to have a meaningful name, but it is
strongly recommended. For example, if you made a change to pjsip, you
may have something like this "res_pjsip_relative_title", where
"relative_title" is something more descriptive than that. Inside each
file, you will need a subject line for your change, followed by a
description. There can be multiple subject lines. The file may look
something like this:
Subject: res_pjsip
Subject: Core
A description that explains the changes made and why. The release
script will handle the bulleting and section separators!
You can still separate with new lines within your description.
The headers ("Subject" and "Master-Only") are case sensative, but the
value for "Master-Only" ("true" or "True") is not.
For more information, check out the wiki page:
https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt
When extconfig.conf file is parsed, the code previously searched for
character comma without verifying if error (null or blank). This caused
a segmentation error.
sungtae kim [Thu, 21 Mar 2019 23:09:14 +0000 (00:09 +0100)]
main/json.c: Added app_name, app_data to channel type
It was difficult to check the channel's current application and
parameters using ARI for current channels. Added app_name, app_data
items to show the current application information.
Ben Ford [Wed, 3 Apr 2019 15:24:06 +0000 (10:24 -0500)]
build: Fix compiler warnings/errors.
The compiler complained about a couple of variables that weren't
initialized but were being used. Initializing them to NULL resolves the
warnings/errors.
res/res_rtp_asterisk: Enable rxjitter calculation for video
It looks like we're not properly calculating jitter values on received
video streams. This patch enables the code that does jitter calculations
for those streams.
Ben Ford [Tue, 26 Mar 2019 19:56:37 +0000 (14:56 -0500)]
alembic: Fix errors during upgrade head.
When trying to upgrade using alembic, a couple different errors kept
popping up that prevented the upgrade. An additional parameter was
needed when changing the schema for mwi_subscribe_replaces_unsolicited
from an integer to an enum. When changing from a string to an enum, the
type needed to be cast for postgresql. The other issue was a parameter
being used during column creation that did not exist.
After fixing the upgrade process, it revealed errors with the downgrade
process. One was a variable not being defined in the downgrade function,
and the other was tables not existing when using MySQL. This was due to
a context check that should have encompassed MySQL, but in the end was
not doing so.
Ben Ford [Wed, 27 Mar 2019 16:03:40 +0000 (11:03 -0500)]
build: Add staging directories for future changes.
This is the first step in changing the release process so that changes
made to the CHANGES and UPGRADE.txt files do not result in merge
conflicts every time multiple people modify these files. The changes
made will go in these new directories: doc/CHANGES-staging and
doc/UPGRADE-staging. The README.md files explain how things will work,
but here's a little overview. When you make a change that would go in
either CHANGES or UPGRADE.txt, this should instead be documented in a
new file in the doc/CHANGES-staging or doc/UPGRADE-staging directory,
respectively. The format will look like this:
Subject: res_pjsip
A description that explains the changes made and why. The release
script will handle the bulleting and section separators! The
'Subject:' header is case-sensitive.
You can still separate with new lines within your description.
Subject: res_ari
Master-Only: true
You can have more than one subject, and they don't have to be the
same! Also, the 'Master-Only' header should always be true and is
also case-sensitive (but the value is not - you can have 'true' or
'True'). This header will only ever be present in the master branch.
For more information, check out the wiki page:
https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt
This is an initial change for ASTERISK_28111. Functionally, this will
make no difference, but it will prep the directories for when the
changes from CHANGES and UPGRADE.txt are extracted.
Joshua Colp [Mon, 25 Mar 2019 11:34:09 +0000 (08:34 -0300)]
manager: Use separate lock for session event notification.
When notifying a manager session that new events were available
the same lock was used that was also held when doing things within
the session (such as sending events out). If the manager session
blocked for a period of time this would cause a back up of messages
in Stasis and would also block any other sessions from receiving
events.
This change adds a separate lock to the manager session which is
strictly used for notifying it that new events are available.
sungtae kim [Wed, 6 Mar 2019 22:21:42 +0000 (23:21 +0100)]
res/res_ari: Added timestamp as a requirement for all ARI events
Changed to requirement to having timestamp for all of ARI events.
The below ARI events were changed to having timestamp.
PlaybackStarted, PlaybackContinuing, PlaybackFinished,
RecordingStarted, RecordingFinished, RecordingFailed,
ApplicationReplaced, ApplicationMoveFailed
Alexei Gradinari [Mon, 25 Mar 2019 16:42:27 +0000 (12:42 -0400)]
res_config_odbc: set empty extended field as a single whitespace
If Realtime @ variable value is NULL or empty or contains only whitespaces
then when we try to retrieve it using PJSIP_ENDPOINT we get WARNING
pjsip_endpoint_function_read: Unknown property @my_var for PJSIP endpoint.
And the variable is missing in the result of CLI pjsip show endpoint.
projects / thirdparty / asterisk.git / log
