Joshua Colp [Mon, 6 Nov 2017 16:29:33 +0000 (16:29 +0000)]
res_pjsip_pubsub: Ensure remote URI contains URI only.
This change makes it so that any user of the pubsub
API that requests the remote URI receives only the URI.
Previously the entire string was returned, which could
contain a display name.
Corey Farrell [Mon, 6 Nov 2017 09:21:12 +0000 (04:21 -0500)]
configure: Add autoconf check for libopusfile.
This check is being added to make it easier for end-users of third party
open source Opus modules. This was removed by ASTERISK-26426 but only
the module needed to be removed.
Corey Farrell [Fri, 3 Nov 2017 15:35:34 +0000 (11:35 -0400)]
Build System: Fix build failure caused by recent CLI improvements.
We use the editline library to help with filename completion in our CLI
interface. Some systems failed to find the header when included from
loader.c. This is fixed by setting the proper CFLAGS for the build of
loader.o.
Ben Ford [Wed, 1 Nov 2017 16:12:45 +0000 (11:12 -0500)]
res_pjsip: Add to list of valid characters for from_user.
Fixes a regression where some characters were unable to be used in
the from_user field of an endpoint. Additionally, the backtick was
removed from the list of valid characters, since it is not valid,
and it was replaced with a single quote, which is a valid character.
Corey Farrell [Mon, 30 Oct 2017 22:30:18 +0000 (18:30 -0400)]
Modules: Additional improvements to CLI completion.
Replace 'needsreload' argument with a 'type' argument to specify which
type of modules you want completion. This provides more accurate CLI
completion for load and unload commands.
* 'module unload' now excludes modules that have active references or are
not running.
* 'module load' now excludes modules that are already running.
* 'core set debug [atleast] <level> [module]' shows running modules only.
Joshua Colp [Thu, 2 Nov 2017 10:34:46 +0000 (07:34 -0300)]
core: Don't attempt to write to a stream that does not exist.
When a frame is provided to ast_write ensure that a multistream
capable channel has a stream for it before attempting to give it
to the channel driver. In some cases (such as a deferred SDP
negotiation) the stream may not yet exist.
Corey Farrell [Thu, 2 Nov 2017 00:46:11 +0000 (20:46 -0400)]
Prevent unload of modules which implement an Optional API.
Once an Optional API module is loaded it should stay loaded. Unloading
an optional API module runs the risk of a crash if something else is
using it. This patch causes all optional API providers to tell the
module loader not to unload except at shutdown.
Kevin Harwell [Tue, 31 Oct 2017 20:08:38 +0000 (15:08 -0500)]
features: Bridge application's BRIDGERESULT not appropriately set
The dialplan application "Bridge" was not setting the BRIDGERESULT to failure
when a failure did occur. Even worse if it did fail to join the bridge it would
still report success.
This patch now sets the BRIDGERESULT variable to an appropriate value for a
given condition state. Also, removed the value INCOMPATIBLE as a valid result
type since it is no longer used.
Joshua Colp [Wed, 25 Oct 2017 22:31:33 +0000 (22:31 +0000)]
core / pjsip: Add support for grouping streams together.
In WebRTC streams (or media tracks in their world) can be grouped
together using the mslabel. This informs the browser that each
should be synchronized with each other.
This change extends the stream API so this information can
be stored with streams. The PJSIP support has been extended
to use the mslabel to determine grouped streams and store
this association on the streams. Finally when creating the
SDP the group information is used to cause each media stream
to use the same mslabel.
Adds an extra option, --asterisk-bin=<path> to ast_coredumper. If
provided, the binary given to gdb will be the parameter, rather than
asterisk from the PATH.
Corey Farrell [Mon, 30 Oct 2017 05:32:32 +0000 (01:32 -0400)]
Modules: Fix issues with CLI completion.
* Stop using ast_module_helper to check if a module is loaded, use
ast_module_check instead (app_confbridge and app_meetme).
* Stop ast_module_helper from listing reload classes when needsreload
was not requested.
Richard Mudgett [Wed, 25 Oct 2017 19:38:19 +0000 (14:38 -0500)]
codec.c: Defensively check the returned samples.
Earlier versions of the codec_opus samples_count callback can return
negative error values on undecodable frames. This resulted in a divide by
zero exception.
* Added a defensive check in ast_codec_samples_count() for a "negative"
samples count return value. Log the event and set the count to zero.
Joshua Colp [Tue, 24 Oct 2017 15:33:57 +0000 (15:33 +0000)]
res_pjsip: Add 'ip' as a valid option to 'identify_by' on endpoint.
When the identify_by option on an endpoint is set to ip it will
only be identified using the res_pjsip_endpoint_identifier_ip module.
This ensures that it is not mistakenly matched using the username of
the From header. To ensure behavior has not changed the default has
been changed to "username,ip" for the identify_by option.
George Joseph [Wed, 25 Oct 2017 17:26:06 +0000 (11:26 -0600)]
ast_coredumper: Add gzipping of binaries and display of signal info
The --tarball-coredump option now creates a gzipped tarball of
coredumps processed, their results txt files and copies of
/etc/os-release, /usr/sbin/asterisk, /usr/lib(64)/libasterisk* and
/usr/lib(64)/asterisk as those files are needed to properly examine
the coredump. The file will be named
/tmp/asterisk.<timestamp>.coredumps.tar.gz or
/tmp/asterisk-<uniqueid>.coredumps.tar.gz if --tarball-uniqueid was
specified.
Added dumps of *_siginfo to the top of the txt files so you can
tell what signal was invoked.
Ben Ford [Wed, 25 Oct 2017 14:23:55 +0000 (09:23 -0500)]
http.c: Fix http header send content.
Currently ast_http_send barricades a portion of the content that
needs to be sent in order to establish a connection for things
like the ARI client. The conditional and contents have been changed
to ensure that everything that needs to be sent, will be sent.
Corey Farrell [Thu, 30 Mar 2017 14:51:14 +0000 (10:51 -0400)]
Build System: Fix --disable-xmldoc option.
The configure option to disable XML documentation does not currently
work. This patch makes it effective, but also causes an ABI change by
removing the ast_xmldoc_* symbols. Disabling xmldoc also prevents docs
from being automatically generated, but they can still be manually
generated with 'make doc/core-en_US.xml'.
Corey Farrell [Tue, 24 Oct 2017 14:43:15 +0000 (10:43 -0400)]
chan_sip: Fix SUBSCRIBE with missing "Expires" header.
When chan_sip receives a SUBSCRIBE request with no "Expires" header it
processes the request as an unsubscribe. This is incorrect, per RFC3264
when the "Expires" header is missing a default expiry should be used.
Corey Farrell [Mon, 23 Oct 2017 17:42:37 +0000 (13:42 -0400)]
test_config: Fix failure and segfault when config_hook is run twice.
On second run the config_hook test was unexpectedly failing to load
test_config.conf because it was still unmodified since the last load.
This is fixed by not passing CONFIG_FLAG_FILEUNCHANGED for the initial
loads, only using it when we are tested that a reload of unmodified
files do not initiate the hook.
George Joseph [Mon, 23 Oct 2017 17:23:04 +0000 (11:23 -0600)]
res_pjsip_sdp_rtp: Fix setting of address type for rtp_ipv6
create_outgoing_sdp_stream was setting "addr_type = STR_IP6" only
when an ipv6 media_address was specified on the endpoint. If
rtp_ipv6 was set and ast_sip_get_host_ip_string returned an ipv6
address, we were leaving the addr_type set at the default of
STR_IP4. This caused the address type to be set incorrectly on the
"o" and "c" SDP attributes even though the address was set
correctly. Some clients don't like the mismatch.
* Removed the test for endpoint/media_address and now check all
addresses for ipv6.
Joshua Colp [Sun, 22 Oct 2017 22:32:20 +0000 (19:32 -0300)]
res_xmpp: Ensure the connection filter is available.
Users of the API that res_xmpp provides expect that a
filter be available on the client at all times. When
OAuth authentication support was added this requirement
was not maintained.
This change merely moves the OAuth authentication to
after the filter is created, ensuring users of res_xmpp
can add things to the filter as needed.
Corey Farrell [Wed, 18 Oct 2017 18:41:25 +0000 (14:41 -0400)]
res_pjsip_pubsub: Prevent unload except during shutdown.
Prevent unload of the module as certain pjsip initialization functions
cannot be reversed. This required a reorder of the module_load so that
the non-reversable pjsip functions are not called until all potential
errors have been ruled out.
Corey Farrell [Wed, 18 Oct 2017 17:04:29 +0000 (13:04 -0400)]
chan_sip: Fix output of 'sip set debug off'.
When sip.conf contains 'sipdebug=yes' it is impossible to disable it
using CLI 'sip set debug off'. This corrects the output of that CLI
command to instruct the user to turn sipdebug off in the configuration
file.
Matt Jordan [Fri, 6 Oct 2017 15:51:17 +0000 (10:51 -0500)]
res_corosync: Fix linking issue with Corosync 2.x
At some point in time in the history of Corosync (certainly within the
2.x branch), the corosync_cfg_state_track function was removed.
Unfortunately, the cfg library is only linked if this function is
present. Without the cfg library being linked to res_corosync, loading
of res_corosync will fail.
This patch makes it so that detecting corosync's core libraries,
determined by the COROSYNC external library checks, links both the cpg
and cfg libraries with res_corosync.
Joshua Colp [Sat, 14 Oct 2017 19:41:37 +0000 (19:41 +0000)]
bridge_softmix: Reduce topology cloning and improve renegotiation.
As channels join and leave an SFU the bridge_softmix module
needs to renegotiate to add and remove their streams from
the other participants. Previously this was done by constructing
the ideal stream topology every time but in the case of leave
this was incomplete.
This change makes it so bridge_softmix keeps an ideal stream
topology for each channel and uses it when making changes. This
ensures that when we request a renegotiation we are always
certain that we are aiming for the best stream topology
possible. In the case of a channel leaving this ensures that
we try to have an existing participant fill their place if
a participant has a fixed limit on the maximum number of video
streams they allow.
Richard Mudgett [Mon, 2 Oct 2017 22:42:48 +0000 (17:42 -0500)]
cdr.c: Add container to key off of Party B channel names.
The CDR performance gets worse the further it gets behind in processing
stasis messages. One of the reasons is because of a n*m loop used when
processing Party B information.
* Added a new CDR container that is keyed to Party B so we don't need such
a large loop when processing Party B information.
NOTE: To reduce the size of the patch I deferred to another patch the
renaming of the Party A active_cdrs_by_channel container to
active_cdrs_master and renaming the container's hash and cmp functions
appropriately.
Torrey Searle [Wed, 11 Oct 2017 11:04:12 +0000 (13:04 +0200)]
contrib/script/sip_to_pjsip: implement 'all' for allow/disallow
when 'all' is specified in an allow or disallow section, it should erase
all values from the inverse section in the default config. E.G.
allow=all should erase any deny values from default config &
vice-versa
Guido Falsi [Sat, 14 Oct 2017 09:11:24 +0000 (11:11 +0200)]
chan_dahdi: wrap include file which is not present on BSD systems in #ifdef
The sys/sysmacros.h include file does not exist in BSD systems and
is not required to build this module there.
Since an "#if defined(__NetBSD__) || defined(__FreeBSD__)" section
already exist I moved that include line inside it's #else branch.
Alexander Traud [Fri, 13 Oct 2017 14:43:12 +0000 (16:43 +0200)]
res_pjsip_session: Rewrite o= with external_media_address.
PJSIP allows a domain name as external_media_address. This allows chan_pjsip to
be used behind a NAT with changing IP addresses. The IP address of that domain
is resolved to the c= line already. This change sets also the o= line to that
domain.
When making channels compatible the bridge_simple module
will renegotiate one to better match the other. Some
endpoints incorrectly terminate the call if this process
fails.
To better handle this scenario the audio streams present
on the new requested topology will include any existing
negotiated formats that happen to exist on the first
valid audio stream. This ensures formats are persent that
are known to be acceptable to the remote endpoint.
projects / thirdparty / asterisk.git / log
