* 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.
ASTERISK-27378
Change-Id: Iaed8c1e4fcbeb242921dbac7929a0fe75ff4b239
Fix typo, that specify usage wrong option 'dtmf-features' for CHANNEL() function
instead of correct 'dtmf_features'
ASTERISK-27377 #close
Change-Id: I15ecc829c1035b359584673e12cdb5c9291ac930
When allocate_subscription fails to initialize fields of the new sub it
calls destroy_subscription.
Change-Id: I5b79c915ec216dc00c13c1e4172137864a4bec85
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.
ASTERISK-27194
Change-Id: Icf69350307ecbbc80a3d74de46af9bd80ea17819
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.
ASTERISK-27206
Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd
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.
Change-Id: Ib9ee6d83592d4b1bc90cb3419a05376a88d1ded9
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.
ASTERISK-27372
Change-Id: I8816d2d8f80f4fefc6dcae4b5fdfc97f1e46496d
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'.
ASTERISK-26639
Change-Id: Ifac562340c09f80c83e0203de098fcac93bf8c44
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.
ASTERISK-18140
Change-Id: Ibf6dcd4fdd07a32c2bc38be1dd557981f08188b5
A few places in hashtab use free instead of ast_free, remove declaration
of ASTMM_LIBC from hashtab.c as it's no longer needed.
Change-Id: I2ff089bad71640c03c3ce97f1b00fc962ef79427
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.
ASTERISK-25960
Change-Id: Ifd679509a23ed163e5cc647490bf7df4ae3cd856
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.
ASTERISK-27198
Reported by: Martin Cisárik
Change-Id: I5214fc31b728117842243807e7927a319cf77592
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.
ASTERISK-27346
Change-Id: I4ac474afe220e833288ff574e32e2b9a23394886
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.
ASTERISK-24483
Change-Id: Iee900f20bdd6ee1bfe23efdec0d87765eadce8a7
Prevent unload of the module as certain pjsip initialization functions
cannot be reversed.
ASTERISK-24483
Change-Id: I94597ec8b8491f5af9c57bf66dbc3b078fe2d49d
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.
ASTERISK-23462 #close
Change-Id: I1cceade9caa9578e1b060feb832e3495ef5ad318
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.
Change-Id: I674e9e1c8fea11c3bf81154aaa7c1fd43f945465
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.
ASTERISK-27354
Change-Id: I58070f421ddeadd2844a33b869b052630cf2e514
* Rename the Party A CDR container from active_cdrs_by_channel to
active_cdrs_master.
* Renamed the support functions associated with active_cdrs_master
appropriately.
ASTERISK-27335
Change-Id: I6104bb3edc3a0b7243ce502e45e8832b0cff14f7
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.
ASTERISK-27335
Change-Id: I0bf66e8868f8adaa4b5dcf9e682e34951c350249
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
ASTERISK-27333 #close
Change-Id: I99219478fb98f08751d769daaee0b7795118a5a6
