Age | Commit message (Collapse) | Author |
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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.
ASTERISK-27387
Change-Id: Id80c10a644508365c87b3182e99ea49da11b0281
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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.
ASTERISK-27378
Change-Id: Iea3e00054461484196c46f688f02635cc886bad1
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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.
ASTERISK-27364
Change-Id: Icf84ca982a67cdd6e9a71851eb7eb1bd0e865276
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A comment was updated when I ran 'make ari-stubs'.
Change-Id: Ib5154ae3ad72aff53374c28ead540fe349c42175
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Change-Id: Ie083987e69dc43b6861671c218cacacc11b2072f
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into 15
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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.
ASTERISK-27369 #close
Change-Id: I22588e7125a765edf35cff28c98ca143e9927554
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channel_state_invalid leaked a reference to the channel snapshot any
time it was aquired.
ASTERISK-27067 #close
Change-Id: I8c653f00416b39978513c5605c4be0f03b1df29a
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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.
ASTERISK-27379
Change-Id: Id6299aa031efe46254edbdc7973c534d54d641ad
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into 15
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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.
ASTERISK-27380 #close
Change-Id: I25f5b91eb75059b0fb2f142e468c26b283b0a9f3
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The ps_endpoints table was missing the bundle column
introduced with the bundle feature in
commit 065c3005ad92.
ASTERISK-27374 #close
Change-Id: Ic900f4f2c20f64b99ea898d50f5c0a7117472d46
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* 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
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Fix typo, that specify usage wrong option 'dtmf-features' for CHANNEL() function
instead of correct 'dtmf_features'
ASTERISK-27377 #close
Change-Id: I15ecc829c1035b359584673e12cdb5c9291ac930
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When allocate_subscription fails to initialize fields of the new sub it
calls destroy_subscription.
Change-Id: I5b79c915ec216dc00c13c1e4172137864a4bec85
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into 15
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ASTERISK-27181
Change-Id: Ic4468b49860bd7f67e922baf4c9e96828c184d17
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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
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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
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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
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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
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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
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into 15
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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
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into 15
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ASTERISK-23556
Reported by: Marcello Ceschia
Change-Id: Ic27e88e0336a0d83877dc857938659dc5560b93c
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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
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Change-Id: Ibb3e47f27a395d74d8c5263db015b05434f5969b
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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
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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
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