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
channel_state_invalid leaked a reference to the channel snapshot any
time it was aquired.
ASTERISK-27067 #close
Change-Id: I8c653f00416b39978513c5605c4be0f03b1df29a
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
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
The ps_endpoints table was missing the bundle column
introduced with the bundle feature in
commit 065c3005ad.
ASTERISK-27374 #close
Change-Id: Ic900f4f2c20f64b99ea898d50f5c0a7117472d46
* 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
Jenkins2
Kevin Harwell
Corey Farrell
Joshua Colp
Tzafrir Cohen
Florian Floimair
Igor Goncharovskiy
Richard Mudgett
George Joseph
Ben Ford
Alexander Traud