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Checking option_debug directly is incorrect as it ignores file/module
specific debug settings. This system-wide change replaces nearly all
direct checks for option_debug with the DEBUG_ATLEAST macro.
Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0
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ASTERISK-27716
Change-Id: I52525e35e1620341272219911d054a1e3d3ec01e
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ASTERISK-27715
Reported by: Corey Farrell
Change-Id: I0d6d9572d1352dc7ad30c9917173f1e980d8c938
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ASTERISK-27714
Reported by: John Nemeth
Change-Id: I1b84a89315a5f61222123d21bf35c59224da8990
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ASTERISK-27713
Change-Id: I90c769e3c7f8c26de8a3af11335862cec15a1b22
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ASTERISK-27712
Reported by: Joerg Sonnenberger, D'Arcy Cain
Change-Id: Idf1c9d43617a3e13028b95b313415903d80ef807
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ASTERISK-27711
Change-Id: Idc9194035b2958b99f6b01eb5b438d45a074565b
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ASTERISK-27710
Change-Id: I4c777e41b31d4415bbe21cb435ad47b43ebb5467
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ASTERISK-27709
Reported by: John Nemeth
Change-Id: I11b1ae8fd404c04066f1458f5d71f9536359d58d
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The pool cache gets in the way of finding use after free errors of memory
pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a
pool is released because it gets put into the cache instead of being
freed.
* Added the "cache_pools" option to pjproject.conf. Disabling the option
helps track down pool content mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the pool
contents are used after free and who freed it.
To disable the pool caching simply disable the cache_pools option in
pjproject.conf and restart Asterisk.
Sample pjproject.conf setting:
[startup]
cache_pools=no
* Made current users of the caching pool factory initialization and
destruction calls call common routines to create and destroy cached pools.
ASTERISK-27704
Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828
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Change-Id: I459dc320a8c9452a01eed6f403d786741587c890
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Between Asterisk 11 and Asterisk 13 there was a significant increase
in the number of AST_FRAME_NULL frames being processed by app_amd.c's
main loop. Each AST_FRAME_NULL frame was being counted as 100ms
towards the total time and silence. This may have been accurate
when app_amd.c was orginally added, but it is not in Asterisk 13.
As such the total analysis time and silence calculations were way
off effectively breaking app_amd.c
* Additional debug messages were added
* AST_FRAME_NULL are now ignored
ASTERISK-27610
Change-Id: I18aca01af98f87c1e168e6ae0d85c136d1df5ea9
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* Fix --tarball-config so the option doesn't cause an error.
* Allow for missing /etc/os-release.
* Add a sleep between tarballing the coredump and removing the
output directory to allow the filesystem to settle.
Change-Id: I73e03b13087978bcc7f6bc9f45753990f82d9d77
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A couple of additional properties are needed in rtp_engine to enable
support for packet retransmission: AST_RTP_PROPERTY_RETRANS_RECV and
AST_RTP_PROPERTY_RETRANS_SEND. These will both be enabled automatically
if an endpoint has the webrtc option enabled. While this adds no
functionality currently, it will serve as a building block for future
changes for RTP retransmission support.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
Change-Id: Ic598acd042a045f9d10e5bdccb66f4efc9e587cc
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When a line is the maximum length "\n" is found at sizeof(buf) - 2 since
the last character is actually the null terminator. In addition if a
line was exactly 8190 plus a multiple of 8192 characters long the config
parser would skip the following line.
Additionally fix comment in voicemail.conf sample config. It previously
stated that emailbody can only contain up to 512 characters which is
always wrong. The buffer is normally 8192 characters unless LOW_MEMORY
is enabled then it is 512 characters. The updated comment states that
the line can be up to 8190 or 510 characters since the line feed and
NULL terminator each use a character.
ASTERISK-26688 #close
Change-Id: I80864a0d40d2e2d8cd79d72af52a8f0a3a99c015
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The transferrer's session channel was destroyed by the transferrer's
serializer thread in a race condition with the transfer target's
serializer thread during an attended transfer. The transfer target's
serializer was attempting to clean up a deferred end status on behalf of
the transferrer's channel when it should have passed the action to the
transferrer's serializer. When the transfer target's serializer lost the
race then both threads wind up trying to end the transferrer's session.
* Push the ast_sip_session_end_if_deferred() call onto the transferrer's
serializer to avoid a race condition that results in a crash. The
session_end() function that could be called by
ast_sip_session_end_if_deferred() really must be executed by the
transferrer's serializer to avoid this kind of crash.
ASTERISK-27568
Change-Id: Iacda724e7cb24d7520e49b2fd7e504aa398d7238
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deleted
autoconf gcc libsamplerate sqlite
changed
binutils to libbfd
freetds-devel to freetds
gmime2 to gmime26
mysql55-client to mysql57-client
added
alsa-lib bison bzip2 cclient corosync doxygen libedit flex graphviz
libhoard libical libilbc libltdl lua neon newt net-snmp
openldap-client openssl patch pkgconf portaudio postgresql10-client
python radcli speexdsp subversion uriparser xmlstarlet libzip
ASTERISK-27686
Change-Id: Ibe88c9b26e59c30d26cdb313a3ef01c9f37ac80d
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Change-Id: Id52f719078a65c4b2eee7ab99d761eba6b6aed94
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ASTERISK-27659
Change-Id: I04a2705d2cb7df250769967bc59e2b397a49b797
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The handling of stream topologies was not protected by channel locks in
simple_bridge_request_stream_topology_change().
* Fixed topology handling to be protected by channel locks where needed in
simple_bridge_request_stream_topology_change().
ASTERISK-27692
Change-Id: Ica5d78a6c7ecf4f0b95fb16de28d3889b32c4776
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passed up."
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In ast_websocket_read() we were not adequately checking that the
payload_len was non-zero before passing it to ws_safe_read(). Calling
ws_safe_read with a len argument of 0 will result in a busy loop until
the underlying socket is closed.
ASTERISK-27658 #close
Change-Id: I9d59f83bc563f711df1a6197c57de473f6b0663a
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pjproject's fmtp retrieval function failed to catch invalid fmtp attributes.
Because of this Asterisk would crash if given an SDP with an invalid fmtp
attribute.
When retrieving the format this patch now makes sure the fmtp attribute is
available. If not available it now returns an error status.
ASTERISK-27583 #close
Change-Id: I5cebe000ce2d846cae3af33b6d72c416e51caf2f
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pjproject's media format parsing algorithm failed to catch invalid values.
Because of this Asterisk would crash if given an SDP with a invalid media
format description.
When parsing the media format description this patch now properly parses the
value and returns an error status if it can't successfully parse/convert the
value.
ASTERISK-27582 #close
Change-Id: I883b3a4ef85b6972397f7b56bf46c5779c55fdd6
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