AST-2018-002: Crash when given an invalid SDP media format description
AST-2018-003: Crash with an invalid SDP fmtp attribute
AST-2018-004: Crash when receiving SUBSCRIBE request
AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
AST-2018-006: WebSocket frames with 0 sized payload causes DoS
(only 15.x reported as affected by AST-2018-001)
BIO_meth_set_callback_ctrl(). however in this case it is just setting the
callback pointer to NULL, and BIO_meth_new() already returns zeroed space,
so just skip that for now.
Fix #! line for bash in astversion. Not forced in RUN_DEPENDS because in all
the time this has been present only one person noticed, so the script doesn't
seem too popular. (Script also makes some assumptions about library versions
which don't apply here but I don't think it's worth poking at this too far).
Reported by landry@
(other codecs are already disabled in the pjproject build, it's only used for sip
signalling - asterisk has its own codec stack, this doesn't affect use of the
asterisk-g729 package).
AST-2017-012: Remote Crash Vulnerability in RTCP Stack
If a compound RTCP packet is received containing more than one report
(for example a Receiver Report and a Sender Report) the RTCP stack
will incorrectly store report information outside of allocated memory
potentially causing a crash.
AST-2017-014: Crash in PJSIP resource when missing a contact header
A select set of SIP messages create a dialog in Asterisk. Those SIP
messages must contain a contact header. For those messages, if the
header was not present and using the PJSIP channel driver, it would
cause Asterisk to crash. The severity of this vulnerability is somewhat
mitigated if authentication is enabled. If authentication is enabled a
user would have to first be authorized before reaching the crash point.
to AST-2017-005.
The RTP/RTCP stack will now validate RTCP packets before processing
them. Packets failing validation are discarded. RTP stream qualification
now requires the intended series of packets from the same address
without seeing packets from a different source address to accept a new
source address.
AST-2017-005: Media takeover in RTP stack
AST-2017-006: Shell access command injection in app_minivm
AST-2017-007: Remote Crash Vulerability in res_pjsip
also install the basic-pbx sample configs
which are gcc-specific, or clang with -fblocks, which we don't have
working fully yet).
To avoid a C++ standard library conflict, switch to a stripped-down and
patched copy of pjsua/pjsip built as part of the Asterisk build.
Some slight patch gymnastics; Asterisk doesn't distribute pjsua itself
but rather normally downloads, untars and patches as part of the build,
which isn't compatible with the patches we need to apply in order to
fix it with libressl.
