changes aren't too extreme, but upgrading users should review upgrade notes
in /usr/local/share/doc/asterisk (UPGRADE-14.txt, UPGRADE-15.txt, UPGRADE.txt).
fail, reported by naddy@.
set COMPILER_LANGS=c while there, the GNU compiler is only used for C nested
functions (because I still have no ideas about the BlocksRuntime with clang),
c++ is not needed.
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)
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.
add various OPENSSL_VERSION_NUMBER patches now that asterisk supports
openssl 1.1:
- we don't have openssl 1.1's SSL_is_server yet, so use the old check
for ssl->server instead
- we do still need the hack to avoid initing multiple times which is
no longer needed in openssl 1.1
OPENSSL_VERSION_NUMBER < 0x10002000L to see if DTLSv1_method is available;
it's an error at runtime only as it's in a dlopen'd module, and doesn't
crash the process, just fails loading the module, so you don't notice
until you wonder why calls are all failing...)
this is a major update - upgrading users should review UPGRADE-12.txt and
UPGRADE.txt in /usr/local/share/doc/asterisk. some configurations will work
unchanged, but there have been big changes in other areas, notably AMI,
CDR and CEL.
- AST-2014-006: MixMonitor manager action allows arbitrary shell commands
to be called from AMI (management interface) users without having proper
permissions.
- AST-2014-007: add a timeout to mitigate possible DoS on http interface
(connecting but making no request ties up a connection)
- If using ConfBridge, note that the dialplan arguments have changed.
- If using the built-in HTTP server, note that a bindaddr must now be given,
previously the default was 0.0.0.0 but this must now be given explicitly.
- Internal database now uses SQLite3 not BDB, conversion tools are provided.
See share/doc/asterisk/UPGRADE.txt for more.
- strip core-sounds and moh out of the main asterisk package,
they change comparatively rarely.
- provide all available languages.
- provide multiple codecs for all files, replacing the asterisk-native-sounds
package which only provided ulaw versions of the asterisk 1.4 files, ports
laid out to permit parallel building.
- the old asterisk-sounds package providing additional sound files beyond
the core ones is now "extra-sounds" modelled after the filename of the
distributed files.
- includes the iLBC codec which now has a free copyright license; patent
licensing has a "no litigation" clause (see codecs/ilbc/LICENSE_ADDENDUM)
so mark as not permitted for CDs
when forming an outgoing SIP request while in pedantic mode, which
can cause a stack buffer to be made to overflow if supplied with
carefully crafted caller ID information"
http://downloads.asterisk.org/pub/security/AST-2011-001.html
This is also a major version update to the long-term support
1.8 branch, previous versions of this diff have been tested by
various ports@ readers, thanks for testing.
Please review /usr/local/share/doc/asterisk/UPGRADE.txt
(also note that memory use has increased).
ok ajacoutot@ jasper@
