AST-2012-014: crashes due to large stack allocations in TCP;
affects remote unauthenticated SIP *over TCP* and remote authenticated
XMPP/HTTP connections.
AST-2012-015: DoS through resource consumption by exploiting device
state caching; exploitable if anonymous calls are permitted.
- while there, revise pbx_spool.c kevent timeout fix; rather than
clamping the timestamp, in the particular problem situation we hit
the loop (where dirlist is empty), pass in NULL rather than
INT_MAX-timenow similar to what's done in the inotify case.
Note: this port may be removed in the future; users are recommended to
migrate to ConfBridge, which is part of Asterisk itself and has improved
greatly in the rewrite for Asterisk 10.x.
- Fix channel reference leak in ChanSpy.
- dsp.c: Fix multiple issues when no-interdigit delay is present,
and fast DTMF 50ms/50ms.
- Fix bug where final queue member would not be removed from memory.
- Fix memory leak when CEL is successfully written to PostgreSQL database.
- Fix DUNDi message routing bug when neighboring peer is unreachable.
- 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.
Sofia-SIP is an open-source SIP User-Agent library, compliant with the
IETF RFC3261 specification (see the feature table). It can be used as a
building block for SIP client software for uses such as VoIP, IM, and
many other real-time and person-to-person communication services.
ok sthen@
- RTP port exhaustion (DoS) if an endpoint responds to SIP INVITEs with
provisional responses but never sends a final response.
- double free with simultaneous access to a single voicemail account.
HAVE_SCHED_YIELD should be set since we have sched_yield(). The
sched_yield.h compatibility wrapper then does not try to create a
fallback macro which conflicts with the prototype of the syscall
wrapper.
While here, add "ac_cv_header_uuid_uuid_h=no" alongside
"ac_cv_lib_uuid_uuid_generate=no" to fix:
configure: WARNING: uuid/uuid.h: accepted by the compiler, rejected by the preprocessor!
OK fgsch@
AST-2012-007, AST-2012-008 fixed in the short-lived 1.8.12.1 release:
* A remotely exploitable crash vulnerability exists in the IAX2 channel
driver if an established call is placed on hold without a suggested music
class. Asterisk will attempt to use an invalid pointer to the music
on hold class name, potentially causing a crash.
* A remotely exploitable crash vulnerability was found in the Skinny (SCCP)
Channel driver. When an SCCP client closes its connection to the server,
a pointer in a structure is set to NULL. If the client was not in the
on-hook state at the time the connection was closed, this pointer is later
dereferenced. This allows remote authenticated connections the ability to
cause a crash in the server, denying services to legitimate users.
Also from 1.8.12.2
* Resolve crash in subscribing for MWI notifications.
ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the
variable should definitely not be used after that. To solve this in
the two cases that affect subscribing for MWI notifications, we
instead save the ref locally, and unref them in the error
conditions.
- add an extra file to PLIST-calendar
- add comments to the sample sip.conf showing how to hide version numbers
- fix use of _POSIX_THREAD_PRIORITY_SCHEDULING, from Brad
