(core)
- Italian language prompts for core sounds
- missing prompts for VoiceMailMain application in Russian
(extra)
- various fixed files in extra-sounds
- various new files in extra-sounds (some in French, many in English)
- many files duplicated from core-sounds have been removed
- note: "an-error-has-occured" has been renamed "an-error-has-occurred"
Additionally the packages now install the text files detailing changes
and a text description of the font files (renamed to avoid conflicts
between flavours).
sln16 versions have been dropped from packages for now to save a
few hundred MB per arch on the mirrors, g729 versions have been added
in their place.
REVISION doesn't change the stem of the package name. Came after some head
scratching after naddy@ reported a PLIST_DB change to telephony/asterisk's
@depend lines which happened after I bumped REVISION on the asterisk-sounds
ports when I tweaked CATEGORIES.
This module provides an interface to the Asterisk Manager Interface.
Its goal is to provide a flexible, powerful, and reliable way to
interact with Asterisk upon which other applications may be built.
It utilizes AnyEvent and therefore can integrate very easily into
event-based applications, but still provides blocking functions for use
with standard scripting.
* A possible buffer overflow during H.264 (video) format negotiation.
CVE-2013-2685
* A denial of service exists in Asterisk's HTTP server.
CVE-2013-2686
* A potential username disclosure exists in the SIP channel driver.
CVE-2013-2264
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@
