Asterisk News

Asterisk Releases

Asterisk 13.7.2 Now Available

Feb 5, 2016

The Asterisk Development Team has announced the release of Asterisk 13.7.2.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.7.2 resolves an issue reported by the community and would have not been possible without your participation.

Thank you!

The following is the issue resolved in this release:

Bug

  • [ASTERISK-25702] - PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.2

Thank you for your continued support of Asterisk!


Asterisk 11.6-cert12, 11.21.1, 13.1-cert3, 13.7.1 Now Available (Security Release)

Feb 3, 2016

The Asterisk Development Team has announced security releases for Certified Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of these versions resolves the following security vulnerabilities:

  • AST-2016-001: BEAST vulnerability in HTTP server
    The Asterisk HTTP server currently has a default configuration which allows the BEAST vulnerability to be exploited if the TLS functionality is enabled. This can allow a man-in-the-middle attack to decrypt data passing through it.
  • AST-2016-002: File descriptor exhaustion in chan_sip
    Setting the sip.conf timert1 value to a value higher than 1245 can cause an integer overflow and result in large retransmit timeout times. These large timeout values hold system file descriptors hostage and can cause the system to run out of file descriptors.
  • AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data
    If no UDPTL packets are lost there is no problem. However, a lost packet causes Asterisk to use the available error correcting redundancy packets. If those redundancy packets have zero length then Asterisk uses an uninitialized buffer pointer and length value which can cause invalid memory accesses later when the packet is copied.

For a full list of changes in the current releases, please see the ChangeLogs:

The security advisories are available at:

Thank you for your continued support of Asterisk!


Asterisk 13.7.0 Now Available

Jan 15, 2016

The Asterisk Development Team has announced the release of Asterisk 13.7.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.7.0 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release:

Bug

  • [ASTERISK-7803] - [patch] Update the maximum packetization values in frame.c
  • [ASTERISK-24106] - WebSockets Automatically decides what driver it will use
  • [ASTERISK-24146] - [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec
  • [ASTERISK-24543] - Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs
  • [ASTERISK-24779] - Passthrough OPUS codec not working with chan_pjsip
  • [ASTERISK-24958] - Forwarding loop detection inhibits certain desirable scenarios
  • [ASTERISK-25116] - res_pjsip: Two PeerStatus AMI messages are sent for every status change
  • [ASTERISK-25135] - [patch]RTP Timeout hangup cause code missing
  • [ASTERISK-25160] - [patch] Opus Codec: SIP/SDP line fmtp missing when called internally
  • [ASTERISK-25165] - Testsuite - Sorcery memory cache leaks
  • [ASTERISK-25364] - [patch]Issue a TCP connection(kernel) and thread of asterisk is not released
  • [ASTERISK-25373] - add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants
  • [ASTERISK-25391] - AMI GetConfigJSON returns invalid JSON
  • [ASTERISK-25400] - Hints broken when "CustomPresence" doesn't exist in AstDB
  • [ASTERISK-25404] - segfault/crash in chan_pjsip_hangup ... at chan_pjsip.c
  • [ASTERISK-25434] - Compiler flags not reported in 'core show settings' despite usage during compilation
  • [ASTERISK-25435] - Asterisk periodically hangs. UDP Recv-Q greatly exceeds zero.
  • [ASTERISK-25438] - res_rtp_asterisk: ICE role message even when ICE is not enabled
  • [ASTERISK-25441] - Deadlock in res_sorcery_memory_cache.
  • [ASTERISK-25443] - [patch]IPv6 - Potential issue in via header parsing
  • [ASTERISK-25449] - main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny
  • [ASTERISK-25451] - Broken video - erased rtp marker bit
  • [ASTERISK-25455] - Deadlock of PJSIP realtime over res_config_pgsql
  • [ASTERISK-25461] - Nested dialplan #includes don't work as expected.
  • [ASTERISK-25476] - chan_sip loses registrations after a while
  • [ASTERISK-25484] - [patch] autoframing=yes has no effect
  • [ASTERISK-25485] - res_pjsip_outbound_registration: registration stops due to 400 response
  • [ASTERISK-25486] - res_pjsip: Fix deadlock when validating URIs
  • [ASTERISK-25494] - build: GCC 5.1.x catches some new const, array bounds and missing paren issues
  • [ASTERISK-25498] - Asterisk crashes when negotiating g729 without that module installed
  • [ASTERISK-25505] - res_pjsip_pubsub: Crash on off-nominal when UAS dialog can't be created
  • [ASTERISK-25513] - Crash: malloc failed with high load of subscriptions.
  • [ASTERISK-25522] - ARI: Crash when creating channel via ARI originate with requesting channel
  • [ASTERISK-25527] - Quirky xmldoc description wrapping
  • [ASTERISK-25533] - [patch] buffer for ast_format_cap_get_names only 64 bytes
  • [ASTERISK-25535] - [patch] format creation on module load instead of cache
  • [ASTERISK-25537] - [patch] format-attribute module: RFC or internal defaults?
  • [ASTERISK-25545] - [patch] translation module gets cached not joint format
  • [ASTERISK-25546] - threadpool: Race condition between idle timeout and activation
  • [ASTERISK-25552] - hashtab: Improve NULL tolerance
  • [ASTERISK-25561] - app_queue.c line 6503 (try_calling): mutex 'qe->chan' freed more times than we've locked!
  • [ASTERISK-25569] - app_meetme: Audio quality issues
  • [ASTERISK-25573] - [patch] H.264 format attribute module: resets whole SDP
  • [ASTERISK-25575] - res_pjsip: Dynamic outbound registrations created via ARI are not loaded into memory on Asterisk start/restart
  • [ASTERISK-25582] - Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38
  • [ASTERISK-25583] - [patch] format-attribute module: RFC 7587 (Opus Codec)
  • [ASTERISK-25584] - [patch] format-attribute module: VP8 missing
  • [ASTERISK-25585] - [patch]rasterisk never hits most of main(), but it's assumed to
  • [ASTERISK-25590] - CLI Usage info for 'pjsip send notify' references incorrect config
  • [ASTERISK-25593] - fastagi: record file closed after sending result
  • [ASTERISK-25595] - Unescaped : in messge sent to statsd
  • [ASTERISK-25598] - res_pjsip: Contact status messages are printing a hash instead of the uri
  • [ASTERISK-25599] - [patch] SLIN Resampling Codec only 80 msec
  • [ASTERISK-25600] - bridging: Inconsistency in BRIDGEPEER
  • [ASTERISK-25601] - json: Audit reference usage and thread safety
  • [ASTERISK-25608] - res_pjsip/contacts/statsd: Lifecycle events aren't consistent
  • [ASTERISK-25609] - [patch]Asterisk may crash when calling ast_channel_get_t38_state(c)
  • [ASTERISK-25610] - Asterisk crash during "sip reload"
  • [ASTERISK-25615] - res_pjsip: Setting transport async_operations > 1 causes segfault on tls transports
  • [ASTERISK-25616] - Warning with a Codec Module which supports PLC with FEC
  • [ASTERISK-25619] - res_chan_stats not sending the correct information to StatsD
  • [ASTERISK-25625] - res_sorcery_memory_cache: Add full backend caching
  • [ASTERISK-25640] - pbx: Deadlock on features reload and state change hint.
  • [ASTERISK-25664] - ast_format_cap_append_by_type leaks a reference
  • [ASTERISK-25689] - pjsip show contacts not working in Asterisk 13.7rc2

Improvement

New Feature

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0

Thank you for your continued support of Asterisk!


Asterisk 11.21.0 Now Available

Jan 15, 2016

The Asterisk Development Team has announced the release of Asterisk 11.21.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.21.0 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release:

Bug

  • [ASTERISK-7803] - [patch] Update the maximum packetization values in frame.c
  • [ASTERISK-24146] - [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec
  • [ASTERISK-25135] - [patch]RTP Timeout hangup cause code missing
  • [ASTERISK-25364] - [patch]Issue a TCP connection(kernel) and thread of asterisk is not released
  • [ASTERISK-25373] - add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants
  • [ASTERISK-25391] - AMI GetConfigJSON returns invalid JSON
  • [ASTERISK-25400] - Hints broken when "CustomPresence" doesn't exist in AstDB
  • [ASTERISK-25434] - Compiler flags not reported in 'core show settings' despite usage during compilation
  • [ASTERISK-25438] - res_rtp_asterisk: ICE role message even when ICE is not enabled
  • [ASTERISK-25443] - [patch]IPv6 - Potential issue in via header parsing
  • [ASTERISK-25449] - main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny
  • [ASTERISK-25455] - Deadlock of PJSIP realtime over res_config_pgsql
  • [ASTERISK-25461] - Nested dialplan #includes don't work as expected.
  • [ASTERISK-25476] - chan_sip loses registrations after a while
  • [ASTERISK-25494] - build: GCC 5.1.x catches some new const, array bounds and missing paren issues
  • [ASTERISK-25498] - Asterisk crashes when negotiating g729 without that module installed
  • [ASTERISK-25527] - Quirky xmldoc description wrapping
  • [ASTERISK-25537] - [patch] format-attribute module: RFC or internal defaults?
  • [ASTERISK-25552] - hashtab: Improve NULL tolerance
  • [ASTERISK-25569] - app_meetme: Audio quality issues
  • [ASTERISK-25585] - [patch]rasterisk never hits most of main(), but it's assumed to
  • [ASTERISK-25593] - fastagi: record file closed after sending result
  • [ASTERISK-25599] - [patch] SLIN Resampling Codec only 80 msec
  • [ASTERISK-25609] - [patch]Asterisk may crash when calling ast_channel_get_t38_state(c)
  • [ASTERISK-25610] - Asterisk crash during "sip reload"
  • [ASTERISK-25616] - Warning with a Codec Module which supports PLC with FEC
  • [ASTERISK-25640] - pbx: Deadlock on features reload and state change hint.

Improvement

  • [ASTERISK-24718] - [patch]Add inital support of "sanitize" to configure

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0

Thank you for your continued support of Asterisk!


Asterisk 13.7.0-rc3 Now Available

Jan 12, 2016

The Asterisk Development Team has announced the release of Asterisk 13.7.0-rc3.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.7.0-rc3 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release:

Bug

  • [ASTERISK-25640] - pbx: Deadlock on features reload and state change hint.
  • [ASTERISK-25664] - ast_format_cap_append_by_type leaks a reference
  • [ASTERISK-25689] - pjsip show contacts not working in Asterisk 13.7rc2

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0-rc3

Thank you for your continued support of Asterisk!


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