Asterisk News

Asterisk Releases

Asterisk 11.21.0-rc3 Now Available

Jan 12, 2016

The Asterisk Development Team has announced the release of Asterisk 11.21.0-rc3.

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

The release of Asterisk 11.21.0-rc3 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-25640] - pbx: Deadlock on features reload and state change hint.

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0-rc3

Thank you for your continued support of Asterisk!


Asterisk 13.7.0-rc2 Now Available

Dec 18, 2015

The Asterisk Development Team has announced the release of Asterisk 13.7.0-rc2.

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

The release of Asterisk 13.7.0-rc2 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 candidate:

Bug

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0-rc2

Thank you for your continued support of Asterisk!


Asterisk 13.7.0-rc1 Now Available

Dec 15, 2015

The Asterisk Development Team has announced the release of Asterisk 13.7.0-rc1.

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

The release of Asterisk 13.7.0-rc1 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 candidate:

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-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-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

Improvement

New Feature

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0-rc1

Thank you for your continued support of Asterisk!


Asterisk 11.21.0-rc1 Now Available

Dec 15, 2015

The Asterisk Development Team has announced the release of Asterisk 11.21.0-rc1.

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

The release of Asterisk 11.21.0-rc1 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 candidate:

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

Improvement

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

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0-rc1

Thank you for your continued support of Asterisk!


Asterisk 13.6.0 Now Available

Oct 9, 2015

The Asterisk Development Team has announced the release of Asterisk 13.6.0.

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

The release of Asterisk 13.6.0 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-25185] - Segfault in app_queue on transfer scenarios
  • [ASTERISK-25215] - Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember
  • [ASTERISK-25227] - No audio at in-band announcements in ooh323 channel
  • [ASTERISK-25265] - [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1
  • [ASTERISK-25271] - Parking & blind transfer: Transferer channel not hung up if no MOH
  • [ASTERISK-25292] - Testuite: tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
  • [ASTERISK-25295] - res_pjsip crash - pjsip_uri_get_uri at /usr/include/pjsip/sip_uri.h
  • [ASTERISK-25296] - RTP performance issue with several channel drivers.
  • [ASTERISK-25297] - Crashes running channels/pjsip/resolver/srv/failover/in_dialog testsuite tests
  • [ASTERISK-25299] - RTP port leaks with incoming OOH323 calls
  • [ASTERISK-25304] - res_pjsip: XML sanitization may write past buffer
  • [ASTERISK-25305] - Dynamic logger channels can be added multiple times
  • [ASTERISK-25306] - Persistent subscriptions can save multiple SIP messages at once, leading to potential crashes.
  • [ASTERISK-25309] - [patch] iLBC 20 advertised
  • [ASTERISK-25312] - res_http_websocket: Terminate connection on fatal cases
  • [ASTERISK-25315] - DAHDI channels send shortened duration DTMF tones.
  • [ASTERISK-25318] - tests/rest_api/applications/subscribe-endpoint/nominal/resource: Sporadically failing
  • [ASTERISK-25320] - chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite
  • [ASTERISK-25322] - Crash occurs when using MixMonitor with t() or r() options.
  • [ASTERISK-25325] - ARI PUT reload chan_sip HTTP response 404
  • [ASTERISK-25339] - res_pjsip: Empty "auth" sections from non-config backgrounds are interpreted as valid
  • [ASTERISK-25341] - bridge: Hangups may get lost when executing actions
  • [ASTERISK-25342] - res_pjsip: Repeated usage of pj_gethostip may block
  • [ASTERISK-25346] - chan_sip: Overwriting answered elsewhere hangup cause on call pickup
  • [ASTERISK-25353] - [patch] Transcoding while different in Frame size = Frames lost
  • [ASTERISK-25355] - sched: ast_sched_del may return prematurely due to spurious wakeup
  • [ASTERISK-25356] - res_pjsip_sdp_rtp: Multiple keepalive scheduled items may exist
  • [ASTERISK-25362] - Deadlock due to presence state callback
  • [ASTERISK-25365] - Persistent subscriptions have extra Content-Length/corrupted messages
  • [ASTERISK-25367] - pbx: Long pattern match hints may cause "core show hints" to crash
  • [ASTERISK-25369] - res_parking: ParkAndAnnounce - Inheritable variables aren't applied to the announcer channel
  • [ASTERISK-25381] - res_pjsip: AoRs deleted via ARI (or other mechanism) do not destroy their related contacts
  • [ASTERISK-25383] - Core dumps on startup and shutdown with MALLOC_DEBUG enabled
  • [ASTERISK-25384] - Regular Asterisk crashes when using Page application. "user_data is NULL"
  • [ASTERISK-25387] - res_pjsip_nat: Malformed REGISTER request causes NAT'd Contact header to not be rewritten
  • [ASTERISK-25390] - default_from_user can crash with certain configuration backends
  • [ASTERISK-25394] - pbx: Incorrect device and presence state when changing hint details
  • [ASTERISK-25396] - chan_sip: Extremely long callerid name causes invalid SIP
  • [ASTERISK-25399] - app_queue: AgentComplete event has wrong reason
  • [ASTERISK-25407] - Asterisk fails to log to multiple syslog destinations
  • [ASTERISK-25410] - app_record: RECORDED_FILE variable not being populated
  • [ASTERISK-25418] - On-hold channels redirected out of a bridge appear to still be on hold
  • [ASTERISK-25423] - Caller gets no Connected line update during call pickup.
  • [ASTERISK-25438] - res_rtp_asterisk: ICE role message even when ICE is not enabled
  • [ASTERISK-25449] - main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny

Improvement

  • [ASTERISK-24870] - ARI: Subscriptions to bridges generally not super useful
  • [ASTERISK-25310] - [patch]on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED

New Feature

  • [ASTERISK-25252] - ARI: Add the ability to manipulate log channels
  • [ASTERISK-25377] - res_pjsip: Change default "From user" from UUID to something more palatable

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.6.0

Thank you for your continued support of Asterisk!


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