The Asterisk Development Team has announced the first release candidate of Asterisk 11.17.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.17.0-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-15434] – When ast_pbx_start failed, both an error response and BYE are sent to the caller
- [ASTERISK-16779] – Cannot disallow unknown format ”
- [ASTERISK-17721] – Incoming SRTP calls that specify a key lifetime fail
- [ASTERISK-18105] – most of asterisk modules are unbuildable in cygwin environment
- [ASTERISK-18708] – func_curl hangs channel under load
- [ASTERISK-19470] – Documentation on app_amd is incorrect
- [ASTERISK-20850] – Nested functions aren’t portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
- [ASTERISK-21038] – Bad command completion of “core set debug channel”
- [ASTERISK-22436] – No BYE to masqueraded channel on INVITE with replaces
- [ASTERISK-23214] – chan_sip WARNING message ‘We are requesting SRTP for audio, but they responded without it’ is ambiguous and wrong in some cases
- [ASTERISK-23390] – NewExten Event with application AGI shows up before and after AGI runs
- [ASTERISK-24451] – chan_iax2: reference leak in sched_delay_remove
- [ASTERISK-24479] – Enable REF_DEBUG for module references
- [ASTERISK-24701] – Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information
- [ASTERISK-24724] – ‘httpstatus’ Web Page Produces Incomplete HTML
- [ASTERISK-24739] – – Out of files — call fails — numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules
- [ASTERISK-24742] – Fix ast_odbc_find_table function in res_odbc
- [ASTERISK-24772] – ODBC error in realtime sippeers when device unregisters under MariaDB
- [ASTERISK-24786] – – Asterisk terminates when playing a voicemail stored in LDAP
- [ASTERISK-24787] – – Microsoft exchange incompatibility for playing back messages stored in IMAP – play_message: No origtime
- [ASTERISK-24796] – Codecs and bucket schema’s prevent module unload
- [ASTERISK-24797] – bridge_softmix: G.729 codec license held
- [ASTERISK-24799] – make fails with undefined reference to SSLv3_client_method
- [ASTERISK-24800] – Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill
- [ASTERISK-24808] – res_config_odbc: Improper escaping of backslashes occurs with MySQL
- [ASTERISK-24814] – asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers
- [ASTERISK-24817] – init_logger_chain: unreachable code block
- [ASTERISK-24825] – Caller ID not recognized using Centrex/Distinctive dialing
- [ASTERISK-24828] – Fix Frame Leaks
- [ASTERISK-24838] – chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling
- [ASTERISK-24876] – Investigate reference leaks from tests/channels/local/local_optimize_away
- [ASTERISK-24879] – Compilation fails due to 64bit time under OpenBSD
- [ASTERISK-24880] – Compilation under OpenBSD
Improvement
- [ASTERISK-24790] – Reduce spurious noise in logs from voicemail – Couldn’t find mailbox %s in context
New Feature
- [ASTERISK-17899] – Adds a ‘ignorecryptolifetime’ (Ignore Crypto Lifetime) option to sip.conf for SRTP keys specifying optional ‘lifetime’
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0-rc1
Thank you for your continued support of Asterisk!