Asterisk 1.8.23.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.8.23.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.23.0 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:

  • [ASTERISK-17386] – res_config_ldap with malloc_debug produces munmap_chunk(): invalid pointer:
  • [ASTERISK-17436] – random deadlocks – SIP messages not being processed
  • [ASTERISK-17458] – Deadlocks when using pthread timer
  • [ASTERISK-17467] – external moh is blocked when using dahdi timer
  • [ASTERISK-18207] – externnotify script called with (null) context parameter during pollmessages run, essentially stopping it from running.
  • [ASTERISK-19431] – Asterisk Russian language support missing voicemail prompts
  • [ASTERISK-19754] – Deadlock in chan_sip / pthread_timing
  • [ASTERISK-19883] – – RTP packet with Timestamp=0 on Multicast paging
  • [ASTERISK-20225] – Segmentation Fault on manager_play_dtmf sip_senddigit_end
  • [ASTERISK-20577] – Asterisk deadlocks waiting for timer in res_timing_pthread while running AGI script
  • [ASTERISK-21069] – xmpp distributed device states aggregation update fails
  • [ASTERISK-21151] – ‘Squelching’ early media in DAHDI (sig_pri)
  • [ASTERISK-21164] – Need clarification on distributed device state behavior and whether this behavior is a possible regression
  • [ASTERISK-21225] – Setting nat=force_rport in [general] sip.conf will never work
  • [ASTERISK-21243] – Backport Appropiate NAT Setting Cleanups To 1.8
  • [ASTERISK-21246] – use of rtpkeepalive uses CN packet with marker bit set, plus a ULAW payload instead of CN
  • [ASTERISK-21302] – app_voicemail crashes on config error and there are some potential memory leaks
  • [ASTERISK-21329] – chan_alsa: patch for crash when audio device in unexpected state
  • [ASTERISK-21356] – Segfault during bridge channel proxy inspection in a masquerade caused by an AMI Redirect of two channels
  • [ASTERISK-21389] – res_timing_pthread fails to return from write, causing timer dependent operations to block indefinitely
  • [ASTERISK-21394] – – Fundamental changes to CDR within single asterisk family (1.8) during externally initiated blind transfers with an h extension present
  • [ASTERISK-21397] – manager crash on unloading app_queue
  • [ASTERISK-21407] – features_shutdown doesn’t finish cleanup
  • [ASTERISK-21409] – – Race condition with IAX2 transfer, 2 releases happen on same call legs. locks up with many threads blocked by iax2_destroy_helper
  • [ASTERISK-21412] – config.c/config_text_file_load() leaks globbuf
  • [ASTERISK-21429] – Distributed Device State using JABBER/XMPP not working since Secuity Advisory AST-2012-015
  • [ASTERISK-21466] – [crash] command (sip show peers) crashes Asterisk with ~3500 registered peers
  • [ASTERISK-21522] – DTMF end is not always processed, causes one-way audio
  • [ASTERISK-21664] – Asterisk terminates calls if Session-Expires isn’t present on INVITE
  • [ASTERISK-21677] – NOTIFYs for BLF start queuing up and fail to be sent out
  • [ASTERISK-21716] – logger thread sometimes exits with messages still queued
  • [ASTERISK-21719] – res_srtp doesn’t cleanup srtp library
  • [ASTERISK-21723] – pbx cleanup is incomplete
  • [ASTERISK-21724] – __ast_rwlock_destroy can segfault with DEBUG_THREADS
  • [ASTERISK-21742] – SIP Session-Expires: Set timer to correctly expire at (~2/3) of the expiry interval when not the refresher.
  • [ASTERISK-21744] – – fix lower bound check with -ve integer conversion from a float
  • [ASTERISK-21779] – Manager closes connection when a SendText action is requested during hangup
  • [ASTERISK-21782] – Delayed audio to agent when answering a queue call
  • [ASTERISK-21787] – No IAX2 communication either user/peer or friend accounts
  • [ASTERISK-21793] – Segmentation fault when dealing with Agent channels
  • [ASTERISK-21799] – Dropouts/distortion in MixMonitor recording when recording RTP with ptime of 60ms
  • [ASTERISK-21800] – ooh323 channels stuck if no gatekeer or ooh323 reload

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.23.0

Thank you for your continued support of Asterisk!

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