Asterisk Now Available

The Asterisk Development Team has announced the release of Asterisk This release is available for immediate download at

The release of Asterisk 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-13797] – relax badshell tilde test
  • [ASTERISK-15879] – Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
  • [ASTERISK-18923] – res_fax_spandsp usage counter is wrong
  • [ASTERISK-20784] – Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
  • [ASTERISK-21721] – SIP Failed to parse multiple Supported: headers
  • [ASTERISK-22791] – asterisk sends Re-INVITE after receiving a BYE
  • [ASTERISK-22945] – Memory leaks in chan_sip.c with realtime peers
  • [ASTERISK-23768] – Asterisk man page contains a (new) unquoted minus sign
  • [ASTERISK-23846] – Unistim multilines. Loss of voice after second call drops (on a second line).
  • [ASTERISK-24011] – safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM
  • [ASTERISK-24063] – Asterisk does not respect outbound proxy when sending qualify requests
  • [ASTERISK-24190] – IMAP voicemail causes segfault
  • [ASTERISK-24307] – Unintentional memory retention in stringfields
  • [ASTERISK-24325] – res_calendar_ews: cannot be used with neon 0.30
  • [ASTERISK-24335] – [PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of rejected calls
  • [ASTERISK-24348] – Built-in editline tab complete segfault with MALLOC_DEBUG
  • [ASTERISK-24357] – [fax] Out of bounds error in update_modem_bits
  • [ASTERISK-24390] – astobj2: REF_DEBUG reports false leaks with ao2_callback with OBJ_MULTIPLE
  • [ASTERISK-24393] – rtptimeout=0 doesn’t disable rtptimeout
  • [ASTERISK-24406] – Some caller ID strings are parsed differently since 11.13.0
  • [ASTERISK-24425] – jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
  • [ASTERISK-24432] – Install when REF_DEBUG is enabled
  • [ASTERISK-24436] – Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
  • [ASTERISK-24476] – main/app.c / app_voicemail: ast_writestream leaks

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

Thank you for your continued support of Asterisk!

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