[asterisk-dev] Asterisk 1.8.14.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the first release candidate of
Asterisk 1.8.14.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.14.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release candidate:

* --- Relay proper SIP responses on calling side.
  (Closes issue ASTERISK-19914.)

* --- Fix pvt_sip for inbound call to use peer's allowtransfer setting
  (Closes issue ASTERISK-19856. Reported by Jacek)

* --- Prevent sip_pvt refleak when an ast_channel outlasts its
      corresponding sip_pvt.
  (Closes issue ASTERISK-19425.)

* --- format_mp3: Fix a possible crash mp3_read().
  (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk)

* --- Block on frameout if the hardware has enough samples to complete
      a frame.
  (Closes issue ASTERISK-19643. Reported by Shaun Ruffell)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.14.0-rc1

Thank you for your continued support of Asterisk!