[asterisk-dev] Asterisk 1.8.9.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team is pleased to announce the first release candidate
of Asterisk 1.8.9.0. This release candidate is available for immediate download
at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.9.0-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:

* Handling of AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop
  (closes issue ASTERISK-19095)

* Fix for timing source dependency issues involving Music On Hold
  (closes issue ASTERISK-17474)

* Fixes for several issues with audiohooks and ChanSpy

* Multiple fixes in chan_sip relating to TCP settings, bad file descriptors,
  potential deadlocks, and much more
  (closes issue ASTERISK-18837, ASTERISK-18799, ASTERISK-17760, and many others)

* Restore call progress code for analog ports
  (closes issue ASTERISK-18841)

* Fix parsing issues in main/pbx.c
  (closes issue ASTERISK-18909)

* Fix regression in res_rtp_asterisk that rtp/rtcp set debug ip only works
  when port is specified
  (closes issue ASTERISK-18693)

And much more!  For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.9.0-rc1

Thank you for your continued support of Asterisk!