-
    - The release of Asterisk 1.8.0-rc5 was triggered by some last minute platform
    - compatibility IPv6 changes. In addition, the availability of the English sound
    - prompts with Australian accents has been added.
    -
    - A full list of new features can be found in the CHANGES file.
    -
    - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
    -
    - For a full list of changes in the current release candidate, please see the
    - ChangeLog:
    -
    - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc5
    -
    - This release candidate contains fixes since the last release candidate as
    - reported by the community. A sampling of the changes in this release candidate
    - include:
    -
    -  * Additional fixups in chan_gtalk that allow outbound calls to both Google
    -    Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip
    -    and stunaddr.
    -    (Closes issue #13971. Patched by dvossel)
    -
    -  * Resolve manager crash issue.
    -    (Closes issue #17994. Reported by vrban. Patchd by dvossel)
    -
    -  * Documentation updates for sample configuration files.
    -    (Closes issues #18107, #18101. Reported, patched by lathama, lmadsen)
    -
    -  * Resolve issue where faxdetect would only detect the first fax call in
    -    chan_dahdi.
    -    (Closes issue #18116. Reported by seandarcy. Patched by rmudgett)
    -
    -  * Resolve issue where a channel that is setup and torn down *very* quickly may
    -    not have the right call disposition or ${DIALSTATUS}.
    -    (Closes issue #16946. Reported by davidw. Review
    -     https://reviewboard.asterisk.org/r/740/)
    -
    -  * Set TCLASS field of IPv6 header when SIP QoS options are set.
    -    (Closes issue #18099. Reported by jamesnet. Patched by dvossel)
    -
    -  * Resolve issue where Asterisk could crash on shutdown when using SRTP.
    -    (Closes issue #18085. Reported by st. Patched by twilson)
    -
    -  * Fix issue where peers host port would be lost on a SIP reload.
    -    (Closes issue #18135. Reported, tested by lmadsen. Patched by dvossel)
    -
    - A short list of available features includes:
    -
    -   * Secure RTP
    -   * IPv6 Support in the SIP channel driver
    -   * Connected Party Identification Support
    -   * Calendaring Integration
    -   * A new call logging system, Channel Event Logging (CEL)
    -   * Distributed Device State using Jabber/XMPP PubSub
    -   * Call Completion Supplementary Services support
    -   * Advice of Charge support
    -   * Much, much more!
    -
    - A full list of new features can be found in the CHANGES file.
    -
    - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
    -
    - For a full list of changes in the current release candidate, please see the
    - ChangeLog:
    -
    - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4