- This release candidate contains fixes since the release candidate as reported by
    - the community. A sampling of the changes in this release candidate include:
    -
    -  * Still build chan_sip even if res_crypto cannot be built (use, but not depend)
    -    (Reported by a user on the mailing list. Patched by tilghman)
    -
    -  * Get notifications for call files only when a file is closed, not when created
    -    (Closes issue #17924. Reported by mkeuter. Patched by abeldeck)
    -
    -  * Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk
    -    expects the DTMF to arrive on the RTP stream and not via jingle DTMF
    -    signalling.
    -    (Patched by dvossel. Tested by malcolmd)
    -
    -  * Fixes to allow chan_gtalk to communicate with the Gmail web client.
    -    (Patched by phsultan and dvossel)
    -
    -  * Fix to GET DATA to allow audio to be streamed via an AGI.
    -    (Closes issue #18001. Reported by jamicque. Patched by tilghman)
    -
    -  * Resolve dnsmgr memory corruption in chan_iax2.
    -    (Closes issue #17902. Reported by afried. Patched by russell, 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=checkout
    -
    - 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-rc3