Update to Asterisk 11.1.0:  this is a major new long term support release.

As this is a major release, you should read the information about updating:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11

You can also find documentation in:  /usr/pkg/share/doc/asterisk

----- 11.1.0:

The Asterisk Development Team has announced the release of Asterisk 11.1.0.

The release of Asterisk 11.1.0 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:

* --- Fix execution of 'i' extension due to uninitialized variable.

* --- Prevent resetting of NATted realtime peer address on reload.

* --- Fix ConfBridge crash if no timing module loaded.

* --- Fix the Park 'r' option when a channel parks itself.

* --- Fix an issue where outgoing calls would fail to establish audio
      due to ICE negotiation failures.

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.1.0

----- 11.0.1:

The Asterisk Development Team has announced the release of Asterisk 11.0.1.

The release of Asterisk 11.0.1 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:

* --- chan_sip: Fix a bug causing SIP reloads to remove all entries
      from the registry

* --- confbridge: Fix a bug which made conferences not record with
      AMI/CLI commands

* --- Fix an issue with res_http_websocket where the chan_sip
      WebSocket handler could not be registered.

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.1

Thank you for your continued support of Asterisk!

----- 11.0.0:

The Asterisk Development Team is pleased to announce the release of
Asterisk 11.0.0.

Asterisk 11 is the next major release series of Asterisk.  It is a Long Term
Support (LTS) release, similar to Asterisk 1.8.  For more information about
support time lines for Asterisk releases, see the Asterisk versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 11, please see the
Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11

A short list of new features includes:

* A new channel driver named chan_motif has been added which provides support
  for Google Talk and Jingle in a single channel driver.  This new channel
  driver includes support for both audio and video, RFC2833 DTMF, all codecs
  supported by Asterisk, hold, unhold, and ringing notification. It is also
  compliant with the current Jingle specification, current Google Jingle
  specification, and the original Google Talk protocol.

* Support for the WebSocket transport for chan_sip.

* SIP peers can now be configured to support negotiation of ICE candidates.

* The app_page application now no longer depends on DAHDI or app_meetme. It
  has been re-architected to use app_confbridge internally.

* Hangup handlers can be attached to channels using the CHANNEL() function.
  Hangup handlers will run when the channel is hung up similar to the h
  extension; however, unlike an h extension, a hangup handler is associated with
  the actual channel and will execute anytime that channel is hung up,
  regardless of where it is in the dialplan.

* Added pre-dial handlers for the Dial and Follow-Me applications.  Pre-dial
  allows you to execute a dialplan subroutine on a channel before a call is
  placed but after the application performing a dial action is invoked. This
  means that the handlers are executed after the creation of the callee
  channels, but before any actions have been taken to actually dial the callee
  channels.

* Log messages can now be easily associated with a certain call by looking at
  a new unique identifier, "Call Id".  Call ids are attached to log messages for
  just about any case where it can be determined that the message is related
  to a particular call.

* Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
  Asterisk. Unlike traditional ACLs defined in specific module configuration
  files, Named ACLs can be shared across multiple modules.

* The Hangup Cause family of functions and dialplan applications allow for
  inspection of the hangup cause codes for each channel involved in a call.
  This allows a dialplan writer to determine, for each channel, who hung up and
  for what reason(s).

* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
  lets you set some of the configuration options from the general section
  of features.conf on a per-channel basis. FEATUREMAP() lets you customize
  the key sequence used to activate built-in features, such as blindxfer,
  and automon.

* Support for DTLS-SRTP in chan_sip.

* Support for named pickupgroups/callgroups, allowing any number of pickupgroups
  and callgroups to be defined for several channel drivers.

* IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation

A full list of all new features can also be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/11/CHANGES

For a full list of changes in the current release, please see the ChangeLog.

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0

Thank you for your continued support of Asterisk!
Updated comms/asterisk to 11.1.0