asterisk users - Sep 2010 - Asterisk 1.8.0 Release Candidate 2 Now Available

The Asterisk Development Team has announced the second release candidate of
Asterisk 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Asterisk 1.8.0-rc1 was not released due to an issue found prior to release.

  * Make AMI honor enabled=no
    (Closes issue #18040. Reported by: twilson
     Review: https://reviewboard.asterisk.org/r/938/)

All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any issues found to the issue tracker,
https://issues.asterisk.org/. It is also very useful to see successful test
reports. Please post those to the asterisk-dev mailing list.

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long
Term Support (LTS) release, similar to Asterisk 1.4. For more information about
support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

With the availability of the Asterisk 1.8.0 release candidates, the binary
add-on modules for Asterisk produced by Digium have been updated with new
versions that are compatible with Asterisk 1.8. The availability of these
modules will assist with the testing of Asterisk 1.8.0 in a wider variety of
situations.

This release candidate contains fixes since the last beta release as reported by
the community. A sampling of the changes in this release candidate include:

  * Add slin16 support for format_wav (new wav16 file extension)
    (Closes issue #15029. Reported, patched by andrew. Tested by Qwell)

  * Fixes a bug in manager.c where the default configuration values weren't
reset
    when the manager configuration was reloaded.
    (Closes issue #17917. Reported by lmadsen. Patched by bbryant)

  * Various fixes for the calendar modules.
    (Patched by Jan Kalab.
     Reviewboard: https://reviewboard.asterisk.org/r/880/
     Closes issue #17877. Review: https://reviewboard.asterisk.org/r/916/
     Closes issue #17776. Review: https://reviewboard.asterisk.org/r/921/)

  * Add CHANNEL(checkhangup) to check whether a channel is in the process of
    being hung up.
    (Closes issue #17652. Reported, patched by kobaz)

  * Fix a bug with MeetMe where after announcing the amount of time left in a
    conference, if music on hold was playing, it doesn't restart.
    (Closes issue #17408, Reported, patched by sysreq)

  * Fix interoperability problems with session timer behavior in Asterisk.
    (Closes issue #17005. Reported by alexcarey. Patched by dvossel)

  * Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was
    determined to be one of the most significant bottlenecks in SIP registration
    processing. This patch improved the speed of an astdb load test by 50000%
    (yes, Fifty-Thousand Percent). On this particular load test setup, this
    doubled the number of SIP registrations the server could handle.
    (Review: https://reviewboard.asterisk.org/r/825/)

  * Don't clear the username from a realtime database when a registration
    expires. Non-realtime chan_sip does not clear the username from memory when
a
    registration expiries so realtime probably shouldn't either.
    (Closes issue #17551. Reported, patched by: ricardolandim. Patched by
     mnicholson)

  * Don't hang up a call on an SRTP unprotect failure. Also make it more
obvious
    when there is an issue en/decrypting.
    (Closes issue #17563. Reported by Alexcr. Patched by sfritsch. Tested by
     twilson)

  * Many more issues. This is a significant upgrade over Asterisk 1.8.0 beta 5!

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-rc2

Thank you for your continued support of Asterisk!

At 01:14 PM 9/23/2010, you wrote:
>The Asterisk Development Team has announced the second release candidate of
>Asterisk 1.8.0. This release candidate is available for immediate download
at
>http://downloads.asterisk.org/pub/telephony/asterisk/
I downloaded this, ran "./configure" followed by "make
menuselect"
and I don't seem to have SIP as an available protocol. Is there 
something I can do to make it available? It works fine on the most 
recent 1.6 version and it's worked on most of the prior 1.8 versions.

Ira

On 10-09-26 02:55 PM, Ira wrote:
> At 10:37 PM 9/24/2010, you wrote:
>> You probably need to install libssl-dev then rerun ./configure.  At
>> least I did (Debian Lenny).  Seems chan_sip needs res_crypto which
>> needs libssl.
>
> Thanks, I tried to figure out what I needed but I failed. That was
> it, though on CentOS it seems to be openssl-devel.
FYI, this is no longer an issue as of today. I opened an issue per the Asterisk 
development team, and Tilghman fixed the issue.

https://issues.asterisk.org/view.php?id=18062

The next release candidate will allow chan_sip to use, but not require, the 
OpenSSL development libraries.

Thanks!
Leif.