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Asterisk Now Available

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

The release of Asterisk 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:

* AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
Reported by: Catalin Sanda

* Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing
to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
causes the loop to exit prematurely. This causes a variety of negative side
effects, depending on when the loop exits. This patch handles the frame by
essentially swallowing the frame in the local loop, as the current channel
drivers expect the RTP bridge to handle the frame, and, in the case of the
local bridge loop, no additional action is necessary.
(closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
by: Matt Jordan

* Fix timing source dependency issues with MOH. Prior to this patch,
res_musiconhold existed at the same module priority level as the timing
sources that it depends on. This would cause a problem when music on
hold was reloaded, as the timing source could be changed after
res_musiconhold was processed. This patch adds a new module priority
level, AST_MODPRI_TIMING, that the various timing modules are now loaded
at. This now occurs before loading other resource modules, such
that the timing source is guaranteed to be set prior to resolving
the timing source dependencies.
(closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H,
Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patched by elguero

* Fix RTP reference leak. If a blind transfer were initiated using a
REFER without a prior reINVITE to place the call on hold, AND if Asterisk
were sending RTCP reports, then there was a reference leak for the
RTP instance of the transferrer.
(closes issue ASTERISK-19192) Reported by: Tyuta Vitali

* Fix blind transfers from failing if an ‘h’ extension
is present. This prevents the ‘h’ extension from being run on the
transferee channel when it is transferred via a native transfer
mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported
by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
Mark Michelson (license 5049)

* Restore call progress code for analog ports. Extracting sig_analog
from chan_dahdi lost call progress detection functionality. Fix
analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller Patched by Richard Miller

* Fix regression that ‘rtp/rtcp set debup ip’ only works when a port
was also specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by:
Walter Doekes

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


Thank you for your continued support of Asterisk!

H.323 Video Conferencing with GNU Gatekeeper

H.323 Video Conferencing with GNU Gatekeeper has been improved with the new released announced by Jan Willamowius, Founder of the GNU Gatekeeper Project.

A new major release of the GNu Gatekeeper has been anounced. This new version “has many new features that will allow even your legacy endpoints to move to a new age of H.323 audio and video communications.”

New features:
– full traversal zone support (gatekeeper-to-gatekeeper H.460.18/.19)
Now you can place one GnuGk behind a firewall and let it tunnel out the calls for all other devices behind the firewall eg. to a VCS or to another GnuGk.
This was probably the most request feature in the past.

– full IPv6 support (incl. IPv4-IPv6 proxying)
With the proxy function, you can let GnuGk manage a network of IPv6 endpoints and connect them to the IPv4 network or make legacy endpoints reachable for IPv6 calls.

RTP multiplexing (all calls to and from devices supporting H.460.19 will only use 2 sockets total)

– rewrite destination IPs into aliases

– ENUM, SRV and RDS routing policies extended for LRQs, in case the calling gatekeeper isn’t able to do this

– notifications when GnuGk opens listen ports
This allows you to update firewall rules on the fly, so you only have the minimum amount of ports open.

– improved H.235 password authentication with neighbors

– massive performance improvement when (re-)loading large numbers of GW rewrites

– interop fixes for Polycom m100 and Sorenson endpoints

– fixes in the underlying libraries so *BSD systems can get the latest GnuGk features

– a few bug fixes

The project provides executables for Linux (32 and 64 bit), Windows, MacOS X, FreeBSD, OpenBSD, NetBSD and Solaris.

You can download the new version at


Enjoy the new release of such a great software!

Speech recognition in Asterisk using Google Voice API

I’m exited to announce that Lefteris Zafiris has written an agi script that uses Google Voice API for voice recognition.

As the author says, “the script records from the current channel untill the pound key (#) is pressed or the timeout (15 seconds) is reached. The recording is send over to google speech recognition service and the returned text string is assigned to a channel variable.”

More info and dialplan examples can be found in the README file:

The script is available here:

The author reports that the code is still young and not roughly tested so comments, suggestions and bug reports are more than welcome.

Enjoy this code jewel and please provide feedback/comments to the author.

Asterisk Dialstatus

The DIALSTATUS channel variable is created when you attempt to connect to another device or endpoint and bridge the call with the Dial Application. It contains the status of the call reflected in one of the following values:

  • BUSY
  • DONTCALL – For the Privacy and Screening Modes. Will be set if the called party chooses to send the calling party to the ‘Go Away’ script.
  • TORTURE – For the Privacy and Screening Modes. Will be set if the called party chooses to send the calling party to the ‘torture’ script.


New Release: MariaDB, MySQL alternative

The MariaDB project has announced the availability of MariaDB 5.3.3-rc, the first Release Candidate release in the 5.3 series. Several optimization features introduced in MariaDB 5.3 have been thoroughly tested, and switched on by default in 5.3.3.

Performance changes that has been worked:

  • Subquery materialization (materialization=on)
  • Semi-join optimizations (semijoin=on,firstmatch=on,loosescan=on)
  • Derived table optimization (derived_merge=on,derived_with_keys=on -Documentation is being worked on-),
  • Index Condition Pushdown (index_condition_pushdown=on)
  • Nested loop join will use its Block-based variant more aggressively
  • Block-based join for OUTER JOINs (outer_join_with_cache=on)
  • Block-based join for semi-joins (semijoin_with_cache=on)
  • Linked join buffers (more aggressive buffering of multi-way joins) (@@join_cache_level==2)

Also, DISTINCT and GROUP BY clauses are now removed from subqueries when possible. This allows for more efficient query plans. (backported from MySQL 5.6)

About the changes related to usability:

EXPLAIN  output has been improved in MariaDB 5.3.3 to be easier to understand:

  • The select_type column now shows MATERIALIZED for subqueries that are executed with Materialization (it used to show SUBQUERY, which made it hard to distinguish  materialized subqueries from other kinds subqueries).
  • For a Duplicate Elimination strategy, Start temporary is now shown at the first table from the subquery.

For a list of changes, about MariaDB 5.3, please read the following link:

Enjoy MariaDB!

Asterisk Now Available

The Asterisk Development Team has announced the release of Asterisk This release is available for immediate download at

The release of Asterisk corrects two flaws in sip.conf.sample related to AST-2011-013:

* The sample file listed *two* values for the ‘nat’ option as being the default. Only ‘yes’ is the default.

* The warning about having differing ‘nat’ settings confusingly referred to both peers and users.

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


Thank you for your continued support of Asterisk!