About Asterisk 1.6.2

Now that Asterisk 1.6.2.0 (and 1.6.2.1) has recently been released, we thought it wise to create an announcement about some of the new items and changes available in the new feature release, along with where Asterisk is going in the next few months.

As with all new releases, if you plan on upgrading from any previous release be sure you test thoroughly in a test environment, and carefully read both the

CHANGES and the UPGRADE.txt files, which contain useful information about functionality changes between versions.

Some of the new features included in the Asterisk 1.6.2.0 release are:

SIP Changes

• if the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set, the sound will be played to the target of an attended transfer
• Added support for subscribing to MWI on a remote server and making the status available as a mailbox. Please see the sip.conf.sample file for more information.
• Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.

DAHDI Changes

• chan_dahdi now supports MFC/R2 signalling when Asterisk is compiled with support for LibOpenR2. http://www.libopenr2.org/
• As of version 1.6.1, Asterisk no longer depends on DAHDI as the sole timing source. In 1.6.2, an additional timing provider was introduced, res_timing_timerfd, which takes advantage of the timerfd API provided by newer versions of the Linux kernel. This timing provider provides much higher performance than the other non-DAHDI timing module, res_timing_pthread.

Dialplan Functions

• Added a new dialplan function, CURLOPT, which permits setting various options that may be useful with the CURL dialplan function, such as cookies, proxies, connection timeouts, passwords, etc.
• Added debugging CLI functions to func_odbc, ‘odbc read’ and ‘odbc write’.
• func_odbc now may specify an insert query to execute, when the write query affects 0 rows (usually indicating that no such row exists).
• func_odbc now supports database transactions across multiple queries.
• Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better obtaining realtime data from the dialplan.

Dialplan Applications

• Scheduled meetme conferences may now have their end times extended by using MeetMeAdmin.
• A new application, Originate, has been introduced, that allows asynchronous call origination from the dialplan.
• Added ConfBridge dialplan application which does conference bridges without DAHDI.

Additional Features

• extensions.conf now allows you to use keyword “same” to define an extension without actually specifying an extension. It uses exactly the same pattern as previously used on the last “exten” line. For example:

       exten => 123,1,NoOp(something)
       same  =>     n,SomethingElse()

• All deprecated CLI commands are removed from the source code. They are now handled by the new clialiases module. See cli_aliases.conf.sample file.
• The contrib/scripts/ directory now has a script called sip_nat_settings that will give you the correct output for an Asterisk box behind nat. It will give you the externhost and localnet settings.
• The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and can connect calls in passthrough mode, as well as record and play back files.

Version Numbering Changes

Asterisk 1.6.2 will be the last release in the 1.6 series of feature releases.

The original purpose of the Asterisk 1.6 releases was to change how often a

release was created, in order to allow shorter time periods between versions so

the community wouldn’t have to wait 2-3 years for a new feature. The intention

was to release a new 1.6 version, which included feature additions and

performance increases, every 3-4 months.

Unfortunately, the time frame ended up being closer to every 6 months, so the

original intention wasn’t fully developed. In addition, the numbering scheme

was confusing to some users. (Additional information about the how and the why of

the 1.6 numbering scheme can be read at:

http://blogs.asterisk.org/2009/06/24/about-the-new-asterisk-versioning-method/

)

So after about a year and a half of releases (1.6.0.x, 1.6.1.x, and 1.6.2.x),

community feedback was considered, and all that was learned during the 1.6

release cycle was put together to create a better release scheme, which takes

the advantages of both the long stable release cycles, and shorter feature

release cycles of Asterisk, while adding a layer of predictability allowing the

community to plan deployments accordingly.

Whenever a release is created, it will now be tagged either as Standard or LTS

(Long Term Support). Asterisk 1.4 would be considered an LTS release, meaning it

receives bug fixes for a longer period of time, of at least 4 years. A standard

release would have a shorter bug fix release period of at least 1 year.

After the support period expires, that release would then receive security

updates for at least 1 year (Asterisk 1.2 is an example of this), and eventually

would become end-of-life (Asterisk 1.0), thereby no longer receiving updates.

All Asterisk 1.6.x releases are considered Standard releases.

The next LTS (Long Term Support) release will be Asterisk 1.8, slated to be

branched from trunk around Q2 of 2010, at which point time will be spent testing

and getting it ready for full release as Asterisk 1.8.0.

For more information about the current state of releases for versions of

Asterisk currently available see: http://www.asterisk.org/asterisk-versions.

Upcoming Features in Asterisk 1.8

Asterisk 1.8 is shaping up to be an exciting LTS release which will contain

several performance improvements and many new features. Some of the new features

to look forward to in Asterisk 1.8 include:

• The much awaited SRTP support in chan_sip will be added.
• The chan_mgcp module has added PacketCable NCS 1.0 support for Docsis/Eurodocsis Networks. See configs/res_pktccops.conf for more information.
• The chan_spy module has several new options added, including:
  • Added c() option allowing custom DTMF to be set to cycle through the next available channel. By default this is still ‘*’.
  • Added x() option allowing DTMF to be set to exit the application.
  • Added the ‘S’ option, which makes the application automatically exit once it hits a point where no more channels are available to spy on.
  • Added the ‘E’ option, which spies on a single channel and exits when that channel hangs up.
• The MeetMe application now turns on the DENOISE() function by default, for each participant. In our tests, this has significantly decreased background noise (especially noisy data centers).
• A new interface, Channel Event Logging (CEL), is introduced here. CEL logs single events, much like the AMI, but it differs from the AMI in that it logs to database backends much like CDR does.
• A new set of modules were added supporting calendar integration with Asterisk. Dialplan functions for reading from and writing to calendars are included, as well as the ability to execute dialplan logic upon calendar event notifications.
• A new RTP engine and channel driver have been added which supports Multicast RTP. The channel driver can be used with the Page application to perform multicast RTP paging.
• In app_queue, you can now specify a position at which the caller will enter the queue.
• Information regarding the party that a person is currently connected to may be updated dynamically throughout the call. This has the advantage of allowing for the display of a phone to be updated during situations such as a call forward or a transfer.
• A new feature, call completion, will be added. This feature allows for a caller who reaches an unresponsive or busy party to be automatically contacted when the called party becomes available again.