[Asterisk-Users] *** Asterisk Sunday News: Off track with 1.0, moving forward

Sun Jun 13 04:58:54 MST 2004

Thank you very much for all feedback on Asterisk Sunday News!
This is the last issue for June. This week I'll go on holiday
and will be back with more news in early July.

My kids are getting summer leave this week and we'll be
visiting the south of England for a while. Another part of
Europe that still use their own currency.

If you think there's an European standard, you're
wrong. England have different phoneplugs, powerplugs and drives
on the wr..., sorry, the other side of the road. So there's not
only a difference between Europe and the US in ISDN standards,
like PRI/E1 and PRI/T1, but also when it comes down to simple
things like power plugs. But that's another cup of tea. Time to
focus on Asterisk.

This week's topics:
-------------------
* Asterisk - gone fishing again
* Asterisk release plans: What happened with stable?
* The Astricon FAQ
* Chan_sip2 news: The Yngve release
* Recent additions to Asterisk CVS Head

*** Asterisk - gone fishing again
---------------------------------
   Two weeks ago, I wrote about a way to include contexts based
on time and date. Tilghman corrected me, saying that there is an
easier way, using the gotoiftime() application. I stand corrected.
For more information on gotoiftime() see
* The cli command "show application gotoiftime"
* http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20gotoiftime

*** Asterisk release plans: What happened with stable?
------------------------------------------------------
Last week I made a note that the release plans was a bit shaky. The state
as of now is that the stable-1.0 CVS tree will *not* be released as a
1.0. There has been too many bug reports on both the stable-1.0 and the
HEAD branch, and new are coming in.

The decision is to base the future 1.0-release on the CVS head tree.
The current "stable-1.0" tree will be released as something intermediary,
maybe 0.91, and at that point it will be considered end-of-life.
At some point when we have cleared the bug tracker from major issues, we
will fork a new stable-1.0 tree and start working on that.

As a community, we now need to focus on solving all the bugs in the CVS head
tree. We need help, Mark Spencer can't handle all bugs by himself. So when
reporting bugs, make sure you are available for questions and testing.
Any patches in the bug tracker that you can test, test. Report your findings
to the bug tracker, both good and bad.

If you're a bug marshal, this is your call to action. After the new fork,
we will also need release marshals, that are responsible for maintaning
a release and adding bug fixes, no new functionality, to that release for
production systems. It's easy to find people that create new wonderful
stuff, harder to find good maintainers.

This is the life of Open Source. We can't schedule releases, it's an open
process and we will release when we agree that the code is stable.
That's why we can't release a cvs tree that we know has got a lot of bugs
as a 1.0. The changes between the old 1.0-stable and CVS head are too
many to port back, so let's move on, clear outstanding issues and try again.

Let's work together and aim for a stable release soon. That will require
your involvement. You are an important part of this community. Visit the
bug tracker now:

* http://bugs.digium.com

*** Chan_sip2 news: The Yngve release
-------------------------------------
For those of you interested in testing a dangerous development branch
of the SIP channel, test my chan_sip2 code. I started the chan_sip2
project a while ago to be able to test some ideas I had without
distributing a set of patches or having to consider production servers.
It is a test platform for new and changed functionality in the SIP
channel. As a result, a lot of the chan_sip2 code is now integrated
in the CVS head.

The next part to move into both chan_sip and chan_iax2 is the
configuration templates.

The latest release, called Yngve, has a few additions

* SIPAddHeader(): An application to add a SIP header to an outbound call
* SIPGetHeader(): An application to read any SIP header on incoming call

These are really useful if you want to read RPID headers or transfer
an accountcode between two Asterisk servers.

I've also changed the authentication part and added realm based
authentication. This way, you can configure Asterisk to always authenticate
with proper credentials to a SIP realm challenge, regardless of peer or
user - or if you just use a SIP provider for dialing out. Also, one peer
can be configured to use multiple credentials for outbound calls, all
controlled by the realm in the authentication challenge.

If I get positive feedback on these functions, they will be ported back to
the CVS head chan_sip.

* Chan_sip2: http://bugs.digium.com/bug_view_page.php?bug_id=0000759

*** Recent additions to Asterisk CVS Head
-----------------------------------------
Here's some of the additions to Asterisk CVS Head
* res_config: A driver for loading configurations from various sources
* res_config_odbc: A driver for ODBC database access for res_config
* app BackgroundDetect: Background a file with talk detect
* A lot of fixes to support recursive Mutexes on FreeBSD
* NFAS and GR-303 support
* Changes to Voicemail (an exit for VoiceMailMain added)

More on res_config later, when I've tested it. I'm sure that a lot of
you are going to use it soon, when you discover what it can do for
you. Hint, hint :-)

*** The Astricon FAQ
---------------------
Astricon - the first conference for the Asterisk Community, is going to
take place in Atlanta, Georgia, September 22-24 2004.

* When will you open for registration?
   A week from now
* What will it cost?
   A three day conference will cost $400. We will have an early-bird price
   for registrations before july 10, also two-day pricing.
* Any chance of me speaking at Astricon?
   YES, see above!
* Where will it be?
   Atlanta, Georgia, USA. The exact location will be revealed
   next week, hopefully.
* May I speak too?
   YES, mail me now at info at astricon.net
* Any chance of my company being sponsor and exhibitor?
   YES, send mail to info at astricon.net for details
* Will Mark Spencer be there?
   YES, he will be there in the middle of the crowd
   Digium will participate as a partner and exhibitor
* Where can I find more information?
   Check the web site, http://www.astricon.net

*** Useful Asterisk web links:
------------------------------
* Asterisk: http://www.asterisk.org
* Asterisk mailing lists: http://lists.digium.com
   (users, bsd, dev, biz and cvs mailing list)
* Asterisk bug tracker: http://bugs.digium.com
* Asterisk IRC channel: #asterisk on irc.freenode.net
* Digium: http://www.digium.com
* Wiki: http://www.voip-info.org
* Voip Search: http://search.voip-forum.com
* Astricon: http://www.astricon.net
* Asterisk documentation project: http://www.asteriskdocs.org

That's all for this week, no awards and no tutorial. Next issue will
come in early July. While waiting for that issue, please help us resolving
outstanding bugs in the bug tracker.

And if you have topics for me to include in Asterisk Sunday News, mail
me off list :-)

Have a nice Asterisk week!
/Olle