[asterisk-users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *

Tue May 29 02:19:05 MST 2007

Welcome to the Asterisk users community!
----------------------------------------

Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.

Our community is also growing fast and we're having a lot
of interaction, on the IRC and on the mailing lists.

It's great to have you participating in this Open Source project
- building an Open Source PBX. Here are a few things to know and
remember while working with the project.

Last week we had the annual AstriDevCon - the Asterisk Developer's
Conference. At that meeting, core developers and project members
meet to discuss current and future issues, new designs and - equally
important - get a chance to know each other behind the acronyms
in the bug tracker and on the IRC. We had a great week where we
got a lot of important things done, as you can see on the number of
changes that was done to Asterisk during that week.

One decision that we took was to stop maintaining 1.2 as a current
release from August 1st 2007. At that date, we will move 1.2 of  
Asterisk,
Asterisk-addons, libpri and zaptel to security maintenance status.
1.4 will at that point be the recommended release.

Between now and August 1st we will focus on fixing open issues
in 1.4 to make sure it's production quality code. Please help us with
that by  answering questions quickly in the bug tracker, testing
and reporting issues. Together, we'll make sure that 1.4 becomes
a great product.

Again, welcome to the Asterisk.org Open Source PBX Project!

Meet you on the IRC channel, the bug tracker or
on the mailing list!

/oej

** Asterisk European Events

This week, I'll talk on the Open Source VoIP event in Utrecht,  
Netherlands
http://www.mediaplaza.nl/mp.php/overheid/agenda/agenda.php?id=230

June 12th we have a Asterisk BOF at the VON Europe show in Stockholm
http://www.von.com/2007/springEurope_stockholm/html/ 
confSchedule_gvsb1178630538.html#gvsb1178630538

** Asterisk version information

At this moment we have three current versions of Asterisk, the
developer version and the release versions (1.2/1.4). The release
versions are distributed as .tar.gz archives on several servers. The
current released version of Asterisk is 1.2.18 for the 1.2 version
and 1.4.4 for the 1.4 version. The release version
is fixed, we are adding no new functions and only changes it
when bugs are fixed.

The development version is to be used by people that can test
new functions and live with bugs and unexpected shortcomings.
The development version is branded 1.5 and will be the basis
for the next release version, version 1.6.

There are also a lot of development branches in our subversion
repository, hosting new functionality developed for testing by
you, the Asterisk community.

For more information about these, please visit
http://www.voip-forum.com/index.php?p=189&more=1

** The mailing list is growing

Today, we propably have over 10,000 readers on the -users list. This
means that everything anyone write to this mailing list, is sent to
thousands of mailboxes that are already flowing over with messages.
That's why we all need to follow some simple rules on how to use
the mailing list and the other tools that are available.

** Think before sending a message, think twice

I would like to stress the fact that you have to think before you send a
message to such a big list. Do *not* send out personal replies on the  
list.

If you offer services to someone, do *not* CC: or reply to the list, it
will annoy more potential customers than get you new customers. If you
send out a message by mistake, you don't have to apologize to all of us,
we understand you're embarassed. We will get more annoyed by your
apology than over your first message.

And please do not send out "test" messages to the list.

** Try finding the answer first, then ask the list

The Asterisk Wiki at http://www.voip-info.org is an important
knowledge base for the project.

Go there to find your answer first, then search the mailing list
archives (Google or http://search.voip-forum.com) and then
go to the IRC channel. The IRC channel is populated with Asterisk gurus
around the clock (literally) and they'll help you move forward.

* IRC info: http://www.asterisk.org/index.php?menu=support#irc
* There's many links to Asterisk web pages on the documentation
   page at http://www.asterisk.org
* The Asterisk FAQ is found on the wiki
   http://www.voip-info.org/wiki-Asterisk+FAQ
* The Asterisk documentation project (which needs your help)
   is at http://www.asteriskdocs.org
   You can download their new book from the web site or buy
   it from the bookstore.
* Asterisk Daily news is at
   http://www.sineapps.com/news.php

Finally, if you don't find the answer elsewhere, try the list.

** Mailing lists
For developers, there is a developer's list, asterisk-dev.
Do not use this list as a secondary support line if you do
not get an answer on the -users list. It is meant for developer
discussions, not advanced support. If you need answers, there
is a better chance that you will get help on the irc channel.

For BSD users (FreeBSD, NetBSD, OpenBSD and OS/X) there's a
list called asterisk-bsd. There is also a business list
for those that want to ask for commercial services and
inform their community about new services (asterisk-biz).

You'll find all lists on http://lists.digium.com, which is the
site where you manage your subscription to this list as well.

Please, do not crosspost the same message to multiple mailing
lists. It will not help you, it will only add to the mail flow
and get people that read both lists irritated. If you are
unsure which list to use, send only to the -users list.

Make sure that you remove unnecessary text when you reply,
to make it easy to browse the mailing list quickly. And please
do not send HTML mail to a mailing list.

** Reporting bugs

If you think you have found a bug, report it. We need bug reports.
Read this document http://www.digium.com/bugtracker.html and then
go to the bugtracker http://bugs.digium.com to file a report.
If you are unsure, find a bug marshal on the IRC channel to help
you. They're appointed to support you with how to handle bugs.

Please check the bugtracker thoroughly before posting a new bug;
often, your bug or feature already exists but is simply slowly
making it's way through the system.  Duplicate reports slow things
down for everyone, so please spend a few minutes searching first.

The bug tracker is also a place where you add your contribution
to Asterisk. If you have coded extra functionality, make sure you
give it back to the project so it can be added to the code base.
This is how Asterisk grows, free contributions and consultants
that are paid to add functionality on a case by case basis.

** Be a community member - contribute!
The Asterisk software growth is very much based on user contributions.
That's really how we all pay for the software - and get revenue back.
If you develop custom functionality, you can rest assured that there
is someone out there that wants it, needs it and will be helped by it.

Don't forget to contribute. Open Source is both giving and taking.
The financial model behind it all is really cooperative in some way.
As one member to the community said to a contractor:
   "Hey, I'm paying you to deliver code to me, then I'm giving it
    away to the community. How did this happen?"
It's the Open Source business model. And if it didn't work, we
wouldn't have a lot of the software platforms that we all use
in our business systems - Linux, Apache, MySQL, PostgreSQL and
Asterisk.

** Remember: It's Open Source, it's voluntary
Asterisk.org is a Open Source project. This means you can't demand
help from community members, nor demand new functions or support.
However, there are many individuals and companies out there that are
offering services based on Asterisk, from VoIP service providers to
consultants all over the world.

Of course, this is also part of Digium's business, so you have
plenty of help if your willing to pay. Digium is to be found at
http://www.digium.com. Service providers and consultants are
listed on the wiki, where you'll find companies all over the globe
that are willing to set up your PBX, provide training and get you
connected to either the PSTN or the growing telephony network on the
Internet.

* See http://www.voip-info.org/wiki-Asterisk%20consultants
* For training, see http://edvina.net/training

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PS. This message will be sent regularly. If you have any
corrections or additional information that needs to be
included, mail me * off list *. Thank you!