[Asterisk-Users] *** Asterisk sunday news: Read the sample configs, Luke!

[Olle E. Johansson]

* Read the config sample files! (even if you're an Asterisk guru)
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For those of you that have a working installation that you keep using, this is a
reminder to check into the configs/ directory of the Asterisk source tree, regardless
if you downloaded a tar ball or from CVS.

As we add or change features in Asterisk, the sample config files are updated. If you
look there, you might get new insights into how to solve your problems. Also, you
might find new features that you really need.

If you have a new installation "gmake samples" or "make samples" will install these
files for you.

In CVS head, the development source tree, we've added quite a lot of information
recently to these files. They are more educational now and contains a lot of sample
configurations.

*** Check app_groupcount!
-------------------------
There's a new app in Asterisk town. In fact, there are several new applications in
CVS head. One of the major recent additions is app_groupcount, that you can use
to limit the number of calls to, well, just about anything. A SIP peer, a PRI link,
a call center staff member, a conference and calls to or from your boy
and/or girlfriend :-)

The command for setting a group is setgroup(), the command for enforcing a
limitation is checkgroup().

Please start using this instead of the incominglimit and outgoinglimit settings
in sip.conf. These are not working as expected and the more general solution
with app_groupcount is a much better solution that works cross channels. This
is an end-of-life warning for outgoinglimit and incominglimit :-)

As always, the CLI command "show applications" and "show application <name>" is
your best friend.

*** Set your SIP realm!
-----------------------
In CVS head, the SIP channel is now able to use a proper SIP realm for
authentication. The realm is the server group that has a common authentication
for a user. It could be one server or a number of servers that shares a
password/user database.

According to the SIP RFC, it should be set either to a domain or a hostname,
depending on what your realm covers. It should be globally unique.
Up to know, all Asterisk servers used the "asterisk" realm. That made it a bit
hard for some phones to know the difference between one server and another.

Please note that if you are using the "md5secret" setting in sip.conf, this
secret is based on the realm. If you change the realm, you need to rehash
your secrets.

*** Asterisk 1.0: Less than five bugs away
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If you follow the CVS, you will notice that there are very few changes in the
stable part of the source tree. Only bug fixes go in there and Mark have been
working like crazy to fix the major bugs. The bug tracker had almost 300 open
bugs just a while ago, and we are now down to a handful identified bugs.
As usual with Open Source Software, relase is not set to marketing plans.
Release will come when the software is ready to be shipped. So when Mark
decides that we've fixed the bugs that needs fixing, a release candidate
will be made and published for download.

Please plan to help us test the 1.0rc1 real hard. Do whatever you can to
crash it, to make it dial your mother-in-law when you really want to
talk to your husband, to make it connect the whole office to the HR
departments secret conference call by mistake and accidentally fill
your hard disk drive with non-existing voice mail messages. We do not
belive that you can, but if you can, report the bugs and help us move
forward to a 1.0 release!

If you want to start stress-testing it now, download the stable
CVS release. Instructions is to be found at http://www.asterisk.org

*** Astricon: Coming right up, sir!
-----------------------------------
We get a lot of questions about Astricon. To answer a few:
- We're still open of speaker's proposals, even though the time limit is up.
- The conference venue is not set yet, we will add it to the web site as soon
   as we have more information
- Pre-registration will start rsn (real soon now)
- We will find a location with a standard class hotel as well as a
   lower price alternative.
- Yes, we will have the voice of Asterisk there (hint, hint)
Astricon is at http://www.astricon.net

* Useful pointers:
------------------
* Asterisk: http://www.asterisk.org
* Asterisk mailing lists: http://lists.digium.com
   (users, 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

Have a nice Asterisk week!
/Olle