asterisk users - Jul 2004 - *** Asterisk Summer News: The heat is on!

Another issue of Asterisk Summer News, delivered right to your
mailbox! Back here in Sweden, it's finally summer weather.
Sunshine and some heat. It's good for our ice bears and
the snow houses to get some sunshine :-)

Asterisk development and IRC chat has gone into a lazy summer
mode, but the mailing list is still cooking. It's impossible
to keep up with it, for both gurus and newbies, even during
summer holidays.

This issue will be a short issue with just a few articles.
Enjoy!

This week's topics:
-------------------
* Asterisk 1.0rc1: Feedback, please
* Astricon 2004: Early bird discount only applies in July
* Asterisk IRC chatters: BEHAVE!
* Open Source VoIP Watch: SER 0.8.14
* Dialplan updates: The DIAL() application
* Recent CVS changes


*** Asterisk 1.0rc1: Feedback, please
-------------------------------------
So we've had some time to try out the release candidate for
Asterisk 1.0. If you haven't tried it yet, please do. It is very important
for your business and for the Asterisk community that we try to find
and fix as many errors as possible before we release 1.0.

With the success and growth we've been experiencing lately in the
Asterisk.org project, I believe there will be even more success in
the fall. This will certainly lead to more pressure from people
that use Asterisk in production.

In that situation, we need a stable branch code for production use
and a development CVS tree for creative development and dangerous
but exciting code.

In order to get there, we need your help. Test rc1 (or rc2 which is
on it's way) and provide feedback.

* Download mirrors: http://www.voip-info.org/wiki-Asterisk-mirrors
* Linux RPMs: ftp://ftp.nacs.net/asterisk
* Instructions on how to report bugs:
   http://www.digium.com/bugtracker.html

To get better documentation for 1.0, join the asterisk-docs mailing list
and contribute to the effort. Leif Madsen and Jared Smith really
needs your help in order to get a decent handbook out to the 1.0 release.

* http://www.asteriskdocs.org


*** Astricon 2004: Early bird discount only applies in July
-----------------------------------------------------------
Astricon 2004 is getting closer. This is the first Asterisk user's and
developer's conference. During July, you will get an early-bird
discount on the registration fee so please do not forget to register
and pay before july 31.

The conference agenda was published this week. Amongst the speakers
you'll find:

* Mark Spencer, lead developer of Asterisk and founder of Digium
* Ravi Sakaria, founder of VoicePulse
* David Beckemeyer, Distinguished Research Engineer, Earthlink
* Ed Guy, Chief Scientist, Pulver.com

Also, a lot of those Asterisk Guru's you find on the IRC channel
will speak in live sessions:
* bkw_, twisted, blitzrage, jtodd, jsmith

You may register for one, two or three days with hotel room booking
at the web site. We also have information and discounts on
shuttles from the airport.

* http://www.astricon.net

*** Asterisk IRC chatters: BEHAVE!
----------------------------------
The #asterisk IRC channel have had a tendency to fall into nonsense chatting
that has no connection to Asterisk. Also, there's been a number of reports
of bad behaviours toward newbie's. This forced Mark Spencer to ask the
community to remember that they also have been new to Asterisk and behave
friendlier:

     "To everyone who spends time in #asterisk or #asterisk-bugs or
basically
      anything with #asterisk in its name, I want to implore you to please treat
      new users with respect, and act as good representatives of the Asterisk
      community.

      Recently I have had more reports of new users being severely turned off of
the
      project in general due to the comments, reactions and attitudes of a few
      members of the asterisk channels.

      The success of the Asterisk project depends upon users and developers,
      and remember that every one of you, even the most experienced Asterisk
      users were at one point a newbie and needed some hand holding from
someone.

      Finally, I would also ask that the #asterisk channel in particular please
      stay as focused on Asterisk related topics as possible."


*** Open Source VoIP Watch: SER 0.8.14
--------------------------------------
IPtel.org has released a new stable release of the SIP Express Router, the
Open Source SIP Proxy that a lot of commercial service providers use, as
well as many companies. They state that 0.8.14 is more of a maintenance
release than a release with a lot of new features.

So what is the difference between a SIP proxy and Asterisk:
* A SIP proxy is never involved in the media stream, it doesn't answer
   or originate calls
* A SIP proxy supports many more SIP applications than voice

There are many installations using SER as a SIP Proxy and Asterisk
as a feature server for PSTN connectivity, voicemail, conferencing
and call center features.

Read more
* More on Asterisk and SIP Proxy:
http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy
* Release notes: ftp://ftp.berlios.de/pub/ser/0.8.14/doc/NEWS
* Home page: http://iptel.org/ser/


*** Dialplan Update: The DIAL application
-----------------------------------------
The dial() application is the core of the dial plan. Dial() is what you use
to set up a call when Asterisk receives something on a channel, an inbound
call.

It's important to follow up the status of the call in a proper way. Up
until recently, we've only had one way of doing it, but in the spirit
of Perl, Asterisk now has many ways of creating a dial plan entry that
reacts to the status of the call. The new way has a more fine grained
result code.

Dial now returns a text string in the ${DIALSTATUS} variable. This string
can be used in many ways, creating special extensions is one way of doing it.
Here's an explanation of the status codes:

- CANCEL: Call is cancelled
- ANSWER: Call is answered
- NOANSWER: No answer
- BUSY: Busy signal received
- CONGESTION: Congestion
- CHANUNAVAIL: Channel unavailable (On SIP, peer may not be registred)

You can use this to goto special extensions, like

exten =>55555,3,goto(result-${DIALSTATUS})
exten =>result-CHANUNAVAIL,1, playback(channel-unavailable)

This is specially useful in macros. At the same time, two other
channel variables was introduced that reports the length of
the call in seconds.

Read more
* "Show application dial" in your Asterisk CLI
* README.variables in your asterisk source code tree
* Dialstatus documentation:
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
* Dial() documentation: http://www.voip-info.org/wiki-Asterisk+cmd+dial



*** Recent CVS changes
----------------------
Here's a number of additions done to Asterisk CVS head since the last
newsletter.
Bug # identifies the patch/bug report in the bug tracker.

GENERAL/MISC
* Asterisk: Add -U and -G options to set user/group to run Asterisk as if you
   do not want to run Asterisk as root
* New Asterisk manpage - see "man asterisk"
* DSP: Lower default volume
* New sound files
* Belgium tones (bug #2130)
* Fix ADSI prog to only accept 253 (bug #2135)

BILLING/CDR
* CDR: Support for FreeTDS: A library that connects to MS-SQL and Sybase
* CDR: Add Manager CDR (off by default) (bug #2127) courtesy cybershield

APPLICATIONS
* Voicemail: Make the e-mail message-ID more unique
* Background: Ability to play sounds before answering
* PlayBack: Ability to play sounds before answering the call
* Dial: Make '*' count as CANCEL (bug #2098)
* Dial: Enable for both caller and callee to hang up using "*"
* Dial: Copy account code and flags from incoming to outgoing channel for
         purposes of local channel
* Queue: Create option for joining empty queue (bug #2126)
* Queue: Allow optional event whenever an agent is called from a
         queue (bug #2066)
* Queue: Unify queue add/remove from manager and CLI (bug #2125/2123)
* Queue: Allow for both caller and callee to hang up using "*"
* Meetme: New fixes for re-entering pin code
* Meetme: Allow for both caller and callee to hang up using "*"

CHANNELS
* RTP: Add option to disable checksums on RTP UDP ports (bug #2068)
* SIP: Don't consider port number in name of peer in create_addr (bug #1974)
* SIP: Reinitialize user agent on reload
* SIP: Remove quotes from MD5 in digest auth header (bug #2116)
* SIP: Make request URI in CANCEL match that of the original
        INVITE exactly (bug #2134)
* SIP: Add "username" to sip show peer (bug #2163)
* ZAP: Fix signalling for GR303 FXSKS CPE so we can look like a concentrator
* ZAP: Fix chan_zap compiling without libpri
* ZAP: Heavily reduce stack usage, remove ancient and useless tor.h
* ZAP: A lot of locking issues fixed
* ZAP: Fix "ZapOffHook" (bug #2161)
* MGCP: Create one generally useful runtime option and one compile time option
to work
        around bugs in the DPH100M phone (bug #2122)
* MGCP: Turn off DTMF generally in MGCP and make option to enable RFC2833 or
in-band
* ALSA: Updates
* General: Added support to be able to set the channel var TRANSFER_CONTEXT

PORTABILITY
* Fix Yellow Dog Linux (PowerPC) build (bug #2109)
* Debian: Add debian initialization script (bug #2008)
* Improved scripts for Redhat starting/stopping Asterisk
* FreeBSD: Fix astman build on FreeBSD (bug #2119)

NEW APPLICATIONS
- none-

Upgrade your Asterisk now and test all these new functions!
* http://www.asterisk.org/index.php?menu=download



*** 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 2004: http://www.astricon.net
* Asterisk documentation project: http://www.asteriskdocs.org

*** Epilogue: The heat is on
-----------------------------
Asterisk is featured in more magazines and web sites every day.
We now have a polish Asterisk forum, Mac OS X user interfaces
for Asterisk and a lot of other things we couldn't believe a while
ago. It happens fast and it's a lot of fun.

I am personally looking forward to a one-week holiday on the
west coast of Sweden. In a small cottage by the sea with no
fibre connectivity, not even DSL. I'll be off line, preparing
for a stormy fall with a lot of activity and a great conference.

We've already got registrations from all over the world, showing
how global the Asterisk community is - Nigeria, Denmark, India,
US, Malaysia, Colombia.

It'll be a great event and a milestone for the
Asterisk.org project. Make sure you register now!

Have a great Asterisk Week!

/Olle