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 [EMAIL PROTECTED] * Any chance of my company being sponsor and exhibitor? YES, send mail to [EMAIL PROTECTED] 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
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
