Olle E Johansson
2007-May-29 02:19 UTC
[asterisk-users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
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 ------------------------------------------------------------- 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!
Apparently Analagous Threads
- *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
- * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
- FOSDEM in Brussells - Feb 23-24
- [Asterisk-Dev] call pickup with snom function keys now working with cvs-head + patch sipsubscribe-20050715.rev779.txt
- BugTracker Information - REPOST