Asterisk 1.4 delivers many new features. In regards to call state subscriptions, there are many news for you. Call state subscriptions are what makes the lamps blink on your phone when your collegue’s phone rings. In 1.4, you can make it blink based on activity in parking lots and meetme conferences as well. Read on! (more…)
For a long time, we have needed a platform for managing feature requests - things that the community or developers would like to see in Asterisk. We used to have a “feature request” category in the bug tracker, but there was no good way to handle them in the bug tracker and they where in the way for the work done by developers in the tracker. They ended up getting closed, only to be reachable by searching closed bug reports. Not a very good solution for brainstorms and good ideas.
The new site is basically a blog with comments and voting capability. You register on the site to be able to file a feature request. Other people may then add comments or vote for requests.
Hopefully, this will be a repository of ideas and a good discussion platform. Things will be stored and accessible. As usual, filing a feature request is not a guarantee that anything will happen. You still need to make sure developer resources are put to it somehow.
Please also remember that it’s not a support forum. You can’t get help in the idea repository. There are already mailing lists and forums in place for that.
Let’s try this out for a while and see if it’s a good tool that works for us. Register for an account on www.asteriskideas.org today!
Thanks for any feedback!
/Olle
Happy New Year, Asterisk community!
During 2007 we accomplished a lot. We polished Asterisk 1.4 to a new state of readiness for production. We moved Asterisk 1.2 into no-maintenance mode, something I think we might have to reconsider after a discussion on the Asterisk-users mailing list. And we did not release anything new. Which is a good thing.
Why is that a good thing? Well, in an Open Source project you can choose between many different release strategies, depending on the software. Asterisk is a PBX. A PBX is in most cases something you don’t upgrade unless there’s a need to. We see that on the slow uptake everytime we release a new version of Asterisk. One year after the release of Asterisk 1.4, most of the installed base seems to run Asterisk 1.2. And they’re happy with it.
The problem is getting new features out there. We have a policy of not introducing new features to a released version of the software. That means we’re forcing people to upgrade to get new features - and new bugs. Would it be possible to create a new module interface so we can release various modules independently of the core? I don’t know, but that would create more complexity at the same time as it gives us a bit more flexibility for upgrades. At this point, 1.6 modules will not run in an 1.2 or 1.4 environment.
Anyway, I just wanted to write a note to say Happy New 2008! During this year, we hope to release a new version of Asterisk. During next year, you might be interested to put it into production. We developers just have to realize that it takes an awfully long time from idea to implementation in real life in the Open Source PBX market.
During one and a half year I’ve had a great oppurtunity to work with Asterisk development and evangelisation thanks to the Voop team, Thorsten Lockert, Gunnar Helliesen, Morten Reistad and the rest of the Voopers. They’ve paid part of my time so I could focus on long-term Asterisk issues and development, working in the bug tracker and starting the chan_sip3 (Codename Pineapple) project.
I think this sponsorship has made them the largest Asterisk corporate sponsor outside of the US. They have themselves contributed many patches, like the res_snmp module made by Thorsten and a number of fixes to the IAX2 channel.
Now, the contract is ending and I am looking for a new company to work with. If you are interested in sponsoring my work in the Asterisk project, making sure that I can dedicate part of my time to support the project, please contact me at oej (at) edvina.net for an open discussion. The sponsorship has also included my participation in the SIP interoperability events, SIPit.
A huge thank you to the Voop team!
/Olle
For many years, I’ve been working to remove the “type=user” and “type=friend” from chan_sip. In 1.4 you can in fact do everything you can do with a user with a peer. The only difference is the way we match incoming calls.
I’ve created a branch called “kill_the_user” that has no type=friend or type=user, only peers.alls are handled this way
If you can find any way this may give you problems, please inform me now. Otherwise, I’m goingto test this branch with all of you. I don’t foresee any problems going ahead with this in trunk.s compatibility, we will accept “type=friend” and “type=user” with a warning in 1.6,then remove it totally. Old configurations will still work. The longterm goal is to define new objects, please check the docs for Codename Pineapple objects,phone, service and trunk.For realtime, realtime users will not be used any more.
In the stream of updated drafts for the IETF meeting in december, I found a draft namedReal-time text interworking between PSTN and IP networks.This is work-in-progress in Asterisk svn trunk. We already have support for T.140 realtime text in the SIP channel, and support for US TDD phones in the Zap/PSTN channel driver. During this summer, I’ve assisted a developer who implemented a RTT gateway in Asterisk, so you could call with an old PSTN TDD phone and have a conversation with a modern SIP client with support for realtime text.This code will soon be implemented in Asterisk svn trunk, so Asterisk will become the reference implementation for this draft. How many xOIP acronyms do we have now? FOIP, VOIP, TOIP? I surely missed a few there… :-)From the abstract:
IP networks can support real-time text communication. SIP-based real- time text is called Text-over-IP or ToIP. PSTN networks support real-time text using textphones (or TTYs). When real-timetext is supported by different networks, gateways are needed to provide interoperability. Real-time text capable gateways may also support real-time voice.This specification describes procedures for interworking between ToIP and PSTN textphones using a real-time text capable gateway (RTT gateway). It also describes ways to route calls to RTT gateways for several call scenarios. Procedures that support the phased introduction of RTT gateways and procedures that support the invocation of text channels at any time during the call are included. Interworking of PSTN text phones thatdo not support simultaneity of voice and text with IP User Agents that support simultaneous voice and text is also described.
The draft SIP Multiple REFER describes a way to send a call transfer (REFER) indicating setup of a new call to multiple destinations, i.e. initiating a conference. This could be handy between two Asterisk servers, a way to set up a conference on a different server, optimized for conferences.
RFC 3261 (SIP) [RFC3261] is extended by RFC 3515 [RFC3515] with aREFER method that allows a user agent to request a server to send a request to a third party. Still, a number of applications need torequest a server to initiate transactions towards a set of destinations. In one example, the moderator of a conference may wantthe conference server to send BYE requests to a group of participants. In another example, the same moderator may want the conference server to INVITE a set of new participants. We define an extension to the REFER method so that REFER request can be used to refer servers to multiple destinations. In addition, this mechanism uses the suppression of the REFER method implicit subscription specified in RFC 4488 to suppress REFER’s implicit subscription.
The world of realtime technologies on IP networks and the Internet is cooking with ideas, solutions and innovation. Combining it with social networks, and even more is happening. Protocols like SIP, XMPP/Jabber, SIMPLE, Jingle and Open Source solutions like Asterisk, OpenSER, Ekiga, Sofia-SIP, ReSIProcate and others are changing the landscape. It’s high time to get a helicopter view of what’s going on as well as detailed information about the building blocks for this new service delivery network.
Edvina.net is gathering leading experts in this area to the conference The IP Communications Technology Summit 2008 in Stockholm, April 2-4. Tutorials, hands-on labs, conference and user group meetings. Book this date in your almanac today! More information coming soon. If you have an idea or a talk proposal, send e-mail to me, Olle E. Johansson, at oej@edvina.net today!
The Asterisk Developer Team is proud to announce the Asterisk SPE v1.0 Beta Releasefor immediate download on tftp.digium.com.The SPE has been developed as a joint project between Digium, the Asterisk Company,Voop, the European Asterisk Dialtone provider and the Asterisk community.The Asterisk Service Provider Edition is focused on the needs for the new breedof Telecom companies - the Voice over IP Service Providers. It will be availableboth as a free download in Open Source and as a commercial productcalled Asterisk Commercial Service Provider Edition, ACSPE.
- “We felt the need to focus on being an enabler for this new kind of telco ,making sure that Asterisk fits into their network as well as business modelsin a professional way” says Matt Penser, Asterisk innovator. “The previous versions was more targeted to the needs of the business user, a market where Asterisk already is stronger than any other offering on the marke\”.Â
The Asterisk SPE has a number of new features, that makes it the most powerful platform for these companies. No other Open Source package can deliver amatching feature set:
These new solutions will enhance Asterisk and will help the VSP’s toleap light years ahead of their competion.
The VoipRoute SmartRTP bridging technology enhances the Asterisk RTPbridge with a new scheme. In addition to the current RTP bridges - the native bridge,the remote bridge and the hybrid RTP-direct bridge, SmartRTP uses a combinationof the BGP IP routing protocols and the TRIP VoIP routing system to find thebest and fastest way to route calls between IP nodes on the Internet or local network.
- “The SmartRTP bridge system, based on our patented VoipRoute core, makes sure that call latency is minimal. We also enhanced it with a MediaRescue solution that will capture lost media frames and re-insert them in the audio or video stream before it reaches the destination.” says Josua Polk, the Asterisk RTP developer.“This system implements an Asterisk VoipRoute layer on top of the Internet and uses Dundi(TM) to automatically discover new SmartRTP relays and their properties. It practically erases packet loss, jitter and latency from the list of issues for the provider’s support department. We call it SPEake-friendly calls!”Â
The RateRoute(TM) solution is only available in the ACSPE due to licenses fromother vendors, soon to be disclosed. The RateRoute system analyze the call from fifteen distinct properties and use an external hardware accelerator to find the best route to forward the call, be it PSTN or VoIP channels. By using the hardware accelerator RR520P PCI express card, LCR decisions is now down to microseconds without accessing external databases.
- “We’ve implemented this in our commercial VoIP network during development, and it cut our costs by at least 75% and enhanced call quality. Billing and CDRmediation is much easier, since the RateRoute system always picked one outbound service provider that always matched the fifteen criteria for carrier selection” says Anders Runnstam at PulseVoip in Bergen, Norway.Â
The SPITwall(R) technology is developed by Olle E. Johansson, a member of the Asterisk developer team and Senior Technical Advisor for Voop in Bergen, Norway - the Asterisk Dialtone provider.
- “I got more and more annoying calls during development, which disturbed me a lot and caused me to loose concentration. On the other hand, it inspired me to develop SPITwall to be able to filter them out.
I have measured up to 95% success rate on call filtering, which is far beyond any similar products on the market. By not bothering with answering the final 5%, I could concentrate on development again and succesfully finish my development projects.” says Olle.Â
The SPITwall is built on a shared database and use bayesian techniques to analyze the content of the call. It requires Asterisk ChanSpy to be ableto listen in and warn the callee about ongoing unsolicited calls. The callee can also press certain DTMF sequences during the call to mark the call as SPIT. The voice pattern, SPITwall checksums and call properties will then immediately be stored in the Digium SPITcore repository to be available for all other users.
- “Using the community to build a SPIT-fighting database is natural for an Open Source project. The community is the power of Asterisk and by sharing a resource like this, we can make sure that everyone contributes. The SPITshare(r) analyzer makes sure that companiest hat does not contribute will get older data and more SPIT calls” says Jill Timmer, VP or marketing.Â
SPITwall 1.0 is available with English, Norwegian and Swedish language support. Some support for Canadian and southern US dialects is implemented, and will be finished by release time.
In addition to these revolutionary features, Asterisk Service Provider Edition will contain full T.39 faxing (leapfrogging systems that only support up to version 38), the Codename Pomengranade SIP Stack, the Project Okapi IAX3 trunking technology featuring IAX3 over SS7 transport, the VoipVotedigital phone voting system with app_preselect cheating technology and a fully working version of the VirtuAST virtual PBX hosting Asterisk virtualisation core.Â
Asterisk SPE v1.0 beta is available for immediate download. At thistime we’re looking for feedback from service providers.Release date for the 1.0 version is to be released, pending beta tests. ACSP Edition will be available with Telco level 24/8 support May 15th, 2007. The RR520 RateRoute hardware accelerator is in distribution through authorized resellers starting April 10th.Asterisk, Digium, SPITwall, SpitShare, VoipRoute, SmartRTP, RateRoute and Dundi are trademarks that may be registered by Digium, Inc.For immediate release, April 1st 2007. On behalf of the Asterisk development team and project 0401/OlleÂ
One thing I avoided working with for a long time is the Asterisk voicemail code. One module in Asterisk I’ve constantly been naming as one of the worst parts is voicemail. One part of Asterisk that I’ve been kind of avoiding during my trainings is voicemail. And there’s where I’ve spent a lot of time recently… Life is strange.Instead of fixing the current voicemail, I decided to restart. Breakup large apps into small building blocks, allowing Asterisk admins to use the rich dialplan script language or AEL to build a voicemail solution that fits the organization.I’ve named this minivoicemail, which for each addition becomes moreof a bad choice of name for this project. Flexivoicemail could be better… :-)I’ve removed functionality like ODBC and IMAP support, something thatc an be reapplied later. I’ve also not replaced the hooks into other channels for voicemail notification, but that can be done too. I haven’t started replacing voicemailmain(), since I\’ve focused on the need of larger systems where one only supports e-mail notifications of voicemail with audio attached.What I currently have is:Applications
Dialplan functions
CLI commands
New features:
The branch is based on Asterisk 1.2 and can easily be downloaded from svn.digium.com/svn/asterisk/team/oej/minivoicemail I need testers, ideas for new applications and possibly coders that canhelp to complete this.Here’s how you start
And if you want to encourage me further, paypal to info at edvina.net, thanks!
