Digium today released Asterisk 1.8, a major new release of the popular open source telephony platform. According to Digium, it includes more than 200 notable new features.
I spoke with Digium's Product Manager, Steve Sokol, to learn more about this important release.
“Asterisk 1.8 is the first Long Term Support release since version 1.4,” said Russell Bryant, Digium’s lead Asterisk developer. “It includes hundreds of enhancements, many of which will help community members building large-scale solutions. The Asterisk community has, as always, been absolutely integral in the development of this major release. Now that it’s available, we’re rolling our development resources to work on the planned set of Asterisk 1.10 features.”
Stefan Wintermeyer, founder of AMOOMA GmbH, a German systems integrator specializing in Asterisk, contributor to Asterisk 1.8 and Digium partner, commented: “The new ISDN features in Asterisk 1.8 are a big step forward for the European Asterisk community. We’ve been waiting a long time and are pleased that Digium went the extra mile. Even without the ISDN enhancements, the pure performance jump in 1.8 is significant. We can save money on hardware because more calls can be handled by a single Asterisk installation.”
Here's some highlighted new features in Asterisk 1.8:
- Secure real-time transport protocol (RTP) support—New end-to-end VoIP encryption of signaling and media to compliment the existing encrypted signaling support.
Steve explained SRTP support will be huge since it will allow for government and financial deployments of the Asterisk platform.
- Security event framework—Modular capability for collecting and distributing security events within Asterisk.
- Extensive additions to ISDN-BRI functionality—Call completion services, connected party identification, ETSI advice of charge (AOC), message waiting indicator (MWI), call rerouting and call deflection.
- Session initiation protocol (SIP) changes—Substantial increase in the speed of registrations, transport layer security (TLS) improvements and more flexible network address translation (NAT) handling.
- IPv6 support—Although IPv6 has a slow uptake in the enterprise, at the carrier core, IPv6 is progressing very quickly. Steve mentioned that could be the initial place where IPv6 can be leveraged on the Asterisk platform. He said, "I suspect initially you'll see it in the carrier and cable deployments. In fact, we have support now for PacketCable NCS built into Asterisk as well, so those two goes nicely hand-in-hand. There are a lot of cable providers that are rolling out IPV6 for all the set top boxes and other devices that are attached to their network." He added, "The other area IPv6 is big is Asia. There is a lot of IP underruns in Asia where there is billions of people with billions of devices. So it's really caught on there. In a lot of ways that has been a gating factor for Asterisk deployments in Asia. But now that IPv6 is there, we're expecting an even larger percentage of adoption."
- Calendar integration—Support for Microsoft Exchange, CalDav and iCalendar. This is a pretty cool feature. If for instance your Calendar has you listed as in a meeting, Asterisk can automatically send the caller to your voicemail. Or if you are marked as 'or Out of Office' Asterisk can automatically forward to your cell phone.
- Channel event logging (CEL)—Enhanced call tracking and logging for better audit trail and billing purposes.
- XMPP distributed messaging—Better scalability for message waiting and device state.
- Improved internationalization and localization—Asterisk offers improved handling of concatenated audio playback (dates, numbers).
- Google Talk and Google Voice support—Inbound and outbound support for Google Talk and Google Voice calling. Google is serious about voice. Google is offering a free DID and free calling until the end of 2010 via Google Talk. So with the Google Talk integration, you essentially have a FREE trunk to make free calling on your Asterisk PBX. Take that Skype Connect!
- High-resolution timestamps for call data records (CDR)—Carrier and enterprise users can track call times to the microsecond.
- Better support for voice codecs—16 kHz signed linear media streams are now supported. Additional HD voice codecs supported. Siren 7 and Siren 14 codecs from Polycom are also now supported
- PacketCable NCS 1.0 support—Allows cable companies to use Asterisk as an option to create business services.
- Default de-noise for conference bridge calls—Conference calls will sound clearer.
- ConfBridge application enhancements—DAHDI hardware is no longer required to use this software feature. New call conferencing application that does not require the DAHDI kernel interface to operate.
- Pitch shift functions—The pitch of audio, including of callers' voices, can be manipulated.
- Multicast RTP paging—Extremely efficient and scalable method for handset paging. Instead of sending an INVITE to 50 IP phones, you can leverage multicast to send a broadcast to the IP phones. Steve said Polycom, snom, and he said he thought Aastra phones supported this multicast feature. Multicast means a little more work for your switches and routers, but much less stress on the Asterisk server than the traditional way.
- Faster development and more robust unit testing—Digium has implemented Agile development and a new automated testing framework. The Agile process streamlines development and gives Asterisk users a better view into development plans.
AsteriskNOW, the GUI front-end to Asterisk, will need to be updated to take advantage of all these new features. Steve Sokol said they're working on updating AsteriskNOW. However, he said there are additional challenges with providing a GUI to some of these new features. He explained, "The GUI will need to catch up and build implementations of the different features we've built into Asterisk. For example if you want to use the calendaring feature we would need to have the GUI guys come along and build a means of filling out the forms that has your calendar server on it and your username and password to query your calendar." Have no worry you anti-hand-editing-Asterisk-config-files - they're working on it. Steve added that the FreePBX folks have copies of 1.8 and they're hard at work as well to update that popular GUI.
Asterisk 1.8 is released under the GNU General Public License (GPLv2). It is free of charge and available for download at http://www.asterisk.org. Go get your copy today!