09/04/2012 - Digium dials up new IP phones for Asterisk, Switchvox
Digium, which earlier this month announced its new Switchvox business phone system was interoperable with LifeSize's videoconferencing platform, today said the new IP phones are available generally.
Digium said the phones were specifically engineered to leverage Asterisk and Switchvox, Digium's unified communications system.
The company said the new phones were include interactive voicemail, visual call parking, one-touch call recording, a searchable contact directory and presence management. The Digium phones also include an app engine with a JavaScript API that allows programmers to create applications that interface directly with core phone features.
It's not just the devices that have new features, the company said, pointing to updates for both Asterisk and Switchvox that are designed to increase functionality and direct integration with the new Digium phones.
The MSRP for the new phones runs between $129 and $279; the Digium phones are currently available in selected geographies.
For more:
- see this release
Related articles:
Digium Switchvox UC phones now interoperate with LifeSize
Kerio's new Asterisk-based Operator 1.2 adds support for analog lines
Digium adds FMC to Switchvox in latest release
Digium acquired Switchvox back in 2007
Digium added unified communications to Switchvox in 2009
12/03/2012 - Digium Switchvox UC phones now interoperate with LifeSize
Videoconferencing is becoming increasingly popular across all levels of businesses, with an estimated 75 percent of businesses expressing interest in using the technology, according to several recent studies.
Digium, looking to increase the value of its Asterisk-based Swicthvox business phone system, which includes unified communications capabilities, has announced it now integrates with the LifeSize Express 220 HD video conferencing offering.
Digium said the move now allows its Switchvox customers, especially small work groups, teams or individual workers, to use the LifeSize Express 220 system to communicate via video. The LifeSize platform includes support for dual HD displays, full HD camera and phone or microphone options.
"The world is moving to UC, combining many separate applications into a single solution, and one important UC application that business users demand is video," said Leslie Conway, vice president of global marketing at Digium. "We've ensured interoperability with Switchvox and the LifeSize Express 220 system to further allow our business customers to have full access to the way they want to communicate at a price they can afford."
Digium created and owns Asterisk, a popular and widely used open source telephony software.
For more:
- see this release
- see this Channelnomics article
Related articles:
Digium adds FMC to Switchvox in latest release
Digium acquired Switchvox back in 2007
Digium added unified communications to Switchvox in 2009
09/03/2012 - The "Asterisk" Story
Editor's Note: Its been awhile but we are back. Big changes coming soon. Had to take leave but that is now in the past. I wanted to start off with posting this Origin story about Asterisk is you haven't committed their story to memory.
Origin stories are all the rage these days, and while perhaps the origin of Asterisk isn’t as exciting as the genesis of Wolverine, it’s still a pretty interesting tale.
Way back in 1999, Mark Spencer had just started Linux Support Services (LSS), an innovative small business that offered support for the Linux operating system. This was the height of the “Dot Com” era, and many start-up businesses were taking advantage of the open source operating system. LSS took off, and as it grew, Mark found that he needed a phone system.
Back in those days, phone systems were 100-percent proprietary. They were also expensive. Not wanting to take out a loan for a phone system he would probably outgrow in a matter of months, Mark decided to build his own PBX. Unlike proprietary phone systems, Mark’s solution was flexible software that took advantage of the power (and price point) of Linux. Mark named the project “Asterisk,” a reference to the wildcard character.
Within a year, the Dot-com bubble popped and the demand for Linux support dried up. Fortunately for Mark, interest in his software PBX had exploded. Linux Support Services quickly pivoted to focus on the growing demand for hardware and services related to Asterisk. The groundswell of interest in an open source telephony system grew into the Asterisk Community with thousands of developers and users who pitched in, providing patches, enhancements and valuable feedback. What started as a pragmatic solution to a cash-flow problem, turned into a revolution.
By 2003, the business had been renamed “Digium” and was well on its way to becoming the world’s leading purveyor of telephony interface hardware.
In the nearly 13 years since Mark released the initial Asterisk code, the PBX market has undergone a massive shift. Open standards now rule what was once a proprietary market. Expensive, limited proprietary PBX hardware has given way to commodity computers running powerful software. Digium has grown from being a niche player to competing with the biggest names in the PBX market.
So, there you have it. That’s how it all started. By the way, if you have an interesting story about how Asterisk or other open source software changed your life, we would love to hear it.
Source: Digium Inc.
02/02/2012 - Digium Introduces World’s First Phones Designed for Asterisk
Digium introduces
a new family of high-definition IP phones. They are the first that are engineered
to fully leverage the power of Asterisk, the world’s most widely adopted open source
communications software, and Switchvox, Digium’s award-winning unified communications
system. With Digium technology on both the server and the phone, users will benefit
from the best possible performance, unprecedented integration and a uniquely customizable
phone system.
Asterisk has always been about flexibility, allowing integrators and developers to create highly customized solutions. Likewise, Digium phones include an app engine with a simple yet powerful JavaScript API that lets programmers create custom apps that run on the phones. They aren’t simply XML pages; Digium phone apps can interface directly with core phone features.
Digium has leveraged this unique programming interface of the phones to create a suite of productivity applications that work with both Asterisk and Switchvox. Switchvox includes a unique web interface called Switchboard that gives each system user control of their personal communications environment. Digium has extended the capabilities of the Switchboard to the phone, putting advanced features like presence management, searchable contact directory, queue monitoring, recording and voicemail control, all at the user’s fingertips.
The Digium phones include the following models:
- D40—An entry-level HD IP phone with 2-line keys. This is Digium’s best value phone, designed for any employee in the company.
- D50—A mid-level HD IP phone with 4-line keys and 10 rapid dial/busy lamp field keys with an easy to print paper label strip for the user’s most important contacts. This model is perfect for users who spend a lot of time on the phone.
- D70—An executive-level HD IP phone with 6-line keys and 10 rapid dial/BLF keys and real-time status information displayed on an additional LCD screen, allowing users to quickly navigate through up to 100 of their most important contacts. Designed for administrators or executives, the D70 offers top-of-the-line features.
26/01/2012 - Kerio's new Asterisk-based Operator 1.2 adds support for analog lines
UC vendor Kerio Technologies, looking to push deeper into the SMB market, this week rolled out its latest iteration of its IP PBX phone system, Kerio Operator 1.2, which now is based on Asterisk 1.8. Kerio also added support for analog phone lines through a Digium TDM410 card, enabling organizations that have limited SIP connectivity to take advantage of Kerio Operator's VoIP features.
The San Jose,Calif.-based company said the upgrade is aimed at making the system secure and simple for its partners, the IT department and employees making and receiving calls.
Kerio said the new platform, which uses industry-standard SIP VoIP protocol, allows small and midsized organizations to trim telephony costs and stay connected anywhere.
Kerio Operator 1.2 is available as a software appliance,with its own security-hardened operating system, a VMware virtual appliance allowing for rapid deployment in production or evaluation environments on standard PC hardware and in two hardware appliance configurations available in some markets.
For more:
- see this release
Related article:
Kerio upgrades its Operator IP PBX solution for SMBs
15/11/2011 - Digium Releases Octal-Span Digital Card; Connects Traditional Telephony Services with Asterisk Communications Systems
Digium announces
the availability of the TE820 Octal-Span digital card. This new high-density solution
compliments Digium’s existing broad suite of telephony card offerings designed specifically
for Asterisk-based communications systems. The TE820 enables Asterisk integrators
and OEMs to build large scale telephony deployments that are both high performance
and cost-effective.
Asterisk is the most widely used open source software for creating business phone systems and other communications applications. The combination of Digium hardware and Asterisk software provides a cost-effective platform for building numerous communications solutions, from PBX systems and VoIP gateways to IVR servers, call centers and complete unified communications suites. The TE820 supports up to 192 channels (in T1/J1 mode) or 240 channels (in E1 mode) and is available with or without hardware echo cancellation.
The TE820 card supports industry standard telephony protocols, including multiple variants of Primary Rate ISDN. Each span can be configured as either CPE or network for optimal flexibility. The optional VPMOCT256 hardware echo cancellation module, based on the industry-leading Octasic chipset, offloads the task of echo cancellation from the CPU, increasing overall system performance and call quality.
The Octal-Span digital card will be available on November 18, 2011 from Digium and Digium partners.
28/10/2011 - Digium and Open Source Community Release Asterisk 10 at AstriCon
Digium releases Asterisk
10. Asterisk is a communications platform that allows developers to create powerful
business phone systems and unified communications solutions. Since its introduction
12 years ago, Asterisk has been used, free of charge, in nearly every country of the
world to power telephone and other communications systems. It has been downloaded
millions of times, including two million last year alone, establishing Asterisk as
the most popular open source telephony engine.
The most important new feature in Asterisk 10 is its wide-band media engine. Digium has replaced Asterisk’s telephony-grade media engine with a more advanced one, providing support for studio-quality audio and a nearly unlimited number of codecs. By supporting high and ultra high-definition voice, Asterisk can now be used to power communications applications that would have otherwise required specialized or expensive equipment and service in order to convey nuances in speech or emotion. Digium has also updated Asterisk’s media support for Asterisk 10, with several new codecs, including Skype’s SILK codec, 32kHz Speex support and pass-through support for CELT.
Built with open source community support
Digium is advancing Asterisk with version 10, while simultaneously leading work on the Asterisk Scalable Communications Framework. Asterisk SCF will allow developers to create real-time communications applications that include voice, video and text that meet the demands of a full range of uses, from embedded applications to enterprise and carrier solutions.
Asterisk 10 makes its debut at AstriCon, the Asterisk User Conference & Expo, in Denver. Hundreds of attendees, including software and PBX developers, enterprise IT pros, systems integrators and call center and CRM developers, welcomed the announcement. In its eighth year, AstriCon is offering conference tracks focusing on technical information, carriers and call centers, cloud computing, commerce, government, enterprise and the Asterisk ecosystem. Developer conferences geared toward contributors to the Asterisk and Asterisk SCF projects are also taking place during this year’s AstriCon.
Asterisk 10 is available for free download and is licensed under the GNU General Public License v2.
New features in Asterisk 10
Asterisk 10 offers developers, integrators, resellers and telephony pros a range of new capabilities. A few include:
- New media engine—Asterisk 10 supports more media types and virtually any type of audio. The overhaul to the media engine allows Asterisk to support a nearly unlimited number of codecs.
- More codecs—The platform includes new codecs, including the wideband version of Speex, Skype’s super-wideband SILK and pass-through support for several CELT variants.
- Additional sampling rates—Asterisk previously operated on 8 and 16 kHz sampled audio, but now supports super- and ultra-wideband sampling rates as file format types for file playback or recording. Asterisk now supports 8, 12, 16, 24, 32, 44.1, 48, 96 and 192 kHz rates for superb audio quality.
- New conferencing application—Digium replaced the MeetMe conferencing bridge with an HD-capable intelligent bridge application called ConfBridge. It supports all codecs and conference rates and works on any Asterisk 10 system, regardless of operating system or architecture. Intelligent mixing algorithms provide each participant with the optimal audio quality for their connection. Also, ConfBridge is fully customizable, so systems administrators and integrators can configure call-in menus on a caller-by-caller basis.
- Support for videoconferencing—ConfBridge relays video of a designated speaker or the current speaker to other participants in the conference. Video-capable SIP devices that use the same codec are required.
- Significant new fax capabilities—Asterisk 10 includes T.38 gateway capabilities that allow outgoing fax calls from analog fax machines to be connected to T.38 fax endpoints over SIP and incoming T.38 fax calls to be delivered directly to fax machines. This allows for more straightforward integration of fax capabilities into an Asterisk system and allows users to get delivery confirmation from other fax machines.
- Text message routing—Asterisk has long been able to send and receive text messages, but can now route messages as well. Asterisk 10 supports the SIP MESSAGE and XMPP protocols, allowing it to act as a text messaging server and bridge between different messaging protocols.
11/08/2011 - Building the realtime web – now with realtime multimedia
06/05/2011 - Splices – how to manage multiple media sessions
01/04/2011 - The IETF, the ITU and the IPv6 forum launches SIP-six – the new version of SIP – moving from SIP 2.0 to SIP 6.0 in one move!
06/02/2011 - What’s wrong with SIP security?
21/10/2010 - Asterisk 1.8 PBX Now Available For Download
The Asterisk Development Team is proud to announce the release of Asterisk 1.8. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.
http://www.asterisk.org/asterisk-versions
The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release.
You can find a summary of the work involved with the 1.8.0 release in the summary:
http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt
A short list of available features includes:
* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
For a full list of changes in the current release candidate, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0
Thank you for your continued support of Asterisk!
18/10/2010 - Asterisk PBX 1.8 Release Candidate 4 Now Available
http://downloads.asterisk.org/pub/telephony/asterisk/
With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc4 is currently scheduled to become the full release of Asterisk 1.8.0.
All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/. It is also very useful to see successful test
reports. Please post those to the asterisk-dev mailing list.
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.
http://www.asterisk.org/asterisk-versions
With the availability of the Asterisk 1.8.0 release candidates, the binary add-on modules for Asterisk produced by Digium have been updated with new versions that are compatible with Asterisk 1.8. The availability of these
modules will assist with the testing of Asterisk 1.8.0 in a wider variety of situations.
This release candidate contains fixes since the last release candidate as reported by the community. A sampling of the changes in this release candidate include:
* Additional fixups in chan_gtalk that allow outbound calls to both Google
Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip
and stunaddr.
(Closes issue #13971. Patched by dvossel)
* Resolve manager crash issue.
(Closes issue #17994. Reported by vrban. Patchd by dvossel)
* Documentation updates for sample configuration files.
(Closes issues #18107, #18101. Reported, patched by lathama, lmadsen)
* Resolve issue where faxdetect would only detect the first fax call in
chan_dahdi.
(Closes issue #18116. Reported by seandarcy. Patched by rmudgett)
* Resolve issue where a channel that is setup and torn down *very* quickly may not have the right call disposition or ${DIALSTATUS}.
(Closes issue #16946. Reported by davidw. Review
https://reviewboard.asterisk.org/r/740/)
* Set TCLASS field of IPv6 header when SIP QoS options are set.
(Closes issue #18099. Reported by jamesnet. Patched by dvossel)
* Resolve issue where Asterisk could crash on shutdown when using SRTP.
(Closes issue #18085. Reported by st. Patched by twilson)
* Fix issue where peers host port would be lost on a SIP reload.
(Closes issue #18135. Reported, tested by lmadsen. Patched by dvossel)
A short list of available features includes:
* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
For a full list of changes in the current release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4
Thank you for your continued support of Asterisk!
06/10/2010 - New Fedora Linux Project Leader Comes From Asterisk Roots
The Red Hat sponsored Fedora Linux community is an open source development effort that includes a diverse set of participants. At the top of the organizational chart for Fedora is the position of Fedora Project Leader, the person tasked with overseeing the general direction and operations of the Fedora project.
In July, Jared Smith took up the position of Fedora Project Leader, replacing the outgoing Paul Frields. Among Smith's first jobs is to guide the development and release of the upcoming Fedora 14 Linux distribution, set for general availability in November. Building the Fedora Linux distribution is one of Smith's key responsibilities as Fedora Project Leader, but it involves more than just pure code.
"A lot of the time we think of Fedora as just the bits and the bytes that we burn on a CD every six months and ship out, but Fedora is more than that, it has to be a community," Smith said. "As such we have to concentrate on building that community and taking care of the community as much as we take care of the bits and bytes."
Smith's vision for Fedora is about ensuring that the Fedora community is an inclusive place where multiple views and contributions are welcome. Smith doesn't necessarily have any new or unique tools for building community, but he does bring a different background to the position than past Fedora Project Leaders.
24/09/2010 - Asterisk PBX 1.8.0 Release Candidate 2 Now Available
The Asterisk Development Team has announced the second release candidate of Asterisk PBX 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Asterisk 1.8.0-rc1 was not released due to an issue found prior to release.
* Make AMI honor enabled=no
(Closes issue #18040. Reported by: twilson
Review: https://reviewboard.asterisk.org/r/938/)
All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker,
https://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.
http://www.asterisk.org/asterisk-versions
With the availability of the Asterisk 1.8.0 release candidates, the binary add-on modules for Asterisk produced by Digium have been updated with new versions that are compatible with Asterisk 1.8. The availability of these
modules will assist with the testing of Asterisk 1.8.0 in a wider variety of situations.
* Add slin16 support for format_wav (new wav16 file extension)
(Closes issue #15029. Reported, patched by andrew. Tested by Qwell)
* Fixes a bug in manager.c where the default configuration values weren't reset
when the manager configuration was reloaded.
(Closes issue #17917. Reported by lmadsen. Patched by bbryant)
* Various fixes for the calendar modules.
(Patched by Jan Kalab.
Reviewboard: https://reviewboard.asterisk.org/r/880/
Closes issue #17877. Review: https://reviewboard.asterisk.org/r/916/
Closes issue #17776. Review: https://reviewboard.asterisk.org/r/921/)
* Add CHANNEL(checkhangup) to check whether a channel is in the process of
being hung up.
(Closes issue #17652. Reported, patched by kobaz)
* Fix a bug with MeetMe where after announcing the amount of time left in a
conference, if music on hold was playing, it doesn't restart.
(Closes issue #17408, Reported, patched by sysreq)
* Fix interoperability problems with session timer behavior in Asterisk.
(Closes issue #17005. Reported by alexcarey. Patched by dvossel)
* Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was
determined to be one of the most significant bottlenecks in SIP registration
processing. This patch improved the speed of an astdb load test by 50000%
(yes, Fifty-Thousand Percent). On this particular load test setup, this
doubled the number of SIP registrations the server could handle.
(Review: https://reviewboard.asterisk.org/r/825/)
* Don't clear the username from a realtime database when a registration
expires. Non-realtime chan_sip does not clear the username from memory when a
registration expiries so realtime probably shouldn't either.
(Closes issue #17551. Reported, patched by: ricardolandim. Patched by
mnicholson)
* Don't hang up a call on an SRTP unprotect failure. Also make it more obvious
when there is an issue en/decrypting.
(Closes issue #17563. Reported by Alexcr. Patched by sfritsch. Tested by
twilson)
* Many more issues. This is a significant upgrade over Asterisk 1.8.0 beta 5!
A short list of available features includes:
* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
For a full list of changes in the current release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc2
Thank you for your continued support of Asterisk!
15/09/2010 - Asterisk PBX 1.6.2.12 Now Available
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.12 resolves several issues reported by the community and would have not been possible without your participation.
The following is a sample of the issues resolved in this release:
* Fix issue where DNID does not get cleared on a new call when using
immediate=yes with ISDN signaling.
(Closes issue #17568. Reported by wuwu. Patched by rmudgett)
* Several updates to res_config_ldap.
(Closes issue #13573. Reported by navkumar. Patched by navkumar, bencer.
Tested by suretec)
* Prevent loss of Caller ID information set on local channel after masquerade.
(Closes issue #17138. Reported by kobaz, patched by jpeeler)
* Fix SIP peers memory leak.
(Closes issue #17774. Reported, patched by kkm)
* Add Danish support to say.conf.sample
(Closes issue #17836. Reported, patched by RoadKill)
* Ensure SSRC is changed when media source is changed to resolve audio delay.
(Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)
* Only do magic pickup when notifycid is enabled.
A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a
call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber
that a device is ringing. This option should only be enabled when the new
'notifycid' option is set, but this was not the case. Instead the call-id
value was included for every RINGING Notify message, which caused a
regression for people who used other methods for call pickup.
(Closes issue #17633. Reported, patched by urosh. Patched by dvossel.
Tested by: dvossel, urosh, okrief, alecdavis)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.12
Thank you for your continued support of Asterisk!
15/09/2010 - Asterisk PBX 1.4.36 Now Available
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.36 resolves several issues reported by the community and would have not been possible without your participation.
The following is a sample of the issues resolved in this release candidate:
* Fix issue where DNID does not get cleared on a new call when using
immediate=yes with ISDN signaling.
(Closes issue #17568. Reported by wuwu. Patched by rmudgett)
* Fix issue where SIP promiscuous redirect could fail to dial the
redirect (app_queue).
* Fixes issue with translator frame not getting freed. This issue prevented
G.729 licenses from being freed up.
(Closes issue #17630. Reported by manvirr. Patched by dvossel)
* Ensure SSRC is changed when media source is changed to resolve audio delay.
(Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)
* Q931 - Sending PROGRESS after sending ALERTING is a protocol error.
(Closes issue #17874. Reported, patched by nic_bellamy)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.36
Thank you for your continued support of Asterisk!
04/08/2010 - VoicePulse Announces SIP Trunking Interoperability with IPitomy PBX Products
VoicePulse and IPitomy announced that they have successfully completed interoperability testing between SIP products and services. VoicePulse is now interoperable with IPitomy’s Pure IP PBX platform. 23/07/2010 - Asterisk PBX 1.6.2.10 Now Available
he Asterisk Development Team has announced the release of Asterisk PBX 1.6.2.10. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.10 resolves several issues reported by the community and would have not been possible without your participation.
The following are a few of the issues resolved by community developers:
* Allow users to specify a port for DUNDI peers.
(Closes issue #17056. Reported, patched by klaus3000)
* Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is
set.
(Closes issue #16815. Reported, patched by rain)
* If there is realtime configuration, it does not get re-read on reload unless
the config file also changes.
(Closes issue #16982. Reported, patched by dmitri)
* Send AgentComplete manager event for attended transfers.
(Closes issue #16819. Reported, patched by elbriga)
* Correct manager variable 'EventList' case.
(Closes issue #17520. Reported, patched by kobaz)
In addition, changes to res_timing_pthread that should make it more stable have
also been implemented.
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10
Thank you for your continued support of Asterisk!
23/07/2010 - Asterisk PBX 1.4.34 Now Available
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.34 resolves several issues reported by the community and would have not been possible without your participation. Thank you!
The following are a few of the issues resolved by community developers:
* Allow users to specify a port for DUNDi peers.
(Closes issue #17056. Reported, patched by klaus3000)
* Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is
set.
(Closes issue #16815. Reported, patched by rain)
* First caller into a dynamic conference new enters the pin once.
(Closes issue #15878. Reported, patched by pabelanger)
* Send AgentComplete manager events in the event of blind and attended
transfers.
(Closes issue #16819. Reported, patched by elbriga)
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.34
Thank you for your continued support of Asterisk!
30/06/2010 - Asterisk libpri 1.4.11.3 Now Available
http://downloads.asterisk.org/pub/telephony/libpri/
This release fixes a regression in the calling number assignment logic:
* Calling Number assignment logic change in libpri 1.4.11. Restored the old behaviour if there is more than one calling number in the incoming SETUP message. A network provided number is reported as ANI.
(Closes issue #17495. Reported and tested by ibercom. Patched by rmudgett)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11.3
Thank you for your continued support of Asterisk!
23/06/2010 - Asterisk PBX 1.4.33.1 Released
The Asterisk Development Team has announced the release of Asterisk 1.4.33.1. This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.33.1 resolves a regression involving the use of FXO signaling in chan_dahdi where a channel could continue ringing after it has been answered.
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33.1
Thank you for your continued support of Asterisk!
03/06/2010 - Asterisk PBX 1.6.2.8 Now Available
The Asterisk Development Team has announced the release of Asterisk PBX 1.6.2.8. This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.8 resolves several issues reported by the community, and would have not been possible without your participation.
The following are a few of the issues resolved by community developers:
* Enable auto complete for CLI command 'logger set level'.
(Closes issue #17152. Reported, patched by pabelanger)
* Make the mixmonitor thread process audio frames faster.
(Closes issue #17078. Reported, tested by geoff2010. Patched by dhubbard)
* Add missing 'useragent' field to sip-friends.sql file.
(Closes issue #17171. Reported, patched by thehar)
* Add example dialplan for dialing ISN numbers (http://www.freenum.org)
(Closes issue #17058. Reported, patched by pprindeville)
* Fix issue with double "sip:" in header field.
(Closes issue #15847. Reported, patched by ebroad)
* Add ability to generate ASCII documentation from the TeX files by running
'make asterisk.txt'.
(Closes issue #17220. Reported by lmadsen. Tested, patched by pabelanger)
* When StopMonitor() is called, ensure that it will not be restarted by a
channel event.
(Closes issue #16590. Reported, patched by kkm)
* Small error in the T.140 RTP port verbose log.
(Closes issue #16998. Reported, patched by frawd. Tested by russell)
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.8
Thank you for your continued support of Asterisk!
--
24/05/2010 - SIPit26: IPv6 tests and experiences
At SIPit26 we began setting up a series of automated self-tests for IPv6, like we’ve done previously with SIP/TLS. We also integrated IPv6 in as many multiparty tests as possible, to see how IPv4 and IPv6 lived together.Some notes and experiences:
- IPv4-only applications will receive IPv6 in messaging. Even if an application DO NOT support IPv6-native connections, the application will surely get IPv6 addresses in various places in the message. In SIP, a call may traverse an IPv6 proxy before reaching your IPv4 proxy or phone. Via headers will have IPv6 and maybe a record-route header too. All user agents needs to support this. We had at least one crash in a proxy that failed to parse an IPv6 address.
- Placing an IPv4 call to a proxy that forwards the message to an IPv6 phone without handling RTP traversal leads to issues as well. The phone gets an IPv6 address in the Contact: header and failes to send the ACK properly. This happened with Asterisk. Because of parsing failure, the parser gave up and sent ACKs and BYEs to the wrong address.
- We did successfully set up calls between IPv6 user agents using IPv6 proxys. The failures happened in the mixed scenarious.
- When placing a call to a domain that was configured with both A and AAAA records for the SRV records, but only one of them responding, we noticed long timeouts before failover, if that even happened. Many discussions about this followed, which lead to the conclusion that this was a poorly configured domain. Some implementations have hard-coded a preference for IPv4 since IPv6 is mostly used over tunnels and add latency today. This should be user-configurable. An owner of a domain can use SRV record weights to indicate a preference to one or the other protocols, which is a better solution. If you use IPv6 over tunnels, make sure that you separate host records for A and AAAA and have a preference towards the A record hosts in your SRV records.
We do need to continue testing all kinds of migration scenarious to be able to come up with a best current practise document. SIPit26 gave us many good experiences to build from. I hope that testing continues at SIPit27 with the new SIPit IPv6-o-matic(R)(C)(TM) and the prompts from Allison Smith!
© Edvina AB, Sollentuna, Sweden 2010 VoIP-Forum. All Rights Reserved.
.22/04/2010 - Asterisk Billing Software - A2Billing 1.7 released
We have just released A2Billing, version 1.7 codenamed Larch. This version, as well as fixing a number of bugs, adds some new features and is available free of charge under the AGPL license to install yourself at www.a2billing.net. Alternatively, Star2Billing S.L. - The commercial arm of A2Billing – can provide professional installation services, training, ongoing support and development.
· Speed Dial IVR feature, so customers can add speed dial numbers via an IVR
· Added Admin UI notification on New Signup, so that you know when someone has signed up to the system.
· New IVR locking system and IVR information
· Agent and Admin have new controls for CallerID, SpeedDial, SIP/IAX Config and DID Information.
· SOAP enhancements, which allow a remote application to communicate with A2Billing.
· Customize the signup fields, so you only show the fields that are relevant to your customer base for on-line sign-up.
On Saturday, April 17th 2010, at 09:00 Eastern Standard Time, Star2Billing S.L. will be talking in technical detail about one of Star2Billing’s more popular products, the IP-Centrex and Multi-tenant PBX system as described at http://www.star2billing.com/multi-tenant at the Atlanta User’s Group Annual Conference via video link.
The Conference will be broadcasted via the Internet. To sign up to attend in person, or to get the access details to attend remotely, see their link at http://atlaug.com/drupal/fest2010
There are many other speakers during the day covering subjects as diverse as endpoints, security, sales and high availability by some of the leaders of the VoIP industry.
The IP-Centrex and Multi-Tennant system is only one of the systems that Star2Billing S.L can provide installation, training and support services. In addition to our products blended with A2Billing listed on our website at www.star2billing.com, we have also built other solutions for our customers as diverse as web-integrated “Expert on Call” systems to “Power Dialler” platforms for large scale telephony outbound advertising and debt management campaigns and designing CMS (Content Management System) plug-ins to integrate seamlessly with A2Billing to provide a structured, integrated and branded customer facing website which is simple to maintain, and needs no formal technical training.
For more information, please see our website at www.star2billing.com
