Asterisk 10, Beta 1
Posted by admin in digium, Uncategorized on July 23, 2011
On the heels of Kevin Fleming’s announcement yesterday discussing the changes in the Asterisk versioning scheme, we’d like to formally announce that Asterisk 10, Beta 1 is now available for community testing. Asterisk 10, a Standard Support release, will be the next major release of Asterisk and follows the release of Asterisk 1.8 LTS, a Long Term Support release. For more information on the different releases of Asterisk, check out the Asterisk Versionspage on the Wiki.
Let’s talk about some of its new capabilities.
A major focus of the Asterisk 10 development cycle was Asterisk’s support for media types. In versions of Asterisk 1.8 and prior, Asterisk supported a rather limited number of codecs due to some architectural limitations. Plumbing was ripped out, kitchens were remodeled, girders were swapped, and Asterisk 10 now has a media architecture that’s capable of handling both a nearly unlimited number of codecs as well as codecs with more complex parameters. What does this mean for users? First, it means that Asterisk now comes with some additional codecs, including the 32kHz variant of the Speex codec (previous versions of Asterisk only supported the 8kHz or 16kHz variants), Skype’s Superwideband SILK codec, and pass-through support for the 44.1kHz and 48kHz variants of the CELT format.
Astute readers will note that earlier versions of Asterisk were only capable of operating on 8kHz and 16kHz sampled audio, and that the aforementioned newly-supported codecs operate at rates other than these. You’re absolutely correct. In order to support these new codecs, Asterisk 10 has also been provided with support for a variety of super and ultra-wideband sampling rates, all of which are supported as file format types for file playback or recording.
Asterisk 10.0.0 Beta 1 Now Available
The Asterisk Development Team is pleased to announce the first beta release of Asterisk 10.0.0-beta1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
With the release of the Asterisk 10 branch, the preceding ’1.’ has been removed from the version number per the blog post available at http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-a…
All interested users of Asterisk are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.
All Asterisk users are invited to participate in the #asterisk-testing channel on IRC to work together in testing the many parts of Asterisk. Additionally users can make use of the RPM and DEB packages now being built for all Asterisk releases. More information available at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
Asterisk 10 is the next major release series of Asterisk. It will be a Standard support release, similar to Asterisk 1.6.2. For more information about support time lines for Asterisk releases, see the Asterisk
versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
A short list of included features includes:
- T.38 gateway functionality has been added to res_fax.
- Protocol independent out-of-call messaging support. Text messages not associated with an active call can now be routed through the Asterisk dialplan. SIP and XMPP are supported so far.
- New highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8kHz-192kHz
- Addition of video_mode option in confbridge.conf to provide basic video conferencing in the ConfBridge() dialplan application.
- Support for defining hints has been added to pbx_lua.
- Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
- Much, much more!
A full list of new features can be found in the CHANGES file.
For a full list of changes in the current release, please see the ChangeLog:
Thank you for your continued support of Asterisk!
The Evolution of Asterisk (or: How We Arrived at Asterisk 10)
We are fast approaching the seven-year anniversary of the release of Asterisk 1.0.0, which occurred at the first AstriCon in September, 2004. If you look back a little further, there were various “0.x” releases made as early as December of 1999… my, how time has flown!
We’ve had quite a few ‘major’ releases of Asterisk since then, including 1.2, 1.4, and most recently, 1.8. Each of these releases has included significant changes, and sometimes architecture-improving changes. Each of them has also included substantial new functionality for Asterisk users. Along the way, we’ve been asked by many people in the community when we are going to start working on (or release) “Asterisk 2.0.” Typically, we’ve responded by saying that will not happen until we can really justify such a significant change in the version number. Many open source projects have gone through similar progressions, and quite a number of them have in fact undergone complete (or nearly complete) rewrites resulting in new ‘major’ versions.
The Asterisk project, however, has tried to avoid that level of disruption to its users. Instead we’ve focused on attempting to provide as much backwards compatibility between major releases as we could. As a result, each time we’ve released a new, major version, the decision has been made that “No, this isn’t Asterisk 2.0,” and we’ve continued with the version numbering scheme that Mark Spencer started all those years ago.
Over the past few months though, as we’ve approached the first beta release of the next major version of Asterisk, we’ve been having a somewhat unexpected conversation: about just how different this release is going to be from the releases that most users in the community are using on their production Asterisk systems (primarily Asterisk 1.4, although there are still a lot of 1.2 users as well).
In fact, even though it’s been an evolutionary process, not a revolutionary one, the next major Asterisk release really will be substantially different from Asterisk 1.4 in some very noticeable ways: wideband conferencing support, basic video conferencing support, support for a number of additional VoIP technologies, full-fledged FAX support, and many others.
That has raised the question: Is this Asterisk 2.0? If not, will there ever be an Asterisk 2.0? After quite a lot of discussion, we came to the conclusion that this isnot Asterisk 2.0, but that it’s also quite unlikely that there ever will be such a release; it wouldn’t be in the community’s best interests to release something that is fundamentally different (and not compatible) but still call it ‘Asterisk.’ That then leaves the question we’ve been asked by many people: If there’s never going to be an Asterisk 2.0, why continue to call these releases “1.x”? What does the “1″ mean, if it’s never going to change?
The conclusion that we’ve reached, and that we hope you’ll agree with, is that Asterisk is always going to be Asterisk, and that you don’t need a “1.” prefix on the version number to be able to identify it. So, starting with the next major release, we’re going to drop the “1.” completely. The next major release, which was going to be Asterisk 1.10, will now be just “Asterisk 10″ and subsequent major releases will be “Asterisk 11″, “Asterisk 12″, and so forth.
We’ll continue with our plan to have both standard and long-term support releases of Asterisk, and we’ll update the Asterisk Project Wiki with this information as soon as the first Asterisk 10 beta goes out. In fact, this should occur very soon.
As always, thanks to everyone for their continued support of Asterisk. That especially includes the developer community, the people that find and report issues, the people that help test patches and the people that devote their time to answering questions on IRC channels, the mailing lists and the forums. We hope to see everyone trying out the forthcoming beta, and we look forward to seeing you all at AstriCon 2011!
The Evolution of Asterisk (or: How We Arrived at Asterisk 10)
We are fast approaching the seven-year anniversary of the release of Asterisk 1.0.0, which occurred at the first AstriCon in September, 2004. If you look back a little further, there were various “0.x” releases made as early as December of 1999… my, how time has flown!
We’ve had quite a few ‘major’ releases of Asterisk since then, including 1.2, 1.4, and most recently, 1.8. Each of these releases has included significant changes, and sometimes architecture-improving changes. Each of them has also included substantial new functionality for Asterisk users. Along the way, we’ve been asked by many people in the community when we are going to start working on (or release) “Asterisk 2.0.” Typically, we’ve responded by saying that will not happen until we can really justify such a significant change in the version number. Many open source projects have gone through similar progressions, and quite a number of them have in fact undergone complete (or nearly complete) rewrites resulting in new ‘major’ versions.
The Asterisk project, however, has tried to avoid that level of disruption to its users. Instead we’ve focused on attempting to provide as much backwards compatibility between major releases as we could. As a result, each time we’ve released a new, major version, the decision has been made that “No, this isn’t Asterisk 2.0,” and we’ve continued with the version numbering scheme that Mark Spencer started all those years ago.
Over the past few months though, as we’ve approached the first beta release of the next major version of Asterisk, we’ve been having a somewhat unexpected conversation: about just how different this release is going to be from the releases that most users in the community are using on their production Asterisk systems (primarily Asterisk 1.4, although there are still a lot of 1.2 users as well).
In fact, even though it’s been an evolutionary process, not a revolutionary one, the next major Asterisk release really will be substantially different from Asterisk 1.4 in some very noticeable ways: wideband conferencing support, basic video conferencing support, support for a number of additional VoIP technologies, full-fledged FAX support, and many others.
That has raised the question: Is this Asterisk 2.0? If not, will there ever be an Asterisk 2.0? After quite a lot of discussion, we came to the conclusion that this isnot Asterisk 2.0, but that it’s also quite unlikely that there ever will be such a release; it wouldn’t be in the community’s best interests to release something that is fundamentally different (and not compatible) but still call it ‘Asterisk.’ That then leaves the question we’ve been asked by many people: If there’s never going to be an Asterisk 2.0, why continue to call these releases “1.x”? What does the “1″ mean, if it’s never going to change?
The conclusion that we’ve reached, and that we hope you’ll agree with, is that Asterisk is always going to be Asterisk, and that you don’t need a “1.” prefix on the version number to be able to identify it. So, starting with the next major release, we’re going to drop the “1.” completely. The next major release, which was going to be Asterisk 1.10, will now be just “Asterisk 10″ and subsequent major releases will be “Asterisk 11″, “Asterisk 12″, and so forth.
We’ll continue with our plan to have both standard and long-term support releases of Asterisk, and we’ll update the Asterisk Project Wiki with this information as soon as the first Asterisk 10 beta goes out. In fact, this should occur very soon.
As always, thanks to everyone for their continued support of Asterisk. That especially includes the developer community, the people that find and report issues, the people that help test patches and the people that devote their time to answering questions on IRC channels, the mailing lists and the forums. We hope to see everyone trying out the forthcoming beta, and we look forward to seeing you all at AstriCon 2011!
FaxPro Module
It has now been a little over a month since Tony Lewis announced the release of the new FreePBX Market Place. We have had an excellent response from the FreePBX community on the Professional Modules that have been released thus far. With that said it is apparent that the FaxPro module has stirred up the most interest. I wanted to put this blog out to answer the most common questions I have been getting and explain a little more about the module itself.
Here are some of the most frequently asked questions on FaxPro.
Q. What is the FaxPro module?
A. The Fax Pro module is a reliable, robust inbound/outbound faxing server. You simply enable faxing for any user on the system; then you point a phone number to specific fax user and let the FaxPro module answer the fax call.
Q. Does the FaxPro module work with T.38?
A. T.38 is handled by asterisk behind the scenes and has nothing to do with the FaxPro module. As long as your system is configured properly you can utilize both inbound and outbound t.38.
Q. Can I use FaxPro with my Trixbox / Elastix / Brand X deployment?
A. No. The FaxPro module can only be used with the FreePBX Distro.
Q. I have a standard Asterisk server, can I use the FaxPro module?
A. No. Again the FaxPro module can only be used with the FreePBX Distro.
Q. How is outbound faxing handled?
A. Outbound faxing is done through a web interface on your FreePBX Distro server. Simply enter the phone number you are faxing to and upload a PDF you would like to fax.
Q. I have a custom asterisk deployment. Can you work with me to implement a faxing solution?
A. We are always happy to discuss the scope of your project to determine if there is an opportunity for us to help. Please contact us.
AstLinux 0.7.9 Released
The AstLinux Team announces the 0.7.9 release. There are several updates and bug fixes related to Asterisk. All current users are encouraged to upgrade at your earliest convenience.
Asterisk 1.8.5.0 Now Available
The Asterisk Development Team announces the release of Asterisk 1.8.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.5.0 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
- Fix Deadlock with attended transfer of SIP call
(Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, cmaj) - Fixes thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, rossbeer, kowalma, Freddi_Fonet) - Be more tolerant of what URI we accept for call completion PUBLISH requests.
(Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson) - Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.
(Closes issue #18742. Reported by jkister. Tested by jcovert, jrose) - This patch fixes a bug with MeetMe behavior where the ‘P’ option for always prompting for a pin is ignored for the first caller.
(Closes issue #18070. Reported by mav3rick. Patched by bbryant) - Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup.
(Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett) - Resolve issue where leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read.
(Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard Mudgett) - Resolve issue where wait for leader with Music On Hold allows crosstalk between participants. Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits.
(Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)
For a full list of changes in this release, please see the ChangeLog:
Thank you for your continued support of Asterisk!
libpri 1.4.12 Now Available
The Asterisk Development Team announces the release of libpri version 1.4.12. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/
The following are some of the issues resolved in this release:
- Add call transfer exchange of subaddresses support and fix PTMP call transfer signaling.
- Invalid PTMP redirecting signaling as TE towards NT.
- Add Q931_IE_TIME_DATE to CONNECT message when in network mode.
(issue #18047 (JIRA PRI-114). Reported by: wuwu. Patched by rmudgett) - Swap of master/slave in pri_enslave() incorrect.
(issue #18769 (JIRA PRI-120). Reported by: jcollie. Patched by jcollie) - Fix I-frame retransmission quirks.
- Crash if NFAS swaps D channels on a call with an active timer.
- DMS-100 not receiving caller name anymore.
(issue #18822 (JIRA PRI-121). Reported by: cmorford. Patched by rmudgett) - B channel lost by incoming call in BRI NT PTMP mode.
- Implement the mandatory T312 timer for NT PTMP broadcast SETUP calls.
This release contains several new features, among them:
- ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
- ETSI Advice Of Charge (AOC) support
- ETSI Explicit Call Transfer (ECT) support
- ETSI Call Waiting support for ISDN phones
- ETSI Malicious Call ID support
- Add Display IE text handling options.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1….
Thank you for your continued support of Asterisk!
Asterisk 1.8.5-rc1 Now Available
Posted by admin in asterisk, Release Candidates, sip on June 29, 2011
The Asterisk Development Team announces the first release candidate of Asterisk 1.8.5. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.5-rc1 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release candidate:
- Fix Deadlock with attended transfer of SIP call
(Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, cmaj) - Fixes thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, rossbeer, kowalma, Freddi_Fonet) - Be more tolerant of what URI we accept for call completion PUBLISH requests.
(Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson) - Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.
(Closes issue #18742. Reported by jkister. Tested by jcovert, jrose) - This patch fixes a bug with MeetMe behavior where the ‘P’ option for always prompting for a pin is ignored for the first caller.
(Closes issue #18070. Reported by mav3rick. Patched by bbryant) - Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup.
(Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett) - Resolve issue where leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read.
(Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard Mudgett) - Resolve issue where wait for leader with Music On Hold allows crosstalk between participants. Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits.
(Closes issue #18418. Reported by MrHanMan. Patched by rmudgett) - Fix timerfd locking issue.
(Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz)
For a full list of changes in this release candidate, please see the ChangeLog:
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.19 Now Available (Final Maintenance Release)
The Asterisk Development Team has announced the final maintenance release of Asterisk, version 1.6.2.19. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
Please note that Asterisk 1.6.2.19 is the final maintenance release from the 1.6.2 branch. Support for security related issues will continue until April 21, 2012. For more information about support of the various Asterisk branches, see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
The release of Asterisk 1.6.2.19 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
- Don’t broadcast FullyBooted to every AMI connection The FullyBooted event should not be sent to every AMI connection every time someone connects via AMI. It should only be sent to the user who just connected.
(Closes issue #18168. Reported, patched by FeyFre) - Fix thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Tested by vois, rossbeer, kowalma, Freddi_Fonet. Patched by dvossel) - Don’t delay DTMF in core bridge while listening for DTMF features.
(Closes issue #15642, #16625. Reported by jasonshugart, sharvanek. Tested by globalnetinc, jde. Patched by oej, twilson) - Fix chan_local crashs in local_fixup() Thanks OEJ for tracking down the issue and submitting the patch.
(Closes issue #19053. Reported, patched by oej) - Don’t offer video to directmedia callee unless caller offered it as well
(Closes issue #19195. Reported, patched by one47)
Additionally security announcements AST-2011-008, AST-2011-010, and AST-2011-011 have been resolved in this release.
For a full list of changes in this release, please see the ChangeLog:
Thank you for your continued support of Asterisk!



