Asterisk status

Status:

  • 2007-03-07 Asterisk 1.4.1
  • 2006-12-26 Asterisk 1.4.0
  • 2006-12-18 Asterisk 1.4.0beta4 and Zaptel 1.4.0beta3
  • 2006-11-19 Asterisk 1.4.0beta3 and Zaptel 1.4.0beta2
  • 2006-11-09 Zaptel 1.2.11
  • 2006-10-19 Asterisk 1.2.13
  • 2006-10-19 Asterisk 1.0.12
  • 2006-09-21 Asterisk 1.4.0beta2, Zaptel 1.4.0beta1 and libpri 1.4.0beta1
  • 2006-09-15 Asterisk 1.2.12.1 and Zaptel 1.2.9.1
  • 2006-09-09 Asterisk 1.2.12 and Zaptel 1.2.9
  • 2006-08-23 Asterisk 1.2.11 and Zaptel 1.2.8
  • 2006-07-15 Asterisk 1.2.10
  • 2006-06-05 Asterisk 1.2.9.1
  • 2006-04-13 Asterisk 1.2.7.1
  • 2006-03-28 Asterisk 1.2.6 and Zaptel 1.2.5
  • 2006-03-04 Asterisk 1.2.5
  • 2006-02-00 Zaptel 1.2.4
  • 2006-02-00 Asterisk 1.2.4 and Zaptel 1.2.3
  • 2006-01-00 Asterisk 1.2.2
  • 2005-06-07 Asterisk 1.2.1 - read the change log
  • Asterisk 1.2.0 has been released. You can download sources here. Check here about the huge number of new features
  • Asterisk 1.0.10 has been released, see http://ftp.digium.com/pub/asterisk/ChangeLog for the changelog. This is the final release of the 1.0 series.
  • The development version of Asterisk, v1.3dev, is now hosted by Subversion instead of CVS.


SineApps news on Asterisk development:

http://www.sineapps.com/news.php - html
http://www.sineapps.com/rssfeed.php - rss feed

Release cycle update (Jan 2006)

Asterisk 1.2 was released over 1 year after Asterisk 1.0, which resulted in many users trying to run the development version of Asterisk in a production capacity so that they could take advantage of the new features that had been added. This produced a flurry of extraneous bug reports and caused extra work for the developers as they could not work on changes that would actually cause disruption of the development tree.

In an effort to combat this problem, and to give the community a more predictable release cycle, the process is being organized so that such a long time between releases will never happen again.

Beginning in January of 2006, we will produce new major Asterisk releases on a six month cycle.
The development cycle will be organized in this fashion:

MONTHS 1 - 3

The first three months of the development cycle are when the development branch will be changed most drastically. The tree is open to large architectural changes as well as new feature enhancements and bug fixes.

MONTHS 4 - 5

For the next two months, the development branch will no longer receive architectural changes. New features that are ready to be merged will still be accepted at this point.

MONTH 6

The last month is reserved for beta testing. No more features will be accepted for the upcoming release. Beta releases will be made on a weekly cycle, culminating in one (or two) release candidate releases just before the final release.

Asterisk 1.4 is scheduled to be released in the beginning of July, 2006. Once the release is made, a branch will be created. This branch will then receive maintenance for bug fixes only. At that point, the development cycle will start over to prepare for the next major release of Asterisk, scheduled for January of 2007.

The Asterisk Development Team

OEJ's Asterisk News


Stable 1.2.x branch

  • Asterisk v1.2.0 was released on Nov. 16, 2005
  • Asterisk 1.2.1 was released on Dec 7, 2005

Stable 1.0.x branch - development completed

Releases 1.0.5 thru 1.0.10 have been made public, no detailed fix list on this page

November 23, 2005

Changelog of Asterisk 1.0.10

chan_local
- In releases 1.0.8 and 1.0.9, the Local channels that are created would not be masqueraded into the new channel type. This has now been fixed.

chan_sip
- The 'insecure' options have been changed to support matching peers by IP only, not requiring authentication on incoming invites, or both. Before, to not require authentication on incoming invites also required matching peers based on IP only.

chan_zap
- Before, call waiting could occur during the initial ringing on the line. This has now been fixed.

app_disa
- We will now not set the accountcode if one is not supplied.

app_meetme
- If the first caller into a conference hangs up while being prompted for the conference pin number, the conference will no longer be held open.

app_userevent - Events created with this application were indicated as a "call" event instead of a "user" event. This made the "user" event permissions not work correctly.

app_voicemail
When using the externpass option for voicemail, the password will be immediately updated in memory as well, instead of having to wait for the next time the configuration is reloaded.

app_zapras
- We now ensure buffer policy is restored after RAS is done with a channel. This could cause audio problems on the channel after zapras is done with it.

- res_agi
We now unmask the SIGHUP signal before executing an AGI script. This fixes problems where some AGI scripts would continue running long after the call is over.

extensions
- A potential crash has been fixed when calling LEN() to get the length of a string that was 80 characters or larger.

logger
- The Asterisk logger will automatically detect when a log file needs to be rotated. However, this feature could put Asterisk in a nasty loop
that would result in a crash.

general
- Added man pages for astgenkey, autosupport, and safe_asterisk

January 14, 2005

Asterisk 1.0.4 is out. You can get it from the CVS stable branch.
ChangeLog:
— general
— fix memory leak evident with extensive use of variables
— update IAXy firmware to version 22
— enable some special write protection
— enable outbound DTMF
— fix seg fault with incorrect usage of SetVar
— other minor fixes including typos and doc updates
— chan_sip
— fix codecs to not be case sensitive
— Re-use auth credentials
— fix MWI when using type=friend
— fix global NAT option
— chan_agent / chan_local
— fix incorrect use count
— chan_zap
— Allow CID rings to be configured in zapata.conf
— no more patching needed for UK CID
— app_macro
— allow Macros to exit with '*' or '#' like regular extension processing
— app_voicemail
— don't allow '#' as a password
— add option to save voicemail before going to the operator
— fix global operator=yes
— app_read
— return 0 instead of -1 if user enters nothing
— res_agi
— don't exit AGI when file not found to stream
— send script parameter when using FastAGI

December 6,2004

Asterisk 1.0.3 was released. You can get it from the CVS stable branch. Here are the changes:
— chan_zap
— fix seg fault when doing *0 to flash a trunk
— rtp
— seg fault fix
— chan_sip
— fix to prevent seg fault when attempting a transfer
— fix bug with supervised transfers
— fix codec preferences
— chan_h323
— fix compilation problem
— chan_iax2
— avoid a deadlock related to a static config of a BUNCH of peers
— cdr_pgsql
— fix memory leak when reading config
— Numerous other minor bug fixes


September 30, 2004

Hello Everyone!

As I am sure most of you have heard, Asterisk 1.0 was released last week during Astricon. There have already been some bug fixes, so Asterisk 1.0.1 is now available.

Now that we have reached this milestone, there will be some changes in the release structure.

There will be two branches in CVS: stable and development. There will be some major changes coming soon to the development branch, but the stable branch will only include bug fixes.

Nothing will change for getting the source from the development branch. For getting the source from the stable branch, you will issue the following command:

cvs checkout -r v1-0 asterisk zaptel libpri asterisk-sounds asterisk-addons

The stable version of Asterisk and its companions will also be available on the Digium ftp server. They will be regularly updated as bug fixes are added to the current stable release. I am hoping that this will help facilitate the propagation of releases throughout the various distributions. Further instructions for accessing CVS and the ftp server are available on asterisk.org.

If you find any problems with the stable branch, submit them to the bug tracker as normal. However, it is important that you note whether the bug applies to 1.0, development, or both.

If you have any questions or comments regarding the stable branch, don't hesitate to contact me.

Thanks!

Russell Bryant
irc: "drumkilla"

September 23, 2004

Asterisk 1.0.0 is out! Mark Spencer released this version during his keynote to the first Astricon. Source is downloadable at http://www.asterisk.org and mirrors listed http://asterisk.paperwork.com/home/ .

Secret date

Asterisk 1.0 RC2

July 18, 2004

Asterisk 1.0 RC1

We have officially made the first release candidate of Zaptel, Libpri, Asterisk and Gastman available. While there are still open major bugs, they are relatively limited, and it was time to go ahead and get the 1.0 ball rolling in earnest.

USA: ftp://ftp.digium.com/pub/asterisk

Mirrors:
UK: http://www.goods2world.com/downloads/
Italy: http://asterisk.espia-net.net/asteriskRC1/
Germany: http://www.asterisk-support.de/mirror/
Denmark: ftp://debian.marlow.dk/mirrors/digium/
Australia: http://mirror.afoyi.com/asterisk/
Malaysia: http://www.mithotech.com/asterisk/mirror/

Hot on the announcement this morning by Mark, I have updated and rebuilt new Asterisk RPMS for RedHat 7.3, 8, 9 and Fedora Core 1:
ftp://ftp.nacs.net/asterisk

Enjoy the code. Special thanks to all the bug marshals and contributers and to everyone who has supported Asterisk through the purchase of Digium hardware. I apologize in advance for our slow 768k link. Please report bugs at http://bugs.digium.com and *please* read the instructions there before submitting bugs or patches.

I hope everyone can share in my excitement, especially as excited as I'll be once I've actually had some sleep.

-Mark Spencer

Note: As of 7/17/04 CVS the parking.conf file has been renamed to features.conf in order to be a more general config file.

June 18, 2004

The following is pasted from SecurityFocus Newsletter #254:
Asterisk PBX Multiple Logging Format String Vulnerabilities
BugTraq ID: 10569
Remote: Yes
Date Published: Jun 18 2004
Relevant URL: http://www.securityfocus.com/bid/10569
Summary:
It is reported that Asterisk is susceptible to format string
vulnerabilities in its logging functions.

An attacker may use these vulnerabilities to corrupt memory, and read or
write arbitrary memory. Remote code execution is likely possible.

Due to the nature of these vulnerabilities, there may exist many
different avenues of attack. Anything that can potentially call the
logging functions with user-supplied data is vulnerable.

Versions 0.7.0 through to 0.7.2 are reported vulnerable.

This was fixed in cvs HEAD and stable on 4/13/2004 and a new source
release was made at the time (version 0.9.0)

June 13, 2004

The stable-1.0 CVS tree will *not* be released as a 1.0. There has been too many bug reports on both the stable-1.0 and the HEAD branch, and new are coming in.
The decision is to base the future 1.0-release on the CVS head tree. The current "stable-1.0" tree will be released as something intermediary, maybe 0.91, and at that point it will be considered end-of-life. At some point when we have cleared the bug tracker from major issues, we will fork a new stable-1.0 tree and start working on that.

May 9, 2004

Read the config sample files! (even if you're an Asterisk guru)

For those of you that have a working installation that you keep using, this is a reminder to check into the configs/ directory of the Asterisk source tree, regardless if you downloaded a tar ball or from CVS.

As we add or change features in Asterisk, the sample config files are updated. If you look there, you might get new insights into how to solve your problems. Also, you might find new features that you really need.

If you have a new installation "gmake samples" or "make samples" will install these files for you.

In CVS head, the development source tree, we've added quite a lot of information recently to these files. They are more educational now and contains a lot of sample configurations.

Check app_groupcount!

There's a new app in Asterisk town. In fact, there are several new applications in CVS head. One of the major recent additions is app_groupcount, that you can use to limit the number of calls to, well, just about anything. A SIP peer, a PRI link, a call center staff member, a conference and calls to or from your boy and/or girlfriend :-)

The command for setting a group is setgroup(), the command for enforcing a limitation is checkgroup().

Please start using this instead of the incominglimit and outgoinglimit settings in sip.conf. These are not working as expected and the more general solution with app_groupcount is a much better solution that works cross channels. This is an end-of-life warning for outgoinglimit and incominglimit :-)

As always, the CLI command "show applications" and "show application <name>" is your best friend.

Set your SIP realm!

In CVS head, the SIP channel is now able to use a proper SIP realm for authentication. The realm is the server group that has a common authentication for a user. It could be one server or a number of servers that shares a password/user database.

According to the SIP RFC, it should be set either to a domain or a hostname, depending on what your realm covers. It should be globally unique. Up to know, all Asterisk servers used the "asterisk" realm. That made it a bit hard for some phones to know the difference between one server and another.

Please note that if you are using the "md5secret" setting in sip.conf, this secret is based on the realm. If you change the realm, you need to rehash your secrets.

Asterisk 1.0: Less than five bugs away

If you follow the CVS, you will notice that there are very few changes in the stable part of the source tree. Only bug fixes go in there and Mark have been working like crazy to fix the major bugs. The bug tracker had almost 300 open bugs just a while ago, and we are now down to a handful identified bugs. As usual with Open Source Software, relase is not set to marketing plans. Release will come when the software is ready to be shipped. So when Mark decides that we've fixed the bugs that needs fixing, a release candidate will be made and published for download.

Please plan to help us test the 1.0rc1 real hard. Do whatever you can to crash it, to make it dial your mother-in-law when you really want to talk to your husband, to make it connect the whole office to the HR departments secret conference call by mistake and accidentally fill your hard disk drive with non-existing voice mail messages. We do not belive that you can, but if you can, report the bugs and help us move forward to a 1.0 release!

If you want to start stress-testing it now, download the stable CVS release. Instructions is to be found at http://www.asterisk.org

Beginning of May, 2004

A lot of patches and new features were moved into the CVS HEAD branch. This includes an enhanced voicemail (with message envelope, an option to directly call the originator etc). SetGroup and CheckGroup are new applications that for example permit to limit the number of outgoing calls on DSL (often required due its limited bandwidth). Also a SMS application was included in CVS. Finally a lot of work is currently being performed on internationlization of voice prompts.

April 13, 2004: Release 0.9.0

Stable version 0.9.0 has been released

April 11, 2004

We're getting closer and closer to a 1.0 release of Asterisk. In order to
get there, the development is now 110% focused on solving major, critical
and crash bugs. (And yes, if you follow the CVS updates, you'll see the
impossible extra 10% :-)

YOU'RE NOT FORGOTTEN, BUT ON HOLD (WITHOUT ON-HOLD-MUSIC)
If you're reporting other bugs, please don't be disappointed or require
immediate action if nothing happens for a while. We need to focus in order
to get to a 1.0. Those reports will not be ignored forever, but maybe for
a short while. And if you get questions or requests for additional
information from a bug marshal, please answer. The bug marshals are there
to help collect all the information a developer needs in order to locate
the problem and try to fix it.

This also applies to patches adding new functionality. We will consider
them for additions later on, but not now. So hang on to your patch report,
refine your code, make sure the community tests it and remind us later if
nothing happens. While waiting on hold, make sure you've submitted a
disclaimer to Digium and confirm that on the bug report.

THE GRAND PLAN
The plan is first to release a 1.0 stable release. As soon as that is out,
new features added to CVS head but not to stable will be ported into the
stable CVS tree and we're going 1.1 as soon as that code is tested and
considered stable.

After that, work will restart on adding new code, new functions and new
architectures. There's quite a lot of work out there that will be very
good additions to an after-1.1 release of Asterisk, especially in the area
of general functions that will add major flexibility both for applications
and for how me manage configurations. Database driven configuration
handlers, embedded script interpreters and extensions to various API:s. A
lot of code that needs extensive testing before considered stable.

HELP US
In order to help us move ahead, please spend some extra time testing the
stable CVS source tree and help us solve problems. Visit the bug tracker,
see if you recognize some of the unresolved problems discussed there.
Maybe you can help us with additional information.

REFERENCES
Bug tracker: http://bugs.digium.com
CVS stable download instructions:
http://www.asterisk.org/index.php?menu=download

End of March, 2004

After the number of untreated bug reports & feature request amounting to almost 300 Mark S. managed to invite a great number of new "bug marshals" to assume reviewer task within the Mantis bugtracker.
A couple of long-standing significant bug notes have been incorporated into the CVS, for example: #214 (caller position announcement in queues), #1114 and #1289 (several MGCP improvements), whereas bug #156 (enhanced voicemail features) still is on hold.

Feb 06, 2004: Stable CVS branch

1.0 hasn't been released yet, but you are able to check out the future stable 1.0 branch from cvs. Only bug fixes will go in this branch so you can automate the checkout/update and rpm build process.

To check out code from our STABLE 1.0 Branch CVS repository for Asterisk ONLY:

  1. cd /usr/src
  2. export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot
  3. cvs login - the password is anoncvs.
  4. cvs checkout -r v1-0_stable asterisk

Nothing going into stable will break anything else (or it shouldn't). Everything would be tested before being applied to 1.0-stable by bug marshals.

Feb 04, 2004: Release 0.7.2

Asterisk 0.7.2 is now released and contains lots and lots of bug fixes from the bug tracker. Highly recommended for people running 0.7.1.
  • DSP Fixes
  • Fix unloading of Zaptel
  • Pass Caller*ID/ANI properly on call forwarding
  • Add indication for Italy
  • Countless small bug fixes from bug tracker

Jan 14, 2004: Release 0.7.1

Asterisk 0.7.1 has been released with a few bug fixes.
  • Fixed timed include context's and GotoIfTime
  • Fixed chan_h323 it now gets remote ip properly instead of 127.0.0.1

Jan 13, 2004: Release 0.7.0

Asterisk 0.7.0 has been released (the previous release was 0.5.0).

Jan 08, 2004: Roadmap

Prompted by the recent discussion on the mailing list regarding the
Asterisk development and release process (or lack thereof), John Todd,
Thorsten Lockert, Brian K. West, and myself have put together a plan to
address the most significant two legitimate concerns that have been
expressed regarding these processes. Specifically:

Concern #1: Asterisk release schedules and path to 1.0.0

Asterisk version 0.7.0 will be released by Monday Jan 12, 2004. Later
that week, we will create a stable branch from which eventually 1.0.0 will
be tagged. Only bug fixes will go into the release branch, while feature
requests and bug fixes will continue to go into the head branch.

If you are currently using CVS asterisk on a production server, we suggest
that you move to the new stable CVS branch when it becomes available.
Instructions for using the new stable CVS branch will be made available on
asterisk.org next week. Snapshots of the stable branch will also be made
available periodically as Asterisk 0.9.x for those not using CVS. If you
wish to remain on the cutting edge, you may leave your system using the
head CVS as it is currently.

Concern #2: Slow integration of bug fixes and feature requests into CVS

With the assistance of John Todd and Brian West, we have added
documentation about how the bug tracker operates, available at
http://www.digium.com/bugtracker.html. This document should help new
users understand how the process of submitting bugs works, how to properly
follow up on bugs to be sure they get applied, and how to contribute to
the bug tracking process as a Bug Marshal, thus accellerating the process.
In addition, I am commiting 5-10 dedicated hours of my own time per week
to work with Bug Marshals on reviewing bugs, patches and feature requests.

Conclusion:

Hopefully these steps will help improve the quality and stability of the
Asterisk code, and make it easier for people who wish to contribute to
Asterisk to do so, while maintaining Asterisk's availibility to continue
to advance new features and applications.


Mark, John, Thorsten, and Brian

Go back to Asterisk

Status:

  • 2007-03-07 Asterisk 1.4.1
  • 2006-12-26 Asterisk 1.4.0
  • 2006-12-18 Asterisk 1.4.0beta4 and Zaptel 1.4.0beta3
  • 2006-11-19 Asterisk 1.4.0beta3 and Zaptel 1.4.0beta2
  • 2006-11-09 Zaptel 1.2.11
  • 2006-10-19 Asterisk 1.2.13
  • 2006-10-19 Asterisk 1.0.12
  • 2006-09-21 Asterisk 1.4.0beta2, Zaptel 1.4.0beta1 and libpri 1.4.0beta1
  • 2006-09-15 Asterisk 1.2.12.1 and Zaptel 1.2.9.1
  • 2006-09-09 Asterisk 1.2.12 and Zaptel 1.2.9
  • 2006-08-23 Asterisk 1.2.11 and Zaptel 1.2.8
  • 2006-07-15 Asterisk 1.2.10
  • 2006-06-05 Asterisk 1.2.9.1
  • 2006-04-13 Asterisk 1.2.7.1
  • 2006-03-28 Asterisk 1.2.6 and Zaptel 1.2.5
  • 2006-03-04 Asterisk 1.2.5
  • 2006-02-00 Zaptel 1.2.4
  • 2006-02-00 Asterisk 1.2.4 and Zaptel 1.2.3
  • 2006-01-00 Asterisk 1.2.2
  • 2005-06-07 Asterisk 1.2.1 - read the change log
  • Asterisk 1.2.0 has been released. You can download sources here. Check here about the huge number of new features
  • Asterisk 1.0.10 has been released, see http://ftp.digium.com/pub/asterisk/ChangeLog for the changelog. This is the final release of the 1.0 series.
  • The development version of Asterisk, v1.3dev, is now hosted by Subversion instead of CVS.


SineApps news on Asterisk development:

http://www.sineapps.com/news.php - html
http://www.sineapps.com/rssfeed.php - rss feed

Release cycle update (Jan 2006)

Asterisk 1.2 was released over 1 year after Asterisk 1.0, which resulted in many users trying to run the development version of Asterisk in a production capacity so that they could take advantage of the new features that had been added. This produced a flurry of extraneous bug reports and caused extra work for the developers as they could not work on changes that would actually cause disruption of the development tree.

In an effort to combat this problem, and to give the community a more predictable release cycle, the process is being organized so that such a long time between releases will never happen again.

Beginning in January of 2006, we will produce new major Asterisk releases on a six month cycle.
The development cycle will be organized in this fashion:

MONTHS 1 - 3

The first three months of the development cycle are when the development branch will be changed most drastically. The tree is open to large architectural changes as well as new feature enhancements and bug fixes.

MONTHS 4 - 5

For the next two months, the development branch will no longer receive architectural changes. New features that are ready to be merged will still be accepted at this point.

MONTH 6

The last month is reserved for beta testing. No more features will be accepted for the upcoming release. Beta releases will be made on a weekly cycle, culminating in one (or two) release candidate releases just before the final release.

Asterisk 1.4 is scheduled to be released in the beginning of July, 2006. Once the release is made, a branch will be created. This branch will then receive maintenance for bug fixes only. At that point, the development cycle will start over to prepare for the next major release of Asterisk, scheduled for January of 2007.

The Asterisk Development Team

OEJ's Asterisk News


Stable 1.2.x branch

  • Asterisk v1.2.0 was released on Nov. 16, 2005
  • Asterisk 1.2.1 was released on Dec 7, 2005

Stable 1.0.x branch - development completed

Releases 1.0.5 thru 1.0.10 have been made public, no detailed fix list on this page

November 23, 2005

Changelog of Asterisk 1.0.10

chan_local
- In releases 1.0.8 and 1.0.9, the Local channels that are created would not be masqueraded into the new channel type. This has now been fixed.

chan_sip
- The 'insecure' options have been changed to support matching peers by IP only, not requiring authentication on incoming invites, or both. Before, to not require authentication on incoming invites also required matching peers based on IP only.

chan_zap
- Before, call waiting could occur during the initial ringing on the line. This has now been fixed.

app_disa
- We will now not set the accountcode if one is not supplied.

app_meetme
- If the first caller into a conference hangs up while being prompted for the conference pin number, the conference will no longer be held open.

app_userevent - Events created with this application were indicated as a "call" event instead of a "user" event. This made the "user" event permissions not work correctly.

app_voicemail
When using the externpass option for voicemail, the password will be immediately updated in memory as well, instead of having to wait for the next time the configuration is reloaded.

app_zapras
- We now ensure buffer policy is restored after RAS is done with a channel. This could cause audio problems on the channel after zapras is done with it.

- res_agi
We now unmask the SIGHUP signal before executing an AGI script. This fixes problems where some AGI scripts would continue running long after the call is over.

extensions
- A potential crash has been fixed when calling LEN() to get the length of a string that was 80 characters or larger.

logger
- The Asterisk logger will automatically detect when a log file needs to be rotated. However, this feature could put Asterisk in a nasty loop
that would result in a crash.

general
- Added man pages for astgenkey, autosupport, and safe_asterisk

January 14, 2005

Asterisk 1.0.4 is out. You can get it from the CVS stable branch.
ChangeLog:
— general
— fix memory leak evident with extensive use of variables
— update IAXy firmware to version 22
— enable some special write protection
— enable outbound DTMF
— fix seg fault with incorrect usage of SetVar
— other minor fixes including typos and doc updates
— chan_sip
— fix codecs to not be case sensitive
— Re-use auth credentials
— fix MWI when using type=friend
— fix global NAT option
— chan_agent / chan_local
— fix incorrect use count
— chan_zap
— Allow CID rings to be configured in zapata.conf
— no more patching needed for UK CID
— app_macro
— allow Macros to exit with '*' or '#' like regular extension processing
— app_voicemail
— don't allow '#' as a password
— add option to save voicemail before going to the operator
— fix global operator=yes
— app_read
— return 0 instead of -1 if user enters nothing
— res_agi
— don't exit AGI when file not found to stream
— send script parameter when using FastAGI

December 6,2004

Asterisk 1.0.3 was released. You can get it from the CVS stable branch. Here are the changes:
— chan_zap
— fix seg fault when doing *0 to flash a trunk
— rtp
— seg fault fix
— chan_sip
— fix to prevent seg fault when attempting a transfer
— fix bug with supervised transfers
— fix codec preferences
— chan_h323
— fix compilation problem
— chan_iax2
— avoid a deadlock related to a static config of a BUNCH of peers
— cdr_pgsql
— fix memory leak when reading config
— Numerous other minor bug fixes


September 30, 2004

Hello Everyone!

As I am sure most of you have heard, Asterisk 1.0 was released last week during Astricon. There have already been some bug fixes, so Asterisk 1.0.1 is now available.

Now that we have reached this milestone, there will be some changes in the release structure.

There will be two branches in CVS: stable and development. There will be some major changes coming soon to the development branch, but the stable branch will only include bug fixes.

Nothing will change for getting the source from the development branch. For getting the source from the stable branch, you will issue the following command:

cvs checkout -r v1-0 asterisk zaptel libpri asterisk-sounds asterisk-addons

The stable version of Asterisk and its companions will also be available on the Digium ftp server. They will be regularly updated as bug fixes are added to the current stable release. I am hoping that this will help facilitate the propagation of releases throughout the various distributions. Further instructions for accessing CVS and the ftp server are available on asterisk.org.

If you find any problems with the stable branch, submit them to the bug tracker as normal. However, it is important that you note whether the bug applies to 1.0, development, or both.

If you have any questions or comments regarding the stable branch, don't hesitate to contact me.

Thanks!

Russell Bryant
irc: "drumkilla"

September 23, 2004

Asterisk 1.0.0 is out! Mark Spencer released this version during his keynote to the first Astricon. Source is downloadable at http://www.asterisk.org and mirrors listed http://asterisk.paperwork.com/home/ .

Secret date

Asterisk 1.0 RC2

July 18, 2004

Asterisk 1.0 RC1

We have officially made the first release candidate of Zaptel, Libpri, Asterisk and Gastman available. While there are still open major bugs, they are relatively limited, and it was time to go ahead and get the 1.0 ball rolling in earnest.

USA: ftp://ftp.digium.com/pub/asterisk

Mirrors:
UK: http://www.goods2world.com/downloads/
Italy: http://asterisk.espia-net.net/asteriskRC1/
Germany: http://www.asterisk-support.de/mirror/
Denmark: ftp://debian.marlow.dk/mirrors/digium/
Australia: http://mirror.afoyi.com/asterisk/
Malaysia: http://www.mithotech.com/asterisk/mirror/

Hot on the announcement this morning by Mark, I have updated and rebuilt new Asterisk RPMS for RedHat 7.3, 8, 9 and Fedora Core 1:
ftp://ftp.nacs.net/asterisk

Enjoy the code. Special thanks to all the bug marshals and contributers and to everyone who has supported Asterisk through the purchase of Digium hardware. I apologize in advance for our slow 768k link. Please report bugs at http://bugs.digium.com and *please* read the instructions there before submitting bugs or patches.

I hope everyone can share in my excitement, especially as excited as I'll be once I've actually had some sleep.

-Mark Spencer

Note: As of 7/17/04 CVS the parking.conf file has been renamed to features.conf in order to be a more general config file.

June 18, 2004

The following is pasted from SecurityFocus Newsletter #254:
Asterisk PBX Multiple Logging Format String Vulnerabilities
BugTraq ID: 10569
Remote: Yes
Date Published: Jun 18 2004
Relevant URL: http://www.securityfocus.com/bid/10569
Summary:
It is reported that Asterisk is susceptible to format string
vulnerabilities in its logging functions.

An attacker may use these vulnerabilities to corrupt memory, and read or
write arbitrary memory. Remote code execution is likely possible.

Due to the nature of these vulnerabilities, there may exist many
different avenues of attack. Anything that can potentially call the
logging functions with user-supplied data is vulnerable.

Versions 0.7.0 through to 0.7.2 are reported vulnerable.

This was fixed in cvs HEAD and stable on 4/13/2004 and a new source
release was made at the time (version 0.9.0)

June 13, 2004

The stable-1.0 CVS tree will *not* be released as a 1.0. There has been too many bug reports on both the stable-1.0 and the HEAD branch, and new are coming in.
The decision is to base the future 1.0-release on the CVS head tree. The current "stable-1.0" tree will be released as something intermediary, maybe 0.91, and at that point it will be considered end-of-life. At some point when we have cleared the bug tracker from major issues, we will fork a new stable-1.0 tree and start working on that.

May 9, 2004

Read the config sample files! (even if you're an Asterisk guru)

For those of you that have a working installation that you keep using, this is a reminder to check into the configs/ directory of the Asterisk source tree, regardless if you downloaded a tar ball or from CVS.

As we add or change features in Asterisk, the sample config files are updated. If you look there, you might get new insights into how to solve your problems. Also, you might find new features that you really need.

If you have a new installation "gmake samples" or "make samples" will install these files for you.

In CVS head, the development source tree, we've added quite a lot of information recently to these files. They are more educational now and contains a lot of sample configurations.

Check app_groupcount!

There's a new app in Asterisk town. In fact, there are several new applications in CVS head. One of the major recent additions is app_groupcount, that you can use to limit the number of calls to, well, just about anything. A SIP peer, a PRI link, a call center staff member, a conference and calls to or from your boy and/or girlfriend :-)

The command for setting a group is setgroup(), the command for enforcing a limitation is checkgroup().

Please start using this instead of the incominglimit and outgoinglimit settings in sip.conf. These are not working as expected and the more general solution with app_groupcount is a much better solution that works cross channels. This is an end-of-life warning for outgoinglimit and incominglimit :-)

As always, the CLI command "show applications" and "show application <name>" is your best friend.

Set your SIP realm!

In CVS head, the SIP channel is now able to use a proper SIP realm for authentication. The realm is the server group that has a common authentication for a user. It could be one server or a number of servers that shares a password/user database.

According to the SIP RFC, it should be set either to a domain or a hostname, depending on what your realm covers. It should be globally unique. Up to know, all Asterisk servers used the "asterisk" realm. That made it a bit hard for some phones to know the difference between one server and another.

Please note that if you are using the "md5secret" setting in sip.conf, this secret is based on the realm. If you change the realm, you need to rehash your secrets.

Asterisk 1.0: Less than five bugs away

If you follow the CVS, you will notice that there are very few changes in the stable part of the source tree. Only bug fixes go in there and Mark have been working like crazy to fix the major bugs. The bug tracker had almost 300 open bugs just a while ago, and we are now down to a handful identified bugs. As usual with Open Source Software, relase is not set to marketing plans. Release will come when the software is ready to be shipped. So when Mark decides that we've fixed the bugs that needs fixing, a release candidate will be made and published for download.

Please plan to help us test the 1.0rc1 real hard. Do whatever you can to crash it, to make it dial your mother-in-law when you really want to talk to your husband, to make it connect the whole office to the HR departments secret conference call by mistake and accidentally fill your hard disk drive with non-existing voice mail messages. We do not belive that you can, but if you can, report the bugs and help us move forward to a 1.0 release!

If you want to start stress-testing it now, download the stable CVS release. Instructions is to be found at http://www.asterisk.org

Beginning of May, 2004

A lot of patches and new features were moved into the CVS HEAD branch. This includes an enhanced voicemail (with message envelope, an option to directly call the originator etc). SetGroup and CheckGroup are new applications that for example permit to limit the number of outgoing calls on DSL (often required due its limited bandwidth). Also a SMS application was included in CVS. Finally a lot of work is currently being performed on internationlization of voice prompts.

April 13, 2004: Release 0.9.0

Stable version 0.9.0 has been released

April 11, 2004

We're getting closer and closer to a 1.0 release of Asterisk. In order to
get there, the development is now 110% focused on solving major, critical
and crash bugs. (And yes, if you follow the CVS updates, you'll see the
impossible extra 10% :-)

YOU'RE NOT FORGOTTEN, BUT ON HOLD (WITHOUT ON-HOLD-MUSIC)
If you're reporting other bugs, please don't be disappointed or require
immediate action if nothing happens for a while. We need to focus in order
to get to a 1.0. Those reports will not be ignored forever, but maybe for
a short while. And if you get questions or requests for additional
information from a bug marshal, please answer. The bug marshals are there
to help collect all the information a developer needs in order to locate
the problem and try to fix it.

This also applies to patches adding new functionality. We will consider
them for additions later on, but not now. So hang on to your patch report,
refine your code, make sure the community tests it and remind us later if
nothing happens. While waiting on hold, make sure you've submitted a
disclaimer to Digium and confirm that on the bug report.

THE GRAND PLAN
The plan is first to release a 1.0 stable release. As soon as that is out,
new features added to CVS head but not to stable will be ported into the
stable CVS tree and we're going 1.1 as soon as that code is tested and
considered stable.

After that, work will restart on adding new code, new functions and new
architectures. There's quite a lot of work out there that will be very
good additions to an after-1.1 release of Asterisk, especially in the area
of general functions that will add major flexibility both for applications
and for how me manage configurations. Database driven configuration
handlers, embedded script interpreters and extensions to various API:s. A
lot of code that needs extensive testing before considered stable.

HELP US
In order to help us move ahead, please spend some extra time testing the
stable CVS source tree and help us solve problems. Visit the bug tracker,
see if you recognize some of the unresolved problems discussed there.
Maybe you can help us with additional information.

REFERENCES
Bug tracker: http://bugs.digium.com
CVS stable download instructions:
http://www.asterisk.org/index.php?menu=download

End of March, 2004

After the number of untreated bug reports & feature request amounting to almost 300 Mark S. managed to invite a great number of new "bug marshals" to assume reviewer task within the Mantis bugtracker.
A couple of long-standing significant bug notes have been incorporated into the CVS, for example: #214 (caller position announcement in queues), #1114 and #1289 (several MGCP improvements), whereas bug #156 (enhanced voicemail features) still is on hold.

Feb 06, 2004: Stable CVS branch

1.0 hasn't been released yet, but you are able to check out the future stable 1.0 branch from cvs. Only bug fixes will go in this branch so you can automate the checkout/update and rpm build process.

To check out code from our STABLE 1.0 Branch CVS repository for Asterisk ONLY:

  1. cd /usr/src
  2. export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot
  3. cvs login - the password is anoncvs.
  4. cvs checkout -r v1-0_stable asterisk

Nothing going into stable will break anything else (or it shouldn't). Everything would be tested before being applied to 1.0-stable by bug marshals.

Feb 04, 2004: Release 0.7.2

Asterisk 0.7.2 is now released and contains lots and lots of bug fixes from the bug tracker. Highly recommended for people running 0.7.1.
  • DSP Fixes
  • Fix unloading of Zaptel
  • Pass Caller*ID/ANI properly on call forwarding
  • Add indication for Italy
  • Countless small bug fixes from bug tracker

Jan 14, 2004: Release 0.7.1

Asterisk 0.7.1 has been released with a few bug fixes.
  • Fixed timed include context's and GotoIfTime
  • Fixed chan_h323 it now gets remote ip properly instead of 127.0.0.1

Jan 13, 2004: Release 0.7.0

Asterisk 0.7.0 has been released (the previous release was 0.5.0).

Jan 08, 2004: Roadmap

Prompted by the recent discussion on the mailing list regarding the
Asterisk development and release process (or lack thereof), John Todd,
Thorsten Lockert, Brian K. West, and myself have put together a plan to
address the most significant two legitimate concerns that have been
expressed regarding these processes. Specifically:

Concern #1: Asterisk release schedules and path to 1.0.0

Asterisk version 0.7.0 will be released by Monday Jan 12, 2004. Later
that week, we will create a stable branch from which eventually 1.0.0 will
be tagged. Only bug fixes will go into the release branch, while feature
requests and bug fixes will continue to go into the head branch.

If you are currently using CVS asterisk on a production server, we suggest
that you move to the new stable CVS branch when it becomes available.
Instructions for using the new stable CVS branch will be made available on
asterisk.org next week. Snapshots of the stable branch will also be made
available periodically as Asterisk 0.9.x for those not using CVS. If you
wish to remain on the cutting edge, you may leave your system using the
head CVS as it is currently.

Concern #2: Slow integration of bug fixes and feature requests into CVS

With the assistance of John Todd and Brian West, we have added
documentation about how the bug tracker operates, available at
http://www.digium.com/bugtracker.html. This document should help new
users understand how the process of submitting bugs works, how to properly
follow up on bugs to be sure they get applied, and how to contribute to
the bug tracking process as a Bug Marshal, thus accellerating the process.
In addition, I am commiting 5-10 dedicated hours of my own time per week
to work with Bug Marshals on reviewing bugs, patches and feature requests.

Conclusion:

Hopefully these steps will help improve the quality and stability of the
Asterisk code, and make it easier for people who wish to contribute to
Asterisk to do so, while maintaining Asterisk's availibility to continue
to advance new features and applications.


Mark, John, Thorsten, and Brian

Go back to Asterisk
Created by: JustRumours, Last modification: Thu 08 of Mar, 2007 (20:50 UTC) by cmaj
Please update this page with new information, just login and click on the "Edit" or "Discussion" tab. Get a free login here: Register Thanks! - Find us on Google+