Asterisk 1.6.1 Release Candidate 1 Now Available

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 1.6.1, tagged as version 1.6.1-rc1. Release candidate 1.6.1-rc1 is available for immediate download at http://downloads.digium.com/.

This release candidate includes a fix to SIP registrations when using realtime.

Additional crash issues have been resolved, in addition to making chan_sip more robust in handling unique scenarios. Issues found in this release candidate can be reported at http://bugs.digium.com/.

For a full list of changes in this release candidate, please see the

ChangeLog:

http://svn.digium.com/view/asterisk/tags/1.6.1-rc1/ChangeLog?view=co

Also see the CHANGES file for useful information about what is new in Asterisk 1.6.1. See the CHANGES file at:

http://svn.digium.com/view/asterisk/tags/1.6.1-rc1/CHANGES?view=co

Thank you for your continued support of Asterisk!

Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released

The Asterisk.org development team has announced the release of Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5. These releases are available for immediate download from http://downloads.digium.com/.

This update for Asterisk includes a security fix for chan_iax2. Please see the associated security adivisory for more details:

http://downloads.digium.com/pub/security/AST-2009-001.html

These updates are a fix to a previous security release (released as versions 1.2.31, 1.4.22.1, and 1.6.0.3).

The new versions are being released after additional testing revealed some issues with the way that scanning for users was blocked. Those issues have been corrected in this release.

This security issue affects the 1.2, 1.4, and 1.6 series of Asterisk.

Also note, that Asterisk 1.6.0.4-rc1 was released yesterday prior to the security update. That release has been removed as there will be no 1.6.0.4 release, but rather will be reincarnated as 1.6.0.6-rc1. The reason for the dead release is to avoid 5 digit release numbers.

ChangeLogs for the various releases are available at:

http://downloads.digium.com/pub/asterisk/ChangeLog-1.2.31.1

http://downloads.digium.com/pub/asterisk/ChangeLog-1.4.22.2

http://downloads.digium.com/pub/asterisk/ChangeLog-1.4.23.1

http://downloads.digium.com/pub/asterisk/ChangeLog-1.6.0.5

Thank you for your continued support of Asterisk!

Asterisk 1.6.0.4 Release Candidate 1 Now Available

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 1.6.0.4, tagged as version 1.6.0.4-rc1. Release candidate

1.6.0.4-rc1 is available for immediate download at http://downloads.digium.com/.

This release candidate includes fixes for OS build compatibility, terminal compatibility, documentation updates, and resolves several crash issues. Issues found in this release candidate can be reported at http://bugs.digium.com/.

For a full list of changes in this release candidate, please see the

ChangeLog:

http://svn.digium.com/view/asterisk/tags/1.6.0.4-rc1/ChangeLog?view=markup

Thank you for your continued support of Asterisk!

Friday Jan 23 at 12 Noon EST: Open Source vs Commercial

Randal Schwartz (FLOSS Weekly) joins us for this slightly OT discussion about Open Source software and its role in all of our lives. You're all welcome to join in as usual.

We will also be connected to a g722 bridge for more experimentation with the merits of wide band audio.

This should give you all the info needed :

http://www.voipusersconference.org

and jump on IRC.freenode.net on #voip-users-conference

to listen or talk, just call

sip:7463#22622#1@proxy.ideasip.com

See you there.

/r

TWITTER:

'voipusers' , 'VUClmailinglist'

'MLasteriskbiz' subjects of -users ML

"MLasteriskusers' subjects of -biz ML

Aastra Introduces two Entry Level phones to their SIP product lineup

From the press release:

Two new models added to the 67xi Series portfolio offering more choice to users.

Designed to deliver enterprise-grade advanced features and performance normally found at higher price points, the Model 6730i and Model 6731i SIP phones share the same powerful, open-standards firmware design found on all Aastra 67xi telephones. Offering up to 6 lines with call appearances, full duplex speakerphones, intercom, paging and auto answer features as well as XML browser capabilities, these SIP phones are ideally suited for regular telephone usage in small and large business environments as well as enterprise and home-based applications. These models, featuring a slightly smaller design footprint than other 67xi Series phones, are fully compatible with Aastra's IP-PBX Key System, the AastraLink Pro^TM 160.

Please click here to view the data sheets on these two models.

The 6731i telephone, which is available now, is unmatched in the market place for its small desk foot print, price and performance. The product features a Dual 10/100 Mbps switched Ethernet ports (LAN and PC ports) with Power over Ethernet (POE) support, TLS and STRP security, as well as the full range of features offered in the Aastra 2.x SIP firmware release. The lower priced 6730i, which can be ordered now with an expected ship date of April, offers similar features as the 6731i but comes equipped with a single 10/100 Mbps Ethernet port without POE support.

Asterisk 1.4.23 Now Available!

This release is a significant bug fix update for the 1.4 release series.

The following list of bugs were resolved with the participation of the community, and this release would not have been possible without your help!

* Fixed datastore handling crashes in app_queue.c.

- Closes issues 14086. Reported by ZX81. Tested by: ZX81, festr.

* Update autosupport script to supply info for both Zaptel and DAHDI

in 1.4 and be sure to run dahdi_test in 1.6.x and trunk instead of

zttest.

- Closes issue 14132. Reported by and patch contributed by: dsedivec

* Don't overflow when paging more than 128 extensions. The number of available

slots for calls in app_page was hardcoded to 128. Proper bounds checking

was not in place to enforce this limit, so if more than 128 extensions were

passed to the Page() app, Asterisk would crash. This patch instead

dynamically allocates memory for the ast_dial structures and removes the

(non-functional) arbitrary limit. This issue would have special importance

to anyone who is dynamically creating the argument passed to the Page

application and allowing more than 128 extensions to be added by an outside

user via some external interface.

- Closes issue 14217. Reported by and patch contributed by: a_villacis

* Do not crash if we are not passed a followme ID.

- Closes issue 14106. Reported by and patch contributed by: ys

* Change the way the T.38 SDP attribute handling to make it more liberal in

what it accepts, and more strict in what it sends.

- Closes issue 13967. Reported by: linulin. Patches contributed by:

arcivanov

In addition to the issues cited above, several issues related to call parking have been resolved, thereby making Asterisk call parking more robust.

See issues 13820, 13747, 14066, 14228, 13854, 12854, and 13139 for more information. Special thanks to otherwiseguy, davidw, bluefox, waverly360, kobaz, Adam Lee, krisk84, and murf.

For a full list of changes, see the ChangeLog:

http://svn.digium.com/svn-view/asterisk/tags/1.4.23/ChangeLog?view=markup

Thank you for your support of Asterisk!

CRTC Broadcasting Notices of consultation:

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