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Asterisk 1.8.8.0 Now Available

The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.8.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.8.0 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:

* Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required.
(Closes ASTERISK-17288. Reported by: Mikael Carlsson Tested by: Matthew
Jordan Review: https://reviewboard.asterisk.org/r/1416/)

* Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels.
(Closes issue ASTERISK-18090. Patched by Kinsey Moore)

* Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
(Closes issue ASTERISK-18340. Reported by: Thomas Arimont. Closes issue
ASTERISK-17725. Reported by: kwk. Tested by: twilson, jrose)

* Fix crashes in ast_rtcp_write()
(Closes issue ASTERISK-18570)
Related issues that look like they are the same problem:
(Issue ASTERISK-17560, ASTERISK-15406, ASTERISK-15257, ASTERISK-13334, ASTERISK-9977, ASTERISK-9716)
Review: https://reviewboard.asterisk.org/r/1444/
Patched by: Russell Bryant

* Fix for incorrect voicemail duration in external notifications.
This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration.
(Closes ASTERISK-16981. Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443)

* Prevent segfault if call arrives before Asterisk is fully booted.
(Patched by alecdavis. https://reviewboard.asterisk.org/r/1407/)

* Fix remote Crash Vulnerability in SIP channel driver (AST-2011-012)
http://downloads.asterisk.org/pub/security/AST-2011-012.pdf

* Fix locking order in app_queue.c which caused deadlocks
(Closes issue ASTERISK-18101. Reported by Paul Rolfe, patched by Gregory Nietsky)
(Closes issue ASTERISK-18487. Reported by Jason Legault, patched by Gregory Nietsky)

* Fix regression in configure script for libpri capability checks
(Closes issue ASTERISK-18687. Reported by norbert, patched by Richard Mudgett)

* Prevent BLF subscriptions from causing deadlocks.
(Closes issue ASTERISK-18663)
Review: https://reviewboard.asterisk.org/r/1563/

* Fix deadlock if peer is destroyed while sending MWI notice.
(Closes issue ASTERISK-18747)
Reported by: Gregory Hinton Nietsky

* Fix issue with setting defaultenabled on categories that are already enabled by default.
(Closes issue ASTERISK-18738)
Reported by: Paul Belanger

* Don’t crash on INFO automon request with no channel AST-2011-014. When automon was enabled in features.conf, it was possible to crash Asterisk by sending an INFO request if no channel had been created yet.

* Fixed crash from orphaned MWI subscriptions in chan_sip This patch resolves the issue where MWI subscriptions are orphaned by subsequent SIP SUBSCRIBE messages.

* Default to nat=yes; warn when nat in general and peer differ AST-2011-013. It is possible to enumerate SIP usernames when the general and user/peer nat settings differ in whether to respond to the port a request is sent from or the port listed for responses in the Via header.

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0

Thank you for your continued support of Asterisk!

 

Voip: Asterisk Security Releases Available

The Asterisk Development Team has announced security releases for Asterisk 1.4, 1.6.2 and 1.8. The available security releases are released as versions 1.4.43, 1.6.2.21 and 1.8.7.2.

These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk versions 1.4.43, 1.6.2.21, and 1.8.7.2 resolves an issue with possible remote enumeration of SIP endpoints with differing NAT settings.

The release of Asterisk versions 1.6.2.21 and 1.8.7.2 resolves a remote crash possibility with SIP when the “automon” feature is enabled.

The issues and resolutions are described in the AST-2011-013 and AST-2011-014 security advisories.

For more information about the details of these vulnerabilities, please read the security advisories AST-2011-013 and AST-2011-014, which were released at the same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.43
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.21
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.7.2

Security advisory AST-2011-013 is available at:

* http://downloads.asterisk.org/pub/security/AST-2011-013.pdf

Security advisory AST-2011-014 is available at:

* http://downloads.asterisk.org/pub/security/AST-2011-013.pdf

Thank you for your continued support of Asterisk!

Remote crash possibility with SIP and the “automon” feature enabled

Asterisk Project Security Advisory – AST-2011-014

Summary:      Remote crash possibility with SIP and the “automon”

Description:  When the “automon” feature is enabled in features.conf, it is possible to send a sequence of SIP requests that cause Asterisk to dereference a NULL pointer and crash.

Resolution:    Applying the referenced patches that check that the pointer is not NULL before accessing it will resolve the issue. The
“automon” feature can be disabled in features.conf as a workaround.

Patches

Download URL Revision
http://downloads.asterisk.org/pub/security/AST-2011-014-1.6.2.diff 1.6.2.20
http://downloads.asterisk.org/pub/security/AST-2011-014-1.8.diff 1.8.7.1

Links

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security

This document may be superseded by later versions; if so, the latest version will be posted at
http://downloads.digium.com/pub/security/AST-2011-014.pdf and http://downloads.digium.com/pub/security/AST-2011-014.html

Discover the Asterisk-based innovation in ClarityLogic – Digium Innovation Award Winner

Congratulations to Clarity, a division of Plantronics, for being recognized as the Digium Innovation Award winner. The Digium Innovation Award is designed to recognize developers, customers and partners for outstanding achievements that are improving business processes, overcoming technology challenges and enhancing the company’s bottom line.

Based in Chattanooga, Tenn., Clarity is the leading supplier of amplified telephones, notification systems, assistive listening devices and other communications devices for millions with hearing loss. With Asterisk-based technology, the company created ClarityLogic, a first-of-its-kind service that enables customer representatives to remotely retrieve and adjust settings, quickly resolving customer issues. This is critically important for Clarity’s primary customer-senior citizens-as many have a difficult time hearing, remembering and performing complex, technical instructions given over the phone.

Asterisk innovators are invited to participate in the 2012 competition by submitting an application online.

Source: Digium.com

Digium Releases Octal-Span Digital Card

Connecting Traditional Telephony Services with Asterisk Communications Systems

TE820 Offers Highest Single-Card Port Density Available for use with Asterisk

Digium®, Inc., the Asterisk® Company, today announced 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.

(more…)

Asterisk 10.0.0-rc1 Now Available

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

All Asterisk users 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.

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 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 (More information available at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 )
* 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.

http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-rc1

Thank you for your continued support of Asterisk!