Thu, 20 Jan 2011 21:35:00 -0400 The Asterisk Development Team has announced a release for the security issue described in AST-2011-001.
Due to a failed merge, Asterisk 1.8.2.1 which should have included the security fix did not. Asterisk 1.8.2.2 contains the the changes which should have been included in Asterisk 1.8.2.1.
This releases is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases
The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2, 1.8.1.2, and 1.8.2.2 resolve an issue when forming an outgoing SIP request while in pedantic mode, which can cause a stack buffer to be made to overflow if supplied with carefully crafted caller ID information. The issue and resolution are described in the AST-2011-001 security advisory.
For more information about the details of this vulnerability, please read the security advisory AST-2011-001, which was released at the same time as this announcement.
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.2
Security advisory AST-2011-001 is available at:
http://downloads.asterisk.org/pub/security/AST-2011-001.pdf
Thank you for your continued support of Asterisk!
Wednesday, February 16, 2011
Asterisk 1.8.2.2 Now Available (Security Release)
Friday, November 19, 2010
Asterisk PBX 1.8.0 Release Candidate 2 Now Available
* 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!
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!
Friday, November 5, 2010
Release of VoX Mobile Out for Android
According to Companies’ officials, VoX Mobile Out service is a brand new Nimbuzz backed option for cloud-based calling for Android phone users un United States. Pervasip has the long standing records of partnership with Nimbuzz's SIP Premium Affiliate Partner Program that was created to make third party SIP technology providers to deliver their dedicated services in frames of Nimbuzz platform. Pervasip Information Officer, Mark Richards, noted "We are very excited to become the first US-based premier partner to integrate our cloud-based SIP mobile calling software with Nimbuzz. A smooth and easy download, people appreciate having a phone line from VoX, which gives them a second phone number, a second voice mail box and low-priced VoIP calling plans that run over both the 3G and WiFi data networks. We see reports of daily sales for approximately 200,000 Android phones." "The addition of the VoX service supports our mission to provide as wide a range of service to their users as possible, to give them the choice and power to communicate how they want, anywhere at any time and to make international calls at the lowest rates available. We believe the service will be a great success given its relevance to the increasingly savvy and demanding US mobile phone market and dramatic uptake of Android devices," said Evert Jaap Lugt, CEO, Nimbuzz. VoX Communications Corporation provides Voice over IP telephone services to customers across mobile broadband connectivity to the residential and small business markets over fixed broadband connections.
Tuesday, November 2, 2010
German VoIP company TELES release C5 Software
TELES boasts about the latest release, tagging it as ‘ ’ one of the most groundbreaking developments’ ’ in the progress of the product. Company’ s softswitch solution now supports and can be implemented merely for the full integration of an corporate level PBX to a TELES Class 5 softswitch for trunking. The product also will be featured by set of services such as FMC (Functional Management Code) for home based employees, which does not require enterprises to swap legacy PBX equipment thanks to traditional ISDN replacement and emulation of all major ISDN attached services. As per company’ s officials, the latest version of TELES C5 software makes an end-to-end solution and removes all relayed technical compatibility issues. It is aimed to conjoin TELES products just across the board from access gateways, for instance the TELES VoIPBOX series, and, in combination with an enterprise PBX, through C5 with complete set of features and relayed management, followed by the interconnectivity of server with the broad spectrum of codecs for IP and TDM. C5 Software is expected to have highest level of functionalities in synergy with other TELES products. Company said that all major ISDN supplementary services can now be mutually exchanged through the TELES end-to-end solution which enables a full migration of legacy ISDN equipment and a smooth incorporation of this software/ hardware into merely all IP-based networks. “Release C5 is about SIP trunking and what we call the Single Pipe to the Cloud, where all voice and data communications are ultimately transported via the data cloud. Enterprises can now gain access to the full range of communications services, whilst providers and carriers will now have the potential to exploit their customer base by up-selling to PBX customers, also placing these customers on the path towards full cloud-based unified communications,” said TELES Product Marketing Director, Thomas Haydn, in a statement. “We are in a unique position of being able to supply carriers and service providers with an end-to-end solution that avoids complex compatibility problems. This means that they will reap the benefits of a solution which is reliable and stable, and they will also profit from time-saving access gateway zero touch provisioning which will help to reduce operating costs dramatically,” Haydn continued.