Archive for category t.38
Asterisk 10.0.0 Is Released
The Asterisk Development Team is proud to announce the release of Asterisk 10.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/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
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…
The release of Asterisk 10 would not have been possible without the support and contributions of the community.
You can find an overview of the work involved with the 10.0.0 release in the summary:
http://svn.asterisk.org/svn/asterisk/tags/10.0.0/asterisk-10.0.0-summary…
A short list of available 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.
http://svn.asterisk.org/svn/asterisk/branches/10/CHANGES
Also, when upgrading a system between major versions, it is imperative that you read and understand the contents of the UPGRADE.txt file, which is located at:
http://svn.asterisk.org/svn/asterisk/branches/10/UPGRADE.txt
Thank you for your continued support of Asterisk!
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!
Asterisk 1.4.42 Now Available (Final Maintenance Release)
The Asterisk Development Team has announced the final maintenance release of Asterisk, version 1.4.42. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
Please note that Asterisk 1.4.42 is the final maintenance release from the 1.4 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 1.4.42 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:
- Resolve regression with ring groups in the Dial() application
(Closes issue ASTERISK-17874. Reported by mspuhler. Patched by elguero) - Resolve deadlock when using tab completion on the ‘meetme kick’ CLI command when an invalid (non-existent) conference room is specified.
(Closes issue ASTERISK-17771. Reported, patched by zvision) - Resolve issue where voice frames could be dropped when checking for T.38 during early media.
(Closes issue ASTERISK-17705. Reported, patched by oej) - Resolve issue where DYNAMIC_FEATURES would not activate after a recent DTMF fix.
(Closes issue ASTERISK-17914. Reported by vrban. Patched by twilson)
Additionally security announcements 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!
Asterisk 1.4.39.2, Asterisk 1.6.1.22, Asterisk 1.6.2.16.2 and Asterisk 1.8.2.4 Now Available
Posted by admin in asterisk, Asterisk Security Releases, Security Advisories, t.38 on February 22, 2011
The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.
These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases
The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an issue that when decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems doing T.38 pass through or termination are vulnerable. The issue and resolution are described in the AST-2011-002 security advisory.
For more information about the details of this vulnerability, please read the security advisory AST-2011-002, 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-…
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-…
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-…
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-…
Security advisory AST-2011-002 is available at:
Thank you for your continued support of Asterisk!
Asterisk Security Advisory – AST-2011-002: Multiple array overflow and crash vulnerabilities in UDPTL code
Posted by admin in asterisk, Asterisk Security Advisories, Security Advisories, t.38 on February 22, 2011
| Product | Asterisk |
| Summary | Multiple array overflow and crash vulnerabilities in UDPTL code |
| Nature of Advisory | Exploitable Stack and Heap Array Overflows |
| Susceptibility | Remote Unauthenticated Sessions |
| Severity | Critical |
| Exploits Known | No |
| Reported On | January 27, 2011 |
| Reported By | Matthew Nicholson |
| Posted On | February 21, 2011 |
| Last Updated On | February 22, 2011 |
| Advisory Contact | Matthew Nicholson <mnicholson@digium.com> |
| CVE Name |
| Description | When decoding UDPTL packets, multiple stack and heap based arrays can be made to overflow by specially crafted packets. Systems configured for T.38 pass through or termination are vulnerable. |
Asterisk 1.4.35 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.35.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.35 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following are a few of the issues resolved by community developers:
- Ensure channel placed in meetme in ringing state is properly hung up.
(Closes issue #15871. Reported, patched by Ivan) - If all members are paused, the wrong status is indicated.
(Closes issue #17576. Reported, patched by ramonpeek) - Fix logging message for stale nonce.
(Closes issue #17582. Reported, patched by kenner) - Interpret device state AST_DEVICE_UNKNOWN as extension state AST_EXTENSION_NOT_INUSE.
(Closes issue #16035. Reported by francesco_r. Patched by viniciusfontes) - Resolve T.38 negotiation regression.
(Closes issue #16705. Reported by mpiazzatnetbug. Patched by ebroad)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.35
Thank you for your continued support of Asterisk!
More Fax, More Features
Aloha,
In Digium’s continuing quest to deliver you, our users of Free Fax For Asterisk, and our customers of Fax For Asterisk, with the best possible solution, we are pleased to announce the availability of version 1.2 of…(drum roll without suspense) Fax For Asterisk.
How has this version improved? Why, with new features, of course, hence the title of this blog post.
New features include:
- DAHDI Buffer Policy Implementation -Currently requiring the trunk version of Asterisk, in addition to the 1.2 release of Fax For Asterisk, new dialplan functions to allow the setting of buffer policies to prevent fax failures on higher latency systems, e.g.:exten => 1234,Set(CHANNEL(buffers)=”12,half”)where “12″ represents a number of buffers (each buffer is 20ms), configurable between 4 and 32, and where “half” represents the policy implementation, configurable as “immediate,” “full,” or “half.”
To use, simply set your buffer policy in your dialplan before any send/receive fax operation across a DAHDI channel.
- SIP Fax Detection Options -At present, also requiring the trunk version of Asterisk in addition to the 1.2 release of Fax For Asterisk, new options are available related to T.38 session initiation. Older releases of Fax For Asterisk only detect T.38 fax upon the receipt of CNG. In practice, we’ve discovered that a number of T.38 providers send T.38 invites immediately, and never send CNG to initiate a T.38 session. Thus, the faxdetect option in sip.conf can now be set to:no – To disable all fax detectioncng – To trigger fax detection based on the receipt of a CNG tone
t38 – To trigger fax detection based on the receipt of a SIP T.38 invite, without CNG tone
yes – To trigger fax detection based on the receipt of either a CNG tone or a SIP T.38 invite.
These changes should improve our compatibility with the wild, wild west of T.38 implementations.
- New CLI Commands -New Asterisk command line interface commands are available to display the settings configured in res_fax and res_fax_digium, simply run:fax show settingsto see your current settings.
- Asterisk CLI Type Column -A “Type” column is now displayed when “fax show sessions” is run on the Asterisk CLI, informing the user whether the fax is of type “G.711″ or of type “T.38.”
- ECM Configuration per Provider / Peer & Configuration Moved -Error correction mode may now be configured on a per provider / peer basis. This proves useful in the case that a provider does not implemented T.38 ECM properly. Digium has observed that ECM must be disabled for T.38 faxing to work properly with Gafachi.Configuration of error correction mode has moved from res_fax_digium.com into the res_fax.conf configuration file. Note that the default setting is still to enable ECM.
- SendFax initiate T.38 re-invite -Digium observed that a number of providers or far-end systems did not send a T.38 re-invite and instead waited for the local system (Asterisk) to send it instead. The SendFax application now supports the “z” option to enable this feature. If the “z” option is set during a SendFax, then res_fax will initiate the T.38 re-invite if it is not received in 10 (ten) seconds from the far end. Digium observed that the “z” option must be used for T.38 faxing to work properly with Gafachi.
- Send / ReceiveFax G.711 Fallback mode -A new fallback option “f” has been added to the SendFax and ReceiveFax applications. In the event that T.38 negotiation fails, enabling this option will cause Asterisk to revert to audio fax mode. Digium has observed this is required for some providers, like BroadVox, who provide T.38 for inbound faxing, but accept only audio faxing for outbound.Please note that audio faxing over the Internet is very unreliable, and Digium cannot provide support for fax failures due to poor Internet connections.
- New Debugging utilities -In order to make debugging easier, we’ve added two new command line capture options, one for audio faxes and one for T.38 faxes.For audio capture, do “fax set g711cap on” in the Asterisk CLI and a stereo wav file will be created for each fax session. The resulting files will be saved in /var/log/asterisk/g711cap. To stop capture, do “fax set g711cap off.”For T.38 capture, do “fax set t38cap on” in the Asterisk CLI and a Wireshark compatible pcap file will be created for each fax session. The resulting files will be saved in /var/log/asterisk/t38cap. To stop capture, do “fax set t38cap off.”
Ready to upgrade? Run right over to the Fax For Asterisk Download Selector and grab the new release.
As always, we thank you for your support.
Cheers.
Asterisk Security Advisory – AST-2010-001: T.38 Remote Crash Vulnerability
Posted by admin in asterisk, Asterisk Security Advisories, Security Advisories, sip, t.38 on February 3, 2010
Asterisk Project Security Advisory - AST-2010-001
| Product | Asterisk |
| Summary | T.38 Remote Crash Vulnerability |
| Nature of Advisory | Denial of Service |
| Susceptibility | Remote unauthenticated sessions |
| Severity | Critical |
| Exploits Known | No |
| Reported On | 12/03/09 |
| Reported By | issues.asterisk.org users bklang and elsto |
| Posted On | 02/03/10 |
| Last Updated On | February 2, 2010 |
| Advisory Contact | David Vossel < dvossel AT digium DOT com > |
| CVE Name | CVE-2010-0441 |
Asterisk 1.6.0.22, Asterisk 1.6.1.14, Asterisk 1.6.2.2 Released
Posted by admin in asterisk, Asterisk Security Advisories, Releases, Security Advisories, sip, t.38 on February 3, 2010
The Asterisk Development Team has announced security releases for Asterisk as the following versions:
These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The releases of Asterisk 1.6.0.22, 1.6.1.14, and 1.6.2.2 include the fix described in security advisory AST-2010-001.
The issue is that an attacker attempting to negotiate T.38 over SIP can remotely crash Asterisk by modifying the FaxMaxDatagram field of the SDP to contain either a negative or exceptionally large value. The same crash will occur when the FaxMaxDatagram field is omitted from the SDP, as well.
For more information about the details of this vulnerability, please read the security advisory AST-2010-001, which was released at the same time as this announcement.
For a full list of changes in the current releases, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.22
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.14
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.2
Security advisory AST-2010-001 is available at:
http://asterisk.net.ru/en/2010/02/03/asterisk-security-advisory-ast-2010-001-t-38-remote-crash-vulnerability/
Thank you for your continued support of Asterisk!
Asterisk 1.4.29 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.29.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.29 resolved several issues reported by the community, and would have not been possible without your participation. Thank you!
- Fix to Monitor which previously assumed the file to write to did not contain pathing.
(Closes issue #16377, #16376. Reported by bcnit. Patched by dant. - Propertly set T.38 attributes and don’t return before T.38 ports are configured when T.38 is found but no audio stream is found.
(Closes issue #16318. Reported by bird_of_Luck. Tested by vrban, mihaill. Patched by vrban, mnicholson.) - Avoid crashes with large numbers of MeetMe conferences.
(Closes issue #16509. Reported by Kashif Raza. Tested, Patched by seanbright.) - Change in ‘sip show channels’ display format allowing more digits for CID.
(Closes issue #16459. Reported, Patched by Rzadzins. - Revise documentation on disposition values to the actual values used.
(Closes issue #16289. Reported by wdoekes.)
A summary of changes in this release can be found in the release summary:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.4.29-summary.txt
For a full list of changes in this releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.29
Thank you for your continued support of Asterisk!
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