Archive for category sip

Asterisk Security Advisory – AST-2010-001: T.38 Remote Crash Vulnerability

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

Asterisk Project Security AdvisoryAST-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

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Asterisk 1.6.0.22, Asterisk 1.6.1.14, Asterisk 1.6.2.2 Released

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

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:

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!

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Asterisk 1.6.0.21 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

The Asterisk Development Team has announced the release of Asterisk 1.6.0.21.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.6.0.21 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.
  • If EXEC only gets a single argument, don’t crash when the second is used.
    (Closes issue #16504. Reported by bklang. Patched by tilghman.)
  • Avoid a crash with large numbers of MeetMe conferences.
    (Closes issue #16509. Reported by Kashif Raza. Tested, Patched by seanbright.)
  • Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces (for Solaris 10).
    (Patched by seanbright.)
  • Allow “REMAINDER” to function properly in expressions.
    (Closes issue #16427. Reported, Patched by wdoekes.)
  • Shut down the SIP session timers more gracefully, in order to prevent a possible crash.
    (Reported, Tested by corruptor. Patched by tilghman.)
  • Fix channel name comparison for Bridge() application.
    (Closes issue #16528. Reported, Patched by telecos82.)

A summary of changes in this release can be found in the release summary:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.21-summary.txt

For a full list of changes in this releases, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.21

Thank you for your continued support of Asterisk!
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Asterisk 1.4.29 Now Available

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

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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Asterisk Security Advisory – AST-2009-005: Remote Crash Vulnerability in SIP channel driver

Asterisk The Open Source PBX & Telephony Platform

Asterisk The Open Source PBX & Telephony Platform

On certain implementations of libc, the scanf family of functions uses an unbounded amount of stack memory to repeatedly allocate string buffers prior to conversion to the target type. Coupled with Asterisk’s allocation of thread stack sizes that are smaller than the default, an attacker may exhaust stack memory in the SIP stack network thread by presenting excessively long numeric strings in various fields.
Note that while this potential vulnerability has existed in Asterisk for a very long time, it is only potentially exploitable in 1.6.1 and above, since those versions are the first that have allowed SIP packets to exceed 1500 bytes total, which does not permit strings that are large enough to crash Asterisk. (The number strings presented to us by the security researcher were approximately 32,000 bytes long.)

Additionally note that while this can crash Asterisk, execution of arbitrary code is not possible with this vector.

Upgrade Asterisk to one of the releases listed below.

Asterisk Project Security AdvisoryAST-2009-005

Product

Asterisk

Summary

Remote Crash Vulnerability in SIP channel driver

Nature of Advisory

Denial of Service

Susceptibility

Remote Unauthenticated Sessions

Severity

Critical in 1.6.1; minor in lesser versions

Exploits Known

No

Reported On

July 28, 2009

Reported By

Nick Baggott < nbaggott AT mudynamics DOT com >

Posted On

August 10, 2009

Last Updated On

August 10, 2009

Advisory Contact

Tilghman Lesher < tlesher AT digium DOT com >

CVE Name

CVE-2009-2726

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Fax For Asterisk

T.38 fax for Asterisk

T.38 fax for Asterisk

Digium’s Fax For Asterisk is a commercial facsimile (Fax) termination and origination solution designed to enhance the capabilities of Open Source and commercial Asterisk as well as Switchvox. Fax For Asterisk bundles a suite of user-friendly Asterisk applications and a licensed version of the industry’s leading fax modem software from Commetrex. Fax For Asterisk provides low speed (14400bps) PSTN faxing via DAHDI-compatible telephony boards as well as VoIP faxing to T.38-compatible SIP endpoints and service providers. Licensed on a per-channel basis, Digium’s Fax For Asterisk provides a complete, cost-effective, commercial fax solution for Asterisk users.

Fax For Asterisk provides two components: res_fax and res_fax_digium. Res_fax is an Asterisk resource module that adds fax termination and origination functionality in Asterisk. It provides Asterisk dialplan functions and dialplan applications to enable the user to build highly-customizable fax solutions. Res_fax_digium provides core fax processing functionality in the form of several supported fax modems — V.21, V.27ter, V.29, and V.17 — to achieve speeds up to 14400bps.

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Asterisk says Hello to Fax

If you ask Google about faxing for Asterisk, with the search keywords asterisk and fax, and you ask Google to omit similar entries, you’ll end up with 52 pages of results.

If you ask Google how many times fax has been mentioned on an Asterisk mailing list, by setting the site parameter to lists.digium.com, then Google tells you there are 1120 utterances.

Yesterday, if you asked Digium for help in faxing documents through Asterisk, we’d have apologized and said that we didn’t offer a fax solution for Asterisk.

That was yesterday.

Today, Digium is pleased to announce Fax For Asterisk.

Fax For Asterisk is a commercial facsimile (Fax) termination and origination solution designed to enhance the capabilities of Open Source and commercial Asterisk as well as Switchvox. Fax For Asterisk bundles a suite of user-friendly Asterisk applications and a licensed version of the industry’s leading fax modem software from Commetrex. Fax For Asterisk provides low speed (14400bps) PSTN faxing via DAHDI-compatible telephony boards as well as VoIP faxing to T.38-compatible SIP endpoints and service providers. Licensed on a per-channel basis, Digium’s Fax For Asterisk provides a complete, cost-effective, commercial fax solution for Asterisk users.

Wait, I’ve forgotten something…okay, not really. There’s also Free Fax For Asterisk. Free Fax For Asterisk provides a single-channel only, per Asterisk, version of Fax For Asterisk, for free. Want to use Free Fax For Asterisk now? Visit the Digium webstore and get a license, free of charge.

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