The Asterisk Development Team reports :

If a remote user initiates a SIP call and the recipient picks up, the remote user can reply with a malformed Contact header that Asterisk will improperly handle and cause a crash due to a segmentation fault.

Remove systemd dependencies from EL6 and F15 The Asterisk Development Team has announced the release of Asterisk versions 1.4.41.2, 1.6.2.18.2, and 1.8.4.4, which are security releases.

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

The release of Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 resolves the following issue :

AST-2011-011: Asterisk may respond differently to SIP requests from an invalid SIP user than it does to a user configured on the system, even when the alwaysauthreject option is set in the configuration. This can leak information about what SIP users are valid on the Asterisk system.

For more information about the details of this vulnerability, please read the security advisory AST-2011-011, 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/releases/ChangeLo g-1.4.41.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLo g-1.6.2.18.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLo g-1.8.4.4

Security advisory AST-2011-011 is available at :

http://downloads.asterisk.org/pub/security/AST-2011-011.pdf The Asterisk Development Team has announced the release of Asterisk versions 1.4.41.1, 1.6.2.18.1, and 1.8.4.3, which are security releases.

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

The release of Asterisk 1.4.41.1, 1.6.2.18, and 1.8.4.3 resolves several issues as outlined below :

  - AST-2011-008: If a remote user sends a SIP packet     containing a null, Asterisk assumes available data     extends past the null to the end of the packet when the     buffer is actually truncated when copied. This causes     SIP header parsing to modify data past the end of the     buffer altering unrelated memory structures. This     vulnerability does not affect TCP/TLS connections. --     Resolved in 1.6.2.18.1 and 1.8.4.3

  - AST-2011-009: A remote user sending a SIP packet     containing a Contact header with a missing left angle     bracket (<) causes Asterisk to access a NULL pointer. --     Resolved in 1.8.4.3

  - AST-2011-010: A memory address was inadvertently     transmitted over the network via IAX2 via an option     control frame and the remote party would try to access     it. -- Resolved in 1.4.41.1, 1.6.2.18.1, and 1.8.4.3

The issues and resolutions are described in the AST-2011-008, AST-2011-009, and AST-2011-010 security advisories.

For more information about the details of these vulnerabilities, please read the security advisories AST-2011-008, AST-2011-009, and AST-2011-010, which were 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/releases/ChangeLo g-1.4.41.1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLo g-1.6.2.18.1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLo g-1.8.4.3

Security advisories AST-2011-008, AST-2011-009, and AST-2011-010 are available at :

http://downloads.asterisk.org/pub/security/AST-2011-008.pdf http://downloads.asterisk.org/pub/security/AST-2011-009.pdf http://downloads.asterisk.org/pub/security/AST-2011-010.pdf

The Asterisk Development Team has announced the release of Asterisk version 1.8.4.2, which is a security release for Asterisk 1.8.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk 1.8.4.2 resolves an issue with SIP URI parsing which can lead to a remotely exploitable crash :

Remote Crash Vulnerability in SIP channel driver (AST-2011-007)

The issue and resolution is described in the AST-2011-007 security advisory.

For more information about the details of this vulnerability, please read the security advisory AST-2011-007, 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/ChangeLo g-1.8.4.2

Security advisory AST-2011-007 is available at :

http://downloads.asterisk.org/pub/security/AST-2011-007.pdf

The Asterisk Development Team has announced the release of Asterisk 1.8.4.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.4.1 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you!

Below is a list of issues resolved in this release :

  - Fix our compliance with RFC 3261 section 18.2.2. (aka     Cisco phone fix) (Closes issue #18951. Reported by jmls.
    Patched by wdoekes)

  - Resolve a change in IPv6 header parsing due to the Cisco     phone fix issue. This issue was found and reported by     the Asterisk test suite. (Closes issue #18951. Patched     by mnicholson)

  - Resolve potential crash when using SIP TLS support.
    (Closes issue #19192. Reported by stknob. Patched by     Chainsaw. Tested by vois, Chainsaw)

  - Improve reliability when using SIP TLS. (Closes issue     #19182. Reported by st. Patched by mnicholson)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4.1

The Asterisk Development Team has announced the release of Asterisk 1.8.4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.4 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you!

Below is a sample of the issues resolved in this release :

  - Use SSLv23_client_method instead of old SSLv2 only.
    (Closes issue #19095, #19138. Reported, patched by     tzafrir. Tested by russell and chazzam.

  - Resolve crash in ast_mutex_init() (Patched by twilson)

  - Resolution of several DTMF based attended transfer     issues. (Closes issue #17999, #17096, #18395, #17273.
    Reported by iskatel, gelo, shihchuan, grecco. Patched by     rmudgett)

    NOTE: Be sure to read the ChangeLog for more information     about these changes.

  - Resolve deadlocks related to device states in chan_sip     (Closes issue #18310. Reported, patched by one47.
    Patched by jpeeler)

  - Resolve an issue with the Asterisk manager interface     leaking memory when disabled. (Reported internally by     kmorgan. Patched by russellb)

  - Support greetingsfolder as documented in     voicemail.conf.sample. (Closes issue #17870. Reported by     edhorton. Patched by seanbright)

  - Fix channel redirect out of MeetMe() and other issues     with channel softhangup (Closes issue #18585. Reported     by oej. Tested by oej, wedhorn, russellb. Patched by     russellb)

  - Fix voicemail sequencing for file based storage. (Closes     issue #18498, #18486. Reported by JJCinAZ, bluefox.
    Patched by jpeeler)

  - Set hangup cause in local_hangup so the proper return     code of 486 instead of 503 when using Local channels     when the far sides returns a busy. Also affects CCSS in     Asterisk 1.8+. (Patched by twilson)

  - Fix issues with verbose messages not being output to the     console. (Closes issue #18580. Reported by pabelanger.
    Patched by qwell)

  - Fix Deadlock with attended transfer of SIP call (Closes     issue #18837. Reported, patched by alecdavis. Tested by     alecdavid, Irontec, ZX81, cmaj)

Includes changes per AST-2011-005 and AST-2011-006 For a full list of changes in this release candidate, please see the ChangeLog :

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4

Information about the security releases are available at :

http://downloads.asterisk.org/pub/security/AST-2011-005.pdf http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

Note that Tenable Network Security has extracted the preceding description block directly from the Fedora security advisory. Tenable has attempted to automatically clean and format it as much as possible without introducing additional issues.

According to the version in its SIP banner, the version of Asterisk running on the remote host is 1.8.x prior to 1.8.4.2. It, therefore, could be affected by a denial of service vulnerability.

If a remote attacker initiates a SIP call and the recipient picks up, the remote user can reply with a malformed Contact header that will cause Asterisk to crash.

The Asterisk Development Team has announced the release of Asterisk version 1.8.4.2, which is a security release for Asterisk 1.8.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk 1.8.4.2 resolves an issue with SIP URI parsing which can lead to a remotely exploitable crash :

Remote Crash Vulnerability in SIP channel driver (AST-2011-007)

The issue and resolution is described in the AST-2011-007 security advisory.

For more information about the details of this vulnerability, please read the security advisory AST-2011-007, 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/ChangeLo g-1.8.4.2

Security advisory AST-2011-007 is available at :

http://downloads.asterisk.org/pub/security/AST-2011-007.pdf

The Asterisk Development Team has announced the release of Asterisk 1.8.4.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.4.1 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you!

Below is a list of issues resolved in this release :

  - Fix our compliance with RFC 3261 section 18.2.2. (aka     Cisco phone fix) (Closes issue #18951. Reported by jmls.
    Patched by wdoekes)

  - Resolve a change in IPv6 header parsing due to the Cisco     phone fix issue. This issue was found and reported by     the Asterisk test suite. (Closes issue #18951. Patched     by mnicholson)

  - Resolve potential crash when using SIP TLS support.
    (Closes issue #19192. Reported by stknob. Patched by     Chainsaw. Tested by vois, Chainsaw)

  - Improve reliability when using SIP TLS. (Closes issue     #19182. Reported by st. Patched by mnicholson)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4.1

The Asterisk Development Team has announced the release of Asterisk 1.8.4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.4 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you!

Below is a sample of the issues resolved in this release :

  - Use SSLv23_client_method instead of old SSLv2 only.
    (Closes issue #19095, #19138. Reported, patched by     tzafrir. Tested by russell and chazzam.

  - Resolve crash in ast_mutex_init() (Patched by twilson)

  - Resolution of several DTMF based attended transfer     issues. (Closes issue #17999, #17096, #18395, #17273.
    Reported by iskatel, gelo, shihchuan, grecco. Patched by     rmudgett)

    NOTE: Be sure to read the ChangeLog for more information     about these changes.

  - Resolve deadlocks related to device states in chan_sip     (Closes issue #18310. Reported, patched by one47.
    Patched by jpeeler)

  - Resolve an issue with the Asterisk manager interface     leaking memory when disabled. (Reported internally by     kmorgan. Patched by russellb)

  - Support greetingsfolder as documented in     voicemail.conf.sample. (Closes issue #17870. Reported by     edhorton. Patched by seanbright)

  - Fix channel redirect out of MeetMe() and other issues     with channel softhangup (Closes issue #18585. Reported     by oej. Tested by oej, wedhorn, russellb. Patched by     russellb)

  - Fix voicemail sequencing for file based storage. (Closes     issue #18498, #18486. Reported by JJCinAZ, bluefox.
    Patched by jpeeler)

  - Set hangup cause in local_hangup so the proper return     code of 486 instead of 503 when using Local channels     when the far sides returns a busy. Also affects CCSS in     Asterisk 1.8+. (Patched by twilson)

  - Fix issues with verbose messages not being output to the     console. (Closes issue #18580. Reported by pabelanger.
    Patched by qwell)

  - Fix Deadlock with attended transfer of SIP call (Closes     issue #18837. Reported, patched by alecdavis. Tested by     alecdavid, Irontec, ZX81, cmaj)

Includes changes per AST-2011-005 and AST-2011-006 For a full list of changes in this release candidate, please see the ChangeLog :

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4

Information about the security releases are available at :

http://downloads.asterisk.org/pub/security/AST-2011-005.pdf http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

Note that Tenable Network Security has extracted the preceding description block directly from the Fedora security advisory. Tenable has attempted to automatically clean and format it as much as possible without introducing additional issues.

The version of Asterisk running on the remote host is potentially affected by a denial of service vulnerability.  If a remote attacker initiates a SIP call and the recipient picks up, the remote user can reply with a malformed Contact header that will cause Asterisk to crash.

ID	Name	Product	Family	Published	Updated	Severity
54951	FreeBSD : asterisk -- Remote crash vulnerability (34ce5817-8d56-11e0-b5a2-6c626dd55a41)	Nessus	FreeBSD Local Security Checks	6/3/2011	1/6/2021	medium
55582	Fedora 15 : asterisk-1.8.4.4-2.fc15 (2011-8983)	Nessus	Fedora Local Security Checks	7/13/2011	1/11/2021	medium
54971	Asterisk SIP Channel Driver Denial of Service (AST-2011-007)	Nessus	Denial of Service	6/5/2011	4/11/2022	low
55425	Fedora 15 : asterisk-1.8.4.2-1.fc15.1 (2011-8319)	Nessus	Fedora Local Security Checks	6/27/2011	1/11/2021	medium
5939	Asterisk SIP Channel Driver Denial of Service (AST-2011-007)	Nessus Network Monitor	Generic	6/3/2011	3/6/2019	medium

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