-------------------------------------------------------------------------------- Fedora Update Notification FEDORA-2011-9732 2011-07-26 02:57:43 --------------------------------------------------------------------------------
Name : asterisk Product : Fedora 15 Version : 1.8.5.0 Release : 1.fc15.2 URL : http://www.asterisk.org/ Summary : The Open Source PBX Description : Asterisk is a complete PBX in software. It runs on Linux and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. -------------------------------------------------------------------------------- Update Information: The Asterisk Development Team announces the release of Asterisk 1.8.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.5.0 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: * Fix Deadlock with attended transfer of SIP call (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, cmaj) * Fixes thread blocking issue in the sip TCP/TLS implementation. (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, rossbeer, kowalma, Freddi_Fonet) * Be more tolerant of what URI we accept for call completion PUBLISH requests. (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson) * Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call. (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose) * This patch fixes a bug with MeetMe behavior where the 'P' option for always prompting for a pin is ignored for the first caller. (Closes issue #18070. Reported by mav3rick. Patched by bbryant) * Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett) * Resolve issue where leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read. (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard Mudgett) * Resolve issue where wait for leader with Music On Hold allows crosstalk between participants. Parenthesis in the wrong position. Regression from issue #14365 when expanding conference flags to use 64 bits. (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5.0 -------------------------------------------------------------------------------- ChangeLog: * Thu Jul 21 2011 Petr Sabata <[email protected]> - 1.8.5.0-1.2 - Perl mass rebuild * Wed Jul 20 2011 Petr Sabata <[email protected]> - 1.8.5.0-1.1 - Perl mass rebuild * Mon Jul 11 2011 Jeffrey C. Ollie <[email protected]> - 1.8.5.0-1 - The Asterisk Development Team announces the release of Asterisk 1.8.5.0. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.5.0 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: - - * Fix Deadlock with attended transfer of SIP call - (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, - cmaj) - - * Fixes thread blocking issue in the sip TCP/TLS implementation. - (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, - rossbeer, kowalma, Freddi_Fonet) - - * Be more tolerant of what URI we accept for call completion PUBLISH requests. - (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson) - - * Fix a nasty chanspy bug which was causing a channel leak every time a spied on - channel made a call. - (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose) - - * This patch fixes a bug with MeetMe behavior where the 'P' option for always - prompting for a pin is ignored for the first caller. - (Closes issue #18070. Reported by mav3rick. Patched by bbryant) - - * Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If - the call that the dialplan started an AGI script for is hungup while the AGI - script is in the middle of a command then the AGI script is not notified of - the hangup. - (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett) - - * Resolve issue where leaving a voicemail, the MWI message is never sent. The - same thing happens when checking a voicemail and marking it as read. - (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard - Mudgett) - - * Resolve issue where wait for leader with Music On Hold allows crosstalk - between participants. Parenthesis in the wrong position. Regression from issue - #14365 when expanding conference flags to use 64 bits. - (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5.0 * Thu Jul 7 2011 Jeffrey C. Ollie <[email protected]> - 1.8.5-0.2 - Rebuild for net-snmp 5.7 * Fri Jul 1 2011 Jeffrey C. Ollie <[email protected]> - 1.8.5-0.1.rc1 - Fix systemd dependencies in EL6 and F15 * Thu Jun 30 2011 Jeffrey C. Ollie <[email protected]> - 1.8.5-0.1.rc1 - The Asterisk Development Team has announced the first release candidate of - Asterisk 1.8.5. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.5-rc1 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 candidate: - - * Fix Deadlock with attended transfer of SIP call - (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, - cmaj) - - * Fixes thread blocking issue in the sip TCP/TLS implementation. - (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, - rossbeer, kowalma, Freddi_Fonet) - - * Be more tolerant of what URI we accept for call completion PUBLISH requests. - (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson) - - * Fix a nasty chanspy bug which was causing a channel leak every time a spied on - channel made a call. - (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose) - - * This patch fixes a bug with MeetMe behavior where the 'P' option for always - prompting for a pin is ignored for the first caller. - (Closes issue #18070. Reported by mav3rick. Patched by bbryant) - - * Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If - the call that the dialplan started an AGI script for is hungup while the AGI - script is in the middle of a command then the AGI script is not notified of - the hangup. - (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett) - - * Resolve issue where leaving a voicemail, the MWI message is never sent. The - same thing happens when checking a voicemail and marking it as read. - (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard - Mudgett) - - * Resolve issue where wait for leader with Music On Hold allows crosstalk - between participants. Parenthesis in the wrong position. Regression from issue - #14365 when expanding conference flags to use 64 bits. - (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett) - - * Fix timerfd locking issue. - (Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5-rc1 * Thu Jun 30 2011 Jeffrey C. Ollie <[email protected]> - 1.8.4.4-2 - Fedora Directory Server -> 389 Directory Server * Wed Jun 29 2011 Jeffrey C. Ollie <[email protected]> - 1.8.4.4-1 - 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/ChangeLog-1.4.41.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.4 - - Security advisory AST-2011-011 is available at: - - http://downloads.asterisk.org/pub/security/AST-2011-011.pdf * Mon Jun 27 2011 Jeffrey C. Ollie <[email protected]> - 1.8.4.3-3 - Don't forget stereorize * Mon Jun 27 2011 Jeffrey C. Ollie <[email protected]> - 1.8.4.3-2 - Move /var/run/asterisk to /run/asterisk - Add comments to systemd service file on how to mimic safe_asterisk functionality - Build more of the optional binaries - Install the tmpfiles.d configuration on Fedora 15 * Fri Jun 24 2011 Jeffrey C. Ollie <[email protected]> - 1.8.4.3-1 - 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/ChangeLog-1.4.41.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-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 * Tue Jun 21 2011 Jeffrey C. Ollie <[email protected]> - 1.8.4.2-2 - Convert to systemd * Fri Jun 17 2011 Marcela Mašláňová <[email protected]> - 1.8.4.2-1.2 - Perl mass rebuild * Fri Jun 10 2011 Marcela Mašláňová <[email protected]> - 1.8.4.2-1.1 - Perl 5.14 mass rebuild * Fri Jun 3 2011 Jeffrey C. Ollie <[email protected]> - 1.8.4.2-1: - - 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/ChangeLog-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 -------------------------------------------------------------------------------- This update can be installed with the "yum" update program. Use su -c 'yum update asterisk' at the command line. For more information, refer to "Managing Software with yum", available at http://docs.fedoraproject.org/yum/. All packages are signed with the Fedora Project GPG key. More details on the GPG keys used by the Fedora Project can be found at https://fedoraproject.org/keys -------------------------------------------------------------------------------- _______________________________________________ package-announce mailing list [email protected] https://admin.fedoraproject.org/mailman/listinfo/package-announce
