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Update to Asterisk 1.8.5.0: this is a general bug fix release 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 * Fixes thread blocking issue in the sip TCP/TLS implementation. * Be more tolerant of what URI we accept for call completion PUBLISH requests. * Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call. * This patch fixes a bug with MeetMe behavior where the 'P' option for always prompting for a pin is ignored for the first caller. * 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. * 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. * 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. 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 Thank you for your continued support of Asterisk!
2011-07-16 23:35:11 +02:00
$NetBSD: distinfo,v 1.12 2011/07/16 21:35:11 jnemeth Exp $
Import Asterisk 1.8.1: Asterisk is a complete PBX in software. It 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. Asterisk 1.8 is a long term support version (i.e. it will be supported for four years with an additional year of security only fixes). See: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions What's new: Asterisk 1.8 is the next major release series of Asterisk. The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release. You can find a summary of the work involved with the 1.8.0 release in the sumary: http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt 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 ----- The Asterisk Development Team has announced the release of Asterisk 1.8.1. The release of Asterisk 1.8.1 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 issue when using directmedia. Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11) * Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers. (Patched by rmudgett) * Fix playback failure when using IAX with the timerfd module. (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data. (Closes issue #18161. Reported by wdoekes. Patched by tilghman) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
2010-12-15 04:22:43 +01:00
Update to Asterisk 1.8.5.0: this is a general bug fix release 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 * Fixes thread blocking issue in the sip TCP/TLS implementation. * Be more tolerant of what URI we accept for call completion PUBLISH requests. * Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call. * This patch fixes a bug with MeetMe behavior where the 'P' option for always prompting for a pin is ignored for the first caller. * 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. * 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. * 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. 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 Thank you for your continued support of Asterisk!
2011-07-16 23:35:11 +02:00
SHA1 (asterisk-1.8.5.0/asterisk-1.8.5.0.tar.gz) = 9e29581deea773c2537f5c01a43823211688412a
RMD160 (asterisk-1.8.5.0/asterisk-1.8.5.0.tar.gz) = c841993f914bd150696b9163a1cc1d1828f45e4a
Size (asterisk-1.8.5.0/asterisk-1.8.5.0.tar.gz) = 27417584 bytes
SHA1 (asterisk-1.8.5.0/asterisk-extra-sounds-en-gsm-1.4.11.tar.gz) = 8692fa61423b4769dc8bfa78faf9ed5ef7a259b9
RMD160 (asterisk-1.8.5.0/asterisk-extra-sounds-en-gsm-1.4.11.tar.gz) = 68170c769d739d6b5b35b00f999ad6bbf876f9f6
Size (asterisk-1.8.5.0/asterisk-extra-sounds-en-gsm-1.4.11.tar.gz) = 3349898 bytes
SHA1 (asterisk-1.8.5.0/extract-cfile.awk) = c4f08eee1ab83c041bde1ab91672a4a3c43c28b8
RMD160 (asterisk-1.8.5.0/extract-cfile.awk) = cd59f8e5807732023d5aec95187e2d5572f400a4
Size (asterisk-1.8.5.0/extract-cfile.awk) = 667 bytes
SHA1 (asterisk-1.8.5.0/rfc3951.txt) = 1a6c769be750fb02456d60db2470909254496017
RMD160 (asterisk-1.8.5.0/rfc3951.txt) = 15f7ec61653ec9953172f8f2150e7d8f6f620926
Size (asterisk-1.8.5.0/rfc3951.txt) = 373442 bytes
SHA1 (patch-aa) = bdaacb8b7b93886399b53cdda8a4ac7f827f50d5
Import Asterisk 1.8.1: Asterisk is a complete PBX in software. It 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. Asterisk 1.8 is a long term support version (i.e. it will be supported for four years with an additional year of security only fixes). See: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions What's new: Asterisk 1.8 is the next major release series of Asterisk. The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release. You can find a summary of the work involved with the 1.8.0 release in the sumary: http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt 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 ----- The Asterisk Development Team has announced the release of Asterisk 1.8.1. The release of Asterisk 1.8.1 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 issue when using directmedia. Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11) * Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers. (Patched by rmudgett) * Fix playback failure when using IAX with the timerfd module. (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data. (Closes issue #18161. Reported by wdoekes. Patched by tilghman) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
2010-12-15 04:22:43 +01:00
SHA1 (patch-af) = ebad62fcb31b600d30235cc5e93284c93b2c8af9
SHA1 (patch-ag) = c71c61350cefbbe53eefa99245ca7712753f22d5
SHA1 (patch-ai) = e92edab5c1ff323478f41d0b0783102ed527fe39
SHA1 (patch-ak) = adee75b7716a8794de1b8cb054af7a5a8f0e5ffd
SHA1 (patch-al) = b2a1134786d7c3b118ee8c47892f91dd2a4c783a
SHA1 (patch-am) = 5f9cbf47ec1cb66758492a5ed1bf843006eae9b7
SHA1 (patch-an) = 93a5df66fd6459fb76e9191dc3bf37b9ee5483b5
SHA1 (patch-ao) = 0663a698469550b22bb97ee1b18980bc2bc67495
Added a comment that the issue these patches fix (mainly adding support for NetBSD style atomic ops) has been reported upstream. No change to binary package, so no REVISION bump.
2011-01-29 23:50:32 +01:00
SHA1 (patch-ap) = ed22f6483191f429389c0d3198d30c63b96d4df6
Upgrade to 1.8.4.2. This fixes several security issues including: AST-2011-002, AST-2011-003, AST-2011-004, AST-2011-005, AST-2011-006, and AST-2011-007. pkgsrc changes: - add patch for autosupport script; == -> = - patch configure to not unconditionally set PBX_LAUNCHD=1 - this allows res_timing_kqueue.so to build This last change brings a timing source to NetBSD which allows IAX trunking and allows the bridging modules to work, a rather major piece that was missing. Note that I haven't extensively tested it. But, have at it... =========================================================================== 1.8.4.2: The Asterisk Development Team has announced the release of Asterisk version 1.8.4.2, which is a security release for Asterisk 1.8. 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 =========================================================================== 1.8.4.1: The Asterisk Development Team has announced the release of Asterisk 1.8.4.1. 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) * 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. * Resolve potential crash when using SIP TLS support. * Improve reliability when using SIP TLS. 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 =========================================================================== 1.8.4: The Asterisk Development Team has announced the release of Asterisk 1.8.4. 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. * Resolve crash in ast_mutex_init() * Resolution of several DTMF based attended transfer issues. NOTE: Be sure to read the ChangeLog for more information about these changes. * Resolve deadlocks related to device states in chan_sip * Resolve an issue with the Asterisk manager interface leaking memory when disabled. * Support greetingsfolder as documented in voicemail.conf.sample. * Fix channel redirect out of MeetMe() and other issues with channel softhangup * Fix voicemail sequencing for file based storage. * 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+. * Fix issues with verbose messages not being output to the console. * Fix Deadlock with attended transfer of SIP call 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 =========================================================================== 1.8.3.3: 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 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3. The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 resolve two issues: * File Descriptor Resource Exhaustion (AST-2011-005) * Asterisk Manager User Shell Access (AST-2011-006) The issues and resolutions are described in the AST-2011-005 and AST-2011-006 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-005 and AST-2011-006, 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-1.8.3.3 Security advisory AST-2011-005 and AST-2011-006 are available at: http://downloads.asterisk.org/pub/security/AST-2011-005.pdf http://downloads.asterisk.org/pub/security/AST-2011-006.pdf =========================================================================== 1.8.3.2: he Asterisk Development Team has announced security releases for Asterisk branches 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.6.1.24, 1.6.2.17.2, and 1.8.3.2. ** This is a re-release of Asterisk 1.6.1.23, 1.6.2.17.1 and 1.8.3.1 which contained a bug which caused duplicate manager entries (issue #18987). The releases of Asterisk 1.6.1.24, 1.6.2.17.2, and 1.8.3.2 resolve two issues: * Resource exhaustion in Asterisk Manager Interface (AST-2011-003) * Remote crash vulnerability in TCP/TLS server (AST-2011-004) The issues and resolutions are described in the AST-2011-003 and AST-2011-004 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-003 and AST-2011-004, 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-1.8.3.2 Security advisory AST-2011-003 and AST-2011-004 are available at: http://downloads.asterisk.org/pub/security/AST-2011-003.pdf http://downloads.asterisk.org/pub/security/AST-2011-004.pdf =========================================================================== 1.8.3.1: The Asterisk Development Team has announced security releases for Asterisk branches 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.6.1.23, 1.6.2.17.1, and 1.8.3.1. The releases of Asterisk 1.6.1.23, 1.6.2.17.1, and 1.8.3.1 resolve two issues: * Resource exhaustion in Asterisk Manager Interface (AST-2011-003) * Remote crash vulnerability in TCP/TLS server (AST-2011-004) The issues and resolutions are described in the AST-2011-003 and AST-2011-004 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-003 and AST-2011-004, 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-1.8.3.1 Security advisory AST-2011-003 and AST-2011-004 are available at: http://downloads.asterisk.org/pub/security/AST-2011-003.pdf http://downloads.asterisk.org/pub/security/AST-2011-004.pdf =========================================================================== 1.8.3: The Asterisk Development Team has announced the release of Asterisk 1.8.3. The release of Asterisk 1.8.3 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 duplicated data in the AstDB when using DIALGROUP() * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing. * When using cdr_pgsql the billsec field was not populated correctly on unanswered calls. * Resolve memory leak in iCalendar and Exchange calendaring modules. * This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems. * Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi. * Resolve a memory leak when the Asterisk Manager Interface is disabled. * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. * Fix regression that changed behavior of queues when ringing a queue member. * Resolve deadlock involving REFER. Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at http://downloads.asterisk.org/pub/security/AST-2011-002.pdf For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3 =========================================================================== 1.8.2.4: 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. 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 configured for 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-1.8.2.4 Security advisory AST-2011-002 is available at: http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
2011-06-09 11:17:27 +02:00
SHA1 (patch-aq) = b5b448df41c3751dda6340006811cb35dd304d34
Added a comment that the issue these patches fix (mainly adding support for NetBSD style atomic ops) has been reported upstream. No change to binary package, so no REVISION bump.
2011-01-29 23:50:32 +01:00
SHA1 (patch-ar) = da8e614e68e476ce32c66fed5ee9dcb8c5f9a060
SHA1 (patch-as) = b2e1aadf49f20506243ab40796f15aab12d95bad
SHA1 (patch-at) = df318d7b492121ff6f766b0e6ea73415293e96f0
Upgrade to 1.8.4.2. This fixes several security issues including: AST-2011-002, AST-2011-003, AST-2011-004, AST-2011-005, AST-2011-006, and AST-2011-007. pkgsrc changes: - add patch for autosupport script; == -> = - patch configure to not unconditionally set PBX_LAUNCHD=1 - this allows res_timing_kqueue.so to build This last change brings a timing source to NetBSD which allows IAX trunking and allows the bridging modules to work, a rather major piece that was missing. Note that I haven't extensively tested it. But, have at it... =========================================================================== 1.8.4.2: The Asterisk Development Team has announced the release of Asterisk version 1.8.4.2, which is a security release for Asterisk 1.8. 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 =========================================================================== 1.8.4.1: The Asterisk Development Team has announced the release of Asterisk 1.8.4.1. 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) * 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. * Resolve potential crash when using SIP TLS support. * Improve reliability when using SIP TLS. 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 =========================================================================== 1.8.4: The Asterisk Development Team has announced the release of Asterisk 1.8.4. 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. * Resolve crash in ast_mutex_init() * Resolution of several DTMF based attended transfer issues. NOTE: Be sure to read the ChangeLog for more information about these changes. * Resolve deadlocks related to device states in chan_sip * Resolve an issue with the Asterisk manager interface leaking memory when disabled. * Support greetingsfolder as documented in voicemail.conf.sample. * Fix channel redirect out of MeetMe() and other issues with channel softhangup * Fix voicemail sequencing for file based storage. * 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+. * Fix issues with verbose messages not being output to the console. * Fix Deadlock with attended transfer of SIP call 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 =========================================================================== 1.8.3.3: 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 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3. The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 resolve two issues: * File Descriptor Resource Exhaustion (AST-2011-005) * Asterisk Manager User Shell Access (AST-2011-006) The issues and resolutions are described in the AST-2011-005 and AST-2011-006 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-005 and AST-2011-006, 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-1.8.3.3 Security advisory AST-2011-005 and AST-2011-006 are available at: http://downloads.asterisk.org/pub/security/AST-2011-005.pdf http://downloads.asterisk.org/pub/security/AST-2011-006.pdf =========================================================================== 1.8.3.2: he Asterisk Development Team has announced security releases for Asterisk branches 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.6.1.24, 1.6.2.17.2, and 1.8.3.2. ** This is a re-release of Asterisk 1.6.1.23, 1.6.2.17.1 and 1.8.3.1 which contained a bug which caused duplicate manager entries (issue #18987). The releases of Asterisk 1.6.1.24, 1.6.2.17.2, and 1.8.3.2 resolve two issues: * Resource exhaustion in Asterisk Manager Interface (AST-2011-003) * Remote crash vulnerability in TCP/TLS server (AST-2011-004) The issues and resolutions are described in the AST-2011-003 and AST-2011-004 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-003 and AST-2011-004, 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-1.8.3.2 Security advisory AST-2011-003 and AST-2011-004 are available at: http://downloads.asterisk.org/pub/security/AST-2011-003.pdf http://downloads.asterisk.org/pub/security/AST-2011-004.pdf =========================================================================== 1.8.3.1: The Asterisk Development Team has announced security releases for Asterisk branches 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.6.1.23, 1.6.2.17.1, and 1.8.3.1. The releases of Asterisk 1.6.1.23, 1.6.2.17.1, and 1.8.3.1 resolve two issues: * Resource exhaustion in Asterisk Manager Interface (AST-2011-003) * Remote crash vulnerability in TCP/TLS server (AST-2011-004) The issues and resolutions are described in the AST-2011-003 and AST-2011-004 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-003 and AST-2011-004, 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-1.8.3.1 Security advisory AST-2011-003 and AST-2011-004 are available at: http://downloads.asterisk.org/pub/security/AST-2011-003.pdf http://downloads.asterisk.org/pub/security/AST-2011-004.pdf =========================================================================== 1.8.3: The Asterisk Development Team has announced the release of Asterisk 1.8.3. The release of Asterisk 1.8.3 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 duplicated data in the AstDB when using DIALGROUP() * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing. * When using cdr_pgsql the billsec field was not populated correctly on unanswered calls. * Resolve memory leak in iCalendar and Exchange calendaring modules. * This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems. * Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi. * Resolve a memory leak when the Asterisk Manager Interface is disabled. * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. * Fix regression that changed behavior of queues when ringing a queue member. * Resolve deadlock involving REFER. Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at http://downloads.asterisk.org/pub/security/AST-2011-002.pdf For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3 =========================================================================== 1.8.2.4: 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. 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 configured for 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-1.8.2.4 Security advisory AST-2011-002 is available at: http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
2011-06-09 11:17:27 +02:00
SHA1 (patch-au) = 3f69f8bcea685f13008430c0fcb91885b6b72c90
Import Asterisk 1.8.1: Asterisk is a complete PBX in software. It 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. Asterisk 1.8 is a long term support version (i.e. it will be supported for four years with an additional year of security only fixes). See: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions What's new: Asterisk 1.8 is the next major release series of Asterisk. The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release. You can find a summary of the work involved with the 1.8.0 release in the sumary: http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt 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 ----- The Asterisk Development Team has announced the release of Asterisk 1.8.1. The release of Asterisk 1.8.1 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 issue when using directmedia. Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11) * Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers. (Patched by rmudgett) * Fix playback failure when using IAX with the timerfd module. (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data. (Closes issue #18161. Reported by wdoekes. Patched by tilghman) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
2010-12-15 04:22:43 +01:00
SHA1 (patch-av) = 3424013b5bf22624aa664e972e2b495ab3296cbe
SHA1 (patch-aw) = 0534acd67ea5da1eee8cf282035ebf4c559278ab
SHA1 (patch-ax) = 3b41e66a8c926e0afc4f73587e3557370e6c5f6e
Update to Asterisk 1.8.5.0: this is a general bug fix release 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 * Fixes thread blocking issue in the sip TCP/TLS implementation. * Be more tolerant of what URI we accept for call completion PUBLISH requests. * Fix a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call. * This patch fixes a bug with MeetMe behavior where the 'P' option for always prompting for a pin is ignored for the first caller. * 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. * 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. * 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. 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 Thank you for your continued support of Asterisk!
2011-07-16 23:35:11 +02:00
SHA1 (patch-ay) = 7c73c7664ea36b4c34cf38cde8a93c95b55a68aa
Import Asterisk 1.8.1: Asterisk is a complete PBX in software. It 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. Asterisk 1.8 is a long term support version (i.e. it will be supported for four years with an additional year of security only fixes). See: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions What's new: Asterisk 1.8 is the next major release series of Asterisk. The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release. You can find a summary of the work involved with the 1.8.0 release in the sumary: http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt 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 ----- The Asterisk Development Team has announced the release of Asterisk 1.8.1. The release of Asterisk 1.8.1 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 issue when using directmedia. Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11) * Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers. (Patched by rmudgett) * Fix playback failure when using IAX with the timerfd module. (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data. (Closes issue #18161. Reported by wdoekes. Patched by tilghman) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
2010-12-15 04:22:43 +01:00
SHA1 (patch-az) = 64365b12cb47ec0fba358e4326eda172f96068cf
SHA1 (patch-ba) = ffb20f4788f2f253e822fb48c68fec04c31b0619
PR/44257 - Francois Tigeot -- build fixes for DragonFly BSD Don't bother bumping the version since it didn't build on DFBSD before there is no binary package that could have changed, and this doesn't change the binary packages on other systems.
2010-12-22 05:28:52 +01:00
SHA1 (patch-bb) = bf1a2bb2ba1eb2ba44a9b26fa9ae0468510a1575
Import Asterisk 1.8.1: Asterisk is a complete PBX in software. It 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. Asterisk 1.8 is a long term support version (i.e. it will be supported for four years with an additional year of security only fixes). See: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions What's new: Asterisk 1.8 is the next major release series of Asterisk. The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release. You can find a summary of the work involved with the 1.8.0 release in the sumary: http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt 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 ----- The Asterisk Development Team has announced the release of Asterisk 1.8.1. The release of Asterisk 1.8.1 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 issue when using directmedia. Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11) * Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers. (Patched by rmudgett) * Fix playback failure when using IAX with the timerfd module. (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler) * Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) * Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data. (Closes issue #18161. Reported by wdoekes. Patched by tilghman) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
2010-12-15 04:22:43 +01:00
SHA1 (patch-bc) = 0efc99595d1ef82a879361e8bf3b2ef7fd84af62
PR/44257 - Francois Tigeot -- build fixes for DragonFly BSD Don't bother bumping the version since it didn't build on DFBSD before there is no binary package that could have changed, and this doesn't change the binary packages on other systems.
2010-12-22 05:28:52 +01:00
SHA1 (patch-be) = a3d416c097c6aeb0e49dec67a9fc22027d936773
SHA1 (patch-bf) = 67b506d235fabaa73f492d08858407dd9a85fd6e
fix pasto in a DragonFly BSD support patch
2010-12-22 09:25:58 +01:00
SHA1 (patch-bg) = e6dc4b3affdf634efc2b3ee83e81f7ec51ee2e86
PR/44257 - Francois Tigeot -- build fixes for DragonFly BSD Don't bother bumping the version since it didn't build on DFBSD before there is no binary package that could have changed, and this doesn't change the binary packages on other systems.
2010-12-22 05:28:52 +01:00
SHA1 (patch-bh) = 9203ea97daab8c64ea47f236b4961763e76eafe6
SHA1 (patch-bi) = d71662f618a10c3ca4277feb7ad0d659935dee1e
SHA1 (patch-bj) = a184452adf2c883695e3819c13c584a3db9608d7
SHA1 (patch-bk) = 93679dfb04d26c99ac9c2822e0d74d869d16369f
Upgrade to 1.8.4.2. This fixes several security issues including: AST-2011-002, AST-2011-003, AST-2011-004, AST-2011-005, AST-2011-006, and AST-2011-007. pkgsrc changes: - add patch for autosupport script; == -> = - patch configure to not unconditionally set PBX_LAUNCHD=1 - this allows res_timing_kqueue.so to build This last change brings a timing source to NetBSD which allows IAX trunking and allows the bridging modules to work, a rather major piece that was missing. Note that I haven't extensively tested it. But, have at it... =========================================================================== 1.8.4.2: The Asterisk Development Team has announced the release of Asterisk version 1.8.4.2, which is a security release for Asterisk 1.8. 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 =========================================================================== 1.8.4.1: The Asterisk Development Team has announced the release of Asterisk 1.8.4.1. 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) * 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. * Resolve potential crash when using SIP TLS support. * Improve reliability when using SIP TLS. 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 =========================================================================== 1.8.4: The Asterisk Development Team has announced the release of Asterisk 1.8.4. 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. * Resolve crash in ast_mutex_init() * Resolution of several DTMF based attended transfer issues. NOTE: Be sure to read the ChangeLog for more information about these changes. * Resolve deadlocks related to device states in chan_sip * Resolve an issue with the Asterisk manager interface leaking memory when disabled. * Support greetingsfolder as documented in voicemail.conf.sample. * Fix channel redirect out of MeetMe() and other issues with channel softhangup * Fix voicemail sequencing for file based storage. * 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+. * Fix issues with verbose messages not being output to the console. * Fix Deadlock with attended transfer of SIP call 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 =========================================================================== 1.8.3.3: 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 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3. The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 resolve two issues: * File Descriptor Resource Exhaustion (AST-2011-005) * Asterisk Manager User Shell Access (AST-2011-006) The issues and resolutions are described in the AST-2011-005 and AST-2011-006 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-005 and AST-2011-006, 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-1.8.3.3 Security advisory AST-2011-005 and AST-2011-006 are available at: http://downloads.asterisk.org/pub/security/AST-2011-005.pdf http://downloads.asterisk.org/pub/security/AST-2011-006.pdf =========================================================================== 1.8.3.2: he Asterisk Development Team has announced security releases for Asterisk branches 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.6.1.24, 1.6.2.17.2, and 1.8.3.2. ** This is a re-release of Asterisk 1.6.1.23, 1.6.2.17.1 and 1.8.3.1 which contained a bug which caused duplicate manager entries (issue #18987). The releases of Asterisk 1.6.1.24, 1.6.2.17.2, and 1.8.3.2 resolve two issues: * Resource exhaustion in Asterisk Manager Interface (AST-2011-003) * Remote crash vulnerability in TCP/TLS server (AST-2011-004) The issues and resolutions are described in the AST-2011-003 and AST-2011-004 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-003 and AST-2011-004, 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-1.8.3.2 Security advisory AST-2011-003 and AST-2011-004 are available at: http://downloads.asterisk.org/pub/security/AST-2011-003.pdf http://downloads.asterisk.org/pub/security/AST-2011-004.pdf =========================================================================== 1.8.3.1: The Asterisk Development Team has announced security releases for Asterisk branches 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.6.1.23, 1.6.2.17.1, and 1.8.3.1. The releases of Asterisk 1.6.1.23, 1.6.2.17.1, and 1.8.3.1 resolve two issues: * Resource exhaustion in Asterisk Manager Interface (AST-2011-003) * Remote crash vulnerability in TCP/TLS server (AST-2011-004) The issues and resolutions are described in the AST-2011-003 and AST-2011-004 security advisories. For more information about the details of these vulnerabilities, please read the security advisories AST-2011-003 and AST-2011-004, 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-1.8.3.1 Security advisory AST-2011-003 and AST-2011-004 are available at: http://downloads.asterisk.org/pub/security/AST-2011-003.pdf http://downloads.asterisk.org/pub/security/AST-2011-004.pdf =========================================================================== 1.8.3: The Asterisk Development Team has announced the release of Asterisk 1.8.3. The release of Asterisk 1.8.3 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 duplicated data in the AstDB when using DIALGROUP() * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for the function that does the parsing. * When using cdr_pgsql the billsec field was not populated correctly on unanswered calls. * Resolve memory leak in iCalendar and Exchange calendaring modules. * This version of Asterisk includes the new Compiler Flags option BETTER_BACKTRACES which uses libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems. * Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi. * Resolve a memory leak when the Asterisk Manager Interface is disabled. * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. * Fix regression that changed behavior of queues when ringing a queue member. * Resolve deadlock involving REFER. Additionally, this release has the changes related to security bulletin AST-2011-002 which can be found at http://downloads.asterisk.org/pub/security/AST-2011-002.pdf For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3 =========================================================================== 1.8.2.4: 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. 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 configured for 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-1.8.2.4 Security advisory AST-2011-002 is available at: http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
2011-06-09 11:17:27 +02:00
SHA1 (patch-bl) = ae68a81a758e3b49eb54b7400d8d5c6ed4efa51a
Reference in a new issue Copy permalink