asterisk-pjsip-13.7.1-1.fc22$>t+MLe`:u :>9?d  Htx  44 L4 4 4 4 4444l2({8*9,*:U2*G4H4I4XY\(4]4^Cbmdrewfzl|tÜ4ul4v<2w4x4yxCasterisk-pjsip13.7.11.fc22SIP channel based upon the PJSIP librarySIP channel based upon the PJSIP libraryV&arm04-builder06.arm.fedoraproject.org Fedora ProjectFedora ProjectGPLv2Fedora ProjectApplications/Internethttp://www.asterisk.org/linuxarmv7hl|*|*|*\*;T;<k*(*t)*L*M*]I>]I-I-I-I-IH,HCH@HWH@H@H@HkmHQHO@H1kH @Gu@GGG@G@G@G߮G΋@G@G@G@GG{|Gt@Gl@GiGiGg@GcGcG_@G_@G^{G]*@GO@GB@GAzG=@G9G G m@F%@FS@F^F^F @F@FF;@F;@FF@FtF#@F@FEF@F@Fzh@FvsFvsFvsFvsFvsFu"@Fu"@Fr@FAF1F@EWE@E@E8@E7hE6@E6@E6@E2"DDD(@DWIDLDGwDGwDF&@D - 13.7.1-1Fedora Release Engineering - 13.3.2-3.1Jared Smith - 13.3.2-3Robert Scheck - 13.3.2-2Fedora Release Engineering - 13.3.2-1.2Jitka Plesnikova - 13.3.2-1.1Jeffrey C. Ollie - 13.3.2-1:Jeffrey C. Ollie - 13.3.1-1:Jeffrey C. Ollie - 13.3.0-1:Jeffrey C. Ollie - 13.2.0-1:Jeffrey C. Ollie - 13.1.1-1:Jeffrey C. Ollie - 13.1.0-1Peter Robinson 13.0.2-3Tom Callaway - 13.0.2-2Jeffrey C. Ollie - 13.0.2-1Jeffrey C. Ollie - 13.0.1-1Jeffrey C. Ollie - 13.0.0-1Tom Callaway - 11.13.1-2Jeffrey C. Ollie - 11.13.1-1Jeffrey C. Ollie - 11.13.0-1Jeffrey C. Ollie - 11.12.1-1Jeffrey C. Ollie - 11.12.0-1Jeffrey C. Ollie - 11.11.0-1Jitka Plesnikova - 11.10.2-2.2Fedora Release Engineering - 11.10.2-2.1Jeffrey Ollie - 11.10.2-2:Jeffrey Ollie - 11.10.2-1:Jeffrey Ollie - 11.10.1-1:Jeffrey Ollie - 11.10.0-1:Fedora Release Engineering - 11.9.0-2.1Dennis Gilmore - 11.9.0-2Jeffrey Ollie - 11.9.0-1:Jeffrey Ollie - 11.8.1-1:Jeffrey Ollie - 11.8.0-1:Jeffrey Ollie - 11.7.0-1:Jeffrey Ollie - 11.6.1-1:Jeffrey Ollie - 11.6.0-1:Jeffrey Ollie - 11.5.1-3:Jeffrey Ollie - 11.5.1-2:Jeffrey Ollie - 11.5.1-1:Fedora Release Engineering - 11.4.0-2.2Petr Pisar - 11.4.0-2.1Rex Dieter 11.4.0-2Jeffrey Ollie - 11.4.0-1:Tom Callaway - 11.3.0-2:Jeffrey Ollie - 11.3.0-1:Jeffrey Ollie - 11.2.2-1:Jeffrey Ollie - 11.2.1-1:Jeffrey Ollie - 11.2.0-1:Jeffrey Ollie - 11.1.2-1:Jeffrey Ollie - 11.1.1-1:Jeffrey Ollie - 11.1.0-1:Jeffrey Ollie - 11.0.2-1:Dan Horák - 11.0.1-3Dennis Gilmore - 11.0.1-2Jeffrey Ollie - 11.0.1-1Jeffrey Ollie - 11.0.0-1:Jeffrey Ollie - 11.0.0-0.7.rc2:Jeffrey Ollie - 11.0.0-0.6.rc1Jeffrey Ollie - 11.0.0-0.5.beta2Jeffrey Ollie - 11.0.0-0.4.beta2Jeffrey Ollie - 10.8.0-1Dan Horák - 11.0.0-0.3.beta1Dan Horák - 10.7.1-2Jeffrey Ollie - 10.7.1-1Jeffrey Ollie - 10.7.0-1Jeffrey Ollie - 10.6.1-1Jeffrey Ollie - 10.6.0-1Jeffrey Ollie - 11.0.0-0.2.beta1Fedora Release Engineering - 10.5.2-1.2Petr Pisar - 10.5.2-1.1Jeffrey Ollie - 10.5.2-1:Petr Pisar - 10.5.1-1.1Jeffrey Ollie - 10.5.1-1Jeffrey Ollie - 10.5.0-1Petr Pisar - 10.4.2-1.1Jeffrey Ollie - 10.4.2-1Jeffrey Ollie - 10.4.1-1Jeffrey Ollie - 10.4.0-1Jeffrey Ollie - 10.3.1-1Russell Bryant - 10.3.0-1Russell Bryant - 10.2.1-1Jeffrey C. Ollie - 10.1.2-2Jeffrey C. Ollie - 10.1.2-1Jeffrey C. Ollie - 10.1.1-1Jeffrey C. Ollie - 10.1.0-1Russell Bryant - 10.0.0-2Fedora Release Engineering - 10.0.0-1.1Jeffrey C. Ollie - 10.0.0-1Jeffrey C. Ollie - 10.0.0-1Jeffrey C. Ollie - 10.0.0-0.7.rc3Jeffrey C. Ollie - 10.0.0-0.6.rc2Jeffrey C. Ollie - 10.0.0-0.5.rc1Jeffrey C. Ollie - 10.0.0-0.4.beta2Jeffrey C. Ollie - 10.0.0-0.3.beta2Jeffrey C. Ollie - 10.0.0-0.2.beta2Jeffrey C. Ollie - 10.0.0-0.1.beta1Petr Sabata - 1.8.5.0-1.2Petr Sabata - 1.8.5.0-1.1Jeffrey C. Ollie - 1.8.5.0-1Jeffrey C. Ollie - 1.8.5-0.2Jeffrey C. Ollie - 1.8.5-0.1.rc1Jeffrey C. Ollie - 1.8.5-0.1.rc1Jeffrey C. Ollie - 1.8.4.4-2Jeffrey C. Ollie - 1.8.4.4-1Jeffrey C. Ollie - 1.8.4.3-3Jeffrey C. Ollie - 1.8.4.3-2Jeffrey C. Ollie - 1.8.4.3-1Jeffrey C. Ollie - 1.8.4.2-2Marcela Mašláňová - 1.8.4.2-1.2Marcela Mašláňová - 1.8.4.2-1.1Jeffrey C. Ollie - 1.8.4.2-1:Jeffrey C. Ollie - 1.8.3.3-1Jeffrey C. Ollie - 1.8.3.2-2Jeffrey C. Ollie - 1.8.3.2-1Jeffrey C. Ollie - 1.8.3.1-1 - 1.8.3-1 - 1.8.3-0.7.rc3Jeffrey C. Ollie - 1.8.3-0.6.rc2Jeffrey C. Ollie - 1.8.3-0.5.rc2Jeffrey C. Ollie - 1.8.3-0.4.rc2Fedora Release Engineering - 1.8.3-0.3.rc2Jeffrey C. Ollie - 1.8.3-0.2.rc2Jeffrey C. Ollie - 1.8.3-0.1.rc1Jeffrey C. Ollie - 1.8.2.3-1Jeffrey C. Ollie - 1.8.2.2-2Jeffrey C. Ollie - 1.8.2.2-1Jeffrey C. Ollie - 1.8.2.1-1Jeffrey C. Ollie - 1.8.2-1Jeffrey C. Ollie - 1.8.1.1-1Jeffrey C. Ollie - 1.8.1-1Dennis Gilmore - 1.8.0-6Dennis Gilmore - 1.8.0-5Dennis Gilmore - 1.8.0-4Jeffrey C. Ollie - 1.8.0-3Jeffrey C. Ollie - 1.8.0-2Jeffrey C. Ollie - 1.8.0-1Jeffrey C. Ollie - 1.8.0-0.8.rc5:Jeffrey C. Ollie - 1.8.0-0.7.rc3Jeffrey C. Ollie - 1.8.0-0.6.rc2Jeffrey C. Ollie - 1.8.0-0.5.beta5Jeffrey C. Ollie - 1.8.0-0.4.beta4Jeffrey C. Ollie - 1.8.0-0.3.beta3Jeffrey C. Ollie - 1.8.0-0.2.beta2Jeffrey C. Ollie - 1.8.0-0.1.beta2Jeffrey C. Ollie - 1.6.2.10-1Jeffrey C. Ollie - 1.6.2.8-0.3.rc1Marcela Maslanova - 1.6.2.8-0.2.rc1Jeffrey C. Ollie - 1.6.2.7-1Jeffrey C. Ollie - 1.6.2.7-0.2.rc3Jeffrey C. Ollie - 1.6.2.7-0.1.rc2Jeffrey C. Ollie - 1.6.2.6-1Jeffrey C. Ollie - 1.6.2.6-0.1.rc2Jeffrey C. Ollie - 1.6.2.5-2Jeffrey C. Ollie - 1.6.2.5-1Jeffrey C. Ollie - 1.6.2.4-1Jeffrey C. Ollie - 1.6.2.2-1Jeffrey C. Ollie - 1.6.2.1-1Jeffrey C. Ollie - 1.6.2.1-0.1.rc1Jeffrey C. Ollie - 1.6.2.0-1Jeffrey C. Ollie - 1.6.2.0-0.16.rc8Jeffrey C. Ollie - 1.6.2.0-0.15.rc7Jeffrey C. Ollie - 1.6.2.0-0.14.rc6Jeffrey C. Ollie - 1.6.2.0-0.13.rc6Jeffrey C. Ollie - 1.6.2.0-0.12.rc6Jeffrey C. Ollie - 1.6.2.0-0.11.rc5Jeffrey C. Ollie - 1.6.2.0-0.10.rc4Jeffrey C. Ollie - 1.6.2.0-0.9.rc3Jeffrey C. Ollie - 1.6.2.0-0.8.rc3Jeffrey C. Ollie - 1.6.2.0-0.7.rc3Jeffrey C. Ollie - 1.6.2.0-0.6.rc3Jeffrey C. Ollie - 1.6.2.0-0.5.rc3Jeffrey C. Ollie - 1.6.2.0-0.4.rc3Jeffrey C. Ollie - 1.6.2.0-0.3.rc2Jeffrey C. Ollie - 1.6.2.0-0.2.rc2Jeffrey C. Ollie - 1.6.2.0-0.1.rc2Jeffrey C. Ollie - 1.6.1.6-2Jeffrey C. Ollie - 1.6.1.6-1Jeffrey C. Ollie - 1.6.1-0.26.rc1Tomas Mraz - 1.6.1-0.25.rc1Fedora Release Engineering - 1.6.1-0.24.rc1Jeffrey C. Ollie - 1.6.1-0.23.rc1Fedora Release Engineering - 1.6.1-0.22.rc1Jeffrey C. Ollie - 1.6.1-0.21.rc1Tomas Mraz - 1.6.1-0.13.beta4Jeffrey C. Ollie - 1.6.1-0.12.beta4Jeffrey C. Ollie - 1.6.1-0.10.beta4Jeffrey C. Ollie - 1.6.1-0.9.beta4Jeffrey C. Ollie - 1.6.1-0.8.beta4Jeffrey C. Ollie - 1.6.1-0.7.beta3Alex Lancaster - 1.6.1-0.6.beta2Jeffrey C. Ollie - 1.6.1-0.5.beta2Jeffrey C. Ollie - 1.6.1-0.4.beta2Jeffrey C. Ollie - 1.6.1-0.3.beta2Jeffrey C. Ollie - 1.6.1-0.2.beta2Jeffrey C. Ollie - 1.6.0.1-3Jeffrey C. Ollie - 1.6.0.1-2Jeffrey C. Ollie - 1.6.0-1- Bastien Nocera - 1.6.0-0.22.beta9Jeffrey C. Ollie - 1.6.0-0.21.beta9Jeffrey C. Ollie - 1.6.0-0.20.beta9Jeffrey C. Ollie - 1.6.0-0.19.beta9Jeffrey C. Ollie - 1.6.0-0.18.beta9Jeffrey C. Ollie - 1.6.0-0.17.beta9Jeffrey C. Ollie - 1.6.0-0.16.beta9Jeffrey C. Ollie - 1.6.0-0.15.beta9Jeffrey C. Ollie - 1.6.0-0.14.beta9Jeffrey C. Ollie - 1.6.0-0.13.beta8Jeffrey C. Ollie - 1.6.0-0.12.beta7.1Jeffrey C. Ollie - 1.6.0-0.11.beta7.1Jeffrey C. Ollie - 1.6.0-0.10.beta7Jeffrey C. Ollie - 1.6.0-0.9.beta6Jeffrey C. Ollie - 1.6.0-0.8.beta6Jeffrey C. Ollie - 1.6.0-0.6.beta6Tom "spot" Callaway - 1.6.0-0.5.beta5Jeffrey C. Ollie - 1.6.0-0.4.beta5Jeffrey C. Ollie - 1.6.0-0.3.beta4Jeffrey C. Ollie - 1.6.0-0.2.beta4Jeffrey C. Ollie - 1.6.0-0.1.beta4Jeffrey C. Ollie - 1.4.18-1Jeffrey C. Ollie - 1.4.17-1Jeffrey C. Ollie - 1.4.16.2-1Jeffrey C. Ollie - 1.4.16.1-2Jeffrey C. Ollie - 1.4.16.1-1Jeffrey C. Ollie - 1.4.16-2Jeffrey C. Ollie - 1.4.16-1Jeffrey C. Ollie - 1.4.15-7Jeffrey C. Ollie - 1.4.15-6Jeffrey C. Ollie - 1.4.15-5Jeffrey C. Ollie - 1.4.15-4Jeffrey C. Ollie - 1.4.15-3Jeffrey C. Ollie - 1.4.15-2Jeffrey C. Ollie - 1.4.15-1Jeffrey C. Ollie - 1.4.14-2Jeffrey C. Ollie - 1.4.14-1Jeffrey C. Ollie - 1.4.13-7Jeffrey C. Ollie - 1.4.13-6Jeffrey C. Ollie - 1.4.13-1Jeffrey C. Ollie - 1.4.12.1-1Jeffrey C. Ollie - 1.4.11-1Jeffrey C. Ollie - 1.4.10.1-1Jeffrey C. Ollie - 1.4.10-1Jeffrey C. Ollie - 1.4.9-7Jeffrey C. Ollie - 1.4.9-6Jeffrey C. Ollie - 1.4.9-5Jeffrey C. Ollie - 1.4.9-4Jeffrey C. Ollie - 1.4.9-3Jeffrey C. Ollie - 1.4.9-2Jeffrey C. Ollie - 1.4.9-1Jeffrey C. Ollie - 1.4.8-1Jeffrey C. Ollie - 1.4.7.1-1Jeffrey C. Ollie - 1.4.7-1Jeffrey C. Ollie - 1.4.6-4Jeffrey C. Ollie - 1.4.6-3Jeffrey C. Ollie - 1.4.6-2Jeffrey C. Ollie - 1.4.6-1Jeffrey C. Ollie - 1.4.5-10Jeffrey C. Ollie - 1.4.5-9Jeffrey C. Ollie - 1.4.5-8Jeffrey C. Ollie - 1.4.5-7Jeffrey C. Ollie - 1.4.5-6Jeffrey C. Ollie - 1.4.5-5Jeffrey C. Ollie - 1.4.5-4Jeffrey C. Ollie - 1.4.5-3Jeffrey C. Ollie - 1.4.5-1Jeffrey C. Ollie - 1.4.4-2Jeffrey C. Ollie - 1.4.4-1Jeffrey C. Ollie - 1.4.2-1Jeffrey C. Ollie - 1.4.1-2Jeffrey C. Ollie - 1.4.1-1Jeffrey C. Ollie - 1.4.0-6.beta4Jeffrey C. Ollie - 1.4.0-5.beta3Jeffrey C. Ollie - 1.4.0-4.beta3Jeffrey C. Ollie - 1.4.0-3.beta3Jeffrey C. Ollie - 1.4.0-2.beta3Jeffrey C. Ollie - 1.4.0-1.beta3Jeffrey C. Ollie - 1.4.0-0.beta2Jeffrey C. Ollie - 1.2.10-1Jeffrey C. Ollie - 1.2.9.1Jeffrey C. Ollie - 1.2.8Jeffrey C. Ollie - 1.2.7.1-6Jeffrey C. Ollie - 1.2.7.1-5Jeffrey C. Ollie - 1.2.7.1-4Jeffrey C. Ollie - 1.2.7.1-3Jeffrey C. Ollie - 1.2.7.1-2Jeffrey C. Ollie - 1.2.7-1Jeffrey C. Ollie - 1.2.6-3Jeffrey C. Ollie - 1.2.6-2Jeffrey C. Ollie - 1.2.6-1Jeffrey C. Ollie - 1.2.5-1Jeffrey C. Ollie - 1.2.4-4Jeffrey C. Ollie - 1.2.4-3Jeffrey C. Ollie - 1.2.4-2Jeffrey C. Ollie - 1.2.4-1Jeffrey C. Ollie - 1.2.3-4Jeffrey C. Ollie - 1.2.3-3Jeffrey C. Ollie - 1.2.3-2Jeffrey C. Ollie - 1.2.3-1- Update to upstream 13.7.1 release for security fixes - Resolves AST-2016-001: BEAST vulnerability in HTTP server - Resolves AST-2016-002: File descriptor exhaustion in chan_sip - Resolves AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data - Full changelog at http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1 - Also build the 'radius' sub-package against freeradius-client-devel, as the radiusclient-ng project is dead- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild- Remove %defattr macro invocations, as they are no longer needed- Rebuild for libical 2.0.0- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild- Perl 5.22 rebuild- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available - security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11, - 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerability: - - * AST-2015-003: TLS Certificate Common name NULL byte exploit - - When Asterisk registers to a SIP TLS device and and verifies the server, - Asterisk will accept signed certificates that match a common name other than - the one Asterisk is expecting if the signed certificate has a common name - containing a null byte after the portion of the common name that Asterisk - expected. This potentially allows for a man in the middle attack. - - For more information about the details of this vulnerability, please read - security advisory AST-2015-003, which was released at the same time as this - announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2 - - The security advisory is available at: - - * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf- The Asterisk Development Team has announced the release of Asterisk 13.3.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 13.3.1 resolves an issue reported by the - community and would have not been possible without your participation. - Thank you! - - The following is the issue resolved in this release: - - * --- pjsip: resolve compatibility problem with ast_sip_session - (Closes issue ASTERISK-24941. Reported by Matt Jordan) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.1- The Asterisk Development Team has announced the release of Asterisk 13.3.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 13.3.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - New Features made in this release: - ----------------------------------- - * ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a - channel (Reported by Matt Jordan) - * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation - (Reported by Dwayne Hubbard) - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid - string copy (Reported by Yura Kocyuba) - * ASTERISK-24748 - res_pjsip: If wizards explicitly configured in - sorcery.conf false ERROR messages may occur (Reported by Joshua - Colp) - * ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked - (Reported by Matt Jordan) - * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in - res_odbc (Reported by ibercom) - * ASTERISK-24479 - Enable REF_DEBUG for module references - (Reported by Corey Farrell) - * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to - fully disconnect underlying socket, leading to events being - dropped with no additional information (Reported by Matt Jordan) - * ASTERISK-24772 - ODBC error in realtime sippeers when device - unregisters under MariaDB (Reported by Richard Miller) - * ASTERISK-24752 - Crash in bridge_manager_service_req when bridge - is destroyed by ARI during shutdown (Reported by Richard - Mudgett) - * ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported - by Zane Conkle) - * ASTERISK-24015 - app_transfer fails with PJSIP channels - (Reported by Private Name) - * ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk - transfer scenario. (Reported by Mark Michelson) - * ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by - Niklas Larsson) - * ASTERISK-24716 - Improve pjsip log messages for presence - subscription failure (Reported by Rusty Newton) - * ASTERISK-24612 - res_pjsip: No information if a required sorcery - wizard is not loaded (Reported by Joshua Colp) - * ASTERISK-24768 - res_timing_pthread: file descriptor leak - (Reported by Matthias Urlichs) - * ASTERISK-24685 - "pjsip show version" CLI command (Reported by - Joshua Colp) - * ASTERISK-24632 - install_prereq script installs pjproject - without IPv6 support (Reported by Rusty Newton) - * ASTERISK-24085 - Documentation - We should remove or further - document the 'contact' section in pjsip.conf (Reported by Rusty - Newton) - * ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by - JoshE) - * ASTERISK-24700 - CRASH: NULL channel is being passed to - ast_bridge_transfer_attended() (Reported by Zane Conkle) - * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove - (Reported by Corey Farrell) - * ASTERISK-24799 - [patch] make fails with undefined reference to - SSLv3_client_method (Reported by Alexander Traud) - * ASTERISK-22670 - Asterisk crashes when processing ISDN AoC - Events (Reported by klaus3000) - * ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn - call (Reported by Marcel Manz) - * ASTERISK-24740 - [patch]Segmentation fault on aoc-e event - (Reported by Panos Gkikakis) - * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility - for playing back messages stored in IMAP - play_message: No - origtime (Reported by Graham Barnett) - * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc - OSX with 64 bit integers (Reported by Corey Farrell) - * ASTERISK-24796 - Codecs and bucket schema's prevent module - unload (Reported by Corey Farrell) - * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML - (Reported by Ashley Sanders) - * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring - is invalid (Reported by Rusty Newton) - * ASTERISK-24785 - 'Expires' header missing from 200 OK on - REGISTER (Reported by Ross Beer) - * ASTERISK-24677 - ARI GET variable on channel provides unhelpful - response on non-existent variable (Reported by Joshua Colp) - * ASTERISK-24797 - bridge_softmix: G.729 codec license held - (Reported by Kevin Harwell) - * ASTERISK-24812 - ARI: Creating channels through /channels - resource always uses SLIN, which results in unneeded transcoding - (Reported by Matt Jordan) - * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid - thread ID being passed to pthread_kill (Reported by JoshE) - * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime - fail (Reported by Terry Wilson) - * ASTERISK-23214 - chan_sip WARNING message 'We are requesting - SRTP for audio, but they responded without it' is ambiguous and - wrong in some cases (Reported by Rusty Newton) - * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an - error response and BYE are sent to the caller (Reported by - Makoto Dei) - * ASTERISK-18105 - most of asterisk modules are unbuildable in - cygwin environment (Reported by feyfre) - * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell) - * ASTERISK-24751 - Integer values in json payload to ARI cause - asterisk to crash (Reported by jeffrey putnam) - * ASTERISK-24838 - chan_sip: Locking inversion occurs when - building a peer causes a peer poke during request handling - (Reported by Richard Mudgett) - * ASTERISK-24825 - Caller ID not recognized using - Centrex/Distinctive dialing (Reported by Richard Mudgett) - * ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not - HAVE_PJPROJECT (Reported by Stefan Engström) - * ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers - (Reported by Kevin Harwell) - * ASTERISK-24755 - Asterisk sends unexpected early BYE to - transferrer during attended transfer when using a Stasis bridge - (Reported by John Bigelow) - * ASTERISK-24739 - [patch] - Out of files -- call fails -- - numerous files with inodes from under /usr/share/zoneinfo, - mostly posixrules (Reported by Ed Hynan) - * ASTERISK-23390 - NewExten Event with application AGI shows up - before and after AGI runs (Reported by Benjamin Keith Ford) - * ASTERISK-24786 - [patch] - Asterisk terminates when playing a - voicemail stored in LDAP (Reported by Graham Barnett) - * ASTERISK-24808 - res_config_odbc: Improper escaping of - backslashes occurs with MySQL (Reported by Javier Acosta) - * ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported - by Anatoli) - * ASTERISK-20850 - [patch]Nested functions aren't portable. - Adapting RAII_VAR to use clang/llvm blocks to get the - same/similar functionality. (Reported by Diederik de Groot) - * ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI - connection on error (Reported by Dmitriy Serov) - * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported - by Frank DiGennaro) - * ASTERISK-21038 - Bad command completion of "core set debug - channel" (Reported by Richard Kenner) - * ASTERISK-18708 - func_curl hangs channel under load (Reported by - Dave Cabot) - * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by - Atis Lezdins) - * ASTERISK-24876 - Investigate reference leaks from - tests/channels/local/local_optimize_away (Reported by Corey - Farrell) - * ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported - by Corey Farrell) - * ASTERISK-24817 - init_logger_chain: unreachable code block - (Reported by Corey Farrell) - * ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by - snuffy) - * ASTERISK-24879 - [patch]Compilation fails due to 64bit time - under OpenBSD (Reported by snuffy) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes - (Reported by Ben Merrills) - * ASTERISK-24811 - asterisk-publication sorcery object does not - use realtime (Reported by Matt Hoskins) - * ASTERISK-24790 - Reduce spurious noise in logs from voicemail - - Couldn't find mailbox %s in context (Reported by Graham Barnett) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0- The Asterisk Development Team has announced the release of Asterisk 13.2.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 13.2.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them - all at the same time. (Reported by Richard Mudgett) - * ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow - when using non-default sorcery wizard (Reported by Kevin - Harwell) - * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS - from JSSIP (Reported by Badalian Vyacheslav) - * ASTERISK-24607 - res_pjsip_session: re-INVITE with declined - media streams results in 488 (Reported by Matt Jordan) - * ASTERISK-24563 - Direct Media calls within private network - sometimes get one way audio (Reported by Kevin Harwell) - * ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to - race condition in accessing codec in stored ast_frame and codec - core (Reported by Matt Jordan) - * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag - enabled (Reported by Richard Mudgett) - * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is - enabled (Reported by Andreas Steinmetz) - * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly - casts char to unsigned int (Reported by Walter Doekes) - * ASTERISK-24536 - AMI redirect with PJSIP fails to move extra - channel (Reported by Niklas Larsson) - * ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is - chosen for RTP compatible channels when the DTMF mode is not - compatible (Reported by Yaniv Simhi) - * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher - level - 'Remote address is null, most likely RTP has been - stopped' (Reported by Rusty Newton) - * ASTERISK-24513 - Local channel apparently leaked in off-nominal - DTMF attended transfer (Reported by Mark Michelson) - * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present - on startup (Reported by Richard Kenner) - * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong - destination when 'sendrpid=yes' (in proxy environment) (Reported - by Karsten Wemheuer) - * ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall - calls to the transferrer. (Reported by Richard Mudgett) - * ASTERISK-24376 - res_pjsip_refer: REFER request for remote - session attempts to direct channel to external_replaces - extension instead of context, without providing for the - Referred-To SIP URI (Reported by Matt Jordan) - * ASTERISK-24591 - Stasis() side of an ARI originated channel - cannot be Redirected (Reported by Kinsey Moore) - * ASTERISK-24049 - Asterisk Manager Interface: A number of list - type responses aren't using astman_send_listack (Reported by - Jonathan Rose) - * ASTERISK-24637 - Channel re-enters Stasis() when it should not - (Reported by John Bigelow) - * ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does - not function (Reported by John Kiniston) - * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT - (Reported by Kristian Høgh) - * ASTERISK-20744 - [patch] Security event logging does not work - over syslog (Reported by Michael Keuter) - * ASTERISK-24665 - Configure check required for - pjsip_get_dest_info() (Reported by Mark Michelson) - * ASTERISK-23850 - Park Application does not respect Return - Context Priority (Reported by Andrew Nagy) - * ASTERISK-23991 - [patch]asterisk.pc file contains a small error - in the CFlags returned (Reported by Diederik de Groot) - * ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown - while attempting to publish (Reported by Kevin Harwell) - * ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown - (Reported by Corey Farrell) - * ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails - on cross compilation (Reported by abelbeck) - * ASTERISK-24624 - Transfer to invalid extension results in hung - channel. (Reported by Zane Conkle) - * ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf, - Incorrect External Addresses is Used in SIP Packets When - Responding to INVITE (Reported by David Justl) - * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail - - voicemail is not deleted after review, hangup (Reported by LEI - FU) - * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects - 32-bit packages on 64-bit hosts (Reported by Ben Klang) - * ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding - to most traffic, potential deadlock (Reported by Jeff Collell) - * ASTERISK-24560 - Creating a named ARI bridge twice causes a - crash (Reported by Kinsey Moore) - * ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when - MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported - by Matt Jordan) - * ASTERISK-24640 - Registration pending stays forever after sip - reload (Reported by Max Man) - * ASTERISK-24673 - outgoing sip registers cannot be removed or - modified without doing restart (or doing module unload - chan_sip.so) (Reported by Stefan Engström) - * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor - m() option does not queue an MWI event (Reported by Gareth - Palmer) - * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis - fails to get app name (Reported by John Bigelow) - * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive - column comparison for 'defaultuser' (Reported by - HZMI8gkCvPpom0tM) - * ASTERISK-24693 - Investigate and fix memory leaks in Asterisk - (Reported by Kevin Harwell) - * ASTERISK-24626 - Voicemail passwords not being stored in ARA - (Reported by Paddy Grice) - * ASTERISK-24539 - Compile fails on OSX because of sem_timedwait - in bridge_channel.c (Reported by George Joseph) - * ASTERISK-24544 - Compile fails on OSX Yosemite because of - incorrect detection of htonll and ntohll (Reported by George - Joseph) - * ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX' - no longer displays user menus (Reported by Matt Jordan) - * ASTERISK-24721 - manager: ModuleLoad action incorrectly reports - 'module not found' during a Reload operation (Reported by Matt - Jordan) - * ASTERISK-24719 - ConfBridge recording channels get stuck when - recording started/stopped more than once (Reported by Richard - Mudgett) - * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported - by Kevin Harwell) - * ASTERISK-24728 - tcptls: Bad file descriptor error when - reloading chan_sip (Reported by Kevin Harwell) - * ASTERISK-24729 - Outbound registration not occuring on new - registrations after reload. (Reported by Richard Mudgett) - * ASTERISK-24676 - Security Vulnerability: URL request injection - in libCURL (CVE-2014-8150) (Reported by Matt Jordan) - * ASTERISK-24666 - Security Vulnerability: RTP not closed after - sip call using unsupported codec (Reported by Y Ateya) - * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL - versions (Reported by Jared Biel) - * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by - Stephan Eisvogel) - * ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson) - * ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no response - is ever received (Reported by Marco Paland) - * ASTERISK-24737 - When agent not logged in, agent status shows - unavailable, queue status shows agent invalid (Reported by - Richard Mudgett) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-24552 - ARI: Allow associating a channel as an - initiator of an Origination for record keeping purposes - (Reported by Matt Jordan) - * ASTERISK-24553 - ARI/AMI: Include language in standard channel - snapshot output (Reported by Matt Jordan) - * ASTERISK-24643 - res_pjsip: Add user=phone option (Reported by - Matt Jordan) - * ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for - connection-oriented transports. (Reported by Matt Jordan) - * ASTERISK-24412 - [patch]Incomplete channel originate/continue - handling with ARI (Reported by Nir Simionovich (GreenfieldTech - - Israel)) - * ASTERISK-24678 - [PATCH] Added atxfer* settings to - features.conf.sample (Reported by Niklas Larsson) - * ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported - by cloos) - * ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by - Dan Jenkins) - * ASTERISK-24316 - For httpd server, need option to define server - name for security purposes (Reported by Andrew Nagy) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available - security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10, - 11.15.1, 12.8.1, and 13.1.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerabilities: - - * AST-2015-001: File descriptor leak when incompatible codecs are offered - - Asterisk may be configured to only allow specific audio or - video codecs to be used when communicating with a - particular endpoint. When an endpoint sends an SDP offer - that only lists codecs not allowed by Asterisk, the offer - is rejected. However, in this case, RTP ports that are - allocated in the process are not reclaimed. - - This issue only affects the PJSIP channel driver in - Asterisk. Users of the chan_sip channel driver are not - affected. - - * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability - - CVE-2014-8150 reported an HTTP request injection - vulnerability in libcURL. Asterisk uses libcURL in its - func_curl.so module (the CURL() dialplan function), as well - as its res_config_curl.so (cURL realtime backend) modules. - - Since Asterisk may be configured to allow for user-supplied - URLs to be passed to libcURL, it is possible that an - attacker could use Asterisk as an attack vector to inject - unauthorized HTTP requests if the version of libcURL - installed on the Asterisk server is affected by - CVE-2014-8150. - - For more information about the details of these vulnerabilities, please read - security advisory AST-2015-001 and AST-2015-002, which were released at the same - time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf - * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf- The Asterisk Development Team has announced the release of Asterisk 13.1.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 13.1.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - New Features made in this release: - ----------------------------------- - * ASTERISK-24554 - AMI/ARI: Generate events on connected line - changes (Reported by Matt Jordan) - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling - against libsrtp-1.5.0 (Reported by Patrick Laimbock) - * ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported by - Corey Farrell) - * ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing - leak (Reported by Corey Farrell) - * ASTERISK-24430 - missing letter "p" in word response in - OriginateResponse event documentation (Reported by Dafi Ni) - * ASTERISK-24437 - Review implementation of ast_bridge_impart for - leaks and document proper usage (Reported by Scott Griepentrog) - * ASTERISK-24453 - manager: acl_change_sub leaks (Reported by - Corey Farrell) - * ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by - Corey Farrell) - * ASTERISK-24458 - chan_phone fails to build on big endian systems - (Reported by Tzafrir Cohen) - * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers - (Reported by Olle Johansson) - * ASTERISK-24304 - asterisk crashing randomly because of unistim - channel (Reported by dhanapathy sathya) - * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by - Nick Adams) - * ASTERISK-24462 - res_pjsip: Stale qualify statistics after - disablementation (Reported by Kevin Harwell) - * ASTERISK-24465 - audiohooks list leaks reference to formats - (Reported by Corey Farrell) - * ASTERISK-24466 - app_queue: fix a couple leaks to struct - call_queue (Reported by Corey Farrell) - * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled - (Reported by Corey Farrell) - * ASTERISK-24411 - [patch] Status of outbound registration is not - changed upon unregistering. (Reported by John Bigelow) - * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream - leaks (Reported by Corey Farrell) - * ASTERISK-24480 - res_http_websockets: Module reference decrease - below zero (Reported by Corey Farrell) - * ASTERISK-24482 - func_talkdetect: Fix stasis message leak in - audiohook callback (Reported by Corey Farrell) - * ASTERISK-24487 - configuration: sections should be loadable as - template even when not marked (Reported by Scott Griepentrog) - * ASTERISK-20127 - [Regression] Config.c config_text_file_load() - unescapes semicolons ("\;" -> ";") turning them into comments - (corruption) on rewrite of a config file (Reported by George - Joseph) - * ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload - when DNS settings invalid (Reported by Melissa Shepherd) - * ASTERISK-24307 - Unintentional memory retention in stringfields - (Reported by Etienne Lessard) - * ASTERISK-24491 - Memory leak in res_hep (Reported by Zane - Conkle) - * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes - extra calls to ast_module_unref (Reported by Corey Farrell) - * ASTERISK-24447 - Bridge DTMF hooks: Audio doesn't pass when - waiting for more matching digits. (Reported by Richard Mudgett) - * ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to - queue caller (Reported by Steve Pitts) - * ASTERISK-24504 - chan_console: Fix reference leaks to pvt - (Reported by Corey Farrell) - * ASTERISK-24250 - [patch] Voicemail with multi-recipients To: - header fix (Reported by abelbeck) - * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS - length exceeds 50 (roughly) national symbols (Reported by - Dmitriy Bubnov) - * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN - revision r227276 (Reported by Xavier Hienne) - * ASTERISK-24505 - manager: http connections leak references - (Reported by Corey Farrell) - * ASTERISK-24502 - Build fails when dev-mode, dont optimize and - coverage are enabled (Reported by Corey Farrell) - * ASTERISK-24444 - PBX: Crash when generating extension for - pattern matching hint (Reported by Leandro Dardini) - * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP - packet to JSON for res_hep_rtcp and report blocks are greater - than 1 (Reported by Gregory Malsack) - * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended - transfer (Reported by Beppo Mazzucato) - * ASTERISK-24501 - ARI: Moving a channel between bridges followed - by a hangup can cause an ARI client to not receive an expected - ChannelLeftBridge event before StasisEnd (Reported by Matt - Jordan) - * ASTERISK-24336 - PJSIP timer_min_se value under 90 causes crash - (Reported by Leon Rowland) - * ASTERISK-23651 - Reloading some modules that are loaded already, - results in 'No such module' before a successful reload (Reported - by Rusty Newton) - * ASTERISK-24522 - ConfBridge: delay occurs between kicking all - endmarked users when last marked user leaves (Reported by Matt - Jordan) - * ASTERISK-15242 - transmit_refer leaks sip_refer structures - (Reported by David Woolley) - * ASTERISK-24508 - pjsip - REFER request from SNOM is rejected - with "400 bad request" - DEBUG shows "Received a REFER without a - parseable Refer-To" (Reported by Beppo Mazzucato) - * ASTERISK-24535 - stringfields: Fix regression from fix for - unintentional memory retention and another issue exposed by the - fix (Reported by Corey Farrell) - * ASTERISK-24471 - Crash - assert_fail in libc in - pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2 - (Reported by yaron nahum) - * ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces - in-dialog with invalid target causes crash (Reported by Joshua - Colp) - * ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial - module load (Reported by Matt Jordan) - * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs - allow blocked addresses through (Reported by Matt Jordan) - * ASTERISK-24542 - [patch]Failure showing codecs via 'core show - channeltype ' (Reported by snuffy) - * ASTERISK-24533 - 2 threads created per chan_sip entry (Reported - by xrobau) - * ASTERISK-24516 - [patch]Asterisk segfaults when playing back - voicemail under high concurrency with an IMAP backend (Reported - by David Duncan Ross Palmer) - * ASTERISK-24572 - [patch]App_meetme is loaded without its - defaults when the configuration file is missing (Reported by - Nuno Borges) - * ASTERISK-24573 - [patch]Out of sync conversation recording when - divided in multiple recordings (Reported by Nuno Borges) - * ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not - reliably transmitted during transfers (Reported by Matt Jordan) - * ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsip - extension to another pjsip extension (Reported by Abhay Gupta) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR - property 'unanswered' (Reported by Matt Jordan) - * ASTERISK-24283 - [patch]Microseconds precision in the eventtime - column in the cel_odbc module (Reported by Etienne Lessard) - * ASTERISK-24530 - [patch] app_record stripping 1/4 second from - recordings (Reported by Ben Smithurst) - * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded - lookups (Reported by Birger "WIMPy" Harzenetter) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.1.0- Add speexdsp as build dep as speex_echo.h has moved - rhbz 1181021- update for lua 5.3- The Asterisk Development Team has announced security releases for Certified - Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are - released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerability: - - * AST-2014-019: Remote Crash Vulnerability in WebSocket Server - - When handling a WebSocket frame the res_http_websocket module dynamically - changes the size of the memory used to allow the provided payload to fit. If a - payload length of zero was received the code would incorrectly attempt to - resize to zero. This operation would succeed and end up freeing the memory but - be treated as a failure. When the session was subsequently torn down this - memory would get freed yet again causing a crash. - - For more information about the details of this vulnerability, please read - security advisory AST-2014-019, which was released at the same time as this - announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2 - - The security advisory is available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available - security releases are released as versions 1.8.28-cert3, 11.6-cert8, 1.8.32.1, - 11.14.1, 12.7.1, and 13.0.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerabilities: - - * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP - address families - - Many modules in Asterisk that service incoming IP traffic have ACL options - ("permit" and "deny") that can be used to whitelist or blacklist address - ranges. A bug has been discovered where the address family of incoming - packets is only compared to the IP address family of the first entry in the - list of access control rules. If the source IP address for an incoming - packet is not of the same address as the first ACL entry, that packet - bypasses all ACL rules. - - * AST-2014-018: Permission Escalation through DB dialplan function - - The DB dialplan function when executed from an external protocol, such as AMI, - could result in a privilege escalation. Users with a lower class authorization - in AMI can access the internal Asterisk database without the required SYSTEM - class authorization. - - In addition, the release of 11.6-cert8 and 11.14.1 resolves the following - security vulnerability: - - * AST-2014-014: High call load with ConfBridge can result in resource exhaustion - - The ConfBridge application uses an internal bridging API to implement - conference bridges. This internal API uses a state model for channels within - the conference bridge and transitions between states as different things - occur. Unload load it is possible for some state transitions to be delayed - causing the channel to transition from being hung up to waiting for media. As - the channel has been hung up remotely no further media will arrive and the - channel will stay within ConfBridge indefinitely. - - In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves - the following security vulnerability: - - * AST-2014-017: Permission Escalation via ConfBridge dialplan function and - AMI ConfbridgeStartRecord Action - - The CONFBRIDGE dialplan function when executed from an external protocol (such - as AMI) can result in a privilege escalation as certain options within that - function can affect the underlying system. Additionally, the AMI - ConfbridgeStartRecord action has options that would allow modification of the - underlying system, and does not require SYSTEM class authorization in AMI. - - Finally, the release of 12.7.1 and 13.0.1 resolves the following security - vulnerabilities: - - * AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack - - The Asterisk module res_pjsip provides the ability to configure ACLs that may - be used to reject SIP requests from various hosts. However, the module - currently fails to create and apply the ACLs defined in its configuration - file on initial module load. - - * AST-2014-015: Remote crash vulnerability in PJSIP channel driver - - The chan_pjsip channel driver uses a queue approach for relating to SIP - sessions. There exists a race condition where actions may be queued to answer - a session or send ringing after a SIP session has been terminated using a - CANCEL request. The code will incorrectly assume that the SIP session is still - active and attempt to send the SIP response. The PJSIP library does not - expect the SIP session to be in the disconnected state when sending the - response and asserts. - - * AST-2014-016: Remote crash vulnerability in PJSIP channel driver - - When handling an INVITE with Replaces message the res_pjsip_refer module - incorrectly assumes that it will be operating on a channel that has just been - created. If the INVITE with Replaces message is sent in-dialog after a session - has been established this assumption will be incorrect. The res_pjsip_refer - module will then hang up a channel that is actually owned by another thread. - When this other thread attempts to use the just hung up channel it will end up - using a freed channel which will likely result in a crash. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015, - AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the same - time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert3 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-013.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-015.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-016.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf- The Asterisk Development Team is pleased to announce the release of - Asterisk 13.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - Asterisk 13 is the next major release series of Asterisk. It is a Long Term - Support (LTS) release, similar to Asterisk 11. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 13, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13 - - A short list of new features includes: - - * Asterisk security events are now provided via AMI, allowing end users to - monitor their Asterisk system in real time for security related issues. - - * Both AMI and ARI now allow external systems to control the state of a mailbox. - Using AMI actions or ARI resources, external systems can programmatically - trigger Message Waiting Indicators (MWI) on subscribed phones. This is of - particular use to those who want to build their own VoiceMail application - using ARI. - - * ARI now supports the reception/transmission of out of call text messages using - any supported channel driver/protocol stack through ARI. Users receive out of - call text messages as JSON events over the ARI websocket connection, and can - send out of call text messages using HTTP requests. - - * The PJSIP stack now supports RFC 4662 Resource Lists, allowing Asterisk to act - as a Resource List Server. This includes defining lists of presence state, - mailbox state, or lists of presence state/mailbox state; managing - subscriptions to lists; and batched delivery of NOTIFY requests to - subscribers. - - * The PJSIP stack can now be used as a means of distributing device state or - mailbox state via PUBLISH requests to other Asterisk instances. This is - analogous to Asterisk's clustering support using XMPP or Corosync; unlike - existing clustering mechanisms, using the PJSIP stack to perform the - distribution of state does not rely on another daemon or server to perform the - work. - - And much more! - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Documentation - - A full list of all new features can also be found in the CHANGES file: - - http://svnview.digium.com/svn/asterisk/branches/13/CHANGES - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0- rebuild for new libsrtp- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available - security releases are released as versions 1.8.28-cert2, 11.6-cert7, 1.8.31.1, - 11.13.1, 12.6.1, and 13.0.0-beta3. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerability: - - * AST-2014-011: Asterisk Susceptibility to POODLE Vulnerability - - Asterisk is susceptible to the POODLE vulnerability in two ways: - 1) The res_jabber and res_xmpp module both use SSLv3 exclusively for their - encrypted connections. - 2) The core TLS handling in Asterisk, which is used by the chan_sip channel - driver, Asterisk Manager Interface (AMI), and Asterisk HTTP Server, by - default allow a TLS connection to fallback to SSLv3. This allows for a - MITM to potentially force a connection to fallback to SSLv3, exposing it - to the POODLE vulnerability. - - These issues have been resolved in the versions released in conjunction with - this security advisory. - - For more information about the details of this vulnerability, please read - security advisory AST-2014-011, which was released at the same time as this - announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert2 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert7 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.31.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.13.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.6.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0-beta3 - - The security advisory is available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-011.pdf- The Asterisk Development Team has announced the release of Asterisk 11.13.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.13.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-24032 - Gentoo compilation emits warning: - "_FORTIFY_SOURCE" redefined (Reported by Kilburn) - * ASTERISK-24225 - Dial option z is broken (Reported by - dimitripietro) - * ASTERISK-24178 - [patch]fromdomainport used even if not set - (Reported by Elazar Broad) - * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload - warnings and ref leaks (Reported by Walter Doekes) - * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP - ICE candidates in SDP answer (Reported by Badalian Vyacheslav) - * ASTERISK-24019 - When a Music On Hold stream starts it restarts - at beginning of file. (Reported by Jason Richards) - * ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying - if ever not able to resolve (Reported by David Herselman) - * ASTERISK-24211 - testsuite: Fix the dial_LS_options test - (Reported by Matt Jordan) - * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash - Mohod) - * ASTERISK-23577 - res_rtp_asterisk: Crash in - ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by - Jay Jideliov) - * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10) - concurrent WebRTC (avpg/encryption/icesupport) calls (Reported - by Roman Skvirsky) - * ASTERISK-24301 - Security: Out of call MESSAGE requests - processed via Message channel driver can crash Asterisk - (Reported by Matt Jordan) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-24171 - [patch] Provide a manpage for the aelparse - utility (Reported by Jeremy Lainé) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.13.0- The Asterisk Development Team has announced security releases for Certified - Asterisk 11.6 and Asterisk 11 and 12. The available security releases are - released as versions 11.6-cert6, 11.12.1, and 12.5.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - Please note that the release of these versions resolves the following security - vulnerability: - - * AST-2014-010: Remote Crash when Handling Out of Call Message in Certain - Dialplan Configurations - - Additionally, the release of Asterisk 12.5.1 resolves the following security - vulnerability: - - * AST-2014-009: Remote Crash Based on Malformed SIP Subscription Requests - - Note that the crash described in AST-2014-010 can be worked around through - dialplan configuration. Given the likelihood of the issue, an advisory was - deemed to be warranted. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-009 and AST-2014-010, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert6 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.12.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.5.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-009.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-010.pdf- The Asterisk Development Team has announced the release of Asterisk 11.12.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.12.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an - empty string is a bit over zealous (Reported by Matt Jordan) - * ASTERISK-23985 - PresenceState Action response does not contain - ActionID; duplicates Message Header (Reported by Matt Jordan) - * ASTERISK-23814 - No call started after peer dialed (Reported by - Igor Goncharovsky) - * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy - should not call sip_destroy (Reported by Corey Farrell) - * ASTERISK-23818 - PBX_Lua: after asterisk startup module is - loaded, but dialplan not available (Reported by Dennis Guse) - * ASTERISK-18345 - [patch] sips connection dropped by asterisk - with a large INVITE (Reported by Stephane Chazelas) - * ASTERISK-23508 - Memory Corruption in - __ast_string_field_ptr_build_va (Reported by Arnd Schmitter) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-21178 - Improve documentation for manager command - Getvar, Setvar (Reported by Rusty Newton) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.12.0- The Asterisk Development Team has announced the release of Asterisk 11.11.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.11.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-22551 - Session timer : UAS (Asterisk) starts counting - at Invite, UAC starts counting at 200 OK. (Reported by i2045) - * ASTERISK-23792 - Mutex left locked in chan_unistim.c (Reported - by Peter Whisker) - * ASTERISK-23582 - [patch]Inconsistent column length in *odbc - (Reported by Walter Doekes) - * ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all - categories but the requested one (Reported by zvision) - * ASTERISK-23035 - ConfBridge with name longer than max (32 chars) - results in several bridges with same conf_name (Reported by - Iñaki Cívico) - * ASTERISK-23824 - ConfBridge: Users cannot be muted via CLI or - AMI when waiting to enter a conference (Reported by Matt Jordan) - * ASTERISK-23683 - #includes - wildcard character in a path more - than one directory deep - results in no config parsing on module - reload (Reported by tootai) - * ASTERISK-23827 - autoservice thread doesn't exit at shutdown - (Reported by Corey Farrell) - * ASTERISK-23609 - Security: AMI action MixMonitor allows - arbitrary programs to be run (Reported by Corey Farrell) - * ASTERISK-23673 - Security: DOS by consuming the number of - allowed HTTP connections. (Reported by Richard Mudgett) - * ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite - a DEBUG level of zero (Reported by Rusty Newton) - * ASTERISK-23766 - [patch] Specify timeout for database write in - SQLite (Reported by Igor Goncharovsky) - * ASTERISK-23844 - Load of pbx_lua fails on sample extensions.lua - with Lua 5.2 or greater due to addition of goto statement - (Reported by Rusty Newton) - * ASTERISK-23818 - PBX_Lua: after asterisk startup module is - loaded, but dialplan not available (Reported by Dennis Guse) - * ASTERISK-23834 - res_rtp_asterisk debug message gives wrong - length if ICE (Reported by Richard Kenner) - * ASTERISK-23790 - [patch] - SIP From headers longer than 256 - characters result in dropped call and 'No closing bracket' - warnings. (Reported by uniken1) - * ASTERISK-23917 - res_http_websocket: Delay in client processing - large streams of data causes disconnect and stuck socket - (Reported by Matt Jordan) - * ASTERISK-23908 - [patch]When using FEC error correction, - asterisk tries considers negative sequence numbers as missing - (Reported by Torrey Searle) - * ASTERISK-23921 - refcounter.py uses excessive ram for large refs - files (Reported by Corey Farrell) - * ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against - objects that were already freed (Reported by Corey Farrell) - * ASTERISK-23916 - [patch]SIP/SDP fmtp line may include whitespace - between attributes (Reported by Alexander Traud) - * ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite() - (Reported by Steve Davies) - * ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking - PI) in revision 413765 breaks working environments (Reported by - Pavel Troller) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-23492 - Add option to safe_asterisk to disable - backgrounding (Reported by Walter Doekes) - * ASTERISK-22961 - [patch] DTLS-SRTP not working with SHA-256 - (Reported by Jay Jideliov) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.11.0- Perl 5.20 rebuild- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild- Drop the 389 directory server schema (1061414)- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security - releases are released as versions 1.8.15-cert7, 11.6-cert4, 1.8.28.2, 11.10.2, - and 12.3.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - These releases resolve security vulnerabilities that were previously fixed in - 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1, and 12.3.1. Unfortunately, the fix - for AST-2014-007 inadvertently introduced a regression in Asterisk's TCP and TLS - handling that prevented Asterisk from sending data over these transports. This - regression and the security vulnerabilities have been fixed in the versions - specified in this release announcement. - - The security patches for AST-2014-007 have been updated with the fix for the - regression, and are available at http://downloads.asterisk.org/pub/security - - Please note that the release of these versions resolves the following security - vulnerabilities: - - * AST-2014-005: Remote Crash in PJSIP Channel Driver's Publish/Subscribe - Framework - - * AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized - Shell Access - - * AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP - Connections - - * AST-2014-008: Denial of Service in PJSIP Channel Driver Subscriptions - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-005, AST-2014-006, AST-2014-007, and AST-2014-008, - which were released with the previous versions that addressed these - vulnerabilities. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert7 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.2 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert4 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.3.2 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-005.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-006.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-007.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-008.pdf- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security - releases are released as versions 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1, - and 12.3.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following issue: - - * AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP - Connections - - Establishing a TCP or TLS connection to the configured HTTP or HTTPS port - respectively in http.conf and then not sending or completing a HTTP request - will tie up a HTTP session. By doing this repeatedly until the maximum number - of open HTTP sessions is reached, legitimate requests are blocked. - - Additionally, the release of 11.6-cert3, 11.10.1, and 12.3.1 resolves the - following issue: - - * AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized - Shell Access - - Manager users can execute arbitrary shell commands with the MixMonitor manager - action. Asterisk does not require system class authorization for a manager - user to use the MixMonitor action, so any manager user who is permitted to use - manager commands can potentially execute shell commands as the user executing - the Asterisk process. - - Additionally, the release of 12.3.1 resolves the following issues: - - * AST-2014-005: Remote Crash in PJSIP Channel Driver's Publish/Subscribe - Framework - - A remotely exploitable crash vulnerability exists in the PJSIP channel - driver's pub/sub framework. If an attempt is made to unsubscribe when not - currently subscribed and the endpoint's “sub_min_expiry” is set to zero, - Asterisk tries to create an expiration timer with zero seconds, which is not - allowed, so an assertion raised. - - * AST-2014-008: Denial of Service in PJSIP Channel Driver Subscriptions - - When a SIP transaction timeout caused a subscription to be terminated, the - action taken by Asterisk was guaranteed to deadlock the thread on which SIP - requests are serviced. Note that this behavior could only happen on - established subscriptions, meaning that this could only be exploited if an - attacker bypassed authentication and successfully subscribed to a real - resource on the Asterisk server. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-005, AST-2014-006, AST-2014-007, and AST-2014-008, - which were released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert6 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.1 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert3 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.3.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-005.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-006.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-007.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-008.pdf- The Asterisk Development Team has announced the release of Asterisk 11.10.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.10.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-23547 - [patch] app_queue removing callers from queue - when reloading (Reported by Italo Rossi) - * ASTERISK-23559 - app_voicemail fails to load after fix to - dialplan functions (Reported by Corey Farrell) - * ASTERISK-22846 - testsuite: masquerade super test fails on all - branches (still) (Reported by Matt Jordan) - * ASTERISK-23545 - Confbridge talker detection settings - configuration load bug (Reported by John Knott) - * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think - (Reported by Walter Doekes) - * ASTERISK-23620 - Code path in app_stack fails to unlock list - (Reported by Bradley Watkins) - * ASTERISK-23616 - Big memory leak in logger.c (Reported by - ibercom) - * ASTERISK-23576 - Build failure on SmartOS / Illumos / SunOS - (Reported by Sebastian Wiedenroth) - * ASTERISK-23550 - Newer sound sets don't show up in menuselect - (Reported by Rusty Newton) - * ASTERISK-18331 - app_sms failure (Reported by David Woodhouse) - * ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by - Krzysztof Chmielewski) - * ASTERISK-23605 - res_http_websocket: Race condition in shutting - down websocket causes crash (Reported by Matt Jordan) - * ASTERISK-23707 - Realtime Contacts: Apparent mismatch between - PGSQL database state and Asterisk state (Reported by Mark - Michelson) - * ASTERISK-23381 - [patch]ChanSpy- Barge only works on the initial - 'spy', if the spied-on channel makes a new call, unable to - barge. (Reported by Robert Moss) - * ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+) - (Reported by Guillaume Maudoux) - * ASTERISK-23664 - Incorrect H264 specification in SDP. (Reported - by Guillaume Maudoux) - * ASTERISK-22977 - chan_sip+CEL: missing ANSWER and PICKUP event - for INVITE/w/replaces pickup (Reported by Walter Doekes) - * ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone - (Reported by Steve Davies) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-23649 - [patch]Support for DTLS retransmission - (Reported by NITESH BANSAL) - * ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently - available in a CLI command (Reported by Patrick Laimbock) - * ASTERISK-23754 - [patch] Use var/lib directory for log file - configured in asterisk.conf (Reported by Igor Goncharovsky) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.10.0- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild- build against gmime-devel not gmime22-devel - do not use -m64 on aarch64- The Asterisk Development Team has announced the release of Asterisk 11.9.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.9.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-22790 - check_modem_rate() may return incorrect rate - for V.27 (Reported by Paolo Compagnini) - * ASTERISK-23034 - [patch] manager Originate doesn't abort on - failed format_cap allocation (Reported by Corey Farrell) - * ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in - sip.conf.sample (Reported by Eugene) - * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted - minus signs (Reported by Jeremy Lainé) - * ASTERISK-23046 - Custom CDR fields set during a GoSUB called - from app_queue are not inserted (Reported by Denis Pantsyrev) - * ASTERISK-23027 - [patch] Spelling typo "transfered" instead of - "transferred" (Reported by Jeremy Lainé) - * ASTERISK-23008 - Local channels loose CALLERID name when DAHDI - channel connects (Reported by Michael Cargile) - * ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted - request and request queue may differ - fix for locking (Reported - by adomjan) - * ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image - media offer due to invalid or unsupported syntax (Reported by - adomjan) - * ASTERISK-22861 - [patch]Specifying a null time as parameter to - GotoIfTime or ExecIfTime causes segmentation fault (Reported by - Sebastian Murray-Roberts) - * ASTERISK-17837 - extconfig.conf - Maximum Include level (1) - exceeded (Reported by pz) - * ASTERISK-22662 - Documentation fix? - queues.conf says - persistentmembers defaults to yes, it appears to lie (Reported - by Rusty Newton) - * ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot - handle selinux port restrictions (Reported by Corey Farrell) - * ASTERISK-23220 - STACK_PEEK function with no arguments causes - crash/core dump (Reported by James Sharp) - * ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload' - command multiple times on cli_aliases (Reported by Joel Vandal) - * ASTERISK-22757 - segfault in res_clialiases.so on reload when - mapping "module reload" command (Reported by Gareth Blades) - * ASTERISK-17727 - [patch] TLS doesn't get all certificate chain - (Reported by LN) - * ASTERISK-23178 - devicestate.h: device state setting functions - are documented with the wrong return values (Reported by - Jonathan Rose) - * ASTERISK-23232 - LocalBridge AMI Event LocalOptimization value - is opposite to what's expected (Reported by Leon Roy) - * ASTERISK-23098 - [patch]possible null pointer dereference in - format.c (Reported by Marcello Ceschia) - * ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if - res_parking.so is not loaded, or if res_parking.conf has no - configuration (Reported by CJ Oster) - * ASTERISK-23069 - Custom CDR variable not recorded when set in - macro called from app_queue (Reported by Bryan Anderson) - * ASTERISK-19499 - ConfBridge MOH is not working for transferee - after attended transfer (Reported by Timo Teräs) - * ASTERISK-23261 - [patch]Output mixup in - ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686) - * ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic - payload change in rtp mapping in the 200 OK response (Reported - by NITESH BANSAL) - * ASTERISK-23255 - UUID included for Redhat, but missing for - Debian distros in install_prereq script (Reported by Rusty - Newton) - * ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR - variables for subsequent records (Reported by zvision) - * ASTERISK-23141 - Asterisk crashes on Dial(), in - pbx_find_extension at pbx.c (Reported by Maxim) - * ASTERISK-23336 - Asterisk warning "Don't know how to indicate - condition 33 on ooh323c" on outgoing calls from H323 to SIP peer - (Reported by Alexander Semych) - * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set - to minrate=2400, then res_fax refuse to load (Reported by David - Brillert) - * ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set - - probably introduced in 11.7.0 (Reported by OK) - * ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in - handle_response_invite (Reported by Walter Doekes) - * ASTERISK-23406 - [patch]Fix typo in "sip show peer" (Reported by - ibercom) - * ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write - (Reported by Jeremy Lainé) - * ASTERISK-22911 - [patch]Asterisk fails to resume WebRTC call - from hold (Reported by Vytis Valentinavičius) - * ASTERISK-23104 - Specifying the SetVar AMI without a Channel - cause Asterisk to crash (Reported by Joel Vandal) - * ASTERISK-21930 - [patch]WebRTC over WSS is not working. - (Reported by John) - * ASTERISK-23383 - Wrong sense test on stat return code causes - unchanged config check to break with include files. (Reported by - David Woolley) - * ASTERISK-20149 - Crash when faxing SIP to SIP with strictrtp set - to yes (Reported by Alexandr Gordeev) - * ASTERISK-17523 - Qualify for static realtime peers does not work - (Reported by Maciej Krajewski) - * ASTERISK-21406 - [patch] chan_sip deadlock on monlock between - unload_module and do_monitor (Reported by Corey Farrell) - * ASTERISK-23373 - [patch]Security: Open FD exhaustion with - chan_sip Session-Timers (Reported by Corey Farrell) - * ASTERISK-23340 - Security Vulnerability: stack allocation of - cookie headers in loop allows for unauthenticated remote denial - of service attack (Reported by Matt Jordan) - * ASTERISK-23311 - Manager - MoH Stop Event fails to show up when - leaving Conference (Reported by Benjamin Keith Ford) - * ASTERISK-23420 - [patch]Memory leak in manager_add_filter - function in manager.c (Reported by Etienne Lessard) - * ASTERISK-23488 - Logic error in callerid checksum processing - (Reported by Russ Meyerriecks) - * ASTERISK-23461 - Only first user is muted when joining - confbridge with 'startmuted=yes' (Reported by Chico Manobela) - * ASTERISK-20841 - fromdomain not honored on outbound INVITE - request (Reported by Kelly Goedert) - * ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f) - at astobj2.c:120 (Reported by Jamuel Starkey) - * ASTERISK-23509 - [patch]SayNumber for Polish language tries to - play empty files for numbers divisible by 100 (Reported by - zvision) - * ASTERISK-23103 - [patch]Crash in ast_format_cmp, in ao2_find - (Reported by JoshE) - * ASTERISK-23391 - Audit dialplan function usage of channel - variable (Reported by Corey Farrell) - * ASTERISK-23548 - POST to ARI sometimes returns no body on - success (Reported by Scott Griepentrog) - * ASTERISK-23460 - ooh323 channel stuck if call is placed directly - and gatekeeper is not available (Reported by Dmitry Melekhov) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius - against libfreeradius-client (Reported by Jeremy Lainé) - * ASTERISK-22661 - Unable to exit ChanSpy if spied channel does - not have a call in progress (Reported by Chris Hillman) - * ASTERISK-23099 - [patch] WSS: enable ast_websocket_read() - function to read the whole available data at first and then wait - for any fragmented packets (Reported by Thava Iyer)- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security - releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1, - and 12.1.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following issues: - - * AST-2014-001: Stack overflow in HTTP processing of Cookie headers. - - Sending a HTTP request that is handled by Asterisk with a large number of - Cookie headers could overflow the stack. - - Another vulnerability along similar lines is any HTTP request with a - ridiculous number of headers in the request could exhaust system memory. - - * AST-2014-002: chan_sip: Exit early on bad session timers request - - This change allows chan_sip to avoid creation of the channel and - consumption of associated file descriptors altogether if the inbound - request is going to be rejected anyway. - - Additionally, the release of 12.1.1 resolves the following issue: - - * AST-2014-003: res_pjsip: When handling 401/407 responses don't assume a - request will have an endpoint. - - This change removes the assumption that an outgoing request will always - have an endpoint and makes the authenticate_qualify option work once again. - - Finally, a security advisory, AST-2014-004, was released for a vulnerability - fixed in Asterisk 12.1.0. Users of Asterisk 12.0.0 are encouraged to upgrade to - 12.1.1 to resolve both vulnerabilities. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-001, AST-2014-002, AST-2014-003, and AST-2014-004, - which were released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert5 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.26.1 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.8.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.1.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-001.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-002.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-003.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-004.pdf- The Asterisk Development Team has announced the release of Asterisk 11.8.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.8.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-22544 - Italian prompt vm-options has advertisement in - it (Reported by Rusty Newton) - * ASTERISK-21383 - STUN Binding Requests Not Being Sent Back from - Asterisk to Chrome (Reported by Shaun Clark) - * ASTERISK-22478 - [patch]Can't use pound(hash) symbol for custom - DTMF menus in ConfBridge (processed as directive) (Reported by - Nicolas Tanski) - * ASTERISK-12117 - chan_sip creates a new local tag (from-tag) for - every register message (Reported by Pawel Pierscionek) - * ASTERISK-20862 - Asterisk min and max member penalties not - honored when set with 0 (Reported by Schmooze Com) - * ASTERISK-22746 - [patch]Crash in chan_dahdi during caller id - read (Reported by Michael Walton) - * ASTERISK-22788 - [patch] main/translate.c: access to variable f - after free in ast_translate() (Reported by Corey Farrell) - * ASTERISK-21242 - Segfault when T.38 re-invite retransmission - receives 200 OK (Reported by Ashley Winters) - * ASTERISK-22590 - BufferOverflow in unpacksms16() when receiving - 16 bit multipart SMS with app_sms (Reported by Jan Juergens) - * ASTERISK-22905 - Prevent Asterisk functions that are 'dangerous' - from being executed from external interfaces (Reported by Matt - Jordan) - * ASTERISK-23021 - Typos in code : "avaliable" instead of - "available" (Reported by Jeremy Lainé) - * ASTERISK-22970 - [patch]Documentation fix for QUOTE() (Reported - by Gareth Palmer) - * ASTERISK-21960 - ooh323 channels stuck (Reported by Dmitry - Melekhov) - * ASTERISK-22350 - DUNDI - core dump on shutdown - segfault in - sqlite3_reset from /usr/lib/libsqlite3.so.0 (Reported by Birger - "WIMPy" Harzenetter) - * ASTERISK-22942 - [patch] - Asterisk crashed after - Set(FAXOPT(faxdetect)=t38) (Reported by adomjan) - * ASTERISK-22856 - [patch]SayUnixTime in polish reads minutes - instead of seconds (Reported by Robert Mordec) - * ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and - core_event_dispatcher taskprocessor thread (Reported by Etienne - Lessard) - * ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping - memory when is empty (Reported by Gareth Palmer) - * ASTERISK-22871 - cel_pgsql module not loading after "reload" or - "reload cel_pgsql.so" command (Reported by Matteo) - * ASTERISK-23084 - [patch]rasterisk needlessly prints the - AST-2013-007 warning (Reported by Tzafrir Cohen) - * ASTERISK-17138 - [patch] Asterisk not re-registering after it - receives "Forbidden - wrong password on authentication" - (Reported by Rudi) - * ASTERISK-23011 - [patch]configure.ac and pbx_lua don't support - lua 5.2 (Reported by George Joseph) - * ASTERISK-22834 - Parking by blind transfer when lot full orphans - channels (Reported by rsw686) - * ASTERISK-23047 - Orphaned (stuck) channel occurs during a failed - SIP transfer to parking space (Reported by Tommy Thompson) - * ASTERISK-22946 - Local From tag regression with sipgate.de - (Reported by Stephan Eisvogel) - * ASTERISK-23010 - No BYE message sent when sip INVITE is received - (Reported by Ryan Tilton) - * ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set - - probably introduced in 11.7.0 (Reported by OK) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-22728 - [patch] Improve Understanding Of 'Forcerport' - When Running "sip show peers" (Reported by Michael L. Young) - * ASTERISK-22659 - Make a new core and extra sounds release - (Reported by Rusty Newton) - * ASTERISK-22919 - core show channeltypes slicing (Reported by - outtolunc) - * ASTERISK-22918 - dahdi show channels slices PRI channel dnid on - output (Reported by outtolunc) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0- The Asterisk Development Team has announced the release of Asterisk 11.7.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.7.0 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: - - * --- app_confbridge: Can now set the language used for announcements - to the conference. - (Closes issue ASTERISK-19983. Reported by Jonathan White) - - * --- app_queue: Fix CLI "queue remove member" queue_log entry. - (Closes issue ASTERISK-21826. Reported by Oscar Esteve) - - * --- chan_sip: Do not increment the SDP version between 183 and 200 - responses. - (Closes issue ASTERISK-21204. Reported by NITESH BANSAL) - - * --- chan_sip: Allow a sip peer to accept both AVP and AVPF calls - (Closes issue ASTERISK-22005. Reported by Torrey Searle) - - * --- chan_sip: Fix Realtime Peer Update Problem When Un-registering - And Expires Header In 200ok - (Closes issue ASTERISK-22428. Reported by Ben Smithurst) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.7.0- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security - releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, - 10.12.4-digiumphones, and 11.6.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following issues: - - * A buffer overflow when receiving odd length 16 bit messages in app_sms. An - infinite loop could occur which would overwrite memory when a message is - received into the unpacksms16() function and the length of the message is an - odd number of bytes. - - * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk - now marks certain individual dialplan functions as 'dangerous', which will - inhibit their execution from external sources. - - A 'dangerous' function is one which results in a privilege escalation. For - example, if one were to read the channel variable SHELL(rm -rf /) Bad - Things(TM) could happen; even if the external source has only read - permissions. - - Execution from external sources may be enabled by setting 'live_dangerously' - to 'yes' in the [options] section of asterisk.conf. Although doing so is not - recommended. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2013-006 and AST-2013-007, which were - released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert4 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.2-cert3 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.24.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.4 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.4-digiumphones - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf - * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf- The Asterisk Development Team has announced the release of Asterisk 11.6.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.6.0 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: - - * --- Confbridge: empty conference not being torn down - (Closes issue ASTERISK-21859. Reported by Chris Gentle) - - * --- Let Queue wrap up time influence member availability - (Closes issue ASTERISK-22189. Reported by Tony Lewis) - - * --- Fix a longstanding issue with MFC-R2 configuration that - prevented users - (Closes issue ASTERISK-21117. Reported by Rafael Angulo) - - * --- chan_iax2: Fix saving the wrong expiry time in astdb. - (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) - - * --- Fix segfault for certain invalid WebSocket input. - (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0- Disable hardened build, as it's apparently causing problems loading modules.- Enable hardened build BZ#954338 - Significant clean ups- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases - are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, - and 11.5.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following issues: - - * A remotely exploitable crash vulnerability exists in the SIP channel driver if - an ACK with SDP is received after the channel has been terminated. The - handling code incorrectly assumes that the channel will always be present. - - * A remotely exploitable crash vulnerability exists in the SIP channel driver if - an invalid SDP is sent in a SIP request that defines media descriptions before - connection information. The handling code incorrectly attempts to reference - the socket address information even though that information has not yet been - set. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2013-004 and AST-2013-005, which were - released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert3 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.2-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.23.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.3 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.3-digiumphones - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.5.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2013-004.pdf - * http://downloads.asterisk.org/pub/security/AST-2013-005.pdf - - The Asterisk Development Team has announced the release of Asterisk 11.5.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.5.0 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 Segfault In app_queue When "persistentmembers" Is Enabled - And Using Realtime - (Closes issue ASTERISK-21738. Reported by JoshE) - - * --- IAX2: fix race condition with nativebridge transfers. - (Closes issue ASTERISK-21409. Reported by alecdavis) - - * --- Fix The Payload Being Set On CN Packets And Do Not Set Marker - Bit - (Closes issue ASTERISK-21246. Reported by Peter Katzmann) - - * --- Fix One-Way Audio With auto_* NAT Settings When SIP Calls - Initiated By PBX - (Closes issue ASTERISK-21374. Reported by Michael L. Young) - - * --- chan_sip: NOTIFYs for BLF start queuing up and fail to be sent - out after retries fail - (Closes issue ASTERISK-21677. Reported by Dan Martens) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.5.0- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild- Perl 5.18 rebuild- rebuild (libical)- The Asterisk Development Team has announced the release of Asterisk 11.4.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.4.0 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 Sorting Order For Parking Lots Stored In Static Realtime - (Closes issue ASTERISK-21035. Reported by Alex Epshteyn) - - * --- Fix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On - A Channel - (Closes issue ASTERISK-21294. Reported by daroz) - - * --- When a session timer expires during a T.38 call, re-invite with - correct SDP - (Closes issue ASTERISK-21232. Reported by Nitesh Bansal) - - * --- Fix white noise on SRTP decryption - (Closes issue ASTERISK-21323. Reported by andrea) - - * --- Fix reload skinny with active devices. - (Closes issue ASTERISK-16610. Reported by wedhorn) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.4.0- fix build with lua 5.2- The Asterisk Development Team has announced the release of Asterisk 11.3.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.3.0 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 issue where chan_mobile fails to bind to first available - port - (Closes issue ASTERISK-16357. Reported by challado) - - * --- Fix Queue Log Reporting Every Call COMPLETECALLER With "h" - Extension Present - (Closes issue ASTERISK-20743. Reported by call) - - * --- Retain XMPP filters across reconnections so external modules - continue to function as expected. - (Closes issue ASTERISK-20916. Reported by kuj) - - * --- Ensure that a declined media stream is terminated with a '\r\n' - (Closes issue ASTERISK-20908. Reported by Dennis DeDonatis) - - * --- Fix pjproject compilation in certain circumstances - (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.3.0- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases - are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones, - and 11.2.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following issues: - - * A possible buffer overflow during H.264 format negotiation. The format - attribute resource for H.264 video performs an unsafe read against a media - attribute when parsing the SDP. - - This vulnerability only affected Asterisk 11. - - * A denial of service exists in Asterisk's HTTP server. AST-2012-014, fixed - in January of this year, contained a fix for Asterisk's HTTP server for a - remotely-triggered crash. While the fix prevented the crash from being - triggered, a denial of service vector still exists with that solution if an - attacker sends one or more HTTP POST requests with very large Content-Length - values. - - This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11 - - * A potential username disclosure exists in the SIP channel driver. When - authenticating a SIP request with alwaysauthreject enabled, allowguest - disabled, and autocreatepeer disabled, Asterisk discloses whether a user - exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. - - This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11 - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2013-001, AST-2013-002, and AST-2013-003, which were - released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.20.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.2-digiumphones - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.2.2 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2013-001.pdf - * http://downloads.asterisk.org/pub/security/AST-2013-002.pdf - * http://downloads.asterisk.org/pub/security/AST-2013-003.pdf- The Asterisk Development Team has announced the release of Asterisk 11.2.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.2.1 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - * --- Fix astcanary startup problem due to wrong pid value from before - daemon call - (Closes issue ASTERISK-20947. Reported by Jakob Hirsch) - - * --- Update init.d scripts to handle stderr; readd splash screen for - remote consoles - (Closes issue ASTERISK-20945. Reported by Warren Selby) - - * --- Reset RTP timestamp; sequence number on SSRC change - (Closes issue ASTERISK-20906. Reported by Eelco Brolman) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.1- The Asterisk Development Team has announced the release of Asterisk 11.2.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.2.0 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: - - * --- app_meetme: Fix channels lingering when hung up under certain - conditions - (Closes issue ASTERISK-20486. Reported by Michael Cargile) - - * --- Fix stuck DTMF when bridge is broken. - (Closes issue ASTERISK-20492. Reported by Jeremiah Gowdy) - - * --- Add missing support for "who hung up" to chan_motif. - (Closes issue ASTERISK-20671. Reported by Matt Jordan) - - * --- Remove a fixed size limitation for producing SDP and change how - ICE support is disabled by default. - (Closes issue ASTERISK-20643. Reported by coopvr) - - * --- Fix chan_sip websocket payload handling - (Closes issue ASTERISK-20745. Reported by Iñaki Baz Castillo) - - * --- Fix pjproject compilation in certain circumstances - (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.0- The Asterisk Development Team has announced a security release for Asterisk 11, - Asterisk 11.1.2. This release addresses the security vulnerabilities reported in - AST-2012-014 and AST-2012-015, and replaces the previous version of Asterisk 11 - released for these security vulnerabilities. The prior release left open a - vulnerability in res_xmpp that exists only in Asterisk 11; as such, other - versions of Asterisk were resolved correctly by the previous releases. - - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following two issues: - - * Stack overflows that occur in some portions of Asterisk that manage a TCP - connection. In SIP, this is exploitable via a remote unauthenticated session; - in XMPP and HTTP connections, this is exploitable via remote authenticated - sessions. The vulnerabilities in SIP and HTTP were corrected in a prior - release of Asterisk; the vulnerability in XMPP is resolved in this release. - - * A denial of service vulnerability through exploitation of the device state - cache. Anonymous calls had the capability to create devices in Asterisk that - would never be disposed of. Handling the cachability of device states - aggregated via XMPP is handled in this release. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-014 and AST-2012-015. - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.2 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-014.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-015.pdf - - Thank you for your continued support of Asterisk - and we apologize for having - to do this twice!- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.11 and Asterisk 1.8, 10, and 11. The available security releases - are released as versions 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones, - and 11.1.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following two issues: - - * Stack overflows that occur in some portions of Asterisk that manage a TCP - connection. In SIP, this is exploitable via a remote unauthenticated session; - in XMPP and HTTP connections, this is exploitable via remote authenticated - sessions. - - * A denial of service vulnerability through exploitation of the device state - cache. Anonymous calls had the capability to create devices in Asterisk that - would never be disposed of. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-014 and AST-2012-015, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert10 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.19.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1-digiumphones - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-014.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-015.pdf- The Asterisk Development Team has announced the release of Asterisk 11.1.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.1.0 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 execution of 'i' extension due to uninitialized variable. - (Closes issue ASTERISK-20455. Reported by Richard Miller) - - * --- Prevent resetting of NATted realtime peer address on reload. - (Closes issue ASTERISK-18203. Reported by daren ferreira) - - * --- Fix ConfBridge crash if no timing module loaded. - (Closes issue ASTERISK-19448. Reported by feyfre) - - * --- Fix the Park 'r' option when a channel parks itself. - (Closes issue ASTERISK-19382. Reported by James Stocks) - - * --- Fix an issue where outgoing calls would fail to establish audio - due to ICE negotiation failures. - (Closes issue ASTERISK-20554. Reported by mmichelson) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.1.0- The Asterisk Development Team has announced the release of Asterisk 11.0.2. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.0.2 resolves an issue reported by the - community and would have not been possible without your participation. - Thank you! - - The following is the issue resolved in this release: - - * --- chan_local: Fix local_pvt ref leak in local_devicestate(). - (Closes issue ASTERISK-20769. Reported by rmudgett) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.2- simplify LDFLAGS setting- clean up things to allow building on arm arches- The Asterisk Development Team has announced the release of Asterisk 11.0.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.0.1 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - * --- chan_sip: Fix a bug causing SIP reloads to remove all entries - from the registry - (Closes issue ASTERISK-20611. Reported by Alisher) - - * --- confbridge: Fix a bug which made conferences not record with - AMI/CLI commands - (Closes issue ASTERISK-20601. Reported by Vilius) - - * --- Fix an issue with res_http_websocket where the chan_sip - WebSocket handler could not be registered. - (Closes issue ASTERISK-20631. Reported by danjenkins) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.1- The Asterisk Development Team is pleased to announce the release of - Asterisk 11.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - Asterisk 11 is the next major release series of Asterisk. It is a Long Term - Support (LTS) release, similar to Asterisk 1.8. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 11, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 - - A short list of new features includes: - - * A new channel driver named chan_motif has been added which provides support - for Google Talk and Jingle in a single channel driver. This new channel - driver includes support for both audio and video, RFC2833 DTMF, all codecs - supported by Asterisk, hold, unhold, and ringing notification. It is also - compliant with the current Jingle specification, current Google Jingle - specification, and the original Google Talk protocol. - - * Support for the WebSocket transport for chan_sip. - - * SIP peers can now be configured to support negotiation of ICE candidates. - - * The app_page application now no longer depends on DAHDI or app_meetme. It - has been re-architected to use app_confbridge internally. - - * Hangup handlers can be attached to channels using the CHANNEL() function. - Hangup handlers will run when the channel is hung up similar to the h - extension; however, unlike an h extension, a hangup handler is associated with - the actual channel and will execute anytime that channel is hung up, - regardless of where it is in the dialplan. - - * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial - allows you to execute a dialplan subroutine on a channel before a call is - placed but after the application performing a dial action is invoked. This - means that the handlers are executed after the creation of the callee - channels, but before any actions have been taken to actually dial the callee - channels. - - * Log messages can now be easily associated with a certain call by looking at - a new unique identifier, "Call Id". Call ids are attached to log messages for - just about any case where it can be determined that the message is related - to a particular call. - - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in - Asterisk. Unlike traditional ACLs defined in specific module configuration - files, Named ACLs can be shared across multiple modules. - - * The Hangup Cause family of functions and dialplan applications allow for - inspection of the hangup cause codes for each channel involved in a call. - This allows a dialplan writer to determine, for each channel, who hung up and - for what reason(s). - - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() - lets you set some of the configuration options from the general section - of features.conf on a per-channel basis. FEATUREMAP() lets you customize - the key sequence used to activate built-in features, such as blindxfer, - and automon. - - * Support for DTLS-SRTP in chan_sip. - - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups - and callgroups to be defined for several channel drivers. - - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation - - A full list of all new features can also be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES - - For a full list of changes in the current release, please see the ChangeLog. - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0- The Asterisk Development Team has announced the second release candidate of - Asterisk 11.0.0. This release candidate is available for immediate - download at http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.0.0-rc2 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release candidate: - - * --- Fix an issue where outgoing calls would fail to establish audio - due to ICE negotiation failures. - (Closes issue ASTERISK-20554. Reported by mmichelson) - - * --- Ensure Asterisk fails TCP/TLS SIP calls when certificate - checking fails - (Closes issue ASTERISK-20559. Reported by kmoore) - - * --- Don't make chan_sip export global symbols. - (Closes issue ASTERISK-20545. Reported by kmoore) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.0-rc2- The Asterisk Development Team is pleased to announce the first release candidate - of Asterisk 11.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - All interested users of Asterisk are encouraged to participate in the - Asterisk 11 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - All Asterisk users are invited to participate in the #asterisk-testing channel - on IRC to work together in testing the many parts of Asterisk. - - Asterisk 11 is the next major release series of Asterisk. It will be a Long - Term Support (LTS) release, similar to Asterisk 1.8. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 11, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 - - A short list of new features includes: - - * A new channel driver named chan_motif has been added which provides support - for Google Talk and Jingle in a single channel driver. This new channel - driver includes support for both audio and video, RFC2833 DTMF, all codecs - supported by Asterisk, hold, unhold, and ringing notification. It is also - compliant with the current Jingle specification, current Google Jingle - specification, and the original Google Talk protocol. - - * Support for the WebSocket transport for chan_sip. - - * SIP peers can now be configured to support negotiation of ICE candidates. - - * The app_page application now no longer depends on DAHDI or app_meetme. It - has been re-architected to use app_confbridge internally. - - * Hangup handlers can be attached to channels using the CHANNEL() function. - Hangup handlers will run when the channel is hung up similar to the h - extension; however, unlike an h extension, a hangup handler is associated with - the actual channel and will execute anytime that channel is hung up, - regardless of where it is in the dialplan. - - * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial - allows you to execute a dialplan subroutine on a channel before a call is - placed but after the application performing a dial action is invoked. This - means that the handlers are executed after the creation of the callee - channels, but before any actions have been taken to actually dial the callee - channels. - - * Log messages can now be easily associated with a certain call by looking at - a new unique identifier, "Call Id". Call ids are attached to log messages for - just about any case where it can be determined that the message is related - to a particular call. - - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in - Asterisk. Unlike traditional ACLs defined in specific module configuration - files, Named ACLs can be shared across multiple modules. - - * The Hangup Cause family of functions and dialplan applications allow for - inspection of the hangup cause codes for each channel involved in a call. - This allows a dialplan writer to determine, for each channel, who hung up and - for what reason(s). - - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() - lets you set some of the configuration options from the general section - of features.conf on a per-channel basis. FEATUREMAP() lets you customize - the key sequence used to activate built-in features, such as blindxfer, - and automon. - - * Support for DTLS-SRTP in chan_sip. - - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups - and callgroups to be defined for several channel drivers. - - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation - - A full list of all new features can also be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES - - For a full list of changes in the current release, please see the ChangeLog. - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-rc1- Don't forget format_ilbc module- The Asterisk Development Team is pleased to announce the second beta release of - Asterisk 11.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - All interested users of Asterisk are encouraged to participate in the - Asterisk 11 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - All Asterisk users are invited to participate in the #asterisk-testing channel - on IRC to work together in testing the many parts of Asterisk. - - Asterisk 11 is the next major release series of Asterisk. It will be a Long - Term Support (LTS) release, similar to Asterisk 1.8. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 11, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 - - A short list of new features includes: - - * A new channel driver named chan_motif has been added which provides support - for Google Talk and Jingle in a single channel driver. This new channel - driver includes support for both audio and video, RFC2833 DTMF, all codecs - supported by Asterisk, hold, unhold, and ringing notification. It is also - compliant with the current Jingle specification, current Google Jingle - specification, and the original Google Talk protocol. - - * Support for the WebSocket transport for chan_sip. - - * SIP peers can now be configured to support negotiation of ICE candidates. - - * The app_page application now no longer depends on DAHDI or app_meetme. It - has been re-architected to use app_confbridge internally. - - * Hangup handlers can be attached to channels using the CHANNEL() function. - Hangup handlers will run when the channel is hung up similar to the h - extension; however, unlike an h extension, a hangup handler is associated with - the actual channel and will execute anytime that channel is hung up, - regardless of where it is in the dialplan. - - * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial - allows you to execute a dialplan subroutine on a channel before a call is - placed but after the application performing a dial action is invoked. This - means that the handlers are executed after the creation of the callee - channels, but before any actions have been taken to actually dial the callee - channels. - - * Log messages can now be easily associated with a certain call by looking at - a new unique identifier, "Call Id". Call ids are attached to log messages for - just about any case where it can be determined that the message is related - to a particular call. - - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in - Asterisk. Unlike traditional ACLs defined in specific module configuration - files, Named ACLs can be shared across multiple modules. - - * The Hangup Cause family of functions and dialplan applications allow for - inspection of the hangup cause codes for each channel involved in a call. - This allows a dialplan writer to determine, for each channel, who hung up and - for what reason(s). - - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() - lets you set some of the configuration options from the general section - of features.conf on a per-channel basis. FEATUREMAP() lets you customize - the key sequence used to activate built-in features, such as blindxfer, - and automon. - - * Support for DTLS-SRTP in chan_sip. - - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups - and callgroups to be defined for several channel drivers. - - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation - - A full list of all new features can also be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES - - For a full list of changes in the current release, please see the ChangeLog. - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta2- The Asterisk Development Team has announced the release of Asterisk 10.8.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.8.0 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: - - * --- AST-2012-012: Resolve AMI User Unauthorized Shell Access through - ExternalIVR - (Closes issue ASTERISK-20132. Reported by Zubair Ashraf of IBM X-Force Research) - - * --- AST-2012-013: Resolve ACL rules being ignored during calls by - some IAX2 peers - (Closes issue ASTERISK-20186. Reported by Alan Frisch) - - * --- Handle extremely out of order RFC 2833 DTMF - (Closes issue ASTERISK-18404. Reported by Stephane Chazelas) - - * --- Resolve severe memory leak in CEL logging modules. - (Closes issue AST-916. Reported by Thomas Arimont) - - * --- Only re-create an SRTP session when needed - (Issue ASTERISK-20194. Reported by Nicolo Mazzon) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.8.0- fix build on s390- fix build on s390- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are - released as versions 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones - resolve the following two issues: - - * A permission escalation vulnerability in Asterisk Manager Interface. This - would potentially allow remote authenticated users the ability to execute - commands on the system shell with the privileges of the user running the - Asterisk application. Please note that the README-SERIOUSLY.bestpractices.txt - file delivered with Asterisk has been updated due to this and other related - vulnerabilities fixed in previous versions of Asterisk. - - * When an IAX2 call is made using the credentials of a peer defined in a - dynamic Asterisk Realtime Architecture (ARA) backend, the ACL rules for that - peer are not applied to the call attempt. This allows for a remote attacker - who is aware of a peer's credentials to bypass the ACL rules set for that - peer. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-012 and AST-2012-013, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert7 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.15.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.7.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.7.1-digiumphones - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-012.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-013.pdf- The Asterisk Development Team has announced the release of Asterisk 10.7.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.7.0 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 potential with ast_set_hangupsource() calls. - (Closes issue ASTERISK-19801. Reported by Alec Davis) - - * --- Fix request routing issue when outboundproxy is used. - (Closes issue ASTERISK-20008. Reported by Marcus Hunger) - - * --- Set the Caller ID "tag" on peers even if remote party - information is present. - (Closes issue ASTERISK-19859. Reported by Thomas Arimont) - - * --- Fix NULL pointer segfault in ast_sockaddr_parse() - (Closes issue ASTERISK-20006. Reported by Michael L. Young) - - * --- Do not perform install on existing directories - (Closes issue ASTERISK-19492. Reported by Karl Fife) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.7.0- The Asterisk Development Team has announced the release of Asterisk 10.6.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.6.1 resolves an issue reported by the - community and would have not been possible without your participation. - Thank you! - - The following is the issue resolved in this release: - - * --- Remove a superfluous and dangerous freeing of an SSL_CTX. - (Closes issue ASTERISK-20074. Reported by Trevor Helmsley) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.1- The Asterisk Development Team has announced the release of Asterisk 10.6.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.6.0 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: - - * --- format_mp3: Fix a possible crash in mp3_read(). - (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk) - - * --- Fix local channel chains optimizing themselves out of a call. - (Closes issue ASTERISK-16711. Reported by Alec Davis) - - * --- Re-add LastMsgsSent value for SIP peers - (Closes issue ASTERISK-17866. Reported by Steve Davies) - - * --- Prevent sip_pvt refleak when an ast_channel outlasts its - corresponding sip_pvt. - (Closes issue ASTERISK-19425. Reported by David Cunningham) - - * --- Send more accurate identification information in dialog-info SIP - NOTIFYs. - (Closes issue ASTERISK-16735. Reported by Maciej Krajewski) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.0- The Asterisk Development Team is pleased to announce the first beta release of - Asterisk 11.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - All interested users of Asterisk are encouraged to participate in the - Asterisk 11 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - All Asterisk users are invited to participate in the #asterisk-testing channel - on IRC to work together in testing the many parts of Asterisk. - - Asterisk 11 is the next major release series of Asterisk. It will be a Long - Term Support (LTS) release, similar to Asterisk 1.8. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 11, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 - - A short list of new features includes: - - * A new channel driver named chan_motif has been added which provides support - for Google Talk and Jingle in a single channel driver. This new channel - driver includes support for both audio and video, RFC2833 DTMF, all codecs - supported by Asterisk, hold, unhold, and ringing notification. It is also - compliant with the current Jingle specification, current Google Jingle - specification, and the original Google Talk protocol. - - * Support for the WebSocket transport for chan_sip. - - * SIP peers can now be configured to support negotiation of ICE candidates. - - * The app_page application now no longer depends on DAHDI or app_meetme. It - has been re-architected to use app_confbridge internally. - - * Hangup handlers can be attached to channels using the CHANNEL() function. - Hangup handlers will run when the channel is hung up similar to the h - extension; however, unlike an h extension, a hangup handler is associated with - the actual channel and will execute anytime that channel is hung up, - regardless of where it is in the dialplan. - - * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial - allows you to execute a dialplan subroutine on a channel before a call is - placed but after the application performing a dial action is invoked. This - means that the handlers are executed after the creation of the caller/callee - channels, but before any actions have been taken to actually dial the callee - channels. - - * Log messages can now be easily associated with a certain call by looking at - a new unique identifier, "Call Id". Call ids are attached to log messages for - just about any case where it can be determined that the message is related - to a particular call. - - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in - Asterisk. Unlike traditional ACLs defined in specific module configuration - files, Named ACLs can be shared across multiple modules. - - * The Hangup Cause family of functions and dialplan applications allow for - inspection of the hangup cause codes for each channel involved in a call. - This allows a dialplan writer to determine, for each channel, who hung up and - for what reason(s). - - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() - lets you set some of the configuration options from the general section - of features.conf on a per-channel basis. FEATUREMAP() lets you customize - the key sequence used to activate built-in features, such as blindxfer, - and automon. - - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups - and callgroups to be defined for several channel drivers. - - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation - - A full list of all new features can also be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES - - For a full list of changes in the current release, please see the ChangeLog. - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta1- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild- Perl 5.16 rebuild- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are - released as versions 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones - resolve the following two issues: - - * If Asterisk sends a re-invite and an endpoint responds to the re-invite with - a provisional response but never sends a final response, then the SIP dialog - structure is never freed and the RTP ports for the call are never released. If - an attacker has the ability to place a call, they could create a denial of - service by using all available RTP ports. - - * If a single voicemail account is manipulated by two parties simultaneously, - a condition can occur where memory is freed twice causing a crash. - - These issues and their resolution are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-010 and AST-2012-011, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert4 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.13.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2-digiumphones - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-010.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-011.pdf- Perl 5.16 rebuild- The Asterisk Development Team has announced a security release for Asterisk 10. - This security release is released as version 10.5.1. - - The release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 10.5.1 resolves the following issue: - - * A remotely exploitable crash vulnerability was found in the Skinny (SCCP) - Channel driver. When an SCCP client sends an Off Hook message, followed by - a Key Pad Button Message, a structure that was previously set to NULL is - dereferenced. This allows remote authenticated connections the ability to - cause a crash in the server, denying services to legitimate users. - - This issue and its resolution is described in the security advisory. - - For more information about the details of this vulnerability, please read - security advisory AST-2012-009, 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-10.5.1 - - The security advisory is available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-009.pdf- The Asterisk Development Team has announced the release of Asterisk 10.5.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.5.0 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: - - * --- Turn off warning message when bind address is set to any. - (Closes issue ASTERISK-19456. Reported by Michael L. Young) - - * --- Prevent overflow in calculation in ast_tvdiff_ms on 32-bit - machines - (Closes issue ASTERISK-19727. Reported by Ben Klang) - - * --- Make DAHDISendCallreroutingFacility wait 5 seconds for a reply - before disconnecting the call. - (Closes issue ASTERISK-19708. Reported by mehdi Shirazi) - - * --- Fix recalled party B feature flags for a failed DTMF atxfer. - (Closes issue ASTERISK-19383. Reported by lgfsantos) - - * --- Fix DTMF atxfer running h exten after the wrong bridge ends. - (Closes issue ASTERISK-19717. Reported by Mario) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.5.0- Perl 5.16 rebuild- The Asterisk Development Team has announced the release of Asterisk 10.4.2. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.4.2 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - * --- Resolve crash in subscribing for MWI notifications - (Closes issue ASTERISK-19827. Reported by B. R) - - * --- Fix crash in ConfBridge when user announcement is played for - more than 2 users - (Closes issue ASTERISK-19899. Reported by Florian Gilcher) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.2- The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are - released as versions 1.8.11-cert2, 1.8.12.1, and 10.4.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.8.11-cert2, 1.8.12.1, and 10.4.1 resolve the following - two issues: - - * A remotely exploitable crash vulnerability exists in the IAX2 channel - driver if an established call is placed on hold without a suggested music - class. Asterisk will attempt to use an invalid pointer to the music - on hold class name, potentially causing a crash. - - * A remotely exploitable crash vulnerability was found in the Skinny (SCCP) - Channel driver. When an SCCP client closes its connection to the server, - a pointer in a structure is set to NULL. If the client was not in the - on-hook state at the time the connection was closed, this pointer is later - dereferenced. This allows remote authenticated connections the ability to - cause a crash in the server, denying services to legitimate users. - - These issues and their resolution are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-007 and AST-2012-008, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.12.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.4.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-007.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-008.pdf- The Asterisk Development Team has announced the release of Asterisk 10.4.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.4.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - * --- Prevent chanspy from binding to zombie channels - (Closes issue ASTERISK-19493. Reported by lvl) - - * --- Fix Dial m and r options and forked calls generating warnings - for voice frames. - (Closes issue ASTERISK-16901. Reported by Chris Gentle) - - * --- Remove ISDN hold restriction for non-bridged calls. - (Closes issue ASTERISK-19388. Reported by Birger Harzenetter) - - * --- Fix copying of CDR(accountcode) to local channels. - (Closes issue ASTERISK-19384. Reported by jamicque) - - * --- Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors - (Closes issue ASTERISK-19303. Reported by Jon Tsiros) - - * --- Eliminate double close of file descriptor in manager.c - (Closes issue ASTERISK-18453. Reported by Jaco Kroon) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.0- The Asterisk Development Team has announced security releases for Asterisk 1.6.2, - 1.8, and 10. The available security releases are released as versions 1.6.2.24, - 1.8.11.1, and 10.3.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.6.2.24, 1.8.11.1, and 10.3.1 resolve the following two - issues: - - * A permission escalation vulnerability in Asterisk Manager Interface. This - would potentially allow remote authenticated users the ability to execute - commands on the system shell with the privileges of the user running the - Asterisk application. - - * A heap overflow vulnerability in the Skinny Channel driver. The keypad - button message event failed to check the length of a fixed length buffer - before appending a received digit to the end of that buffer. A remote - authenticated user could send sufficient keypad button message events that the - buffer would be overrun. - - In addition, the release of Asterisk 1.8.11.1 and 10.3.1 resolve the following - issue: - - * A remote crash vulnerability in the SIP channel driver when processing UPDATE - requests. If a SIP UPDATE request was received indicating a connected line - update after a channel was terminated but before the final destruction of the - associated SIP dialog, Asterisk would attempt a connected line update on a - non-existing channel, causing a crash. - - These issues and their resolution are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-004, AST-2012-005, and AST-2012-006, which were - released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.24 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.11.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.3.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-004.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-005.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-006.pdf- Update to 10.3.0- Update to 10.2.1 from upstream. - Fix remote stack overflow in app_milliwatt. - Fix remote stack overflow, including possible code injection, in HTTP digest authentication handling. - Disable asterisk-corosync package, as it doesn't build right now. - Resolves: rhbz#804045, rhbz#804038, rhbz#804042- * Add patch extracted from upstream to build with Corosync since - OpenAIS is no longer available. - * Add PrivateTmp=true to systemd service file (#782478) - * Add some macros to make it easier to build with fewer dependencies - (with corresponding less functionality) (#787389) - * Add isa macros in a few places plus a few other changes to make it - easier to cross-compile. (#787779)- The Asterisk Development Team has announced the release of Asterisk 10.1.2. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.1.2 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - * --- Fix SIP INFO DTMF handling for non-numeric codes --- - (Closes issue ASTERISK-19290. Reported by: Ira Emus) - - * --- Fix crash in ParkAndAnnounce --- - (Closes issue ASTERISK-19311. Reported-by: tootai) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.2- The Asterisk Development Team has announced the release of Asterisk 10.1.1. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.1.1 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: - - * --- Fixes deadlocks occuring in chan_agent --- - (Closes issue ASTERISK-19285. Reported by: Alex Villacis Lasso) - - * --- Ensure entering T.38 passthrough does not cause an infinite loop --- - (Closes issue ASTERISK-18951. Reported-by: Kristijan Vrban) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.1- The Asterisk Development Team is pleased to announce the release of - Asterisk 10.1.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.1.0 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: - - * AST-2012-001: prevent crash when an SDP offer - is received with an encrypted video stream when support for video - is disabled and res_srtp is loaded. (closes issue ASTERISK-19202) - Reported by: Catalin Sanda - - * Allow playback of formats that don't support seeking. ast_streamfile - previously did unconditional seeking on files that broke playback of - formats that don't support that functionality. This patch avoids the - seek that was causing the problem. - (closes issue ASTERISK-18994) Patched by: Timo Teras - - * Add pjmedia probation concepts to res_rtp_asterisk's learning mode. In - order to better handle RTP sources with strictrtp enabled (which is the - default setting in 10) using the learning mode to figure out new sources - when they change is handled by checking for a number of consecutive (by - sequence number) packets received to an rtp struct based on a new - configurable value called 'probation'. Also, during learning mode instead - of liberally accepting all packets received, we now reject packets until a - clear source has been determined. - - * Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing - to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop - causes the loop to exit prematurely. This causes a variety of negative side - effects, depending on when the loop exits. This patch handles the frame by - essentially swallowing the frame in the local loop, as the current channel - drivers expect the RTP bridge to handle the frame, and, in the case of the - local bridge loop, no additional action is necessary. - (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested - by: Matt Jordan - - * Fix timing source dependency issues with MOH. Prior to this patch, - res_musiconhold existed at the same module priority level as the timing - sources that it depends on. This would cause a problem when music on - hold was reloaded, as the timing source could be changed after - res_musiconhold was processed. This patch adds a new module priority - level, AST_MODPRI_TIMING, that the various timing modules are now loaded - at. This now occurs before loading other resource modules, such - that the timing source is guaranteed to be set prior to resolving - the timing source dependencies. - (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, - Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont - Patched by elguero - - * Fix RTP reference leak. If a blind transfer were initiated using a - REFER without a prior reINVITE to place the call on hold, AND if Asterisk - were sending RTCP reports, then there was a reference leak for the - RTP instance of the transferrer. - (closes issue ASTERISK-19192) Reported by: Tyuta Vitali - - * Fix blind transfers from failing if an 'h' extension - is present. This prevents the 'h' extension from being run on the - transferee channel when it is transferred via a native transfer - mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported - by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by - Mark Michelson (license 5049) - - * Restore call progress code for analog ports. Extracting sig_analog - from chan_dahdi lost call progress detection functionality. Fix - analog ports from considering a call answered immediately after - dialing has completed if the callprogress option is enabled. - (closes issue ASTERISK-18841) - Reported by: Richard Miller Patched by Richard Miller - - * Fix regression that 'rtp/rtcp set debup ip' only works when a port - was also specified. - (closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by: - Walter Doekes - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.0- Remove asterisk-ais. OpenAIS was removed from Fedora.- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild- Don't build API docs as the build never finishes- The Asterisk Development Team is proud to announce the release of - Asterisk 10.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - Asterisk 10 is the next major release series of Asterisk. It will be a - Standard support release, similar to Asterisk 1.6.2. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - With the release of the Asterisk 10 branch, the preceding '1.' has been removed - from the version number per the blog post available at - - - http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ - - The release of Asterisk 10 would not have been possible without the support and - contributions of the community. - - You can find an overview of the work involved with the 10.0.0 release in the - summary: - - http://svn.asterisk.org/svn/asterisk/tags/10.0.0/asterisk-10.0.0-summary.txt - - A short list of available features includes: - - * T.38 gateway functionality has been added to res_fax. - * Protocol independent out-of-call messaging support. Text messages not - associated with an active call can now be routed through the Asterisk - dialplan. SIP and XMPP are supported so far. - * New highly optimized and customizable ConfBridge application capable of mixing - audio at sample rates ranging from 8kHz-192kHz - * Addition of video_mode option in confbridge.conf to provide basic video - conferencing in the ConfBridge() dialplan application. - * Support for defining hints has been added to pbx_lua. - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svn.asterisk.org/svn/asterisk/branches/10/CHANGES - - Also, when upgrading a system between major versions, it is imperative that you - read and understand the contents of the UPGRADE.txt file, which is located at: - - http://svn.asterisk.org/svn/asterisk/branches/10/UPGRADE.txt- The Asterisk Development Team has announced the third release candidate of - Asterisk 10.0.0. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.0.0-rc3 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: - - * Add ASTSBINDIR to the list of configurable paths - - This patch also makes astdb2sqlite3 and astcanary use the configured - directory instead of relying on $PATH. - - * Don't crash on INFO automon request with no channel - - AST-2011-014. When automon was enabled in features.conf, it was possible - to crash Asterisk by sending an INFO request if no channel had been - created yet. - - * Fixed crash from orphaned MWI subscriptions in chan_sip - - This patch resolves the issue where MWI subscriptions are orphaned - by subsequent SIP SUBSCRIBE messages. - - * Fix a change in behavior in 'database show' from 1.8. - - In 1.8 and previous versions, one could use any fullword portion of - the key name, including the full key, to obtain the record. Until this - patch, this did not work for the full key. - - * Default to nat=yes; warn when nat in general and peer differ - - AST-2011-013. It is possible to enumerate SIP usernames when the general and - user/peer nat settings differ in whether to respond to the port a request is - sent from or the port listed for responses in the Via header. In 1.4 and - 1.6.2, this would mean if one setting was nat=yes or nat=route and the other - was either nat=no or nat=never. In 1.8 and 10, this would mean when one - was nat=force_rport and the other was nat=no. - - In order to address this problem, it was decided to switch the default - behavior to nat=yes/force_rport as it is the most commonly used option - and to strongly discourage setting nat per-peer/user when at all - possible. - - * Fixed SendMessage stripping extension from To: header in SIP MESSAGE - - When using the MessageSend application to send a SIP MESSAGE to a - non-peer, chan_sip stripped off the extension and failed to add it back - to the sip_pvt structure before transmitting. This patch adds the full - URI passed in from the message core to the sip_pvt structure. - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc3- The Asterisk Development Team has announced the second release candidate of - Asterisk 10.0.0. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.0.0-rc2 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: - - * Ensure that a null vmexten does not cause a segfault - - * Fix issue with ConfBridge participants hanging up during DTMF feature - menu usage getting stuck in conference forever - (closes issue ASTERISK-18829) - Reported by: zvision - - * Fix app_macro.c MODULEINFO section termination - (closes issue ASTERISK-18848) - Reported by: Tony Mountifield - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc2- The Asterisk Development Team is pleased to announce the first release candidate - of Asterisk 10.0.0. This release candidate is available for immediate download - at http://downloads.asterisk.org/pub/telephony/asterisk/ - - All Asterisk users are encouraged to participate in the Asterisk 10 testing - process. Please report any issues found to the issue tracker, - https://issues.asterisk.org/jira. It is also very useful to see successful test - reports. Please post those to the asterisk-dev mailing list. - - All Asterisk users are invited to participate in the #asterisk-testing - channel on IRC to work together in testing the many parts of Asterisk. - - Asterisk 10 is the next major release series of Asterisk. It will be a - Standard support release, similar to Asterisk 1.6.2. For more - information about support time lines for Asterisk releases, see the Asterisk - versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - A short list of features includes: - - * T.38 gateway functionality has been added to res_fax. - * Protocol independent out-of-call messaging support. Text messages not - associated with an active call can now be routed through the Asterisk - dialplan. SIP and XMPP are supported so far. - * New highly optimized and customizable ConfBridge application capable of mixing - audio at sample rates ranging from 8kHz-192kHz - (More information available at - https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 ) - * Addition of video_mode option in confbridge.conf to provide basic video - conferencing in the ConfBridge() dialplan application. - * Support for defining hints has been added to pbx_lua. - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/10/CHANGES - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-rc1- Add patch from upstream SVN to fix AST-2011-012- Patch cleanup day- The Asterisk Development Team is pleased to announce the second beta release of - Asterisk 10.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - With the release of the Asterisk 10 branch, the preceding '1.' has been removed - from the version number per the blog post available at - http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ - - All interested users of Asterisk are encouraged to participate in the - Asterisk 10 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - - All Asterisk users are invited to participate in the #asterisk-testing - channel on IRC to work together in testing the many parts of Asterisk. - - Asterisk 10 is the next major release series of Asterisk. It will be a - Standard support release, similar to Asterisk 1.6.2. For more - information about support time lines for Asterisk releases, see the Asterisk - versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - A short list of features includes: - - * T.38 gateway functionality has been added to res_fax. - - * Protocol independent out-of-call messaging support. Text messages not - associated with an active call can now be routed through the Asterisk - dialplan. SIP and XMPP are supported so far. - - * New highly optimized and customizable ConfBridge application capable of mixing - audio at sample rates ranging from 8kHz-192kHz - - * Addition of video_mode option in confbridge.conf to provide basic video - conferencing in the ConfBridge() dialplan application. - - * Support for defining hints has been added to pbx_lua. - - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). - - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/10/CHANGES - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta2- - The Asterisk Development Team is pleased to announce the first beta release of - Asterisk 10.0.0-beta1. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - With the release of the Asterisk 10 branch, the preceding '1.' has been removed - from the version number per the blog post available at - http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ - - All interested users of Asterisk are encouraged to participate in the - Asterisk 10 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - - All Asterisk users are invited to participate in the #asterisk-testing - channel on IRC to work together in testing the many parts of Asterisk. - Additionally users can make use of the RPM and DEB packages now being built for - all Asterisk releases. More information available at - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages - - Asterisk 10 is the next major release series of Asterisk. It will be a - Standard support release, similar to Asterisk 1.6.2. For more - information about support time lines for Asterisk releases, see the Asterisk - versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - A short list of included features includes: - - * T.38 gateway functionality has been added to res_fax. - * Protocol independent out-of-call messaging support. Text messages not - associated with an active call can now be routed through the Asterisk - dialplan. SIP and XMPP are supported so far. - * New highly optimized and customizable ConfBridge application capable of mixing - audio at sample rates ranging from 8kHz-192kHz - * Addition of video_mode option in confbridge.conf to provide basic video - conferencing in the ConfBridge() dialplan application. - * Support for defining hints has been added to pbx_lua. - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svn.digium.com/view/asterisk/branches/10/CHANGES?view=checkout - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta1- Perl mass rebuild- Perl mass rebuild- 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- Rebuild for net-snmp 5.7- Fix systemd dependencies in EL6 and F15- 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- Fedora Directory Server -> 389 Directory Server- 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- Don't forget stereorize- 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- 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- Convert to systemd- Perl mass rebuild- Perl 5.14 mass rebuild- - 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- 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. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - 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-1.4.40.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.25 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.3 - 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- Bump release and rebuild for mysql 5.5.10 soname change.- 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 1.6.1.24, 1.6.2.17.2, and 1.8.3.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - ** 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-1.6.1.24 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.2 - 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- 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. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - 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-1.6.1.23 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.1 - 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- The Asterisk Development Team has announced the release of Asterisk 1.8.3. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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() - (Closes issue #18091. Reported by bunny. Patched by tilghman) - - * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. - (Closes issue #18464. Reported, patched by IgorG) - - * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of - unit tests for the function that does the parsing. - (Closes issue #18350. Reported by gbour. Patched by Marquis) - - * When using cdr_pgsql the billsec field was not populated correctly on - unanswered calls. - (Closes issue #18406. Reported by joscas. Patched by tilghman) - - * Resolve memory leak in iCalendar and Exchange calendaring modules. - (Closes issue #18521. Reported, patched by pitel. Tested by cervajs) - - * 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. - (Patched by tilghman) - - * Resolve issue where no Music On Hold may be triggered when using - res_timing_dahdi. - (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested - by francesco_r, rfrantik, one47) - - * Resolve a memory leak when the Asterisk Manager Interface is disabled. - (Reported internally by kmorgan. Patched by russellb) - - * Reimplemented fax session reservation to reverse the ABI breakage introduced - in r297486. - (Reported internally. Patched by mnicholson) - - * Fix regression that changed behavior of queues when ringing a queue member. - (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.) - - * Resolve deadlock involving REFER. - (Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.) - - 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- - The Asterisk Development Team has announced the third release candidate of - Asterisk 1.8.3. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.3-rc3 resolves the following issues in addition to - those included in 1.8.3-rc1 and 1.8.3-rc2: - - * Fix regression that changed behavior of queues when ringing a queue member. - (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.) - - * Resolve deadlock involving REFER. - (Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3-rc3- Bump release to build for F15- Remove isa macros- Make library dependencies architecture specific- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_RebuildThe Asterisk Development Team has announced the second release candidate of Asterisk 1.8.3. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.3-rc2 resolves the following issues in addition to those included in 1.8.3-rc1: * Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi. (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47) * Resolve a memory leak when the Asterisk Manager Interface is disabled. (Reported internally by kmorgan. Patched by russellb) * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported internally. 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-1.8.3-rc2- - The Asterisk Development Team has announced the first release candidate of - Asterisk 1.8.3. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.3-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: - - * Resolve duplicated data in the AstDB when using DIALGROUP() - (Closes issue #18091. Reported by bunny. Patched by tilghman) - - * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. - (Closes issue #18464. Reported, patched by IgorG) - - * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of - unit tests for the function that does the parsing. - (Closes issue #18350. Reported by gbour. Patched by Marquis) - - * When using cdr_pgsql the billsec field was not populated correctly on - unanswered calls. - (Closes issue #18406. Reported by joscas. Patched by tilghman) - - * Resolve memory leak in iCalendar and Exchange calendaring modules. - (Closes issue #18521. Reported, patched by pitel. Tested by cervajs) - - * 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. - (Patched by tilghman)- - The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.2.3 resolves the following issue: - - * Reimplemented fax session reservation to reverse the ABI breakage introduced - in r297486. - (Reported by Jeremy Kister on the asterisk-users mailing list. Patched by - mnicholson) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2.3- Build with SRTP support- - The Asterisk Development Team has announced a release for the security issue - described in AST-2011-001. - - Due to a failed merge, Asterisk 1.8.2.1 which should have included the security - fix did not. Asterisk 1.8.2.2 contains the the changes which should have been - included in Asterisk 1.8.2.1. - - This releases is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2, - 1.8.1.2, and 1.8.2.2 resolve an issue when forming an outgoing SIP request while - in pedantic mode, which can cause a stack buffer to be made to overflow if - supplied with carefully crafted caller ID information. The issue and resolution - are described in the AST-2011-001 security advisory. - - For more information about the details of this vulnerability, please read the - security advisory AST-2011-001, 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-1.8.2.2 - - Security advisory AST-2011-001 is available at: - - http://downloads.asterisk.org/pub/security/AST-2011-001.pdf- - The Asterisk Development Team has announced security releases for the following - versions of Asterisk: - - * 1.4.38.1 - * 1.4.39.1 - * 1.6.1.21 - * 1.6.2.15.1 - * 1.6.2.16.1 - * 1.8.1.2 - * 1.8.2.1 - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2, - 1.8.1.2, and 1.8.2.1 resolve an issue when forming an outgoing SIP request while - in pedantic mode, which can cause a stack buffer to be made to overflow if - supplied with carefully crafted caller ID information. The issue and resolution - are described in the AST-2011-001 security advisory. - - For more information about the details of this vulnerability, please read the - security advisory AST-2011-001, which was released at the same time as this - announcement. - - For a full list of changes in the current releases, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.38.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.39.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.21 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.15.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.16.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.1.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.1 - - Security advisory AST-2011-001 is available at: - - http://downloads.asterisk.org/pub/security/AST-2011-001.pdf- - The Asterisk Development Team has announced the release of Asterisk 1.8.2. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.2 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: - - * 'sip notify clear-mwi' needs terminating CRLF. - (Closes issue #18275. Reported, patched by klaus3000) - - * Patch for deadlock from ordering issue between channel/queue locks in - app_queue (set_queue_variables). - (Closes issue #18031. Reported by rain. Patched by bbryant) - - * Fix cache of device state changes for multiple servers. - (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested - by russellb) - - * Resolve issue where channel redirect function (CLI or AMI) hangs up the call - instead of redirecting the call. - (Closes issue #18171. Reported by: SantaFox) - (Closes issue #18185. Reported by: kwemheuer) - (Closes issue #18211. Reported by: zahir_koradia) - (Closes issue #18230. Reported by: vmarrone) - (Closes issue #18299. Reported by: mbrevda) - (Closes issue #18322. Reported by: nerbos) - - * Fix reloading of peer when a user is requested. Prevent peer reloading from - causing multiple MWI subscriptions to be created when using realtime. - (Closes issue #18342. Reported, patched by nivek.) - - * Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0 - so res_jabber doesn't think there is already an XMPP connection sending - device state. Also clean up CLI commands a bit. - (Closes issue #18272. Reported by klaus3000. Patched by Marquis42) - - * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of - setting peer->cdr = NULL, set it to not post. - (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares) - - * Fixes issue with outbound google voice calls not working. Thanks to az1234 - and nevermind_quack for their input in helping debug the issue. - (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2- - The Asterisk Development Team has announced the release of Asterisk 1.8.1.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.1.1 resolves two issues reported by the community - since the release of Asterisk 1.8.1. - - * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of - setting peer->cdr = NULL, set it to not post. - (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares) - - * Fixes issue with outbound google voice calls not working. Thanks to az1234 - and nevermind_quack for their input in helping debug the issue. - (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1.1- - The Asterisk Development Team has announced the release of Asterisk 1.8.1. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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- dont package up the ices bits on el the client doesnt exist for us- dont build the 389 directory server package its not available on rhel6- dont always build AIS modules we dont have the BuildRequires on epel- Rebuild for new net-snmp.- Always build AIS modules- The Asterisk Development Team is proud to announce the release of Asterisk - 1.8.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - Asterisk 1.8 is the next major release series of Asterisk. It will be a Long - Term Support (LTS) release, similar to Asterisk 1.4. For more information about - support time lines for Asterisk releases, see the Asterisk versions page. - - http://www.asterisk.org/asterisk-versions - - 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 - - Thank you for your continued support of Asterisk!- - The release of Asterisk 1.8.0-rc5 was triggered by some last minute platform - compatibility IPv6 changes. In addition, the availability of the English sound - prompts with Australian accents has been added. - - 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-rc5 - - This release candidate contains fixes since the last release candidate as - reported by the community. A sampling of the changes in this release candidate - include: - - * Additional fixups in chan_gtalk that allow outbound calls to both Google - Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip - and stunaddr. - (Closes issue #13971. Patched by dvossel) - - * Resolve manager crash issue. - (Closes issue #17994. Reported by vrban. Patchd by dvossel) - - * Documentation updates for sample configuration files. - (Closes issues #18107, #18101. Reported, patched by lathama, lmadsen) - - * Resolve issue where faxdetect would only detect the first fax call in - chan_dahdi. - (Closes issue #18116. Reported by seandarcy. Patched by rmudgett) - - * Resolve issue where a channel that is setup and torn down *very* quickly may - not have the right call disposition or ${DIALSTATUS}. - (Closes issue #16946. Reported by davidw. Review - https://reviewboard.asterisk.org/r/740/) - - * Set TCLASS field of IPv6 header when SIP QoS options are set. - (Closes issue #18099. Reported by jamesnet. Patched by dvossel) - - * Resolve issue where Asterisk could crash on shutdown when using SRTP. - (Closes issue #18085. Reported by st. Patched by twilson) - - * Fix issue where peers host port would be lost on a SIP reload. - (Closes issue #18135. Reported, tested by lmadsen. Patched by dvossel) - - 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-rc4- This release candidate contains fixes since the release candidate as reported by - the community. A sampling of the changes in this release candidate include: - - * Still build chan_sip even if res_crypto cannot be built (use, but not depend) - (Reported by a user on the mailing list. Patched by tilghman) - - * Get notifications for call files only when a file is closed, not when created - (Closes issue #17924. Reported by mkeuter. Patched by abeldeck) - - * Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk - expects the DTMF to arrive on the RTP stream and not via jingle DTMF - signalling. - (Patched by dvossel. Tested by malcolmd) - - * Fixes to allow chan_gtalk to communicate with the Gmail web client. - (Patched by phsultan and dvossel) - - * Fix to GET DATA to allow audio to be streamed via an AGI. - (Closes issue #18001. Reported by jamicque. Patched by tilghman) - - * Resolve dnsmgr memory corruption in chan_iax2. - (Closes issue #17902. Reported by afried. Patched by russell, dvossel) - - 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=checkout - - 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-rc3- This release candidate contains fixes since the last beta release as reported by - the community. A sampling of the changes in this release candidate include: - - * Add slin16 support for format_wav (new wav16 file extension) - (Closes issue #15029. Reported, patched by andrew. Tested by Qwell) - - * Fixes a bug in manager.c where the default configuration values weren't reset - when the manager configuration was reloaded. - (Closes issue #17917. Reported by lmadsen. Patched by bbryant) - - * Various fixes for the calendar modules. - (Patched by Jan Kalab. - Reviewboard: https://reviewboard.asterisk.org/r/880/ - Closes issue #17877. Review: https://reviewboard.asterisk.org/r/916/ - Closes issue #17776. Review: https://reviewboard.asterisk.org/r/921/) - - * Add CHANNEL(checkhangup) to check whether a channel is in the process of - being hung up. - (Closes issue #17652. Reported, patched by kobaz) - - * Fix a bug with MeetMe where after announcing the amount of time left in a - conference, if music on hold was playing, it doesn't restart. - (Closes issue #17408, Reported, patched by sysreq) - - * Fix interoperability problems with session timer behavior in Asterisk. - (Closes issue #17005. Reported by alexcarey. Patched by dvossel) - - * Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was - determined to be one of the most significant bottlenecks in SIP registration - processing. This patch improved the speed of an astdb load test by 50000% - (yes, Fifty-Thousand Percent). On this particular load test setup, this - doubled the number of SIP registrations the server could handle. - (Review: https://reviewboard.asterisk.org/r/825/) - - * Don't clear the username from a realtime database when a registration - expires. Non-realtime chan_sip does not clear the username from memory when a - registration expiries so realtime probably shouldn't either. - (Closes issue #17551. Reported, patched by: ricardolandim. Patched by - mnicholson) - - * Don't hang up a call on an SRTP unprotect failure. Also make it more obvious - when there is an issue en/decrypting. - (Closes issue #17563. Reported by Alexcr. Patched by sfritsch. Tested by - twilson) - - * Many more issues. This is a significant upgrade over Asterisk 1.8.0 beta 5! - - 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=checkout - - 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-rc2- This release contains fixes since the last beta release as reported by the - community. A sampling of the changes in this release include: - - * Fix issue where TOS is no longer set on RTP packets. - (Closes issue #17890. Reported, patched by elguero) - - * Change pedantic default value in chan_sip from 'no' to 'yes' - - * Asterisk now dynamically builds the "Supported" header depending on what is - enabled/disabled in sip.conf. Session timers used to always be advertised as - being supported even when they were disabled in the configuration. - (Related to issue #17005. Patched by dvossel) - - * Convert MOH to use generic timers. - (Closes issue #17726. Reported by lmadsen. Patched by tilghman) - - * Fix SRTP for changing SSRC and multiple a=crypto SDP lines. Adding code to - Asterisk that changed the SSRC during bridges and masquerades broke SRTP - functionality. Also broken was handling the situation where an incoming - INVITE had more than one crypto offer. - (Closes issue #17563. Reported by Alexcr. Patched by twilson) - - Asterisk 1.8 contains many new features over previous releases of Asterisk. - A short list of included features includes: - - * Secure RTP - * IPv6 Support in the SIP Channel - * 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=checkout - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta5- This release contains fixes since the last beta release as reported by the - community. A sampling of the changes in this release include: - - * Fix parsing of IPv6 address literals in outboundproxy - (Closes issue #17757. Reported by oej. Patched by sperreault) - - * Change the default value for alwaysauthreject in sip.conf to "yes". - (Closes issue #17756. Reported by oej) - - * Remove current STUN support from chan_sip.c. This change removes the current - broken/useless STUN support from chan_sip. - (Closes issue #17622. Reported by philipp2. - Review: https://reviewboard.asterisk.org/r/855/) - - * PRI CCSS may use a stale dial string for the recall dial string. If an - outgoing call negotiates a different B channel than initially requested, the - saved original dial string was not transferred to the new B channel. CCSS - uses that dial string to generate the recall dial string. - (Patched by rmudgett) - - * Split _all_ arguments before parsing them. This fixes multicast RTP paging - using linksys mode. - (Patched by russellb) - - * Expand cel_custom.conf.sample. Include the usage of CSV_QUOTE() to ensure - data has valid CSV formatting. Also list the special CEL variables that are - available for use in the mapping. There are also several other CEL fixes in - this release. - (Patched by russellb) - - Asterisk 1.8 contains many new features over previous releases of Asterisk. - A short list of included features includes: - - * Secure RTP - * IPv6 Support in the SIP Channel - * 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=checkout - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta4- - This release contains fixes since the last beta release as reported by the - community. A sampling of the changes in this release include: - - * Fix a regression where HTTP would always be enabled regardless of setting. - (Closes issue #17708. Reported, patched by pabelanger) - - * ACL errors displayed on screen when using dynamic_exclude_static in sip.conf - (Closes issue #17717. Reported by Dennis DeDonatis. Patched by mmichelson) - - * Support "channels" in addition to "channel" in chan_dahdi.conf. - (https://reviewboard.asterisk.org/r/804) - - * Fix parsing error in sip_sipredirect(). The code was written in a way that - did a bad job of parsing the port out of a URI. Specifically, it would do - badly when dealing with an IPv6 address. - (Closes issue #17661. Reported by oej. Patched by mmichelson) - - * Fix inband DTMF detection on outgoing ISDN calls. - (Patched by russellb and rmudgett) - - * Fixes issue with translator frame not getting freed. This issue prevented - g729 licenses from being freed up. - (Closes issue #17630. Reported by manvirr. Patched by dvossel) - - * Fixed IPv6-related SIP parsing bugs and updated documention. - (Reported by oej. Patched by sperreault) - - * Add new, self-contained feature FIELDNUM(). Returns a 1-based index into a - list of a specified item. Matches up with FIELDQTY() and CUT(). - (Closes #17713. Reported, patched by gareth. Tested by tilghman) - - Asterisk 1.8 contains many new features over previous releases of Asterisk. - A short list of included features includes: - - * Secure RTP - * IPv6 Support - * 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=checkout - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta3- Rebuild against libpri 1.4.12- Update to 1.8.0-beta2 - Disable building chan_misdn until compilation errors are figured out (https://issues.asterisk.org/view.php?id=14333) - Start stripping tarballs again because Digium added MP3 code :(- - The following are a few of the issues resolved by community developers: - - * Allow users to specify a port for DUNDI peers. - (Closes issue #17056. Reported, patched by klaus3000) - - * Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is - set. - (Closes issue #16815. Reported, patched by rain) - - * If there is realtime configuration, it does not get re-read on reload unless - the config file also changes. - (Closes issue #16982. Reported, patched by dmitri) - - * Send AgentComplete manager event for attended transfers. - (Closes issue #16819. Reported, patched by elbriga) - - * Correct manager variable 'EventList' case. - (Closes issue #17520. Reported, patched by kobaz) - - In addition, changes to res_timing_pthread that should make it more stable have - also been implemented. - - For a full list of changes in the current release, please see the - ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10- Add patch to remove requirement on latex2html- Mass rebuild with perl-5.12.0- * Fix building CDR and CEL SQLite3 modules. - (Closes issue #17017. Reported by alephlg. Patched by seanbright) - - * Resolve crash in SLAtrunk when the specified trunk doesn't exist. - (Reported in #asterisk-dev by philipp64. Patched by seanbright) - - * Include an extra newline after "Aliased CLI command" to get back the prompt. - (Issue #16978. Reported by jw-asterisk. Tested, patched by seanbright) - - * Prevent segfault if bad magic number is encountered. - (Issue #17037. Reported, patched by alecdavis) - - * Update code to reflect that handle_speechset has 4 arguments. - (Closes issue #17093. Reported, patched by gpatri. Tested by pabelanger, - mmichelson) - - * Resolve a deadlock in chan_local. - (Closes issue #16840. Reported, patched by bzing2, russell. Tested by bzing2)- Update to 1.6.2.7-rc3- Update to 1.6.2.7-rc2- Update to final 1.6.2.6 - - The following are a few of the issues resolved by community developers: - - * Make sure to clear red alarm after polarity reversal. - (Closes issue #14163. Reported, patched by jedi98. Tested by mattbrown, - Chainsaw, mikeeccleston) - - * Fix problem with duplicate TXREQ packets in chan_iax2 - (Closes issue #16904. Reported, patched by rain. Tested by rain, dvossel) - - * Fix crash in app_voicemail related to message counting. - (Closes issue #16921. Reported, tested by whardier. Patched by seanbright) - - * Overlap receiving: Automatically send CALL PROCEEDING when dialplan starts - (Reported, Patched, and Tested by alecdavis) - - * For T.38 reINVITEs treat a 606 the same as a 488. - (Closes issue #16792. Reported, patched by vrban) - - * Fix ConfBridge crash when no timing module is loaded. - (Closes issue #16471. Reported, tested by kjotte. Patched, tested by junky) - - For a full list of changes in this releases, please see the ChangeLog: - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.6- Update to 1.6.2.6-rc2- Add a patch that fixes CLI history when linking against external libedit.- Update to 1.6.2.5 - - * AST-2010-002: Invalid parsing of ACL rules can compromise security- Update to 1.6.2.4 - - * AST-2010-002: This security release is intended to raise awareness - of how it is possible to insert malicious strings into dialplans, - and to advise developers to read the best practices documents so - that they may easily avoid these dangers.- Update to 1.6.2.2 - - * AST-2010-001: 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 occurs when the FaxMaxDatagram field is omitted from - the SDP as well.- Update to 1.6.2.1 final: - - * CLI 'queue show' formatting fix. - (Closes issue #16078. Reported by RoadKill. Tested by dvossel. Patched by - ppyy.) - - * Fix misreverting from 177158. - (Closes issue #15725. Reported, Tested by shanermn. Patched by dimas.) - - * Fixes subscriptions being lost after 'module reload'. - (Closes issue #16093. Reported by jlaroff. Patched by dvossel.) - - * app_queue segfaults if realtime field uniqueid is NULL - (Closes issue #16385. Reported, Tested, Patched by haakon.) - - * 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.- Update to 1.6.2.1-rc1- Released version of 1.6.2.0- Update to 1.6.2.0-rc8- Update to 1.6.2.0-rc7- Change the logrotate and the init scripts so that Asterisk doesn't try and write to / or /root- Make dependency on uw-imap conditional and some other changes to make building on RHEL5 easier.- Update to 1.6.2.0-rc6- Update to 1.6.2.0-rc5- Update to 1.6.2.0-rc4- Add patch from upstream to fix how res_http_post forms paths.- Add an AST_EXTRA_ARGS option to the init script - have the init script to cd to /var/spool/asterisk to prevent annoying message- Compile against gmime 2.2 instead of gmime 2.4 because the patch to convert the API calls from 2.2 to 2.4 caused crashes.- Require latex2html used in static-http documents- Change ownership and permissions on config files to protect them.- Update to 1.6.2.0-rc3- Merge firmware subpackage back into the main package.- Package internal help. - Fix up some more paths in the configs so that everything ends up where we want them.- Update to 1.6.2.0-rc2 - We no longer need to strip the tarball as it no longer includes non-free items.- Enable building of API docs. - Depend on version 1.2 or newer of speex- Update to 1.6.1.6 - Drop patches that are too troublesome to maintain anymore or have been integrated upstream.- Add a patch from Quentin Armitage and rebuld.- rebuilt with new openssl- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild- Rebuild to pick up new AIS and ODBC deps. - Update script that strips out bad content from tarball to do the download and to check the GPG signature.- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild- Update to 1.6.1-rc1 - Add backport of conference bridging that is slated for 1.6.2 - Add patches to conference bridging that implement CLI apps- rebuild with new openssl- Fedora Directory Server compatibility patch/subpackage.- Fix up paths. BZ#477238- Update patches- Update to 1.6.1-beta4- Update to 1.6.1-beta3- Rebuild for new gmime- Add patch to fix missing variable on PPC.- Update PPC systems don't have sys/io.h patch.- PPC systems don't have sys/io.h- Update to 1.6.1 beta 2- Fix issue with init script giving wrong path to config file.- Explicitly require dahdi-tools-libs to see if we can get this to build.- Update to final release.- Rebuild- Replace app_rxfax/app_txfax with app_fax taken from upstream SVN.- Bump release and rebuild with new libpri and zaptel.- Add patch pulled from upstream SVN that fixes AST-2008-010 and AST-2008-011.- Add patch for LDAP extracted from upstream SVN (#442011)- Add patch that unbreaks cdr_tds with FreeTDS 0.82. - Properly obsolete conference subpackage.- Disable building cdr_tds since new FreeTDS in rawhide no longer provides needed library.- Bump release and rebuild to fix libtds breakage.- Update to 1.6.0-beta9. - Update patches so that they apply cleanly. - Temporarily disable app_conference patch as it doesn't compile - config/scripts/postgres_cdr.sql has been merged into realtime_pgsql.sql - Re-add the asterisk-strip.sh script as a source file.- Update to 1.6.0-beta8 - Contains fixes for AST-2008-006 / CVE-2008-1897- Return to stripped tarballs since there's more non-free content in the Asterisk tarballs than I thought.- Update to 1.6.0-beta7.1 - Update patches - Back out some changes that were made because beta7 was tagged from the wrong branch.- Update to 1.6.0-beta7 - The Asterisk tarball no longer contains the iLBC code, so we can directly use the upstream tarball without having to modify it. - Get rid of the asterisk-strip.sh script since it's no longer needed. - Diable build of codec_ilbc and format_ilbc (these do not contain any legally suspect code so they can be included in the tarball but it's pointless building them). - Update chan_mobile patch to fix API breakages. - Add a patch to chan_usbradio to fix API breakages.- Add Postgresql schemas from contrib as documentation to the Postgresql subpackage.- Update patches. - Add patch to compile against external libedit rather than using the in-tree version. - Add -Werror-implicit-function-declaration to optflags. - Get rid of hashtest and hashtest2 binaries that link to unfortified versions of *printf functions. They are compiled with -O0 which somehow pulls in the wrong versions. These programs aren't necessary to the operation of the package anyway.- Update to 1.6.0-beta6 to fix some security issues. - - AST-2008-002 details two buffer overflows that were discovered in - RTP codec payload type handling. - * http://downloads.digium.com/pub/security/AST-2008-002.pdf - * All users of SIP in Asterisk 1.4 and 1.6 are affected. - - AST-2008-003 details a vulnerability which allows an attacker to - bypass SIP authentication and to make a call into the context - specified in the general section of sip.conf. - * http://downloads.digium.com/pub/security/AST-2008-003.pdf - * All users of SIP in Asterisk 1.0, 1.2, 1.4, or 1.6 are affected. - - AST-2008-004 Logging messages displayed using the ast_verbose - logging API call are not displayed as a character string, they are - displayed as a format string. - * http://downloads.digium.com/pub/security/AST-2008-004.pdf - - AST-2008-005 details a problem in the way manager IDs are caculated. - * http://downloads.digium.com/pub/security/AST-2008-005.pdf- add Requires for versioned perl (libperl.so)- Update to 1.6.0-beta5 - Remove upstreamed patches.- Package the directory used to store monitor recordings.- Add patch from David Woodhouse that fixes building on PPC64.- Update to 1.6.0 beta 4- Update to 1.4.18. - Use -march=i486 on i386 builds for atomic operations (GCC 4.3 compatibility). - Use "logger reload" instead of "logger rotate" in logrotate file (#432197). - Don't explicitly specify a group in in the init script to prevent Zaptel breakage (#426629). - Split app_ices out to a separate package so that the ices package can be required. - pbx_kdeconsole has been dropped, don't specifically exclude it from the build anymore. - Update app_conference patch. - Drop upstreamed libcap patch.- Update to 1.4.17 to fix AST-2008-001.- Update to 1.4.16.2- Bump release and rebuild to fix broken dep on uw-imap.- Update to the real 1.4.16.1.- Add patch to bring source up to version 1.4.16.1 which will be released shortly to fix some crasher bugs introduced by 1.4.16.- Update to 1.4.16 to fix security bug.- Really, really fix the build problems on devel.- Tweaks to get to build on x86_64- Exclude PPC64- Don't build apidocs by default since there's a problem building on x86_64.- Really get rid of zero length map files.- Get rid of zero length map files. - Shorten descriptions of voicemail subpackages- Update to 1.4.15- Fix license and other rpmlint warnings.- Update to 1.4.14- Add chan_mobile- Don't build cdr_sqlite because sqlite2 has been orphaned. - Rebase local patches to latest upstream SVN - Update app_conference patch to latest from upstream SVN - Apply post-1.4.13 patches from upstream SVN- Update to 1.4.13- Update to 1.4.12.1- Update to 1.4.11- Update to 1.4.10.1.- Update to 1.4.10 (security update).- Add a patch that allows alternate extensions to be defined in users.conf- Update app_conference patch. Enter/leave sounds are now possible.- Update patches so we don't need to run auto* tools, because autoconf 2.60 is required and FC-6 and RHEL5 only have autoconf 2.59.- Don't build app_mp3- Add app_conference- Use plain useradd/groupadd rather than the fedora-usermgmt - Clean up requirements - Clean up build requirements by moving them to package sections- Update to 1.4.9- Update to 1.4.8 - Drop ixjuser patch.- Update to 1.4.7.1- Update to 1.4.7 - RxFAX/TxFAX applications- It's "sbin", not "bin" silly.- Add patch that lets us change TOS bits even when running non-root- voicemail needs to require /usr/bin/sox and /usr/bin/sendmail- Update to 1.4.6 - Remove upstreamed patch.- Build the IMAP and ODBC storage options of voicemail and split voicemail out into subpackages. - Apply patch so that the system UW IMAP libray can be linked against. - Patch modules.conf.sample so that alternal voicemail modules don't get loaded simultaneously. - Link against system GSM library rather than internal copy. - Patch the Makefile so that it doesn't add redundant/wrong compiler options. - Force building with the standard RPM optimization flags. - Install the Asterisk MIB in a location that net-snmp can find it. - Only package docs in the main package that are relevant and that haven't been packaged by a subpackage. - Other minor cleanups.- Move sounds- Update some more ownership/permissions- Fix some permissions.- Update init script patch - Move pid file to subdir of /var/run- Update init script patch to run as non-root- Build modules that depend on FreeTDS. - Don't build voicemail with ODBC storage.- Have the build output the commands executing, rather than covering them up.- Update to 1.4.5 - Remove upstreamed patch.- Add a patch to fix CVE-2007-2488/ASA-2007-013- Update to 1.4.4- Update to 1.4.2- Package the IAXy firmware - Minor clean-ups in files- Update to 1.4.1 - Don't build/package codec_zap (zaptel 1.4.0 doesn't support it)- Update to 1.4.0-beta4 - Various cleanups.- Don't package IAXy firmware because of license - Don't build app_rpt - Don't BR lm_sensors on PPC - Better way to prevent download/installation of sound archives - Redo tarball to eliminate non-free items- Remove explicit dependency on glibc-kernheaders. - Build jabber modules on PPC- *Really* update to beta3 - chan_jingle has been taken out of 1.4 - Move misplaced binaries to where they should be- Remove requirement on asterisk-sounds-core until licensing can be figured out.- Update to 1.4.0-beta3- Update to 1.4.0-beta2- Update to 1.2.10.- Update to 1.2.9.1- Update to 1.2.8 - Add misdn.conf to list of configs. - Drop chan_bluetooth patch for now...- Zaptel subpackage shouldn't obsolete the sqlite subpackage. - Remove mISDN until build issues can be figured out.- Build mISDN channel drivers, modelled after spec file from David Woodhouse- Update chan_bluetooth patch with some additional information as to it's source and comment out more in the configuration file.- Add chan_bluetooth- Split off more stuff into subpackages.- Update to 1.2.7- Fix detection of libpri on 64 bit arches (taken from Matthias Saou's rpmforge package) - Change sqlite subpackage name to sqlite2 (there are sqlite3 modules in development).- Don't build GTK 1.X console since GTK 1.X is being moved out of core...- Update to 1.2.6- Update to 1.2.5. - Removed upstreamed MOH patch. - Add full urls to the app_(r|t)xfax.c sources. - Update spandsp patch.- Actually apply the patch.- Add patch to keep Asterisk from crashing when using MOH inside a MeetMe conference.- BR sqlite2-devel- Update to 1.2.4.- Took some tricks from Asterisk packages by Roy-Magne Mo. - Enable gtk console module. - BR gtk+-devel. - Add logrotate script. - BR sqlite2-devel and new sqlite subpackage. - BR doxygen and graphviz for building duxygen documentation. (But don't build it yet.)- Completely eliminate the "asterisk" user from the spec file. - Move more config files to subpackages. - Consolidate two patches that patch the init script. - BR curl-devel - BR alsa-lib-devel - alsa, curl, oss subpackages- Do not run as user "asterisk" as that prevents setting of IP TOS (which is bad for quality of service). - Add patch for setting TOS separately for SIP and RTP packets.- First version for Fedora Extras.  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