Changelog for asterisk-voicemail-odbc-13.7.1-1.fc23.i686.rpm :

* Thu Feb 04 2016 Jared Smith - 13.7.1-1- 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
* Wed Feb 03 2016 Fedora Release Engineering - 13.3.2-3.1- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild
* Mon Jan 25 2016 Jared Smith - 13.3.2-3- Remove %defattr macro invocations, as they are no longer needed
* Sat Jan 23 2016 Robert Scheck - 13.3.2-2- Rebuild for libical 2.0.0
* Wed Jun 17 2015 Fedora Release Engineering - 13.3.2-1.2- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild
* Sat Jun 06 2015 Jitka Plesnikova - 13.3.2-1.1- Perl 5.22 rebuild
* Thu Apr 09 2015 Jeffrey C. Ollie - 13.3.2-1:- 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
* Thu Apr 09 2015 Jeffrey C. Ollie - 13.3.1-1:- 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
* Wed Apr 01 2015 Jeffrey C. Ollie - 13.3.0-1:- 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
* Wed Apr 01 2015 Jeffrey C. Ollie - 13.2.0-1:- 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
* Fri Jan 30 2015 Jeffrey C. Ollie - 13.1.1-1:- 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
* Fri Jan 30 2015 Jeffrey C. Ollie - 13.1.0-1- 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
* Thu Jan 29 2015 Peter Robinson 13.0.2-3- Add speexdsp as build dep as speex_echo.h has moved - rhbz 1181021
* Thu Jan 15 2015 Tom Callaway - 13.0.2-2- update for lua 5.3
* Wed Dec 10 2014 Jeffrey C. Ollie - 13.0.2-1- 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
* Thu Nov 20 2014 Jeffrey C. Ollie - 13.0.1-1- 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
* Thu Nov 20 2014 Jeffrey C. Ollie - 13.0.0-1- 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
* Fri Nov 14 2014 Tom Callaway - 11.13.1-2- rebuild for new libsrtp
* Mon Oct 20 2014 Jeffrey C. Ollie - 11.13.1-1- 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
* Mon Oct 20 2014 Jeffrey C. Ollie - 11.13.0-1- 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
* Mon Oct 20 2014 Jeffrey C. Ollie - 11.12.1-1- 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
* Mon Oct 20 2014 Jeffrey C. Ollie - 11.12.0-1- 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
* Mon Oct 20 2014 Jeffrey C. Ollie - 11.11.0-1- 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
* Thu Aug 28 2014 Jitka Plesnikova - 11.10.2-2.2- Perl 5.20 rebuild
* Fri Aug 15 2014 Fedora Release Engineering - 11.10.2-2.1- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild
* Thu Jun 19 2014 Jeffrey Ollie - 11.10.2-2:- Drop the 389 directory server schema (1061414)
* Thu Jun 19 2014 Jeffrey Ollie - 11.10.2-1:- 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
* Thu Jun 19 2014 Jeffrey Ollie - 11.10.1-1:- 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
* Thu Jun 19 2014 Jeffrey Ollie - 11.10.0-1:- 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
* Sat Jun 07 2014 Fedora Release Engineering - 11.9.0-2.1- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild
* Thu May 15 2014 Dennis Gilmore - 11.9.0-2- build against gmime-devel not gmime22-devel- do not use -m64 on aarch64
* Wed Apr 23 2014 Jeffrey Ollie - 11.9.0-1:- 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)
* Tue Mar 11 2014 Jeffrey Ollie - 11.8.1-1:- 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
* Tue Mar 04 2014 Jeffrey Ollie - 11.8.0-1:- 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
* Sat Dec 28 2013 Jeffrey Ollie - 11.7.0-1:- 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
* Sat Dec 28 2013 Jeffrey Ollie - 11.6.1-1:- 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
* Sat Dec 28 2013 Jeffrey Ollie - 11.6.0-1:- 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
* Mon Oct 21 2013 Jeffrey Ollie - 11.5.1-3:- Disable hardened build, as it\'s apparently causing problems loading modules.
* Thu Aug 29 2013 Jeffrey Ollie - 11.5.1-2:- Enable hardened build BZ#954338- Significant clean ups
* Thu Aug 29 2013 Jeffrey Ollie - 11.5.1-1:- 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
* Sat Aug 03 2013 Fedora Release Engineering - 11.4.0-2.2- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild
* Wed Jul 17 2013 Petr Pisar - 11.4.0-2.1- Perl 5.18 rebuild
* Fri May 24 2013 Rex Dieter 11.4.0-2- rebuild (libical)
* Mon May 20 2013 Jeffrey Ollie - 11.4.0-1:- 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
* Fri May 10 2013 Tom Callaway - 11.3.0-2:- fix build with lua 5.2
* Tue Apr 23 2013 Jeffrey Ollie - 11.3.0-1:- 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\
\'- (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
* Thu Mar 28 2013 Jeffrey Ollie - 11.2.2-1:- 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
* Sun Feb 10 2013 Jeffrey Ollie - 11.2.1-1:- 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
* Fri Jan 18 2013 Jeffrey Ollie - 11.2.0-1:- 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
* Thu Jan 03 2013 Jeffrey Ollie - 11.1.2-1:- 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!
* Wed Jan 02 2013 Jeffrey Ollie - 11.1.1-1:- 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
* Wed Dec 12 2012 Jeffrey Ollie - 11.1.0-1:- 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
* Fri Dec 07 2012 Jeffrey Ollie - 11.0.2-1:- 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
* Wed Dec 05 2012 Dan Horák - 11.0.1-3- simplify LDFLAGS setting
* Fri Nov 30 2012 Dennis Gilmore - 11.0.1-2- clean up things to allow building on arm arches
* Mon Nov 05 2012 Jeffrey Ollie - 11.0.1-1- 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
* Tue Oct 30 2012 Jeffrey Ollie - 11.0.0-1:- 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
* Wed Oct 17 2012 Jeffrey Ollie - 11.0.0-0.7.rc2:- 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
* Tue Oct 09 2012 Jeffrey Ollie - 11.0.0-0.6.rc1- 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
* Wed Sep 26 2012 Jeffrey Ollie - 11.0.0-0.5.beta2- Don\'t forget format_ilbc module
* Wed Sep 26 2012 Jeffrey Ollie - 11.0.0-0.4.beta2- 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
* Wed Sep 26 2012 Jeffrey Ollie - 10.8.0-1- 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
* Tue Sep 04 2012 Dan Horák - 11.0.0-0.3.beta1- fix build on s390
* Tue Sep 04 2012 Dan Horák - 10.7.1-2- fix build on s390
* Thu Aug 30 2012 Jeffrey Ollie - 10.7.1-1- 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
* Thu Aug 30 2012 Jeffrey Ollie - 10.7.0-1- 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
* Thu Aug 30 2012 Jeffrey Ollie - 10.6.1-1- 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
* Thu Aug 30 2012 Jeffrey Ollie - 10.6.0-1- 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
* Sat Aug 18 2012 Jeffrey Ollie - 11.0.0-0.2.beta1- 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
* Wed Jul 18 2012 Fedora Release Engineering - 10.5.2-1.2- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild
* Mon Jul 09 2012 Petr Pisar - 10.5.2-1.1- Perl 5.16 rebuild
* Thu Jul 05 2012 Jeffrey Ollie - 10.5.2-1:- 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
* Thu Jun 28 2012 Petr Pisar - 10.5.1-1.1- Perl 5.16 rebuild
* Fri Jun 15 2012 Jeffrey Ollie - 10.5.1-1- 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
* Fri Jun 15 2012 Jeffrey Ollie - 10.5.0-1- 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
* Mon Jun 11 2012 Petr Pisar - 10.4.2-1.1- Perl 5.16 rebuild
* Wed May 30 2012 Jeffrey Ollie - 10.4.2-1- 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
* Wed May 30 2012 Jeffrey Ollie - 10.4.1-1- 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
* Fri May 04 2012 Jeffrey Ollie - 10.4.0-1- 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
* Tue Apr 24 2012 Jeffrey Ollie - 10.3.1-1- 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
* Thu Mar 29 2012 Russell Bryant - 10.3.0-1- Update to 10.3.0
* Fri Mar 16 2012 Russell Bryant - 10.2.1-1- 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
* Thu Feb 16 2012 Jeffrey C. Ollie - 10.1.2-2-
* 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)
* Thu Feb 16 2012 Jeffrey C. Ollie - 10.1.2-1- 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
* Thu Feb 16 2012 Jeffrey C. Ollie - 10.1.1-1- 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
* Thu Feb 16 2012 Jeffrey C. Ollie - 10.1.0-1- 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
* Thu Feb 16 2012 Russell Bryant - 10.0.0-2- Remove asterisk-ais. OpenAIS was removed from Fedora.
* Thu Jan 12 2012 Fedora Release Engineering - 10.0.0-1.1- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild
* Tue Jan 03 2012 Jeffrey C. Ollie - 10.0.0-1- Don\'t build API docs as the build never finishes
* Thu Dec 15 2011 Jeffrey C. Ollie - 10.0.0-1- 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
* Fri Dec 09 2011 Jeffrey C. Ollie - 10.0.0-0.7.rc3- 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
* Wed Nov 16 2011 Jeffrey C. Ollie - 10.0.0-0.6.rc2- 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
* Fri Nov 11 2011 Jeffrey C. Ollie - 10.0.0-0.5.rc1- 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
* Tue Oct 18 2011 Jeffrey C. Ollie - 10.0.0-0.4.beta2- Add patch from upstream SVN to fix AST-2011-012
* Fri Oct 14 2011 Jeffrey C. Ollie - 10.0.0-0.3.beta2- Patch cleanup day
* Thu Sep 29 2011 Jeffrey C. Ollie - 10.0.0-0.2.beta2- 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
* Mon Jul 25 2011 Jeffrey C. Ollie - 10.0.0-0.1.beta1-- 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
* Thu Jul 21 2011 Petr Sabata - 1.8.5.0-1.2- Perl mass rebuild
* Wed Jul 20 2011 Petr Sabata - 1.8.5.0-1.1- Perl mass rebuild
* Mon Jul 11 2011 Jeffrey C. Ollie - 1.8.5.0-1- The Asterisk Development Team announces the release of Asterisk 1.8.5.0. This- release is available for immediate download at- http://downloads.asterisk.org/pub/telephony/asterisk/-- The release of Asterisk 1.8.5.0 resolves several issues reported by the- community and would have not been possible without your participation.- Thank you!-- The following is a sample of the issues resolved in this release:--
* Fix Deadlock with attended transfer of SIP call- (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,- cmaj)--
* Fixes thread blocking issue in the sip TCP/TLS implementation.- (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois,- rossbeer, kowalma, Freddi_Fonet)--
* Be more tolerant of what URI we accept for call completion PUBLISH requests.- (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)--
* Fix a nasty chanspy bug which was causing a channel leak every time a spied on- channel made a call.- (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)--
* This patch fixes a bug with MeetMe behavior where the \'P\' option for always- prompting for a pin is ignored for the first caller.- (Closes issue #18070. Reported by mav3rick. Patched by bbryant)--
* Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If- the call that the dialplan started an AGI script for is hungup while the AGI- script is in the middle of a command then the AGI script is not notified of- the hangup.- (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)--
* Resolve issue where leaving a voicemail, the MWI message is never sent. The- same thing happens when checking a voicemail and marking it as read.- (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard- Mudgett)--
* Resolve issue where wait for leader with Music On Hold allows crosstalk- between participants. Parenthesis in the wrong position. Regression from issue- #14365 when expanding conference flags to use 64 bits.- (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)-- For a full list of changes in this release, please see the ChangeLog:-- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5.0
* Thu Jul 07 2011 Jeffrey C. Ollie - 1.8.5-0.2- Rebuild for net-snmp 5.7
* Fri Jul 01 2011 Jeffrey C. Ollie - 1.8.5-0.1.rc1- Fix systemd dependencies in EL6 and F15
* Thu Jun 30 2011 Jeffrey C. Ollie - 1.8.5-0.1.rc1- The Asterisk Development Team has announced the first release candidate of- Asterisk 1.8.5. This release candidate is available for immediate download at- http://downloads.asterisk.org/pub/telephony/asterisk/-- The release of Asterisk 1.8.5-rc1 resolves several issues reported by the- community and would have not been possible without your participation.- Thank you!-- The following is a sample of the issues resolved in this release candidate:--
* Fix Deadlock with attended transfer of SIP call- (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,- cmaj)--
* Fixes thread blocking issue in the sip TCP/TLS implementation.- (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois,- rossbeer, kowalma, Freddi_Fonet)--
* Be more tolerant of what URI we accept for call completion PUBLISH requests.- (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)--
* Fix a nasty chanspy bug which was causing a channel leak every time a spied on- channel made a call.- (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)--
* This patch fixes a bug with MeetMe behavior where the \'P\' option for always- prompting for a pin is ignored for the first caller.- (Closes issue #18070. Reported by mav3rick. Patched by bbryant)--
* Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If- the call that the dialplan started an AGI script for is hungup while the AGI- script is in the middle of a command then the AGI script is not notified of- the hangup.- (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)--
* Resolve issue where leaving a voicemail, the MWI message is never sent. The- same thing happens when checking a voicemail and marking it as read.- (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard- Mudgett)--
* Resolve issue where wait for leader with Music On Hold allows crosstalk- between participants. Parenthesis in the wrong position. Regression from issue- #14365 when expanding conference flags to use 64 bits.- (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)--
* Fix timerfd locking issue.- (Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz)-- For a full list of changes in this release candidate, please see the ChangeLog:-- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5-rc1
* Thu Jun 30 2011 Jeffrey C. Ollie - 1.8.4.4-2- Fedora Directory Server -> 389 Directory Server
* Wed Jun 29 2011 Jeffrey C. Ollie - 1.8.4.4-1- The Asterisk Development Team has announced the release of Asterisk- versions 1.4.41.2, 1.6.2.18.2, and 1.8.4.4, which are security- releases.-- These releases are available for immediate download at- http://downloads.asterisk.org/pub/telephony/asterisk/releases-- The release of Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 resolves the- following issue:-- AST-2011-011: Asterisk may respond differently to SIP requests from an- invalid SIP user than it does to a user configured on the system, even- when the alwaysauthreject option is set in the configuration. This can- leak information about what SIP users are valid on the Asterisk- system.-- For more information about the details of this vulnerability, please- read the security advisory AST-2011-011, which was released at the- same time as this announcement.-- For a full list of changes in the current releases, please see the ChangeLog:-- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.41.2- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.2- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.4-- Security advisory AST-2011-011 is available at:-- http://downloads.asterisk.org/pub/security/AST-2011-011.pdf
* Mon Jun 27 2011 Jeffrey C. Ollie - 1.8.4.3-3- Don\'t forget stereorize
* Mon Jun 27 2011 Jeffrey C. Ollie - 1.8.4.3-2- Move /var/run/asterisk to /run/asterisk- Add comments to systemd service file on how to mimic safe_asterisk functionality- Build more of the optional binaries- Install the tmpfiles.d configuration on Fedora 15
* Fri Jun 24 2011 Jeffrey C. Ollie - 1.8.4.3-1- The Asterisk Development Team has announced the release of Asterisk versions- 1.4.41.1, 1.6.2.18.1, and 1.8.4.3, which are security releases.-- These releases are available for immediate download at- http://downloads.asterisk.org/pub/telephony/asterisk/releases-- The release of Asterisk 1.4.41.1, 1.6.2.18, and 1.8.4.3 resolves several issues- as outlined below:--
* AST-2011-008: If a remote user sends a SIP packet containing a null,- Asterisk assumes available data extends past the null to the- end of the packet when the buffer is actually truncated when- copied. This causes SIP header parsing to modify data past- the end of the buffer altering unrelated memory structures.- This vulnerability does not affect TCP/TLS connections.- -- Resolved in 1.6.2.18.1 and 1.8.4.3--
* AST-2011-009: A remote user sending a SIP packet containing a Contact header- with a missing left angle bracket (<) causes Asterisk to- access a null pointer.- -- Resolved in 1.8.4.3--
* AST-2011-010: A memory address was inadvertently transmitted over the- network via IAX2 via an option control frame and the remote party would try- to access it.- -- Resolved in 1.4.41.1, 1.6.2.18.1, and 1.8.4.3-- The issues and resolutions are described in the AST-2011-008, AST-2011-009, and- AST-2011-010 security advisories.-- For more information about the details of these vulnerabilities, please read- the security advisories AST-2011-008, AST-2011-009, and AST-2011-010, which were- released at the same time as this announcement.-- For a full list of changes in the current releases, please see the ChangeLog:-- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.41.1- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.1- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.3-- Security advisories AST-2011-008, AST-2011-009, and AST-2011-010 are available- at:-- http://downloads.asterisk.org/pub/security/AST-2011-008.pdf- http://downloads.asterisk.org/pub/security/AST-2011-009.pdf- http://downloads.asterisk.org/pub/security/AST-2011-010.pdf
* Tue Jun 21 2011 Jeffrey C. Ollie - 1.8.4.2-2- Convert to systemd
* Fri Jun 17 2011 Marcela Mašláňová - 1.8.4.2-1.2- Perl mass rebuild
* Fri Jun 10 2011 Marcela Mašláňová - 1.8.4.2-1.1- Perl 5.14 mass rebuild
* Fri Jun 03 2011 Jeffrey C. Ollie - 1.8.4.2-1:-- The Asterisk Development Team has announced the release of Asterisk- version 1.8.4.2, which is a security release for Asterisk 1.8.-- This release is available for immediate download at- http://downloads.asterisk.org/pub/telephony/asterisk/releases-- The release of Asterisk 1.8.4.2 resolves an issue with SIP URI- parsing which can lead to a remotely exploitable crash:-- Remote Crash Vulnerability in SIP channel driver (AST-2011-007)-- The issue and resolution is described in the AST-2011-007 security- advisory.-- For more information about the details of this vulnerability, please- read the security advisory AST-2011-007, which was released at the- same time as this announcement.-- For a full list of changes in the current release, please see the ChangeLog:-- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.2-- Security advisory AST-2011-007 is available at:-- http://downloads.asterisk.org/pub/security/AST-2011-007.pdf-- The Asterisk Development Team has announced the release of Asterisk 1.8.4.1.- This release is available for immediate download at- http://downloads.asterisk.org/pub/telephony/asterisk/-- The release of Asterisk 1.8.4.1 resolves several issues reported by the- community. Without your help this release would not have been possible.- Thank you!-- Below is a list of issues resolved in this release:--
* Fix our compliance with RFC 3261 section 18.2.2. (aka Cisco phone fix)- (Closes issue #18951. Reported by jmls. Patched by wdoekes)--
* Resolve a change in IPv6 header parsing due to the Cisco phone fix issue.- This issue was found and reported by the Asterisk test suite.- (Closes issue #18951. Patched by mnicholson)--
* Resolve potential crash when using SIP TLS support.- (Closes issue #19192. Reported by stknob. Patched by Chainsaw. Tested by- vois, Chainsaw)--
* Improve reliability when using SIP TLS.- (Closes issue #19182. Reported by st. Patched by mnicholson)--- For a full list of changes in this release candidate, please see the ChangeLog:-- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4.1- The Asterisk Development Team has announced the release of Asterisk 1.8.4. This- release is available for immediate download at- http://downloads.asterisk.org/pub/telephony/asterisk/-- The release of Asterisk 1.8.4 resolves several issues reported by the community.- Without your help this release would not have been possible. Thank you!-- Below is a sample of the issues resolved in this release:--
* Use SSLv23_client_method instead of old SSLv2 only.- (Closes issue #19095, #19138. Reported, patched by tzafrir. Tested by russell- and chazzam.--
* Resolve crash in ast_mutex_init()- (Patched by twilson)--
* Resolution of several DTMF based attended transfer issues.- (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo,- shihchuan, grecco. Patched by rmudgett)-- NOTE: Be sure to read the ChangeLog for more information about these changes.--
* Resolve deadlocks related to device states in chan_sip- (Closes issue #18310. Reported, patched by one47. Patched by jpeeler)--
* Resolve an issue with the Asterisk manager interface leaking memory when- disabled.- (Reported internally by kmorgan. Patched by russellb)--
* Support greetingsfolder as documented in voicemail.conf.sample.- (Closes issue #17870. Reported by edhorton. Patched by seanbright)--
* Fix channel redirect out of MeetMe() and other issues with channel softhangup- (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb.- Patched by russellb)--
* Fix voicemail sequencing for file based storage.- (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by- jpeeler)--
* Set hangup cause in local_hangup so the proper return code of 486 instead of- 503 when using Local channels when the far sides returns a busy. Also affects- CCSS in Asterisk 1.8+.- (Patched by twilson)--
* Fix issues with verbose messages not being output to the console.- (Closes issue #18580. Reported by pabelanger. Patched by qwell)--
* Fix Deadlock with attended transfer of SIP call- (Closes issue #18837. Reported, patched by alecdavis. Tested by- alecdavid, Irontec, ZX81, cmaj)-- Includes changes per AST-2011-005 and AST-2011-006- For a full list of changes in this release candidate, please see the ChangeLog:-- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4-- Information about the security releases are available at:-- http://downloads.asterisk.org/pub/security/AST-2011-005.pdf- http://downloads.asterisk.org/pub/security/AST-2011-006.pdf
* Thu Apr 21 2011 Jeffrey C. Ollie - 1.8.3.3-1- 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
* Wed Mar 23 2011 Jeffrey C. Ollie - 1.8.3.2-2- Bump release and rebuild for mysql 5.5.10 soname change.
* Thu Mar 17 2011 Jeffrey C. Ollie - 1.8.3.2-1- The Asterisk Development Team has announced security releases for Asterisk- branches 1.6.1, 1.6.2, and 1.8. The available security releases are- released as versions 1.6.1.24, 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
* Thu Mar 17 2011 Jeffrey C. Ollie - 1.8.3.1-1- The Asterisk Development Team has announced security releases for Asterisk- branches 1.6.1, 1.6.2, and 1.8. The available security releases are- released as versions 1.6.1.23, 1.6.2.17.1, and 1.8.3.1.-- 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
* Mon Feb 28 2011 - 1.8.3-1- 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
* Wed Feb 16 2011 - 1.8.3-0.7.rc3-- 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
* Fri Feb 11 2011 Jeffrey C. Ollie - 1.8.3-0.6.rc2- Bump release to build for F15
* Wed Feb 09 2011 Jeffrey C. Ollie - 1.8.3-0.5.rc2- Remove isa macros
* Wed Feb 09 2011 Jeffrey C. Ollie - 1.8.3-0.4.rc2- Make library dependencies architecture specific
* Mon Feb 07 2011 Fedora Release Engineering - 1.8.3-0.3.rc2- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild
* Wed Jan 26 2011 Jeffrey C. Ollie - 1.8.3-0.2.rc2The Asterisk Development Team has announced the second release candidate ofAsterisk 1.8.3. This release candidate is available for immediate download athttp://downloads.asterisk.org/pub/telephony/asterisk/The release of Asterisk 1.8.3-rc2 resolves the following issues in addition tothose 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
* Wed Jan 26 2011 Jeffrey C. Ollie - 1.8.3-0.1.rc1-- 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)
* Wed Jan 26 2011 Jeffrey C. Ollie - 1.8.2.3-1-- 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
* Mon Jan 24 2011 Jeffrey C. Ollie - 1.8.2.2-2- Build with SRTP support
* Mon Jan 24 2011 Jeffrey C. Ollie - 1.8.2.2-1-- 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
* Mon Jan 24 2011 Jeffrey C. Ollie - 1.8.2.1-1-- 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
* Mon Jan 24 2011 Jeffrey C. Ollie - 1.8.2-1-- 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
* Mon Jan 24 2011 Jeffrey C. Ollie - 1.8.1.1-1-- 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
* Mon Jan 24 2011 Jeffrey C. Ollie - 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
* Tue Jan 18 2011 Dennis Gilmore - 1.8.0-6- dont package up the ices bits on el the client doesnt exist for us
* Tue Jan 18 2011 Dennis Gilmore - 1.8.0-5- dont build the 389 directory server package its not available on rhel6
* Fri Dec 10 2010 Dennis Gilmore - 1.8.0-4- dont always build AIS modules we dont have the BuildRequires on epel
* Fri Oct 29 2010 Jeffrey C. Ollie - 1.8.0-3- Rebuild for new net-snmp.
* Tue Oct 26 2010 Jeffrey C. Ollie - 1.8.0-2- Always build AIS modules
* Thu Oct 21 2010 Jeffrey C. Ollie - 1.8.0-1- 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!
* Mon Oct 18 2010 Jeffrey C. Ollie - 1.8.0-0.8.rc5:-- 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
* Fri Oct 08 2010 Jeffrey C. Ollie - 1.8.0-0.7.rc3- 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
* Wed Oct 06 2010 Jeffrey C. Ollie - 1.8.0-0.6.rc2- 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
* Thu Sep 09 2010 Jeffrey C. Ollie - 1.8.0-0.5.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 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
* Tue Aug 24 2010 Jeffrey C. Ollie - 1.8.0-0.4.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 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
* Wed Aug 11 2010 Jeffrey C. Ollie - 1.8.0-0.3.beta3-- 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
* Mon Aug 02 2010 Jeffrey C. Ollie - 1.8.0-0.2.beta2- Rebuild against libpri 1.4.12
* Mon Aug 02 2010 Jeffrey C. Ollie - 1.8.0-0.1.beta2- 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 :(
* Sat Jul 31 2010 Jeffrey C. Ollie - 1.6.2.10-1-- 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
* Wed Jul 14 2010 Jeffrey C. Ollie - 1.6.2.8-0.3.rc1- Add patch to remove requirement on latex2html
* Tue Jun 01 2010 Marcela Maslanova - 1.6.2.8-0.2.rc1- Mass rebuild with perl-5.12.0
* Tue May 04 2010 Jeffrey C. Ollie - 1.6.2.7-1-
* 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)
* Mon May 03 2010 Jeffrey C. Ollie - 1.6.2.7-0.2.rc3- Update to 1.6.2.7-rc3
* Thu Apr 15 2010 Jeffrey C. Ollie - 1.6.2.7-0.1.rc2- Update to 1.6.2.7-rc2
* Fri Mar 12 2010 Jeffrey C. Ollie - 1.6.2.6-1- 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
* Mon Mar 08 2010 Jeffrey C. Ollie - 1.6.2.6-0.1.rc2- Update to 1.6.2.6-rc2
* Mon Mar 08 2010 Jeffrey C. Ollie - 1.6.2.5-2- Add a patch that fixes CLI history when linking against external libedit.
* Thu Feb 25 2010 Jeffrey C. Ollie - 1.6.2.5-1- Update to 1.6.2.5--
* AST-2010-002: Invalid parsing of ACL rules can compromise security
* Thu Feb 18 2010 Jeffrey C. Ollie - 1.6.2.4-1- 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.
* Wed Feb 03 2010 Jeffrey C. Ollie - 1.6.2.2-1- 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.
* Fri Jan 15 2010 Jeffrey C. Ollie - 1.6.2.1-1- 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.
* Tue Jan 12 2010 Jeffrey C. Ollie - 1.6.2.1-0.1.rc1- Update to 1.6.2.1-rc1
* Sat Dec 19 2009 Jeffrey C. Ollie - 1.6.2.0-1- Released version of 1.6.2.0
* Wed Dec 09 2009 Jeffrey C. Ollie - 1.6.2.0-0.16.rc8- Update to 1.6.2.0-rc8
* Wed Dec 02 2009 Jeffrey C. Ollie - 1.6.2.0-0.15.rc7- Update to 1.6.2.0-rc7
* Tue Dec 01 2009 Jeffrey C. Ollie - 1.6.2.0-0.14.rc6- Change the logrotate and the init scripts so that Asterisk doesn\'t try and write to / or /root
* Thu Nov 19 2009 Jeffrey C. Ollie - 1.6.2.0-0.13.rc6- Make dependency on uw-imap conditional and some other changes to make building on RHEL5 easier.
* Fri Nov 13 2009 Jeffrey C. Ollie - 1.6.2.0-0.12.rc6- Update to 1.6.2.0-rc6
* Mon Nov 09 2009 Jeffrey C. Ollie - 1.6.2.0-0.11.rc5- Update to 1.6.2.0-rc5
* Fri Nov 06 2009 Jeffrey C. Ollie - 1.6.2.0-0.10.rc4- Update to 1.6.2.0-rc4
* Tue Oct 27 2009 Jeffrey C. Ollie - 1.6.2.0-0.9.rc3- Add patch from upstream to fix how res_http_post forms paths.
* Sat Oct 24 2009 Jeffrey C. Ollie - 1.6.2.0-0.8.rc3- Add an AST_EXTRA_ARGS option to the init script- have the init script to cd to /var/spool/asterisk to prevent annoying message
* Sat Oct 24 2009 Jeffrey C. Ollie - 1.6.2.0-0.7.rc3- 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.
* Fri Oct 09 2009 Jeffrey C. Ollie - 1.6.2.0-0.6.rc3- Require latex2html used in static-http documents
* Wed Oct 07 2009 Jeffrey C. Ollie - 1.6.2.0-0.5.rc3- Change ownership and permissions on config files to protect them.
* Tue Oct 06 2009 Jeffrey C. Ollie - 1.6.2.0-0.4.rc3- Update to 1.6.2.0-rc3
* Wed Sep 30 2009 Jeffrey C. Ollie - 1.6.2.0-0.3.rc2- Merge firmware subpackage back into the main package.
* Wed Sep 30 2009 Jeffrey C. Ollie - 1.6.2.0-0.2.rc2- Package internal help.- Fix up some more paths in the configs so that everything ends up where we want them.
* Wed Sep 30 2009 Jeffrey C. Ollie - 1.6.2.0-0.1.rc2- Update to 1.6.2.0-rc2- We no longer need to strip the tarball as it no longer includes non-free items.
* Wed Sep 09 2009 Jeffrey C. Ollie - 1.6.1.6-2- Enable building of API docs.- Depend on version 1.2 or newer of speex
* Sun Sep 06 2009 Jeffrey C. Ollie - 1.6.1.6-1- Update to 1.6.1.6- Drop patches that are too troublesome to maintain anymore or have been integrated upstream.
* Tue Sep 01 2009 Jeffrey C. Ollie - 1.6.1-0.26.rc1- Add a patch from Quentin Armitage and rebuld.
* Fri Aug 21 2009 Tomas Mraz - 1.6.1-0.25.rc1- rebuilt with new openssl
* Fri Jul 24 2009 Fedora Release Engineering - 1.6.1-0.24.rc1- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild
* Thu Mar 05 2009 Jeffrey C. Ollie - 1.6.1-0.23.rc1- 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.
* Mon Feb 23 2009 Fedora Release Engineering - 1.6.1-0.22.rc1- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild
* Sun Feb 08 2009 Jeffrey C. Ollie - 1.6.1-0.21.rc1- 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
* Thu Jan 15 2009 Tomas Mraz - 1.6.1-0.13.beta4- rebuild with new openssl
* Sun Jan 04 2009 Jeffrey C. Ollie - 1.6.1-0.12.beta4- Fedora Directory Server compatibility patch/subpackage.
* Sun Jan 04 2009 Jeffrey C. Ollie - 1.6.1-0.10.beta4- Fix up paths. BZ#477238
* Sat Jan 03 2009 Jeffrey C. Ollie - 1.6.1-0.9.beta4- Update patches
* Sat Jan 03 2009 Jeffrey C. Ollie - 1.6.1-0.8.beta4- Update to 1.6.1-beta4
* Tue Dec 09 2008 Jeffrey C. Ollie - 1.6.1-0.7.beta3- Update to 1.6.1-beta3
* Tue Dec 09 2008 Alex Lancaster - 1.6.1-0.6.beta2- Rebuild for new gmime
* Fri Nov 07 2008 Jeffrey C. Ollie - 1.6.1-0.5.beta2- Add patch to fix missing variable on PPC.
* Fri Nov 07 2008 Jeffrey C. Ollie - 1.6.1-0.4.beta2- Update PPC systems don\'t have sys/io.h patch.
* Fri Nov 07 2008 Jeffrey C. Ollie - 1.6.1-0.3.beta2- PPC systems don\'t have sys/io.h
* Fri Nov 07 2008 Jeffrey C. Ollie - 1.6.1-0.2.beta2- Update to 1.6.1 beta 2
* Wed Nov 05 2008 Jeffrey C. Ollie - 1.6.0.1-3- Fix issue with init script giving wrong path to config file.
* Thu Oct 16 2008 Jeffrey C. Ollie - 1.6.0.1-2- Explicitly require dahdi-tools-libs to see if we can get this to build.
* Fri Oct 10 2008 Jeffrey C. Ollie - 1.6.0-1- Update to final release.
* Thu Sep 11 2008 - Bastien Nocera - 1.6.0-0.22.beta9- Rebuild
* Wed Jul 30 2008 Jeffrey C. Ollie - 1.6.0-0.21.beta9- Replace app_rxfax/app_txfax with app_fax taken from upstream SVN.
* Tue Jul 29 2008 Jeffrey C. Ollie - 1.6.0-0.20.beta9- Bump release and rebuild with new libpri and zaptel.
* Fri Jul 25 2008 Jeffrey C. Ollie - 1.6.0-0.19.beta9- Add patch pulled from upstream SVN that fixes AST-2008-010 and AST-2008-011.
* Fri Jul 25 2008 Jeffrey C. Ollie - 1.6.0-0.18.beta9- Add patch for LDAP extracted from upstream SVN (#442011)
* Wed Jul 02 2008 Jeffrey C. Ollie - 1.6.0-0.17.beta9- Add patch that unbreaks cdr_tds with FreeTDS 0.82.- Properly obsolete conference subpackage.
* Thu Jun 12 2008 Jeffrey C. Ollie - 1.6.0-0.16.beta9- Disable building cdr_tds since new FreeTDS in rawhide no longer provides needed library.
* Wed Jun 11 2008 Jeffrey C. Ollie - 1.6.0-0.15.beta9- Bump release and rebuild to fix libtds breakage.
* Mon May 19 2008 Jeffrey C. Ollie - 1.6.0-0.14.beta9- 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.
* Tue Apr 22 2008 Jeffrey C. Ollie - 1.6.0-0.13.beta8- Update to 1.6.0-beta8- Contains fixes for AST-2008-006 / CVE-2008-1897
* Wed Apr 02 2008 Jeffrey C. Ollie - 1.6.0-0.12.beta7.1- Return to stripped tarballs since there\'s more non-free content in the Asterisk tarballs than I thought.
* Sun Mar 30 2008 Jeffrey C. Ollie - 1.6.0-0.11.beta7.1- Update to 1.6.0-beta7.1- Update patches- Back out some changes that were made because beta7 was tagged from the wrong branch.
* Fri Mar 28 2008 Jeffrey C. Ollie - 1.6.0-0.10.beta7- 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.
* Thu Mar 27 2008 Jeffrey C. Ollie - 1.6.0-0.9.beta6- Add Postgresql schemas from contrib as documentation to the Postgresql subpackage.
* Tue Mar 25 2008 Jeffrey C. Ollie - 1.6.0-0.8.beta6- 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.
* Wed Mar 19 2008 Jeffrey C. Ollie - 1.6.0-0.6.beta6- 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
* Tue Mar 18 2008 Tom \"spot\" Callaway - 1.6.0-0.5.beta5- add Requires for versioned perl (libperl.so)
* Wed Mar 05 2008 Jeffrey C. Ollie - 1.6.0-0.4.beta5- Update to 1.6.0-beta5- Remove upstreamed patches.
* Mon Mar 03 2008 Jeffrey C. Ollie - 1.6.0-0.3.beta4- Package the directory used to store monitor recordings.
* Tue Feb 26 2008 Jeffrey C. Ollie - 1.6.0-0.2.beta4- Add patch from David Woodhouse that fixes building on PPC64.
* Tue Feb 26 2008 Jeffrey C. Ollie - 1.6.0-0.1.beta4- Update to 1.6.0 beta 4
* Wed Feb 13 2008 Jeffrey C. Ollie - 1.4.18-1- 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.
* Wed Jan 02 2008 Jeffrey C. Ollie - 1.4.17-1- Update to 1.4.17 to fix AST-2008-001.
* Fri Dec 28 2007 Jeffrey C. Ollie - 1.4.16.2-1- Update to 1.4.16.2
* Sat Dec 22 2007 Jeffrey C. Ollie - 1.4.16.1-2- Bump release and rebuild to fix broken dep on uw-imap.
* Wed Dec 19 2007 Jeffrey C. Ollie - 1.4.16.1-1- Update to the real 1.4.16.1.
* Wed Dec 19 2007 Jeffrey C. Ollie - 1.4.16-2- 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.
* Tue Dec 18 2007 Jeffrey C. Ollie - 1.4.16-1- Update to 1.4.16 to fix security bug.
* Sat Dec 15 2007 Jeffrey C. Ollie - 1.4.15-7- Really, really fix the build problems on devel.
* Sat Dec 15 2007 Jeffrey C. Ollie - 1.4.15-6- Tweaks to get to build on x86_64
* Wed Dec 12 2007 Jeffrey C. Ollie - 1.4.15-5- Exclude PPC64
* Wed Dec 12 2007 Jeffrey C. Ollie - 1.4.15-4- Don\'t build apidocs by default since there\'s a problem building on x86_64.
* Tue Dec 11 2007 Jeffrey C. Ollie - 1.4.15-3- Really get rid of zero length map files.
* Mon Dec 10 2007 Jeffrey C. Ollie - 1.4.15-2- Get rid of zero length map files.- Shorten descriptions of voicemail subpackages
* Fri Nov 30 2007 Jeffrey C. Ollie - 1.4.15-1- Update to 1.4.15
* Tue Nov 20 2007 Jeffrey C. Ollie - 1.4.14-2- Fix license and other rpmlint warnings.
* Mon Nov 19 2007 Jeffrey C. Ollie - 1.4.14-1- Update to 1.4.14
* Fri Nov 16 2007 Jeffrey C. Ollie - 1.4.13-7- Add chan_mobile
* Tue Nov 13 2007 Jeffrey C. Ollie - 1.4.13-6- 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
* Wed Oct 10 2007 Jeffrey C. Ollie - 1.4.13-1- Update to 1.4.13
* Tue Oct 09 2007 Jeffrey C. Ollie - 1.4.12.1-1- Update to 1.4.12.1
* Wed Aug 22 2007 Jeffrey C. Ollie - 1.4.11-1- Update to 1.4.11
* Fri Aug 10 2007 Jeffrey C. Ollie - 1.4.10.1-1- Update to 1.4.10.1.
* Tue Aug 07 2007 Jeffrey C. Ollie - 1.4.10-1- Update to 1.4.10 (security update).
* Tue Aug 07 2007 Jeffrey C. Ollie - 1.4.9-7- Add a patch that allows alternate extensions to be defined in users.conf
* Mon Aug 06 2007 Jeffrey C. Ollie - 1.4.9-6- Update app_conference patch. Enter/leave sounds are now possible.
* Fri Jul 27 2007 Jeffrey C. Ollie - 1.4.9-5- 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.
* Thu Jul 26 2007 Jeffrey C. Ollie - 1.4.9-4- Don\'t build app_mp3
* Wed Jul 25 2007 Jeffrey C. Ollie - 1.4.9-3- Add app_conference
* Wed Jul 25 2007 Jeffrey C. Ollie - 1.4.9-2- Use plain useradd/groupadd rather than the fedora-usermgmt- Clean up requirements- Clean up build requirements by moving them to package sections
* Tue Jul 24 2007 Jeffrey C. Ollie - 1.4.9-1- Update to 1.4.9
* Tue Jul 17 2007 Jeffrey C. Ollie - 1.4.8-1- Update to 1.4.8- Drop ixjuser patch.
* Tue Jul 10 2007 Jeffrey C. Ollie - 1.4.7.1-1- Update to 1.4.7.1
* Mon Jul 09 2007 Jeffrey C. Ollie - 1.4.7-1- Update to 1.4.7- RxFAX/TxFAX applications
* Sun Jul 01 2007 Jeffrey C. Ollie - 1.4.6-4- It\'s \"sbin\", not \"bin\" silly.
* Sat Jun 30 2007 Jeffrey C. Ollie - 1.4.6-3- Add patch that lets us change TOS bits even when running non-root
* Fri Jun 29 2007 Jeffrey C. Ollie - 1.4.6-2- voicemail needs to require /usr/bin/sox and /usr/bin/sendmail
* Fri Jun 29 2007 Jeffrey C. Ollie - 1.4.6-1- Update to 1.4.6- Remove upstreamed patch.
* Thu Jun 21 2007 Jeffrey C. Ollie - 1.4.5-10- 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.
* Mon Jun 18 2007 Jeffrey C. Ollie - 1.4.5-9- Move sounds
* Mon Jun 18 2007 Jeffrey C. Ollie - 1.4.5-8- Update some more ownership/permissions
* Mon Jun 18 2007 Jeffrey C. Ollie - 1.4.5-7- Fix some permissions.
* Mon Jun 18 2007 Jeffrey C. Ollie - 1.4.5-6- Update init script patch- Move pid file to subdir of /var/run
* Mon Jun 18 2007 Jeffrey C. Ollie - 1.4.5-5- Update init script patch to run as non-root
* Sun Jun 17 2007 Jeffrey C. Ollie - 1.4.5-4- Build modules that depend on FreeTDS.- Don\'t build voicemail with ODBC storage.
* Sun Jun 17 2007 Jeffrey C. Ollie - 1.4.5-3- Have the build output the commands executing, rather than covering them up.
* Fri Jun 15 2007 Jeffrey C. Ollie - 1.4.5-1- Update to 1.4.5- Remove upstreamed patch.
* Wed May 09 2007 Jeffrey C. Ollie - 1.4.4-2- Add a patch to fix CVE-2007-2488/ASA-2007-013
* Fri Apr 27 2007 Jeffrey C. Ollie - 1.4.4-1- Update to 1.4.4
* Wed Mar 21 2007 Jeffrey C. Ollie - 1.4.2-1- Update to 1.4.2
* Tue Mar 06 2007 Jeffrey C. Ollie - 1.4.1-2- Package the IAXy firmware- Minor clean-ups in files
* Mon Mar 05 2007 Jeffrey C. Ollie - 1.4.1-1- Update to 1.4.1- Don\'t build/package codec_zap (zaptel 1.4.0 doesn\'t support it)
* Fri Dec 15 2006 Jeffrey C. Ollie - 1.4.0-6.beta4- Update to 1.4.0-beta4- Various cleanups.
* Fri Oct 20 2006 Jeffrey C. Ollie - 1.4.0-5.beta3- 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
* Thu Oct 19 2006 Jeffrey C. Ollie - 1.4.0-4.beta3- Remove explicit dependency on glibc-kernheaders.- Build jabber modules on PPC
* Wed Oct 18 2006 Jeffrey C. Ollie - 1.4.0-3.beta3-
*Really
* update to beta3- chan_jingle has been taken out of 1.4- Move misplaced binaries to where they should be
* Wed Oct 18 2006 Jeffrey C. Ollie - 1.4.0-2.beta3- Remove requirement on asterisk-sounds-core until licensing can be figured out.
* Wed Oct 18 2006 Jeffrey C. Ollie - 1.4.0-1.beta3- Update to 1.4.0-beta3
* Sun Oct 15 2006 Jeffrey C. Ollie - 1.4.0-0.beta2- Update to 1.4.0-beta2
* Tue Jul 25 2006 Jeffrey C. Ollie - 1.2.10-1- Update to 1.2.10.
* Wed Jun 07 2006 Jeffrey C. Ollie - 1.2.9.1- Update to 1.2.9.1
* Fri Jun 02 2006 Jeffrey C. Ollie - 1.2.8- Update to 1.2.8- Add misdn.conf to list of configs.- Drop chan_bluetooth patch for now...
* Tue May 02 2006 Jeffrey C. Ollie - 1.2.7.1-6- Zaptel subpackage shouldn\'t obsolete the sqlite subpackage.- Remove mISDN until build issues can be figured out.
* Mon Apr 24 2006 Jeffrey C. Ollie - 1.2.7.1-5- Build mISDN channel drivers, modelled after spec file from David Woodhouse
* Thu Apr 20 2006 Jeffrey C. Ollie - 1.2.7.1-4- Update chan_bluetooth patch with some additional information as to it\'s source and comment out more in the configuration file.
* Thu Apr 20 2006 Jeffrey C. Ollie - 1.2.7.1-3- Add chan_bluetooth
* Wed Apr 19 2006 Jeffrey C. Ollie - 1.2.7.1-2- Split off more stuff into subpackages.
* Wed Apr 12 2006 Jeffrey C. Ollie - 1.2.7-1- Update to 1.2.7
* Mon Apr 10 2006 Jeffrey C. Ollie - 1.2.6-3- 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).
* Thu Apr 06 2006 Jeffrey C. Ollie - 1.2.6-2- Don\'t build GTK 1.X console since GTK 1.X is being moved out of core...
* Mon Mar 27 2006 Jeffrey C. Ollie - 1.2.6-1- Update to 1.2.6
* Mon Mar 06 2006 Jeffrey C. Ollie - 1.2.5-1- Update to 1.2.5.- Removed upstreamed MOH patch.- Add full urls to the app_(r|t)xfax.c sources.- Update spandsp patch.
* Mon Feb 13 2006 Jeffrey C. Ollie - 1.2.4-4- Actually apply the patch.
* Mon Feb 13 2006 Jeffrey C. Ollie - 1.2.4-3- Add patch to keep Asterisk from crashing when using MOH inside a MeetMe conference.
* Mon Feb 06 2006 Jeffrey C. Ollie - 1.2.4-2- BR sqlite2-devel
* Tue Jan 31 2006 Jeffrey C. Ollie - 1.2.4-1- Update to 1.2.4.
* Wed Jan 25 2006 Jeffrey C. Ollie - 1.2.3-4- 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.)
* Wed Jan 25 2006 Jeffrey C. Ollie - 1.2.3-3- 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
* Wed Jan 25 2006 Jeffrey C. Ollie - 1.2.3-2- 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.
* Wed Jan 25 2006 Jeffrey C. Ollie - 1.2.3-1- First version for Fedora Extras.