--- - branch: MAIN date: Thu Sep 23 23:30:38 UTC 2010 files: - new: '1.21' old: '1.20' path: pkgsrc/comms/asterisk16/Makefile pathrev: pkgsrc/comms/asterisk16/Makefile@1.21 type: modified - new: '1.15' old: '1.14' path: pkgsrc/comms/asterisk16/PLIST pathrev: pkgsrc/comms/asterisk16/PLIST@1.15 type: modified - new: '1.16' old: '1.15' path: pkgsrc/comms/asterisk16/distinfo pathrev: pkgsrc/comms/asterisk16/distinfo@1.16 type: modified - new: '1.8' old: '1.7' path: pkgsrc/comms/asterisk16/options.mk pathrev: pkgsrc/comms/asterisk16/options.mk@1.8 type: modified - new: '1.7' old: '1.6' path: pkgsrc/comms/asterisk16/patches/patch-aa pathrev: pkgsrc/comms/asterisk16/patches/patch-aa@1.7 type: modified - new: '1.7' old: '1.6' path: pkgsrc/comms/asterisk16/patches/patch-aq pathrev: pkgsrc/comms/asterisk16/patches/patch-aq@1.7 type: modified - new: '1.3' old: '1.2' path: pkgsrc/comms/asterisk16/patches/patch-al pathrev: pkgsrc/comms/asterisk16/patches/patch-al@1.3 type: modified - new: '1.3' old: '1.2' path: pkgsrc/comms/asterisk16/patches/patch-ba pathrev: pkgsrc/comms/asterisk16/patches/patch-ba@1.3 type: modified - new: '1.3' old: '1.2' path: pkgsrc/comms/asterisk16/patches/patch-bd pathrev: pkgsrc/comms/asterisk16/patches/patch-bd@1.3 type: modified - new: '1.2' old: 1.1.1.1 path: pkgsrc/comms/asterisk16/patches/patch-au pathrev: pkgsrc/comms/asterisk16/patches/patch-au@1.2 type: modified - new: '1.5' old: '1.4' path: pkgsrc/comms/asterisk16/patches/patch-ax pathrev: pkgsrc/comms/asterisk16/patches/patch-ax@1.5 type: modified id: 20100923T233038Z.3b7a23e5b4e6ce0926f063ea64063a0d6af2be52 log: |2 Update to the 1.6.2 series (specifically 1.6.2.13). This is a feature update, so users that are upgrading should read UPDATE.txt. pkgsrc changes: - update to 1.6.2.13 - bury the asterisk-sounds-extra inside this one to keep it in sync - handle sound tarballs directly (upstream had changed this to do a download during the install phase and dump files in $HOME) - add new documentation files: - asterisk.txt - building_queues.txt - database_transactions.txt - followme.txt ======== 1.6.2.13 ======== This release resolves an issue where the .version and ChangeLog files were not updated for 1.6.2.12. Asterisk 1.6.2.13 has no additional changes from 1.6.2.12 other than the .version, ChangeLog and summary files. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.13 ======== 1.6.2.12 ======== The release of Asterisk 1.6.2.12 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 DNID does not get cleared on a new call when using immediate=yes with ISDN signaling. (Closes issue #17568. Reported by wuwu. Patched by rmudgett) * Several updates to res_config_ldap. (Closes issue #13573. Reported by navkumar. Patched by navkumar, bencer. Tested by suretec) * Prevent loss of Caller ID information set on local channel after masquerade. (Closes issue #17138. Reported by kobaz, patched by jpeeler) * Fix SIP peers memory leak. (Closes issue #17774. Reported, patched by kkm) * Add Danish support to say.conf.sample (Closes issue #17836. Reported, patched by RoadKill) * Ensure SSRC is changed when media source is changed to resolve audio delay. (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler) * Only do magic pickup when notifycid is enabled. A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber that a device is ringing. This option should only be enabled when the new 'notifycid' option is set, but this was not the case. Instead the call-id value was included for every RINGING Notify message, which caused a regression for people who used other methods for call pickup. (Closes issue #17633. Reported, patched by urosh. Patched by dvossel. Tested by: dvossel, urosh, okrief, alecdavis) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.12 ======== 1.6.2.11 ======== The release of Asterisk 1.6.2.11 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Send DialPlanComplete as a response, not as a separate event. Otherwise, it goes to all manager sessions and may exclude the current session, if the Events mask excludes it. (Closes issue #17504. Reported, patched by rrb3942) * Allow the "useragent" value to be restored into memory from the realtime backend. This value is purely informational. It does not alter configuration at all. (Closes issue #16029. Reported, patched by Guggemand) * Fix rt(c)p set debug ip taking wrong argument Also clean up some coding errors. (Closes issue #17469. Reported, patched by wdoekes) * Ensure channel placed in meetme in ringing state is properly hung up. An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed. (Closes issue #15871. Reported, patched by Ivan) * Correct how 100, 200, 300, etc. is said. Also add the crazy British numbers. (Closes issue #16102. Reported, patched by Delvar) * cdr_pgsql does not detect when a table is found. This change adds an ERROR message to let you know when a failure exists to get the columns from the pgsql database, which typically means that the table does not exist. (Closes issue #17478. Reported, patched by kobaz) * Avoid crashing when installing a duplicate translation path with a lower cost. (Closes issue #17092. Reported, patched by moy) * Add missing handling for ringing state for use with queue empty options. (Closes issue #17471. Reported, patched by jazzy) * Fix reporting estimated queue hold time. Just say the number of seconds (after minutes) rather than doing some incorrect calculation with respect to minutes. (Closes issue #17498. Reported, patched by corruptor) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.11 ======== 1.6.2.10 ======== The release of Asterisk 1.6.2.10 resolves several issues reported by the community and would have not been possible without your participation. Thank you! 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 ======= 1.6.2.9 ======= The release of Asterisk 1.6.2.9 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Fix the PickupChan() application (Closes issue #16863. Reported, patched by schern. Patched by cjacobsen. Tested by Graber, cjacobsen, lathama, rickead2000, dvossel) * Improve logging by displaying line number (Closes issue #16303. Reported by dant. Patched by pabelanger. Tested by dant, pabelanger, lmadsen) * Notify CLI when modules are loaded/unloaded (Closes issue #17308. Reported, patched by pabelanger. Tested by russell) * Make the Makefile logic more explicit and move the Snow Leopard logic down to where it's not executed on non-Darwin systems (Closes issue #17028. Reported by pabelanger. Patched by seanbright, tilghman. Tested by pabelanger) * Manager cookies are not compatible with RFC2109. Make that no longer true. (Closes issue #17231. Reported, patched by ecarruda) * With IMAP backend, messages in INBOX were counted twice for MWI (Closes issue #17135. Reported by edhorton. Patched by ebroad, tilghman) * Fix possible segfault when logging (Closes issue #17331. Reported, patched by under. Patched by dvossel) * Fix memory hogging behavior of app_queue (Closes issue #17081. Reported by wliegel. Patched by mmichelson) * Allow type=user SIP endpoints to be loaded properly from realtime (Closes issue #16021. Reported, patched by Guggemand) Additionally, the following issue may be of interest: * Fix transcode_via_sln option with SIP calls and improve PLC usage (Review: https://reviewboard.asterisk.org/r/622/) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9 ======= 1.6.2.8 ======= The release of Asterisk 1.6.2.8 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Enable auto complete for CLI command 'logger set level'. (Closes issue #17152. Reported, patched by pabelanger) * Make the mixmonitor thread process audio frames faster. (Closes issue #17078. Reported, tested by geoff2010. Patched by dhubbard) * Add missing 'useragent' field to sip-friends.sql file. (Closes issue #17171. Reported, patched by thehar) * Add example dialplan for dialing ISN numbers (http://www.freenum.org) (Closes issue #17058. Reported, patched by pprindeville) * Fix issue with double "sip:" in header field. (Closes issue #15847. Reported, patched by ebroad) * Add ability to generate ASCII documentation from the TeX files by running 'make asterisk.txt'. (Closes issue #17220. Reported by lmadsen. Tested, patched by pabelanger) * When StopMonitor() is called, ensure that it will not be restarted by a channel event. (Closes issue #16590. Reported, patched by kkm) * Small error in the T.140 RTP port verbose log. (Closes issue #16998. Reported, patched by frawd. Tested by russell) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.8 ======= 1.6.2.7 ======= The release of Asterisk 1.6.2.7 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * 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) For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.7 ======= 1.6.2.6 ======= The release of Asterisk 1.6.2.6 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! 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 ======= 1.6.2.5 ======= The Asterisk Development Team has announced security releases for the following versions of Asterisk: * 1.6.2.5 The releases of Asterisk 1.6.0.25, 1.6.1.17, and 1.6.2.5 resolve an issue with invalid parsing of ACL (Access Control List) rules leading to a possible compromise in security. The issue and resolution are described in the AST-2010-003 security advisory. For more information about the details of this vulnerability, please read the security advisory AST-2010-003, 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/ChangeLog-1.6.2.5 Security advisory AST-2010-003 is available at: http://downloads.asterisk.org/pub/security/AST-2010-003.pdf ======= 1.6.2.4 ======= The Asterisk Development Team has announced security releases for the following versions of Asterisk: * 1.6.2.4 The releases of Asterisk 1.2.40, 1.4.29.1, 1.6.0.24, 1.6.1.16, and 1.6.2.4 include documention describing a possible dialplan string injection with common usage of the ${EXTEN} (and other expansion variables). The issue and resolution are described in the AST-2010-002 security advisory. If you have a channel technology which can accept characters other than numbers and letters (such as SIP) it may be possible to craft an INVITE which sends data such as 300&Zap/g1/4165551212 which would create an additional outgoing channel leg that was not originally intended by the dialplan programmer. Please note that this is not limited to an specific protocol or the Dial() application. The expansion of variables into programmatically-interpreted strings is a common behavior in many script or script-like languages, Asterisk included. The ability for a variable to directly replace components of a command is a feature, not a bug - that is the entire point of string expansion. However, it is often the case due to expediency or design misunderstanding that a developer will not examine and filter string data from external sources before passing it into potentially harmful areas of their dialplan. With the flexibility of the design of Asterisk come these risks if the dialplan designer is not suitably cautious as to how foreign data is allowed to enter the system unchecked. 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. For more information about the details of this vulnerability, please read the security advisory AST-2010-002, 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/ChangeLog-1.6.2.4 Security advisory AST-2010-002 is available at: http://downloads.asterisk.org/pub/security/AST-2010-002.pdf The README-SERIOUSLY.bestpractices.txt document is available in the top-level directory of your Asterisk sources, or available in all Asterisk branches from 1.2 and up. http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt ======= 1.6.2.3 ======= Was never released. ======= 1.6.2.2 ======= The Asterisk Development Team has announced security releases for Asterisk as the following versions: * 1.6.2.2 The releases of Asterisk 1.6.0.22, 1.6.1.14, and 1.6.2.2 include the fix described in security advisory AST-2010-001. The issue is that 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 will occur when the FaxMaxDatagram field is omitted from the SDP, as well. For more information about the details of this vulnerability, please read the security advisory AST-2009-009, 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/ChangeLog-1.6.2.2 Security advisory AST-2010-001 is available at: http://downloads.asterisk.org/pub/security/AST-2010-001.pdf ======= 1.6.2.1 ======= The release of Asterisk 1.6.2.1 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * 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. A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.2.1-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.1 ======= 1.6.2.0 ======= The release of Asterisk 1.6.2.0 is the first feature release since Asterisk 1.6.1.0, which was released April 27, 2009. Many new features have been included in this release. For a complete list of changes, please see the CHANGES file. For those upgrading from a previous release, please see UPGRADE.txt It should be explicitly stated that Asterisk 1.6.2.0 is a major upgrade over any previous release, and special care should be taken when upgrading existing systems. Please see the UPGRADE.txt file for more information, available at: http://svn.asterisk.org/svn/asterisk/tags/1.6.2.0/UPGRADE.txt A detailed overview to the new features available in Asterisk 1.6.2.0 are forthcoming within the next few days. Please watch http://blogs.asterisk.org for further information! Below is a summary of several new features available in this release: * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with support for LibOpenR2. http://www.libopenr2.org/ * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this option is enabled, Asterisk will watch for a CNG tone in the incoming audio for a received call. If it is detected, the channel will jump to the 'fax' extension in the dialplan. * A new application, Originate, has been introduced, that allows asynchronous call origination from the dialplan. * Added ConfBridge dialplan application which does conference bridges without DAHDI. For information on its use, please see the output of "core show application ConfBridge" from the CLI. * extensions.conf now allows you to use keyword "same" to define an extension without actually specifying an extension. It uses exactly the same pattern as previously used on the last "exten" line. For example: exten => 123,1,NoOp(something) same => n,SomethingElse() * Asterisk now provides the ability to define custom CLI aliases. For example, if you would like to define short form aliases for frequently used commands, such as "sh ch" for "core show channels", that is now possible. See the cli_aliases.conf configuration file for more information. * Asterisk now has support for subscribing to the state of remote voice mailboxes via SIP. * Asterisk now includes expanded HD codec support. G.722.1 and G.722.1C (Siren7/Siren14) passthrough, recording, and playback is now supported. Transcoding will be made available via add-on modules soon for this version of Asterisk. This is just a subset of the changes available in this release. Please see the CHANGES file for additional information, available at: http://svn.asterisk.org/svn/asterisk/tags/1.6.2.0/CHANGES A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.2.0-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.0 module: pkgsrc subject: 'CVS commit: pkgsrc/comms/asterisk16' unixtime: '1285284638' user: jnemeth