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asterisk-debuginfo-11.7.0-1.fc18 RPM for armv5tel

From Fedora 18 updates for arm / debug

Name: asterisk-debuginfo Distribution: Fedora Project
Version: 11.7.0 Vendor: Fedora Project
Release: 1.fc18 Build date: Mon Jan 6 18:45:46 2014
Group: Development/Debug Build host: arm01-builder22.arm.fedoraproject.org
Size: 53640714 Source RPM: asterisk-11.7.0-1.fc18.src.rpm
Packager: Fedora Project
Url: http://www.asterisk.org/
Summary: Debug information for package asterisk
This package provides debug information for package asterisk.
Debug information is useful when developing applications that use this
package or when debugging this package.

Provides

Requires

License

GPLv2

Changelog

* Sat Dec 28 2013 Jeffrey Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 11.5.1-3:
  - Disable hardened build, as it's apparently causing problems loading modules.
* Thu Aug 29 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.5.1-2:
  - Enable hardened build BZ#954338
  - Significant clean ups
* Thu Aug 29 2013 Jeffrey Ollie <jeff@ocjtech.us> - 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 <rel-eng@lists.fedoraproject.org> - 11.4.0-2.2
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild
* Wed Jul 17 2013 Petr Pisar <ppisar@redhat.com> - 11.4.0-2.1
  - Perl 5.18 rebuild
* Fri May 24 2013 Rex Dieter <rdieter@fedoraproject.org> 11.4.0-2
  - rebuild (libical)
* Mon May 20 2013 Jeffrey Ollie <jeff@ocjtech.us> - 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 <spot@fedoraproject.org> - 11.3.0-2:
  - fix build with lua 5.2
* Tue Apr 23 2013 Jeffrey Ollie <jeff@ocjtech.us> - 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\n'
  -   (Closes issue ASTERISK-20908. Reported by Dennis DeDonatis)
  -
  - * --- Fix pjproject compilation in certain circumstances
  -   (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.3.0
* Thu Mar 28 2013 Jeffrey Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <dan[at]danny.cz> - 11.0.1-3
  - simplify LDFLAGS setting
* Fri Nov 30 2012 Dennis Gilmore <dennis@ausil.us> - 11.0.1-2
  - clean up things to allow building on arm arches
* Mon Nov 05 2012 Jeffrey Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 11.0.0-0.5.beta2
  - Don't forget format_ilbc module
* Wed Sep 26 2012 Jeffrey Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <dan[at]danny.cz> - 11.0.0-0.3.beta1
  - fix build on s390
* Tue Sep 04 2012 Dan Horák <dan[at]danny.cz> - 10.7.1-2
  - fix build on s390
* Thu Aug 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <rel-eng@lists.fedoraproject.org> - 10.5.2-1.2
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild
* Mon Jul 09 2012 Petr Pisar <ppisar@redhat.com> - 10.5.2-1.1
  - Perl 5.16 rebuild
* Thu Jul 05 2012 Jeffrey Ollie <jeff@ocjtech.us> - 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 <ppisar@redhat.com> - 10.5.1-1.1
  - Perl 5.16 rebuild
* Fri Jun 15 2012 Jeffrey Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <ppisar@redhat.com> - 10.4.2-1.1
  - Perl 5.16 rebuild
* Wed May 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <russell@russellbryant.net> - 10.3.0-1
  - Update to 10.3.0
* Fri Mar 16 2012 Russell Bryant <russell@russellbryant.net> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <russellb@fedoraproject.org> - 10.0.0-2
  - Remove asterisk-ais.  OpenAIS was removed from Fedora.
* Thu Jan 12 2012 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 10.0.0-1.1
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild
* Tue Jan 03 2012 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-1
  - Don't build API docs as the build never finishes
* Thu Dec 15 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 10.0.0-0.4.beta2
  - Add patch from upstream SVN to fix AST-2011-012
* Fri Oct 14 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.3.beta2
  - Patch cleanup day
* Thu Sep 29 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <contyk@redhat.com> - 1.8.5.0-1.2
  - Perl mass rebuild
* Wed Jul 20 2011 Petr Sabata <contyk@redhat.com> - 1.8.5.0-1.1
  - Perl mass rebuild
* Mon Jul 11 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.5-0.2
  - Rebuild for net-snmp 5.7
* Fri Jul 01 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.5-0.1.rc1
  - Fix systemd dependencies in EL6 and F15
* Thu Jun 30 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.4.4-2
  - Fedora Directory Server -> 389 Directory Server
* Wed Jun 29 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.4.3-3
  - Don't forget stereorize
* Mon Jun 27 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.4.2-2
  - Convert to systemd
* Fri Jun 17 2011 Marcela MaÅ¡lÃ¡Åłová <mmaslano@redhat.com> - 1.8.4.2-1.2
  - Perl mass rebuild
* Fri Jun 10 2011 Marcela MaÅ¡lÃ¡Åłová <mmaslano@redhat.com> - 1.8.4.2-1.1
  - Perl 5.14 mass rebuild
* Fri Jun 03 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.3.2-2
  - Bump release and rebuild for mysql 5.5.10 soname change.
* Thu Mar 17 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.3-0.6.rc2
  - Bump release to build for F15
* Wed Feb 09 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3-0.5.rc2
  - Remove isa macros
* Wed Feb 09 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3-0.4.rc2
  - Make library dependencies architecture specific
* Mon Feb 07 2011 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 1.8.3-0.3.rc2
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild
* Wed Jan 26 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3-0.2.rc2
  The Asterisk Development Team has announced the second release candidate of
  Asterisk 1.8.3. This release candidate is available for immediate download at
  http://downloads.asterisk.org/pub/telephony/asterisk/
  
  The release of Asterisk 1.8.3-rc2 resolves the following issues in addition to
  those included in 1.8.3-rc1:
  
   * Resolve issue where no Music On Hold may be triggered when using
    res_timing_dahdi.
    (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested
     by francesco_r, rfrantik, one47)
  
   * Resolve a memory leak when the Asterisk Manager Interface is disabled.
    (Reported internally by kmorgan. Patched by russellb)
  
   * Reimplemented fax session reservation to reverse the ABI breakage introduced
    in r297486.
    (Reported internally. Patched by mnicholson)
  
  For a full list of changes in this release candidate, please see the ChangeLog:
  
  http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3-rc2
* Wed Jan 26 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.2.2-2
  - Build with SRTP support
* Mon Jan 24 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <dennis@ausil.us> - 1.8.0-6
  - dont package up the ices bits on el the client doesnt exist for us
* Tue Jan 18 2011 Dennis Gilmore <dennis@ausil.us> - 1.8.0-5
  - dont build the 389 directory server package its not available on rhel6
* Fri Dec 10 2010 Dennis Gilmore <dennis@ausil.us> - 1.8.0-4
  - dont always build AIS modules we dont have the BuildRequires on epel
* Fri Oct 29 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-3
  - Rebuild for new net-snmp.
* Tue Oct 26 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-2
  - Always build AIS modules
* Thu Oct 21 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.0-0.2.beta2
  - Rebuild against libpri 1.4.12
* Mon Aug 02 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.2.8-0.3.rc1
  - Add patch to remove requirement on latex2html
* Tue Jun 01 2010 Marcela Maslanova <mmaslano@redhat.com> - 1.6.2.8-0.2.rc1
  - Mass rebuild with perl-5.12.0
* Tue May 04 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.2.7-0.2.rc3
  - Update to 1.6.2.7-rc3
* Thu Apr 15 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.7-0.1.rc2
  - Update to 1.6.2.7-rc2
* Fri Mar 12 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.2.6-0.1.rc2
  - Update to 1.6.2.6-rc2
* Mon Mar 08 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.5-2
  - Add a patch that fixes CLI history when linking against external libedit.
* Thu Feb 25 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.2.1-0.1.rc1
  - Update to 1.6.2.1-rc1
* Sat Dec 19 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-1
  - Released version of 1.6.2.0
* Wed Dec 09 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.16.rc8
  - Update to 1.6.2.0-rc8
* Wed Dec 02 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.15.rc7
  - Update to 1.6.2.0-rc7
* Tue Dec 01 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.2.0-0.12.rc6
  - Update to 1.6.2.0-rc6
* Mon Nov 09 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.11.rc5
  - Update to 1.6.2.0-rc5
* Fri Nov 06 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.10.rc4
  - Update to 1.6.2.0-rc4
* Tue Oct 27 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.2.0-0.6.rc3
  - Require latex2html used in static-http documents
* Wed Oct 07 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.5.rc3
  - Change ownership and permissions on config files to protect them.
* Tue Oct 06 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.4.rc3
  - Update to 1.6.2.0-rc3
* Wed Sep 30 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.3.rc2
  - Merge firmware subpackage back into the main package.
* Wed Sep 30 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.1-0.26.rc1
  - Add a patch from Quentin Armitage and rebuld.
* Fri Aug 21 2009 Tomas Mraz <tmraz@redhat.com> - 1.6.1-0.25.rc1
  - rebuilt with new openssl
* Fri Jul 24 2009 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 1.6.1-0.24.rc1
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild
* Thu Mar 05 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <rel-eng@lists.fedoraproject.org> - 1.6.1-0.22.rc1
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild
* Sun Feb 08 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <tmraz@redhat.com> - 1.6.1-0.13.beta4
  - rebuild with new openssl
* Sun Jan 04 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.12.beta4
  - Fedora Directory Server compatibility patch/subpackage.
* Sun Jan 04 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.10.beta4
  - Fix up paths. BZ#477238
* Sat Jan 03 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.9.beta4
  - Update patches
* Sat Jan 03 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.8.beta4
  - Update to 1.6.1-beta4
* Tue Dec 09 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.7.beta3
  - Update to 1.6.1-beta3
* Tue Dec 09 2008 Alex Lancaster <alexlan[AT]fedoraproject org> - 1.6.1-0.6.beta2
  - Rebuild for new gmime
* Fri Nov 07 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.5.beta2
  - Add patch to fix missing variable on PPC.
* Fri Nov 07 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.4.beta2
  - Update PPC systems don't have sys/io.h patch.
* Fri Nov 07 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.3.beta2
  - PPC systems don't have sys/io.h
* Fri Nov 07 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.2.beta2
  - Update to 1.6.1 beta 2
* Wed Nov 05 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0.1-3
  - Fix issue with init script giving wrong path to config file.
* Thu Oct 16 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.0-1
  - Update to final release.
* Thu Sep 11 2008 - Bastien Nocera <bnocera@redhat.com> - 1.6.0-0.22.beta9
  - Rebuild
* Wed Jul 30 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.0-0.20.beta9
  - Bump release and rebuild with new libpri and zaptel.
* Fri Jul 25 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.0-0.18.beta9
  - Add patch for LDAP extracted from upstream SVN (#442011)
* Wed Jul 02 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.0-0.15.beta9
  - Bump release and rebuild to fix libtds breakage.
* Mon May 19 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.0-0.9.beta6
  - Add Postgresql schemas from contrib as documentation to the Postgresql subpackage.
* Tue Mar 25 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <tcallawa@redhat.com> - 1.6.0-0.5.beta5
  - add Requires for versioned perl (libperl.so)
* Wed Mar 05 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.4.beta5
  - Update to 1.6.0-beta5
  - Remove upstreamed patches.
* Mon Mar 03 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.3.beta4
  - Package the directory used to store monitor recordings.
* Tue Feb 26 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.2.beta4
  - Add patch from David Woodhouse that fixes building on PPC64.
* Tue Feb 26 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.1.beta4
  - Update to 1.6.0 beta 4
* Wed Feb 13 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.17-1
  - Update to 1.4.17 to fix AST-2008-001.
* Fri Dec 28 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.16.2-1
  - Update to 1.4.16.2
* Sat Dec 22 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.16.1-2
  - Bump release and rebuild to fix broken dep on uw-imap.
* Wed Dec 19 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.16.1-1
  - Update to the real 1.4.16.1.
* Wed Dec 19 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.16-1
  - Update to 1.4.16 to fix security bug.
* Sat Dec 15 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-7
  - Really, really fix the build problems on devel.
* Sat Dec 15 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-6
  - Tweaks to get to build on x86_64
* Wed Dec 12 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-5
  - Exclude PPC64
* Wed Dec 12 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.15-3
  - Really get rid of zero length map files.
* Mon Dec 10 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-2
  - Get rid of zero length map files.
  - Shorten descriptions of voicemail subpackages
* Fri Nov 30 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-1
  - Update to 1.4.15
* Tue Nov 20 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.14-2
  - Fix license and other rpmlint warnings.
* Mon Nov 19 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.14-1
  - Update to 1.4.14
* Fri Nov 16 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.13-7
  - Add chan_mobile
* Tue Nov 13 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.13-1
  - Update to 1.4.13
* Tue Oct 09 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.12.1-1
  - Update to 1.4.12.1
* Wed Aug 22 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.11-1
  - Update to 1.4.11
* Fri Aug 10 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.10.1-1
  - Update to 1.4.10.1.
* Tue Aug 07 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.10-1
  - Update to 1.4.10 (security update).
* Tue Aug 07 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.9-7
  - Add a patch that allows alternate extensions to be defined in users.conf
* Mon Aug 06 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.9-6
  - Update app_conference patch. Enter/leave sounds are now possible.
* Fri Jul 27 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.9-4
  - Don't build app_mp3
* Wed Jul 25 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.9-3
  - Add app_conference
* Wed Jul 25 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.9-1
  - Update to 1.4.9
* Tue Jul 17 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.8-1
  - Update to 1.4.8
  - Drop ixjuser patch.
* Tue Jul 10 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.7.1-1
  - Update to 1.4.7.1
* Mon Jul 09 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.7-1
  - Update to 1.4.7
  - RxFAX/TxFAX applications
* Sun Jul 01 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.6-4
  - It's "sbin", not "bin" silly.
* Sat Jun 30 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.6-3
  - Add patch that lets us change TOS bits even when running non-root
* Fri Jun 29 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.6-2
  - voicemail needs to require /usr/bin/sox and /usr/bin/sendmail
* Fri Jun 29 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.6-1
  - Update to 1.4.6
  - Remove upstreamed patch.
* Thu Jun 21 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.5-9
  - Move sounds
* Mon Jun 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-8
  - Update some more ownership/permissions
* Mon Jun 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-7
  - Fix some permissions.
* Mon Jun 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-6
  - Update init script patch
  - Move pid file to subdir of /var/run
* Mon Jun 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-5
  - Update init script patch to run as non-root
* Sun Jun 17 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-4
  - Build modules that depend on FreeTDS.
  - Don't build voicemail with ODBC storage.
* Sun Jun 17 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-3
  - Have the build output the commands executing, rather than covering them up.
* Fri Jun 15 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-1
  - Update to 1.4.5
  - Remove upstreamed patch.
* Wed May 09 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.4-2
  - Add a patch to fix CVE-2007-2488/ASA-2007-013
* Fri Apr 27 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.4-1
  - Update to 1.4.4
* Wed Mar 21 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.2-1
  - Update to 1.4.2
* Tue Mar 06 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.1-2
  - Package the IAXy firmware
  - Minor clean-ups in files
* Mon Mar 05 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.0-6.beta4
  - Update to 1.4.0-beta4
  - Various cleanups.
* Fri Oct 20 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.0-4.beta3
  - Remove explicit dependency on glibc-kernheaders.
  - Build jabber modules on PPC
* Wed Oct 18 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.0-2.beta3
  - Remove requirement on asterisk-sounds-core until licensing can be
    figured out.
* Wed Oct 18 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.0-1.beta3
  - Update to 1.4.0-beta3
* Sun Oct 15 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.0-0.beta2
  - Update to 1.4.0-beta2
* Tue Jul 25 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.10-1
  - Update to 1.2.10.
* Wed Jun 07 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.9.1
  - Update to 1.2.9.1
* Fri Jun 02 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.2.7.1-5
  - Build mISDN channel drivers, modelled after spec file from David Woodhouse
* Thu Apr 20 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.2.7.1-3
  - Add chan_bluetooth
* Wed Apr 19 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.7.1-2
  - Split off more stuff into subpackages.
* Wed Apr 12 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.7-1
  - Update to 1.2.7
* Mon Apr 10 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.2.6-1
  - Update to 1.2.6
* Mon Mar 06 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.2.4-4
  - Actually apply the patch.
* Mon Feb 13 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.2.4-2
  - BR sqlite2-devel
* Tue Jan 31 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.4-1
  - Update to 1.2.4.
* Wed Jan 25 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.2.3-1
  - First version for Fedora Extras.

Files

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Fabrice Bellet, Sat Oct 11 10:05:45 2014