[asterisk/el6/master: 8/8] Merge remote branch 'origin/master' into el6/master
Jeffrey C. Ollie
jcollie at fedoraproject.org
Fri Oct 22 00:40:58 UTC 2010
commit 2d7b88b8447175a30e3d7e614f75269440f92814
Merge: ec15069 038b5db
Author: Jeffrey C. Ollie <jeff at ocjtech.us>
Date: Thu Oct 21 14:00:28 2010 -0500
Merge remote branch 'origin/master' into el6/master
.gitignore | 22 +-
...it-scripts-for-better-Fedora-compatibilty.patch | 71 ++--
...les.conf-so-that-different-voicemail-modu.patch | 8 +-
...king-building-against-an-external-libedit.patch | 55 +-
...rary-function-for-loading-command-history.patch | 8 +-
...igure.ac-to-look-for-pkg-config-gmime-2.0.patch | 12 +-
0006-Fix-up-some-paths.patch | 108 ++--
...hema-that-is-compatible-with-Fedora-Direc.patch | 4 +-
...2html-to-copy-icons-when-building-documen.patch | 6 +-
asterisk.spec | 639 ++++++++++++++++----
menuselect.makeopts | 9 +-
sources | 4 +-
12 files changed, 680 insertions(+), 266 deletions(-)
---
diff --cc asterisk.spec
index 69e2272,04b1be5..5c51415
--- a/asterisk.spec
+++ b/asterisk.spec
@@@ -1042,129 -1144,393 +1151,419 @@@ f
%{_libdir}/asterisk/modules/app_voicemail_plain.so
%changelog
- * Tue Aug 24 2010 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.6.2.12-0.1.rc1
- - The release of Asterisk 1.6.2.12-RC1 resolves several issues reported by the
- - community and would have not been possible without your participation.
- - Thank you!
+ * Thu Oct 21 2010 Jeffrey C. Ollie <jeff at 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/
-
- - The following is a sample of the issues resolved in this release candidate:
+ - 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.
-
- - * Fix issue where DNID does not get cleared on a new call when using
- - immediate=yes with ISDN signaling.
- - (Closes issue #17568. Reported by wuwu. Patched by rmudgett)
+ - http://www.asterisk.org/asterisk-versions
-
- - * Several updates to res_config_ldap.
- - (Closes issue #13573. Reported by navkumar. Patched by navkumar, bencer.
- - Tested by suretec)
+ - 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.
-
- - * Prevent loss of Caller ID information set on local channel after masquerade.
- - (Closes issue #17138. Reported by kobaz, patched by jpeeler)
+ - You can find a summary of the work involved with the 1.8.0 release in the
+ - sumary:
-
- - * Fix SIP peers memory leak.
- - (Closes issue #17774. Reported, patched by kkm)
+ - http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt
-
- - * Add Danish support to say.conf.sample
- - (Closes issue #17836. Reported, patched by RoadKill)
+ - A short list of available features includes:
-
- - * Ensure SSRC is changed when media source is changed to resolve audio delay.
- - (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)
+ - * 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!
-
- - * Only do magic pickup when notifycid is enabled.
- - A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a
- - call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber
- - that a device is ringing. This option should only be enabled when the new
- - 'notifycid' option is set, but this was not the case. Instead the call-id
- - value was included for every RINGING Notify message, which caused a
- - regression for people who used other methods for call pickup.
- - (Closes issue #17633. Reported, patched by urosh. Patched by dvossel.
- - Tested by: dvossel, urosh, okrief, alecdavis)
+ - A full list of new features can be found in the CHANGES file.
-
- - For a full list of changes in the current release, please see the
+ - 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-1.6.2.12-rc1
+ - 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 at 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
- * Wed Aug 11 2010 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.6.2.11-1
+ * Fri Oct 8 2010 Jeffrey C. Ollie <jeff at 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:
-
- - The following are a few of the issues resolved by community developers:
+ - * 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)
-
- - * Send DialPlanComplete as a response, not as a separate event. Otherwise, it
- - goes to all manager sessions and may exclude the current session, if the
- - Events mask excludes it.
- - (Closes issue #17504. Reported, patched by rrb3942)
+ - * Get notifications for call files only when a file is closed, not when created
+ - (Closes issue #17924. Reported by mkeuter. Patched by abeldeck)
-
- - * Allow the "useragent" value to be restored into memory from the realtime
- - backend. This value is purely informational. It does not alter configuration
- - at all.
- - (Closes issue #16029. Reported, patched by Guggemand)
+ - * 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)
-
- - * Fix rt(c)p set debug ip taking wrong argument Also clean up some coding
- - errors.
- - (Closes issue #17469. Reported, patched by wdoekes)
+ - * Fixes to allow chan_gtalk to communicate with the Gmail web client.
+ - (Patched by phsultan and dvossel)
-
- - * Ensure channel placed in meetme in ringing state is properly hung up. An
- - outgoing channel placed in meetme while still ringing which was then hung up
- - would not exit meetme and the channel was not properly destroyed.
- - (Closes issue #15871. Reported, patched by Ivan)
+ - * Fix to GET DATA to allow audio to be streamed via an AGI.
+ - (Closes issue #18001. Reported by jamicque. Patched by tilghman)
-
- - * Correct how 100, 200, 300, etc. is said. Also add the crazy British numbers.
- - (Closes issue #16102. Reported, patched by Delvar)
+ - * Resolve dnsmgr memory corruption in chan_iax2.
+ - (Closes issue #17902. Reported by afried. Patched by russell, dvossel)
-
- - * cdr_pgsql does not detect when a table is found. This change adds an ERROR
- - message to let you know when a failure exists to get the columns from the
- - pgsql database, which typically means that the table does not exist.
- - (Closes issue #17478. Reported, patched by kobaz)
+ - A short list of available features includes:
-
- - * Avoid crashing when installing a duplicate translation path with a lower
- - cost.
- - (Closes issue #17092. Reported, patched by moy)
+ - * 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!
-
- - * Add missing handling for ringing state for use with queue empty options.
- - (Closes issue #17471. Reported, patched by jazzy)
+ - A full list of new features can be found in the CHANGES file.
-
- - * Fix reporting estimated queue hold time. Just say the number of seconds
- - (after minutes) rather than doing some incorrect calculation with respect to
- - minutes.
- - (Closes issue #17498. Reported, patched by corruptor)
+ - 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
+ - For a full list of changes in the current release candidate, please see the
- ChangeLog:
-
- - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.11
+ - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc3
- * Mon Aug 2 2010 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.6.2.10-1.1
- - Disable res_http_post on EPEL until gmime22-devel is available
- - Disable res_ais on EPEL until openais brokenness can be investigated
+ * Wed Oct 6 2010 Jeffrey C. Ollie <jeff at 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 9 2010 Jeffrey C. Ollie <jeff at 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 at 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 at 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 2 2010 Jeffrey C. Ollie <jeff at ocjtech.us> - 1.8.0-0.2.beta2
+ - Rebuild against libpri 1.4.12
+
+ * Mon Aug 2 2010 Jeffrey C. Ollie <jeff at 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 at ocjtech.us> - 1.6.2.10-1
-- Update to 1.6.2.10
+-
+- 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 at ocjtech.us> - 1.6.2.8-0.3.rc1
- Add patch to remove requirement on latex2html
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