[asterisk-commits] lmadsen: tag 1.4.27-rc1 r219175 - /tags/1.4.27-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Sep 17 10:27:18 CDT 2009
Author: lmadsen
Date: Thu Sep 17 10:27:14 2009
New Revision: 219175
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=219175
Log:
Importing files for 1.4.27-rc1 release.
Added:
tags/1.4.27-rc1/.lastclean (with props)
tags/1.4.27-rc1/.version (with props)
tags/1.4.27-rc1/ChangeLog (with props)
Added: tags/1.4.27-rc1/.lastclean
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.4.27-rc1/.lastclean?view=auto&rev=219175
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--- tags/1.4.27-rc1/ChangeLog (added)
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+2009-09-17 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.27-rc1
+
+2009-09-17 14:58 +0000 [r219136] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c, include/asterisk/cdr.h,
+ include/asterisk/channel.h: Prevent a potential race condition
+ and crash when hanging up a channel by removing the channel from
+ the channel list before begining channel tear down. This fix may
+ potentially cause problems with CDR backends that access the
+ channel a CDR is associated with via the channel list. This fix
+ makes the channel unavabile at the time when the CDR backend is
+ invoked. This has been documented in include/asterisk/cdr.h.
+ (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
+ Review: https://reviewboard.asterisk.org/r/362/
+
+2009-09-16 23:21 +0000 [r219023] Tilghman Lesher <tlesher at digium.com>
+
+ * main/config.c, configs/extensions.conf.sample: Properly deal with
+ quotes in the arguments of '#exec' includes. (closes issue
+ #15583) Reported by: pkempgen Patches:
+ 20090726__issue15583.diff.txt uploaded by tilghman (license 14)
+ 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
+ 169) Tested by: pkempgen
+
+2009-09-16 18:00 +0000 [r218867] David Brooks <dbrooks at digium.com>
+
+ * main/pbx.c: Fixes CID pattern matching behavior to mirror that of
+ extension pattern matching. Pattern matching for extensions uses
+ a type of scoring system, giving values for specificity to each
+ character in the pattern. Unfortunately, this is done character
+ by character, in order. This does lead to some less specific
+ patterns being first in line for matching, but it will usually
+ get the job done. This patch merely brings CID matching to the
+ same level as extension matching. This patch does not attempt to
+ tackle the problem shared by extension matching. (closes issue
+ #14708) Reported by: klaus3000
+
+2009-09-16 13:33 +0000 [r218798] Russell Bryant <russell at digium.com>
+
+ * contrib/firmware/iax/iaxy.bin (removed), UPGRADE.txt: Remove the
+ IAXy firmware from Asterisk. The firmware can now be found on
+ downloads.digium.com, where the rest of our binary downloads
+ live. This was the last part of our Asterisk tarballs that was
+ considered non-free by Debian. :-) (closes issue #15838) Reported
+ by: paravoid
+
+2009-09-15 22:27 +0000 [r218730] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: If the user enters the same password as
+ before, don't signal an error when the change does nothing.
+ (closes issue #15492) Reported by: cbbs70a Patches:
+ 20090713__issue15492.diff.txt uploaded by tilghman (license 14)
+
+2009-09-15 16:29 +0000 [r218623] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Fix small memory leak in handle_init_event
+ by always destroying the pthread attr before returning.
+
+2009-09-15 16:03 +0000 [r218578] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Send request contact header field with
+ response to registrer queries instead of the address of record.
+ (closes issue #14438) Reported by: ravindrad Patches:
+ regquerypatch uploaded by ravindrad (license 684) Tested by:
+ ravindrad
+
+2009-09-15 16:01 +0000 [r218577] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_followme.c: Ensure FollowMe sets language in channels it
+ creates. Also, not in the original bug report, but related fields
+ are accountcode and musicclass, and the inheritance of
+ datastores. (closes issue #15372) Reported by: Romik Patches:
+ 20090828__issue15372.diff.txt uploaded by tilghman (license 14)
+ Tested by: cervajs
+
+2009-09-15 14:57 +0000 [r218497-218498] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c: revert accidental commit
+
+ * channels/chan_sip.c, sounds/Makefile: Use proper hostname for
+ downloading sound files.
+
+2009-09-14 21:47 +0000 [r218401] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Fix handling of DAHDI_EVENT_REMOVED event
+ to prevent crash in do_monitor. After talking to rmudgett about
+ some of his recent iflist locking changes, it was determined that
+ the only place that would destroy a channel without being
+ explicitly to do so was in handle_init_event. The loop to walk
+ the interface list has been modified to wait to destroy the
+ channel until the dahdi_pvt of the channel to be destroyed is no
+ longer needed. (closes issue #15378) Reported by: samy
+
+2009-09-14 19:16 +0000 [r218331] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, sounds/Makefile: Don't say "Please try
+ again" if we don't give the user another chance to try again.
+ (issue #15055, SWP-129) Reported by: jthurman
+
+2009-09-14 14:53 +0000 [r218223] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_directed_pickup.c: Ensure we don't pickup ourselves when
+ doing pickup by exten. (closes issue #15100) Reported by:
+ lmsteffan Patches: (modified) pickup.patch uploaded by lmsteffan
+ (license 779)
+
+2009-09-10 23:52 +0000 [r217917-217989] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c: Don't ring another channel, if there's not
+ enough time for a queue member to answer. (Fixes AST-228)
+
+ * contrib/scripts/iax-friends.sql, channels/chan_sip.c,
+ channels/chan_iax2.c: Backport realtime fix to 1.4
+
+2009-09-10 21:06 +0000 [r217806] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: IAX2 encryption regression The IAX2 Call
+ Token security patch inadvertently broke the use of encryption
+ due to the reorganization of code in the socket_process()
+ function. When encryption is used, an incoming full frame must
+ first be decrypted before the information elements can be parsed.
+ The security release mistakenly moved IE parsing before
+ decryption in order to process the new Call Token IE. To resolve
+ this, decryption of full frames is once again done before looking
+ into the frame. This involves searching for an existing callno,
+ checking the pvt to see if encryption is turned on, and
+ decrypting the packet before the internal fields of the full
+ frame are accessed. associated with AST-2009-006 (closes issue
+ #15834) Reported by: karesmakro Patches:
+ iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671)
+ Tested by: dvossel, karesmakro Review:
+ https://reviewboard.asterisk.org/r/355/
+
+2009-09-10 19:52 +0000 [r217668-217735] Olle Johansson <oej at edvina.net>
+
+ * utils/Makefile: Reinstate muted that was removed by mistake.
+ muted doesn't compile any more on os/x, so I have to disable it
+ in order to testcompile other code...
+
+ * utils/Makefile, channels/chan_sip.c: Remove harmful code that
+ causes endless loops. Remove code that causes loops in
+ registrations. We have agreed that the patch that this code was
+ part of was bad. I am ripping out the code that causes the issue.
+ putnopvut needs to check the rest of the patch, if it needs to be
+ changed as well. This solves the issue reported in #15540, but
+ needs more work before we close it (as described above).
+
+2009-09-08 20:01 +0000 [r217156] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_meetme.c: When MOH is playing on the channel,
+ announcements sent through the conference are not heard. (closes
+ issue #14588) Reported by: voipas Patches:
+ 20090716__issue14588__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: lmadsen, twisted, tilghman
+
+2009-09-04 13:56 +0000 [r216432-216435] Michiel van Baak <michiel at vanbaak.info>
+
+ * main/utils.c, include/asterisk/lock.h: make asterisk compile
+ under devmode with DEBUG_THREADS enabled on OpenBSD
+
+ * channels/chan_sip.c: make chan_sip compile under devmode again
+
+2009-09-04 13:45 +0000 [r216430] Olle Johansson <oej at edvina.net>
+
+ * apps/app_playback.c, main/pbx.c, channels/chan_sip.c,
+ apps/app_disa.c, configs/sip.conf.sample: Make apps send PROGRESS
+ control frame for early media and fix too early media issue in
+ SIP The issue at hand is that some legacy (dying) PBX systems
+ send empty media frames on PRI links *before* any call progress.
+ The SIP channel receives these frames and by default signals 183
+ Session progress and starts sending media. This will cause phones
+ to play silence and ignore the later 180 ringing message. A bad
+ user experience. The fix is twofold: - We discovered that
+ asterisk apps that support early media ("noanswer") did not send
+ any PROGRESS frame to indicate early media. Fixed. - We introduce
+ a setting in chan_sip so that users can disable any relay of
+ media frames before the outbound channel actually indicates any
+ sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be
+ disabled for backward compatibility. In later versions of
+ Asterisk, this will be enabled. We don't assume that it will
+ change your Asterisk phone experience - only for the better. We
+ encourage third-party application developers to make sure that if
+ they have applications that wants to send early media, add a
+ PROGRESS control frame transmission to make sure that all channel
+ drivers actually will start sending early media. This has not
+ been the default in Asterisk previous to this patch, so if you
+ got inspiration from our code, you need to update accordingly.
+ Sorry for the trouble and thanks for your support. This code has
+ been running for a few months in a large scale installation (over
+ 250 servers with PRI and/or BRI links to old PBX systems). That's
+ no proof that this is an excellent patch, but, well, it's tested
+ :-)
+
+2009-09-04 13:16 +0000 [r216369] Michiel van Baak <michiel at vanbaak.info>
+
+ * main/astobj2.c: Make sure 'start' is always initialized. This is
+ the same as rev 216222 in trunk but 1.4 is affected as well
+
+2009-09-04 10:48 +0000 [r216008-216263] Russell Bryant <russell at digium.com>
+
+ * doc/IAX2-security.txt (added), /: Merged revisions 216262 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009)
+ | 2 lines Add a plain text version of the IAX2 security document.
+ ........
+
+ * /, UPGRADE.txt: Merged revisions 216080 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009)
+ | 2 lines Add a note about IAX2 to UPGRADE.txt. ........
+
+ * /, doc/IAX2-security.pdf (added): Merged revisions 216005 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009)
+ | 2 lines Add IAX2 security document related to AST-2009-006.
+ ........
+
+2009-09-03 18:32 +0000 [r216000] David Vossel <dvossel at digium.com>
+
+ * channels/iax2-parser.c, main/astobj2.c, configs/iax.conf.sample,
+ include/asterisk/acl.h, channels/iax2-parser.h,
+ include/asterisk/astobj2.h, channels/iax2.h, main/acl.c,
+ channels/chan_iax2.c: Merge code associated with AST-2009-006
+ (closes issue #12912) Reported by: rathaus Tested by: tilghman,
+ russell, dvossel, dbrooks
+
+2009-09-02 21:41 +0000 [r215682] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Re-send non-100 provisional responses to
+ prevent cancellation From section 13.3.1.1 of RFC 3261: If the
+ UAS desires an extended period of time to answer the INVITE, it
+ will need to ask for an "extension" in order to prevent proxies
+ from canceling the transaction. A proxy has the option of
+ canceling a transaction when there is a gap of 3 minutes between
+ responses in a transaction. To prevent cancellation, the UAS MUST
+ send a non-100 provisional response at every minute, to handle
+ the possibility of lost provisional responses. (closes issue
+ #11157) Reported by: rjain Tested by: twilson Review:
+ https://reviewboard.asterisk.org/r/315/
+
+2009-09-01 23:04 +0000 [r215270] Dwayne M. Hubbard <dwayne.hubbard at gmail.com>
+
+ * apps/app_softhangup.c: Use strrchr() so SoftHangup will correctly
+ truncate multi-hyphen channel names In general channel names are
+ in the form Foo/Bar-Z, but the channel name could have multiple
+ hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the
+ channel name at the last hyphen. (closes issue #15810) Reported
+ by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by
+ dhubbard (license 733)
+
+2009-08-31 16:16 +0000 [r214940] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_local.c: Also unlock the "other" channel, when
+ returning, due to glare. (closes issue #15787) Reported by:
+ tim_ringenbach Patches: chan_local.diff uploaded by tim
+ ringenbach (license 540) Tested by: tim_ringenbach
+
+2009-08-28 20:13 +0000 [r214357-214701] Tilghman Lesher <tlesher at digium.com>
+
+ * main/channel.c: Modify comment to be a bit more accurate. We have
+ kept this comment around long enough, that it's pretty clear that
+ we're keeping the code, because changing the code would require a
+ pretty fundamental architectural shift. We've also taken
+ criticism in some quarters, because it was believed that it was
+ referring to the code being nasty. No, the code isn't nasty, just
+ the operation itself is rather odd. Fixed for eternity (probably
+ not).
+
+ * autoconf/libcurl.m4 (added), configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Use autoconf to
+ detect libcurl, as this enables cross-compilation checks,
+ something we didn't allow before. (closes issue #15714) Reported
+ by: pprindeville Patches: 20090813__issue15714.diff.txt uploaded
+ by tilghman (license 14) Tested by: pprindeville
+
+ * autoconf/ast_ext_lib.m4, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: One more build
+ system change, to make the descriptions look better, if we have
+ better information.
+
+ * autoconf/ast_ext_lib.m4, configure,
+ include/asterisk/autoconfig.h.in: Make autoheader descriptions
+ render correctly in our autoconfig.h file. (Figured out while
+ working with issue #14906)
+
+2009-08-26 16:36 +0000 [r214194] David Vossel <dvossel at digium.com>
+
+ * main/channel.c: ast_write() ignores ast_audiohook_write() results
+ In ast_write(), if a channel has a list of audiohooks, those
+ lists are written to and the resulting frame is what ast_write()
+ should continue with. The problem was the returned audiohook
+ frame was not being handled at all, and the original frame passed
+ into it did not contain the mixed audio, so essentially audio was
+ being lost. One result of this was chan_spy's whisper mode no
+ longer worked. To complicate the issue, frames passed into
+ ast_write may either be a single frame, or a list of frames. So,
+ as the list of frames is processed in the audiohook_write, the
+ returned frames had to be added to a new list. (closes issue
+ #15660) Reported by: corruptor Tested by: dvossel
+
+2009-08-25 19:28 +0000 [r213899-214069] Tilghman Lesher <tlesher at digium.com>
+
+ * main/say.c: I should always compile before committing...
+
+ * main/say.c: Fix pronunciation of German dates. (closes issue
+ #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded
+ by Benjamin Kluck (license 803)
+
+ * main/pbx.c: Improve error message by informing user exactly which
+ function is missing a parethesis. (closes issue #15242) Reported
+ by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by
+ dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by
+ loloski (license 68)
+
+ * Makefile: Use the default runlevels for Debian derivatives,
+ instead of making up our own. (closes issue #14730) Reported by:
+ pkempgen
+
+2009-08-21 20:23 +0000 [r213631] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_sip.c: Ensure that T.38 INVITEs generated by
+ Asterisk properly result in T.38 being enabled. (closes issue
+ #15373) Reported by: dcolombo Patches: chan_sip.patch uploaded by
+ mbrancaleoni (license 342) Tested by: dcolombo, mbrancaleoni
+
+2009-08-21 16:52 +0000 [r213559] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk.h: Permit DEBUG_FD_LEAKS to be used with C++
+ source files. (closes issue #15698) Reported by: slavon Patches:
+ 20090817__issue15698.diff.txt uploaded by tilghman (license 14)
+ Tested by: slavon, tilghman
+
+2009-08-21 16:03 +0000 [r213493] Jason Parker <jparker at digium.com>
+
+ * configs/queues.conf.sample: Clarify queues.conf comments to
+ specify that variables should be set in the dialplan. (closes
+ issue #15755) Reported by: trendboy
+
+2009-08-20 20:33 +0000 [r213339] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_features.c: Fix a crash by checking the proper pointer
+ for validity before deferencing it. (closes issue #15751)
+ Reported by: atis Patches: ast_bridge_call_peer_cdr.patch
+ uploaded by atis (license 242)
+
+2009-08-20 19:53 +0000 [r213283] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.exports (added): Make all the symbols for the
+ C-client callbacks global
+
+2009-08-19 21:18 +0000 [r213103] David Vossel <dvossel at digium.com>
+
+ * apps/app_mixmonitor.c: Fixes memory leak caused by incorrectly
+ freeing mixmonitor (closes issue #15699) Reported by: edantie
+ Patches: mixmonitor.patch uploaded by edantie (license 862)
+
+2009-08-18 20:26 +0000 [r212913] Kevin P. Fleming <kpfleming at digium.com>
+
+ * doc/musiconhold-opsound.txt (added), CREDITS, /, UPGRADE.txt,
+ sounds/sounds.xml, build_tools/prep_tarball,
+ doc/musiconhold-fpm.txt (removed), doc/00README.1st,
+ sounds/Makefile: Convert this branch to Opsound music-on-hold.
+ For more details:
+ http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
+
+2009-08-18 16:36 +0000 [r212763] Sean Bright <sean at malleable.com>
+
+ * main/manager.c: Delay the creation of temporary files until we
+ have a valid manager command to handle. Without this patch,
+ asterisk creates a temporary file before determining if the
+ specified command is valid. If invalid, we weren't properly
+ cleaning up the file. (closes issue #15730) Reported by: zmehmood
+ Patches: M15730.diff uploaded by junky (license 177) Tested by:
+ zmehmood
+
+2009-08-18 16:00 +0000 [r212727] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn/isdn_lib.c: Removed some deadwood and added some
+ doxygen comments.
+
+2009-08-17 16:34 +0000 [r212498] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/misdn_config.c: Fix segfault when reloading chan_misdn.
+ If more ports were specified than configured in misdn.conf a
+ reload would crash asterisk. The problem was the unconfigured
+ port was using data from the previously configured port. When the
+ data for an unconfigured port was freed a crash would result from
+ the double free. (closes issue #12113) Reported by: agupta
+ Patches: bug12113.patch uploaded by jpeeler (license 325)
+
+2009-08-17 15:36 +0000 [r212430] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Fix uninitialized variable.
+
+2009-08-12 23:04 +0000 [r211953] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_queue.c: This patch adds additional checking when
+ generating queue log TRANSFER events. The additional checks
+ prevent generation of false TRANSFER events in certain
+ situations. (closes issue #14536) Reported by: aragon Patches:
+ queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
+ Tested by: aragon, mnicholson
+
+2009-08-12 18:46 +0000 [r211807] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Backport fix so that outbound CANCEL
+ requests have same branch as challenged INVITEs. There already
+ was code present to be sure that a CANCEL will contain the same
+ branch-id as the INVITE it is cancelling. However, for INVITES
+ which are challenged downstream, this mechanism did not work
+ properly. Now this is taken care of. This is a backport of a fix
+ already present in all 1.6.X branches and in trunk. It also fixes
+ ABE-1907.
+
+2009-08-10 19:48 +0000 [r211528-211583] Tilghman Lesher <tlesher at digium.com>
+
+ * doc/CODING-GUIDELINES: Conversion specifiers, not format
+ specifiers
+
+ * main/indications.c, main/cli.c, pbx/pbx_loopback.c,
+ channels/chan_dahdi.c, res/res_smdi.c, pbx/pbx_spool.c,
+ channels/chan_skinny.c, pbx/pbx_ael.c, apps/app_dial.c,
+ main/pbx.c, apps/app_privacy.c, codecs/codec_speex.c,
+ funcs/func_math.c, channels/chan_agent.c, apps/app_morsecode.c,
+ apps/app_disa.c, channels/iax2-provision.c, funcs/func_cut.c,
+ pbx/dundi-parser.c, apps/app_talkdetect.c, channels/chan_misdn.c,
+ apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c,
+ pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c,
+ main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c,
+ doc/CODING-GUIDELINES, utils/muted.c, apps/app_meetme.c,
+ main/utils.c, apps/app_followme.c, utils/frame.c,
+ channels/misdn_config.c, main/cdr.c, main/channel.c,
+ channels/chan_phone.c, main/manager.c, apps/app_osplookup.c,
+ apps/app_setcallerid.c, res/res_agi.c, apps/app_rpt.c,
+ channels/chan_mgcp.c, apps/app_adsiprog.c, main/dnsmgr.c,
+ channels/chan_sip.c, apps/app_waitforsilence.c, agi/eagi-test.c,
+ main/acl.c, apps/app_queue.c, channels/chan_oss.c,
+ agi/eagi-sphinx-test.c, channels/chan_h323.c, pbx/pbx_dundi.c,
+ apps/app_sms.c, apps/app_verbose.c, apps/app_dahdibarge.c,
+ funcs/func_rand.c, apps/app_readfile.c, main/frame.c, /,
+ res/res_features.c, apps/app_record.c, funcs/func_strings.c,
+ apps/app_random.c, apps/app_alarmreceiver.c,
+ channels/chan_iax2.c: AST-2009-005
+
+2009-08-09 15:41 +0000 [r211274] Tilghman Lesher <tlesher at digium.com>
+
+ * main/astfd.c: Small oops. Clear the flags which have been
+ checked.
+
+2009-08-07 20:11 +0000 [r211112] Russell Bryant <russell at digium.com>
+
+ * apps/app_chanspy.c: Resolve a deadlock involving app_chanspy and
+ masquerades. (ABE-1936)
+
+2009-08-07 18:16 +0000 [r210913-211038] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_queue.c: QUEUE_MEMBER_LIST _really_ wants the interface
+ name, not the membername. This is a partial revert of revision
+ 82590, which was an attempted cleanup, but in reality, it broke
+ QUEUE_MEMBER_LIST, which has always been intended as a method by
+ which component interfaces could be queried from the queue.
+ Membername isn't useful here, because that field cannot be used
+ to obtain further information about the member. See the
+ documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
+ QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
+ member argument for further justification. (closes issue #15664)
+ Reported by: rain Patches: app_queue-queue_member_list.diff
+ uploaded by rain (license 327)
+
+ * main/channel.c: Because channel information can be accessed
+ outside of the channel thread, we must lock the channel prior to
+ modifying it. (closes issue #15397) Reported by: caspy Patches:
+ 20090714__issue15397.diff.txt uploaded by tilghman (license 14)
+ Tested by: caspy
+
+2009-08-05 19:18 +0000 [r210575] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Dialplan starts execution before the
+ channel setup is complete. * Issue 15655: For the case where
+ dialing is complete for an incoming call, dahdi_new() was asked
+ to start the PBX and then the code set more channel variables. If
+ the dialplan hungup before these channel variables got set,
+ asterisk would likely crash. * Fixed potential for overlap
+ incoming call to erroneously set channel variables as global
+ dialplan variables if the ast_channel structure failed to get
+ allocated. * Added missing set of CALLINGSUBADDR in the dialing
+ is complete case. (closes issue #15655) Reported by: alecdavis
+
+2009-08-05 18:46 +0000 [r210563] Leif Madsen <lmadsen at digium.com>
+
+ * doc/imapstorage.txt: Update imapstorage.txt documentation.
+ Updated the imapstorage.txt documentation to reflect that issues
+ with c-client versions older than 2007 seem to cause crashing
+ issues that are not seen with more recent versions. Documentation
+ has been updated to reflect this. (closes issue #14496) Reported
+ by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+ uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+ dbrooks
+
+2009-08-04 14:51 +0000 [r210237] Kevin P. Fleming <kpfleming at digium.com>
+
+ * Makefile: Eliminate spurious compiler warnings from system
+ headers on *BSD platforms. Ensure that system headers located in
+ /usr/local/include are actually treated as system headers by the
+ compiler, and not as local headers which are subject to warnings
+ from the -Wundef compiler option and others. (closes issue
+ #15606) Reported by: mvanbaak
+
+2009-08-03 16:15 +0000 [r210067] David Brooks <dbrooks at digium.com>
+
+ * channels/chan_dahdi.c: Fixes dialplan wildcard extension taking
+ precedence over call pickup code. Prior to this patch, a wildcard
+ extension in the dialplan (for example, _*.) would take
+ precedence over picking up a call in the channel's pickup group.
+ This patch simply moves the block of code handling pickup group
+ matching to above the extension matching code. (closes issue
+ #14735) Reported by: stevedavies Review:
+ https://reviewboard.asterisk.org/r/319/
+
+2009-08-03 16:11 +0000 [r210064-210066] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_oss.c, apps/app_playback.c, main/asterisk.exports,
+ configure, include/asterisk/autoconfig.h.in,
+ include/asterisk/compat.h, main/strcompat.c, configure.ac,
+ funcs/func_cut.c: Reverting index() fix, applying a different
+ methodology, based upon developer discussions. (related to issue
+ #15639)
+
+ * main/asterisk.exports, include/asterisk/compat.h: Helps if we
+ export the index() function. (Related to issue #15639)
+
+ * configure, include/asterisk/autoconfig.h.in, main/strcompat.c,
+ configure.ac: Apparently, some platforms don't have the index()
+ function. (closes issue #15639) Reported by: nmav
+
+2009-08-01 11:27 +0000 [r209838-209879] Russell Bryant <russell at digium.com>
+
+ * main/db1-ast/mpool/mpool.c: Resolve a valgrind warning about a
+ read from uninitialized memory. (issue #15396) Reported by:
+ aragon
+
+ * apps/app_milliwatt.c: Modify how Playtones() is used in
+ Milliwatt() to resolve gain issue. When Milliwatt() was changed
+ internally to use Playtones() so that the proper tone was used,
+ it introduced a drop in gain in the output signal. So, use the
+ playtones API directly and specify a volume argument such that
+ the output matches the gain of the original Milliwatt() code.
+ (closes issue #15386) Reported by: rue_mohr Patches:
+ issue_15386.rev2.diff uploaded by russell (license 2) Tested by:
+ rue_mohr
+
+2009-08-01 00:52 +0000 [r209759] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/misdn/isdn_lib.c, utils/frame.c, main/Makefile,
+ channels/misdn/ie.c: Minor changes inspired by testing with
+ latest GCC. The latest GCC (what will become 4.5.x) has a few new
+ warnings, that in these cases found some either downright buggy
+ code, or at least seriously poorly designed code that could be
+ improved.
+
+2009-07-28 00:12 +0000 [r209315] Tilghman Lesher <tlesher at digium.com>
+
+ * sounds/sounds.xml: Publish French extra sounds
+
+2009-07-27 17:44 +0000 [r209131] Mark Michelson <mmichelson at digium.com>
+
+ * main/udptl.c, configs/udptl.conf.sample: Allow for UDPTL to use
+ only even-numbered ports if desired. There are some VoIP
+ providers out there that will not accept SDP offers with odd
+ numbered UDPTL ports. While it is my personal opinion that these
+ VoIP providers are misinterpreting RFC 2327, it really is not a
+ big deal to play along with their silly little games. Of course,
+ since restricting UDPTL ports to only even numbers reduces the
+ range of available ports by half, so the option to use only even
+ port numbers is off by default. A user can enable the behavior by
+ setting use_even_ports=yes in udptl.conf. (closes issue #15182)
+ Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson
+ (license 60) Tested by: CGMChris
+
+2009-07-27 09:56 +0000 [r208990] Michiel van Baak <michiel at vanbaak.info>
+
+ * res/res_crypto.c: backport rev 205532 from trunk: pthread_self
+ returns a pthread_t which is not an unsigned int on all pthread
+ implementations. Casting it to an unsigned int fixes compiler
+ warnings.
+
+2009-07-27 01:18 +0000 [r208923] Jeff Peeler <jpeeler at digium.com>
+
+ * main/translate.c, channels/chan_iax2.c: Fix logic errors from
+ 208746
+
+2009-07-25 06:19 +0000 [r208746] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_skinny.c, main/translate.c, channels/chan_iax2.c:
+ Fix compiling under dev-mode with gcc 4.4.0. Mostly trivial
+ changes, but I did not know of any other way to fix the
+ "dereferencing type-punned pointer will break strict-aliasing
+ rules" error without creating a tmp variable in chan_skinny.
+
+2009-07-24 19:24 +0000 [r208622] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Don't impose an arbitrary limit on member lines
+ in queues.conf I know what some of you are thinking: "UGH! Mark,
+ why are you using ast_strdup and ast_free for the string when you
+ can just use ast_strdupa and let the memory free itself?! Have
+ the bats been chewing on your brain again?" Based on past
+ experiences, I don't like using ast_strdupa inside a loop. It's a
+ good way to potentially exhaust stack space. Also, since this
+ only happens when reloading queues, I don't think that heap
+ allocations and frees are going to be a huge problem. (closes
+ issue #15559) Reported by: amorsen
+
+2009-07-24 18:38 +0000 [r208592] Russell Bryant <russell at digium.com>
+
+ * apps/app_dial.c: Do not log an ERROR if autoservice_stop()
+ returns -1. This does not indicate an error. A return of -1 just
+ means that the channel has been hung up. (reported in
+ #asterisk-dev)
+
+2009-07-24 18:26 +0000 [r208386-208587] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Only send a BYE when hanging up a channel
+ that is up. For cases where Asterisk sends an INVITE and receives
+ a non 2XX final response, Asterisk would follow the INVITE
+ transaction by immediately sending a BYE, which was unnecessary.
+ (closes issue #14575) Reported by: chris-mac
+
+ * channels/chan_sip.c: Fix a problem where a 491 response could be
+ sent out of dialog. This generalizes the fix for issue 13849. The
+ initial fix corrected the problem that Asterisk would reply with
+ a 491 if a reinvite were received from an endpoint and we had not
+ yet received an ACK from that endpoint for the initial INVITE it
+ had sent us. This expansion also allows Asterisk to appropriately
+ handle an INVITE with authorization credentials if Asterisk had
+ not received an ACK from the previous transaction in which
+ Asterisk had responded to an unauthorized INVITE with a 407.
+ (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
+ uploaded by mmichelson (license 60) Tested by: klaus3000
+
+2009-07-23 19:19 +0000 [r208380] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Only set the priindication setting when
+ not performing a reload (closes issue #14696) Reported by:
+ fdecher
+
+2009-07-23 16:29 +0000 [r208262-208312] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Remove inaccurate XXX comment.
+
+ * channels/chan_sip.c: Properly handle 183 responses which do not
+ contain an SDP. (closes issue #15442) Reported by: ffloimair
+ Patches: 15442.patch uploaded by mmichelson (license 60) Tested
+ by: tkarl, ffloimair
+
+2009-07-22 20:23 +0000 [r207945-208083] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.exports, include/asterisk/compat.h: Export symbols
+ for functions included in our compatibility headers. (closes
+ issue #15556) Reported by: smw1218
+
+ * funcs/func_strings.c: Force an error if a blank is passed to
+ QUOTE (because the documentation states the argument is not
+ optional). This change makes URIENCODE and QUOTE behave
+ similarly, since the documentation states that the argument is
+ not optional, for both. (closes issue #15439) Reported by:
+ pkempgen Patches: 20090706__issue15439.diff.txt uploaded by
+ tilghman (license 14)
+
+2009-07-21 20:16 +0000 [r207786-207827] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Wait for wink before dialing when using
+ E&M wink signaling There was already code for other signaling
+ types in dahdi_handle_event to handle dialing if a dial operation
+ dial string was present. Simply add SIG_EMWINK to the list.
+ (closes issue #14434) Reported by: araasch
+
+ * channels/chan_dahdi.c: Revert r207573, this approach could
+ potentially block for an unacceptable amount of time.
+
+2009-07-21 14:26 +0000 [r207714] Mark Michelson <mmichelson at digium.com>
+
+ * main/manager.c: Document default timeout for AMI originations.
+ AST-224
+
+2009-07-21 13:04 +0000 [r207647] Kevin P. Fleming <kpfleming at digium.com>
+
+ * codecs/lpc10/Makefile, main/db1-ast/Makefile, Makefile,
+ agi/Makefile, codecs/Makefile, utils/Makefile, main/Makefile,
+ codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules,
+ pbx/Makefile, res/Makefile, channels/Makefile: Ensure that
+ user-provided CFLAGS and LDFLAGS are honored. This commit changes
+ the build system so that user-provided flags (in ASTCFLAGS and
+ ASTLDFLAGS) are supplied to the compiler/linker *after* all flags
+ provided by the build system itself, so that the user can
+ effectively override the build system's flags if desired. In
+ addition, ASTCFLAGS and ASTLDFLAGS can now be provided *either*
+ in the environment before running 'make', or as variable
+ assignments on the 'make' command line. As a result, the use of
+ COPTS and LDOPTS is no longer necessary, so they are no longer
+ documented, but are still supported so as not to break existing
+ build systems that supply them when building Asterisk.
+
+2009-07-20 23:23 +0000 [r207573] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Wait for wink before dialing when using
+ E&M wink signaling This patch adds a new dahdi_wait function to
+ specifically wait for the wink event. If the wink is not
+ eventually received the channel is hung up. (closes issue #14434)
+ Reported by: araasch Patches: emwinkmod uploaded by araasch
+ (license 693)
+
+2009-07-20 19:39 +0000 [r207423] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Answer video SDP offers properly when
+ videosupport is not enabled. Copied from Review board: In issue
+ 12434, the reporter describes a situation in which audio and
+ video is offered on the call, but because videosupport is
+ disabled in sip.conf, Asterisk gives no response at all to the
+ video offer. According to RFC 3264, all media offers should have
+ a corresponding answer. For offers we do not intend to actually
+ reply to with meaningful values, we should still reply with the
+ port for the media stream set to 0. In this patch, we take note
+ of what types of media have been offered and save the information
+ on the sip_pvt. The SDP in the response will take into account
+ whether media was offered. If we are not otherwise going to
+ answer a media offer, we will insert an appropriate m= line with
+ the port set to 0. It is important to note that this patch is
+ pretty much a bandage being applied to a broken bone. The patch
+ *only* helps for situations where video is offered but
+ videosupport is disabled and when udptl_pt is disabled but T.38
+ is offered. Asterisk is not guaranteed to respond to every media
+ offer. Notable cases are when multiple streams of the same type
+ are offered. The 2 media stream limit is still present with this
+ patch, too. In trunk and the 1.6.X branches, things will be a bit
+ different since Asterisk also supports text in SDPs as well.
+ (closes issue #12434) Reported by: mnnojd Review:
+ https://reviewboard.asterisk.org/r/311 Review:
+ https://reviewboard.asterisk.org/r/313
+
+2009-07-20 16:26 +0000 [r207360] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Only do the chan->fdno check in ast_read() in a
+ developer build. I changed this check to only happen in a
+ dev-mode build. I also added a comment explaining what is going
+ on. I also made it so that detection of this situation does not
+ affect ast_read() operation. (closes issue #14723) Reported by:
+ seadweller
+
+2009-07-20 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.26
+
+2009-07-13 Leif Madsen <lmadsen at digium.com>
+
+ * Release Asterisk 1.4.26-rc6
+
+2009-07-13 15:12 +0000 [r206126] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c: Print CID match in "show dialplan". (closes issue
+ #14702) Reported by: klaus3000 Patches:
+ patch_asterisk_1.4.23_CID_matching.txt uploaded by klaus3000
+ (license 65)
+
+2009-07-10 17:39 +0000 [r205877] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Properly ACK 487 responses to canceled
+ INVITEs. From the review board request: The fix from review 298
+ has exposed a new bug in chan_sip. When we hang up an outgoing
+ call, we first will dump all the outstanding packets on the
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