[svn-commits] lmadsen: tag 1.4.31-rc1 r256151 - /tags/1.4.31-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Apr 5 09:34:15 CDT 2010
Author: lmadsen
Date: Mon Apr 5 09:34:13 2010
New Revision: 256151
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=256151
Log:
Importing files for 1.4.31-rc1 release.
Added:
tags/1.4.31-rc1/.lastclean (with props)
tags/1.4.31-rc1/.version (with props)
tags/1.4.31-rc1/ChangeLog (with props)
Added: tags/1.4.31-rc1/.lastclean
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--- tags/1.4.31-rc1/ChangeLog (added)
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+2010-04-05 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.31-rc1 Released
+
+2010-04-02 23:45 +0000 [r256009-256014] Russell Bryant <russell at digium.com>
+
+ * channels/chan_local.c: Resolve a deadlock that occurs due to a
+ pointless call to ast_bridged_channel() (closes issue #16840)
+ Reported by: bzing2 Patches: patch.txt uploaded by bzing2
+ (license 902) issue_16840.rev1.diff uploaded by russell (license
+ 2) Tested by: bzing2, russell
+
+ * main/channel.c: Remove extremely verbose debug message.
+
+2010-03-31 19:09 +0000 [r255591] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Ensure line terminators in email are
+ consistent. Fixes an issue with certain Mail Transport Agents,
+ where attachments are not interpreted correctly. (closes issue
+ #16557) Reported by: jcovert Patches:
+ 20100308__issue16557__1.4.diff.txt uploaded by tilghman (license
+ 14) 20100308__issue16557__1.6.0.diff.txt uploaded by tilghman
+ (license 14) 20100308__issue16557__trunk.diff.txt uploaded by
+ tilghman (license 14) Tested by: ebroad, zktech Reviewboard:
+ https://reviewboard.asterisk.org/r/544/
+
+2010-03-31 17:42 +0000 [r255503] Leif Madsen <lmadsen at digium.com>
+
+ * apps/app_dial.c, configs/sip.conf.sample: Add documentation
+ clarifying when 't' and 'T' can be used. (closes issue #17021)
+ Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad
+
+2010-03-30 20:56 +0000 [r255322-255409] Russell Bryant <russell at digium.com>
+
+ * channels/chan_h323.c: Don't kill Asterisk if the H323 listener
+ does not start.
+
+ * pbx/pbx_dundi.c: Don't make Asterisk not start if pbx_dundi fails
+ to initialize.
+
+2010-03-25 20:41 +0000 [r254714-254800] Jason Parker <jparker at digium.com>
+
+ * utils/Makefile: Don't remove local copies of utils in uninstall.
+
+ * main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS
+ issue with out-of-tree modules. Take 2, without ABI breakage this
+ time. Review: https://reviewboard.asterisk.org/r/588/
+
+2010-03-25 18:51 +0000 [r254639] Russell Bryant <russell at digium.com>
+
+ * Makefile, /: Update Asterisk 1.4 to use menuselect trunk. Review:
+ https://reviewboard.asterisk.org/r/590/
+
+2010-03-25 17:33 +0000 [r254452-254552] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/acl.h: Add doxygen for acl.h Review:
+ https://reviewboard.asterisk.org/r/528
+
+ * main/rtp.c: Several fixes regarding RFC2833 DTMF detection. Here
+ is a copy and paste of the details from my request on reviewboard
+ that dealt with these changes: Fix 1. The first change in place
+ is to fix Mantis issue 15811, which deals with a situation where
+ Asterisk will incorrectly interpret out of order RFC2833 frames
+ as duplicate DTMF digits. For instance, we would receive a
+ sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1
+ seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 seqno 7:
+ DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch when we
+ received the frame with seqno 5, we would interpret this as a new
+ DTMF 1. With this patch, we will check the seqno of the incoming
+ digit and not process the frame if the seqno is lower than the
+ last recorded seqno. Note that we do not record the seqno of the
+ dropped DTMF frame for future processing. While the above
+ situation is what was designed to be fixed, the patch is written
+ in such a way that the following would also be fixed too: seqno
+ 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) seqno 13:
+ DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno 15: DTMF 2
+ (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In this
+ second situation, the beginning of the DTMF 2 arrives before the
+ final end frame of the DTMF 1. With the patch, seqno 12 is no
+ processed and thus we properly interpret the DTMF. Fix 2. The
+ second change in place is to fix an issue like the following:
+ seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet
+ lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end)
+ *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had
+ code in place that was supposed to properly end the previously
+ unended DTMF 1. The problem was that the code was essentially a
+ no-op. The code would set up an end frame for the DTMF 1 but
+ would immediately overwrite the frame with the begin for DTMF 2.
+ I changed process_dtmf_rfc2833() so that instead of returning a
+ single frame, it is given as an output parameter a list of
+ frames. Each frame that needs to be returned is appended to this
+ list. Fix 3. The final change is a minor one where an
+ AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco
+ DTMF or an RFC 3389 frame and no frame was returned, then we
+ would return &ast_null_frame. The problem is that earlier in the
+ function, we may have generated an AST_CONTROL_SRCCHANGE frame
+ and put it in the list of frames we wish to return. This frame
+ would be lost in such a case. The patch fixes this problem
+ Review: https://reviewboard.asterisk.org/r/558
+
+2010-03-25 15:57 +0000 [r254451] Terry Wilson <twilson at digium.com>
+
+ * main/file.c: Handle new SRCCHANGE control message here too
+
+2010-03-24 00:37 +0000 [r254235] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_monitor.c: Ensure that monitor recordings are written to
+ the correct location (again) This is an extension to 248860. As
+ such the dialplan test has been extended: ; non absolute path,
+ not combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
+ exten => 5040, n, dial(sip/5001) ; absolute path, not combined
+ exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten =>
+ 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1,
+ monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
+ exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n,
+ dial(sip/5001) ; combined: changemonitor from no path to non
+ absolute path exten => 5044, 1, monitor(wav,monitor_test6,m)
+ exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this
+ wasn't possible before exten => 5044, n, dial(sip/5001) ; non
+ absolute path, combined exten => 5045, 1,
+ monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n,
+ dial(sip/5001) ; absolute path, combined exten => 5046, 1,
+ monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n,
+ dial(sip/5001) ; no path, combined exten => 5047, 1,
+ monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ;
+ combined: changemonitor from non absolute to absolute (leaves
+ tmp/jeff) exten => 5048, 1,
+ monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n,
+ changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5049, 1,
+ monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n,
+ changemonitor(tmp/jeff/monitor_test14) exten => 5049, n,
+ dial(sip/5001) ; combined: changemonitor from no path to absolute
+ exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n,
+ changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n,
+ dial(sip/5001) ; combined: changemonitor from absolute to no path
+ (leaves /tmp/jeff) exten => 5051, 1,
+ monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n,
+ changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ;
+ not combined: changemonitor from non absolute to no path (leaves
+ tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
+ exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to non
+ absolute exten => 5053, 1, monitor(wav,monitor_test21) exten =>
+ 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n,
+ dial(sip/5001) ; not combined: changemonitor from non absolute to
+ absolute (leaves tmp/jeff) exten => 5054, 1,
+ monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n,
+ changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n,
+ dial(sip/5001) ; not combined: changemonitor from absolute to non
+ absolute (leaves /tmp/jeff) exten => 5055, 1,
+ monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n,
+ changemonitor(tmp/jeff/monitor_test25) exten => 5055, n,
+ dial(sip/5001) ; not combined: changemonitor from no path to
+ absolute exten => 5056, 1, monitor(wav,monitor_test26) exten =>
+ 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056,
+ n, dial(sip/5001) ; not combined: changemonitor from absolute to
+ no path (leaves /tmp/jeff) exten => 5057, 1,
+ monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n,
+ changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001)
+
+2010-03-23 22:45 +0000 [r254046-254161] Jason Parker <jparker at digium.com>
+
+ * main/astobj2.c, main/lock.c (removed), main/channel.c,
+ main/Makefile, include/asterisk/astobj2.h, UPGRADE.txt,
+ include/asterisk/lock.h: Revert revisions 254046 and 254098.
+
+ * UPGRADE.txt: Add note about the out-of-tree module ABI changes.
+
+ * main/astobj2.c, main/lock.c (added), main/channel.c,
+ main/Makefile, include/asterisk/astobj2.h,
+ include/asterisk/lock.h: Allow out-of-tree modules to load,
+ regardless of DEBUG_THREADS/DEBUG_CHANNEL_LOCKS differences. This
+ can be guaranteed by forcing the ABI to no longer change when
+ these compiler flags are set. An unfortunate side-effect to this
+ is that there is an ABI change here. However, there is some
+ mitigation. Existing modules *will* fail to load since they would
+ require functions that no longer exist. Review:
+ https://reviewboard.asterisk.org/r/508/
+
+2010-03-22 19:50 +0000 [r253799] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_features.c: Unconditionally copy the caller's account
+ code to the called party. (related to issue #16331)
+
+2010-03-21 14:26 +0000 [r253631-253670] Russell Bryant <russell at digium.com>
+
+ * main/Makefile: Fix final link on FreeBSD by adding the
+ PTHREAD_CFLAGS.
+
+ * main/sched.c, Makefile, apps/app_dial.c, channels/chan_dahdi.c,
+ main/manager.c, res/res_features.c, main/http.c, main/utils.c,
+ pbx/pbx_dundi.c, apps/app_followme.c: Resolve a number of FreeBSD
+ build issues.
+
+2010-03-18 17:57 +0000 [r253252-253349] Leif Madsen <lmadsen at digium.com>
+
+ * apps/app_userevent.c: Typo found while fixing issue #16961.
+
+ * doc/localchannel.txt: Synchronize text in localchannels.txt and
+ localchannels.tex. (issue #16963)
+
+ * doc/localchannel.txt: Update new Local channel documentation. The
+ original reporter, Kobaz, of an issue with a Local channel that
+ inspired the Local channel documentation provided some tweaks to
+ the documentation after testing what I had written. Hopefully
+ anything that was vague or unclear has been cleaned up by these
+ changes. (closes issue #16963) Reported by: kobaz Patches:
+ localchannel-2.txt uploaded by kobaz (license 834) Tested by:
+ kobaz, lmadsen
+
+2010-03-17 16:25 +0000 [r253158] Terry Wilson <twilson at digium.com>
+
+ * main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c,
+ channels/chan_mgcp.c, channels/chan_sip.c,
+ include/asterisk/rtp.h: Revert API change in release branches
+ This re-renames ast_rtp_update_source to ast_rtp_new_source
+
+2010-03-17 00:26 +0000 [r253018] Leif Madsen <lmadsen at digium.com>
+
+ * configs/say.conf.sample: Add french snipset to say.conf. Add the
+ french snipset to say.conf. (Closes issue #15799)
+
+2010-03-16 20:52 +0000 [r252766-252928] Russell Bryant <russell at digium.com>
+
+ * Makefile.rules: Backport chan_sip build fix for Mac OSX 10.6 from
+ trunk.
+
+ * codecs/gsm/Makefile: Use uname -s, as done in trunk.
+
+ * codecs/gsm/Makefile: Apply codec_gsm Mac OS X 10.6 build fix that
+ is in trunk and 1.6.X.
+
+ * utils/Makefile: Don't treat warnings as errors for muted. muted
+ supports OS X, but uses functions marked as deprecated in 10.6.
+ However, the functions are still supported, so just ignore the
+ warnings for now and allow the build to proceed.
+
+2010-03-16 18:46 +0000 [r252761] Leif Madsen <lmadsen at digium.com>
+
+ * configs/extensions.ael.sample: Additional extensions.ael global
+ variable fixes. Fixing up a couple more overlapping global
+ variable namespaces shared with extensions.conf.sample. Also
+ noticed a few of the lines that were commented out didn't have
+ the closing semi-colon so I added that as well. (issue #17035)
+
+2010-03-15 21:43 +0000 [r252617] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/init.d/org.asterisk.asterisk.plist: Uh, yeah. Umask. I'm
+ stupid.
+
+2010-03-15 20:48 +0000 [r252531-252533] Leif Madsen <lmadsen at digium.com>
+
+ * configs/extensions.ael.sample: Update extensions.ael file to not
+ overlap extensions.conf. Updated the extensions.ael file so the
+ global variables don't overlap those that we have in
+ extensions.conf (sample files). This way unexpected things won't
+ happed hopefully if both pbx_ael and res_config are loaded.
+ (closes issue #17035) Reported by: pprindeville
+
+ * configure, configs/extensions.ael.sample: Revert last commit that
+ had bad changed to configure.
+
+ * configure, configs/extensions.ael.sample: Update extensions.ael
+ file to not overlap extensions.conf. Updated the extensions.ael
+ file so the global variables don't overlap those that we have in
+ extensions.conf (sample files). This way unexpected things won't
+ happed hopefully if both pbx_ael and res_config are loaded.
+ (closes issue #17035) Reported by: pprindeville
+
+2010-03-15 01:39 +0000 [r252361-252366] Tilghman Lesher <tlesher at digium.com>
+
+ * Makefile: Typo
+
+ * main/asterisk.c, Makefile,
+ contrib/init.d/org.asterisk.asterisk.plist (added): Launch
+ Asterisk on Mac OS X with launchd. Reviewboard:
+ https://reviewboard.asterisk.org/r/551/
+
+2010-03-13 00:30 +0000 [r252175] Terry Wilson <twilson at digium.com>
+
+ * main/rtp.c, channels/chan_mgcp.c, main/channel.c,
+ channels/chan_sip.c, channels/chan_skinny.c,
+ include/asterisk/rtp.h, channels/chan_h323.c,
+ configs/sip.conf.sample, include/asterisk/frame.h: Merged
+ revisions 252089 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 |
+ twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
+ Only change the RTP ssrc when we see that it has changed This
+ change basically reverts the change reviewed in
+ https://reviewboard.asterisk.org/r/374/ and instead limits the
+ updating of the RTP synchronization source to only those times
+ when we detect that the other side of the conversation has
+ changed the ssrc. The problem is that SRCUPDATE control frames
+ are sent many times where we don't want a new ssrc, including
+ whenever Asterisk has to send DTMF in a normal bridge. This is
+ also not the first time that this mistake has been made. The
+ initial implementation of the ast_rtp_new_source function also
+ changed the ssrc--and then it was removed because of this same
+ issue. Then, we put it back in again to fix a different issue.
+ This patch attempts to only change the ssrc when we see that the
+ other side of the conversation has changed the ssrc. It also
+ renames some functions to make their purpose more clear. Review:
+ https://reviewboard.asterisk.org/r/540/ ........
+
+2010-03-12 19:58 +0000 [r251986-251997] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Forward declaring dahdi_pri was already
+ done.
+
+ * channels/chan_dahdi.c: Make chan_dahdi wakeup_sub() prototype not
+ conditional.
+
+2010-03-11 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.30 released
+
+2010-03-04 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.4.30-rc3 released
+
+2010-03-03 21:28 +0000 [r250613] Leif Madsen <lmadsen at digium.com>
+
+ * doc/localchannel.txt: Update existing Local channel
+ documentation. A complete re-write of the Local channel
+ documentation has been performed, with the existing information
+ from localchannel.txt and localchannel.tex merged in. (issue
+ #16637) Reported by: kobaz Patches: localchannel.tex uploaded by
+ lmadsen (license 10) localchannel.txt uploaded by lmadsen
+ (license 10) Tested by: lmadsen, jsmith, mmichelson
+
+2010-03-03 19:04 +0000 [r250480] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Make sure to clear red alarm after
+ polarity reversal. From the issue: The automatic overnight line
+ tests (or manual ones) used on UK (BT) lines causes a red alarm
+ on a dahdi / TDM400P connected channel. This is because the line
+ uses voltage tests (battery loss) and polarity reversal. The
+ polarity reversal causes chan_dahdi to initiate v23 CallerID
+ processing but during this the event DAHDI_EVENT_NOALARM is
+ ignored so that the alarm is never cleared. (closes issue #14163)
+ Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded
+ by jedi98 (license 653) Tested by: mattbrown, Chainsaw,
+ mikeeccleston
+
+2010-03-03 18:02 +0000 [r250394] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: fixes problem with duplicate TXREQ packets
+ When Asterisk receives an IAX2 TXREQ packet, try_transfer() will
+ call store_by_transfercallno() to link the chan_iax2_pvt struct
+ into iax_transfercallno_pvts. If a duplicate TXREQ packet is
+ received for the same call, the pvt struct will be linked into
+ iax_transfercallno_pvts multiple times. This patch fixes this.
+ Thanks rain for debugging this and providing a patch! (closes
+ issue #16904) Reported by: rain Patches:
+ iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested
+ by: rain, dvossel
+
+2010-03-02 21:08 +0000 [r250041-250050] Leif Madsen <lmadsen at digium.com>
+
+ * doc/imapstorage.txt: Update IMAP documentation. Update the IMAP
+ documentation to make it clear that storing voicemails in the
+ same folder as a large number of emails could potentially cause
+ significant slow downs when writing or retrieving voicemails.
+ (closes issue #16704) Reported by: TimeHider Tested by: lmadsen,
+ TimeHider
+
+ * configs/cdr.conf.sample: Update documentation to clarify purpose
+ of unanswered option. (closes issue #16267) Reported by: elsto
+ Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license
+ 10) Tested by: davidw, elsto
+
+ * doc/configuration.txt: Update documentation to not imply we
+ support overriding options. (issue #16855) Reported by: davidw
+
+2010-03-02 19:36 +0000 [r249845-249946] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * apps/app_echo.c: revert ability to exit echo app caused a
+ regression, as only supported VOICE, not VIDEO etc. Left in small
+ formatting change. (issue #16880)
+
+ * apps/app_echo.c: fixes ability to exit echo app when called from
+ a ISDN channel, null frames prevent '#' exit. Now only echo back
+ VOICE and DTMF frames (issue #16880) Reported by: alecdavis
+ Patches: based on echo_exit_1-6-1.diff.txt uploaded by alecdavis
+ (license 585) Tested by: alecdavis
+
+2010-03-01 19:35 +0000 [r249671] Sean Bright <sean at malleable.com>
+
+ * apps/app_voicemail.c: Fix crash in app_voicemail related to
+ message counting. We were passing a 'struct inprocess **' and
+ treating it like a 'struct inprocess *' causing a segfault.
+ (closes issue #16921) Reported by: whardier Patches:
+ 20100301_issue16921.patch uploaded by seanbright (license 71)
+ Tested by: whardier
+
+2010-03-01 17:02 +0000 [r249536] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_local.c: Modify queued frames from local channels
+ to not set the other side to up In this case, attended transfers
+ were broken due to ast_feature_request_and_dial detecting the
+ channel being set to up before the answer frame could be read and
+ therefore failing to mark the channel as ready. This fix is a
+ regression fix for 244785, which should continue to work properly
+ as well. (closes issue #16816) Reported by: jamhed Tested by:
+ jamhed, corruptor
+
+2010-02-27 23:51 +0000 [r249365] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/chan_dahdi.c: overlap receiving: automatically send CALL
+ PROCEEDING when dialplan starts Following Q.931 5.2.4 When the
+ user has determined that sufficient call information has been
+ received the user shall stop T302 and send CALL PROCEEDING to the
+ network. Previously timeouts were possible if the dialplan took a
+ long time to issue any response back to the network. Verified
+ that our local TELCO also does the same. (issue #16789) Reported
+ by: alecdavis Patches: based on overlap_receiving_trunk.diff.txt
+ uploaded by alecdavis (license 585) Tested by: alecdavis (closes
+ issue #16789)
+
+2010-02-27 14:07 +0000 [r249234] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_iax2.c: add a reference to the now-published IAX2
+ RFC
+
+2010-02-26 17:04 +0000 [r249100] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: For T.38 reINVITEs treat a 606 the same as a
+ 488. (closes issue #16792) Reported by: vrban Patches:
+ t38_606.patch uploaded by vrban (license 756)
+
+2010-02-25 21:22 +0000 [r248860] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_monitor.c: Ensure that monitor recordings are written to
+ the correct location (again) This is an extension to 248757. As
+ such the dialplan test has been extended: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) exten => 5043, 1,
+ monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n,
+ changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001)
+ exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n,
+ changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by
+ design and emits a warning exten => 5044, n, dial(sip/5001)
+
+2010-02-25 21:21 +0000 [r248859] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: Some platforms clear /var/run at boot, which
+ makes connecting a remote console... difficult. Previously, we
+ only created the default /var/run/asterisk directory at install
+ time. While we could create it in the init script, that would not
+ work for those who start asterisk manually from the command line.
+ So the safest thing to do is to create it as part of the Asterisk
+ boot process. This also changes the ownership of the directory,
+ because the pid and ctl files are created after we setuid/setgid.
+ (closes issue #16802) Reported by: Brian Patches:
+ 20100224__issue16802.diff.txt uploaded by tilghman (license 14)
+ Tested by: tzafrir
+
+2010-02-25 18:06 +0000 [r248668-248757] Jeff Peeler <jpeeler at digium.com>
+
+ * res/res_monitor.c: Ensure that monitor recordings are written to
+ the correct location. Recordings should be placed in the monitor
+ directory when a non-absolute path is used. Exact dialplan used
+ for testing: exten => 5040, 1,
+ monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n,
+ dial(sip/5001) exten => 5041, 1,
+ monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n,
+ dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b)
+ exten => 5042, n, dial(sip/5001) ABE-2101
+
+ * apps/app_voicemail.c: Make deletion of temporary greetings work
+ properly with IMAP_STORAGE This same patch was merged in 220833,
+ but was skipped in this branch erroneously. (closes issue #16170)
+ Reported by: francesco_r
+
+2010-02-24 21:02 +0000 [r248582] Tilghman Lesher <tlesher at digium.com>
+
+ * main/logger.c: Remove color code sequences from verbose messages
+ that go to logfiles. (closes issue #16786) Reported by: dodo
+ Patches: logger2.patch uploaded by dodo (license 989) Tested by:
+ tilghman
+
+2010-02-23 16:26 +0000 [r248396] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: fixes invite with replaces deadlock (closes
+ issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4
+ uploaded by dvossel (license 671) Tested by: pwalker, dvossel
+
+2010-02-22 13:52 +0000 [r248268] Olle Johansson <oej at edvina.net>
+
+ * apps/app_meetme.c: Don't log to debug unless debug is turned on
+
+2010-02-20 22:25 +0000 [r248106] Olle Johansson <oej at edvina.net>
+
+ * main/rtp.c: Make sure we support RTCP compound messages with zero
+ reports
+
+2010-02-19 19:11 +0000 [r248012] Tilghman Lesher <tlesher at digium.com>
+
+ * main/loader.c, /: Backport crash fix from trunk to 1.4, whereby
+ 'core show gracefully' could crash Asterisk. (closes issue
+ #16470) Reported by: kjotte
+
+2010-02-19 17:18 +0000 [r247910] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: Merged revision 247904 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
+ .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri,
+ 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing
+ consistent with other channel technologies. The processing of
+ DTMF tones on the receiving side of an ISDN channel is
+ inconsistent with the way it is handled in other channels,
+ especially DAHDI analog. This causes DTMF tones sent from an ISDN
+ phone to be doubled at the connected party. We are using the
+ following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes
+ Option one is necessary because the asterisk DSP DTMF detection
+ is better than mISDN's internal DSP. Not as many false positives.
+ Option two is necessary to transmit DTMF tones end to end when
+ mISDN channels are connected to SIP channels with out of band
+ DTMF for example. The symptom is that DTMF tones sent by an ISDN
+ phone are doubled on the way through asterisk when two mISDN
+ channels are connected with a Local channel in between or if it
+ is bridged to an analog channel. The doubling of DTMF tones is
+ because DTMF is passed inband to asterisk by the mISDN channel
+ and passed out of band once again after the release of the DTMF
+ tone. Passing it inband is wrong. Neither an analog channel nor
+ SIP channel passes DTMF inband if configured to inband DTMF.
+ Analog and SIP channels filter out the DTMF tones because they
+ use the voice frames returned by ast_dsp_process. But chan_misdn
+ passes the unfiltered input voice frames instead. To overcome one
+ aspect of the problem, the doubling of DTMF tones when two mISDN
+ channels are directly bridged, someone made an 'optimization',
+ where in that case the DTMF tone passed out-of-band to the peer
+ channel is not translated to an inband tone at the transmit side.
+ This optimization is bad because it does not work in general. For
+ example, analog channels or mISDN channels when bridged through
+ an intermediary local channel will generate DTMF tones from
+ out-of-band information. Also, of course, it must not be done
+ when there is no inband DTMF available. This patch fixes the
+ issue. Now chan_misdn will filter the received inband DTMF signal
+ the same as other channel types. Another change included: No need
+ to build an extra translation path because ast_process_dsp does
+ it if required. Patches: misdn-dtmf.patch JIRA ABE-2080
+
+2010-02-18 19:38 +0000 [r247651] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_features.c: Copy the calling party's account code to the
+ called party if they don't already have one. (closes issue
+ #16331) Reported by: bluefox Tested by: mnicholson
+
+2010-02-18 16:53 +0000 [r247502-247508] Leif Madsen <lmadsen at digium.com>
+
+ * README-SERIOUSLY.bestpractices.txt: Add additional link to best
+ practices document per jsmith.
+
+ * README-SERIOUSLY.bestpractices.txt (added): Add best practices
+ documentation. (issue #16808) Reported by: lmadsen (issue #16810)
+ Reported by: Nick_Lewis Tested by: lmadsen Review:
+ https://reviewboard.asterisk.org/r/507/
+
+2010-02-18 04:19 +0000 [r247422] Russell Bryant <russell at digium.com>
+
+ * Makefile, sounds/Makefile: Tweak argument handling for wget in
+ the sounds Makefile. 1) Fix the check to see if we are using wget
+ to not be full of fail. The configure script populates this
+ variable with the absolute path to wget if it is found, so it
+ didn't work. 2) Allow some extra arguments to be passed in for
+ wget. This is just a simple change to allow our Bamboo build
+ script to tell wget to be quiet and not fill up our logs with
+ download status output.
+
+2010-02-17 16:24 +0000 [r247168] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Make sure that when autofill is disabled that
+ callers not in the front of the queue cannot place calls. (closes
+ issue #16834) Reported by: kebl0155 Patches:
+ app_queue_no_autofill.v1.patch uploaded by kebl0155 (license 356)
+
+2010-02-15 23:42 +0000 [r246709] Tilghman Lesher <tlesher at digium.com>
+
+ * Makefile: Make the menuselect instructions correct by allowing
+ 'make menuselect' to actually solve dependency problems.
+ (Previously, it would fail out again with the same message about
+ running 'make menuselect', which was NOT at all helpful.)
+
+2010-02-12 23:30 +0000 [r246545] David Vossel <dvossel at digium.com>
+
+ * main/channel.c: lock channel during datastore removal On channel
+ destruction the channel's datastores are removed and destroyed.
+ Since there are public API calls to find and remove datastores on
+ a channel, a lock should be held whenever datastores are removed
+ and destroyed. This resolves a crash caused by a race condition
+ in app_chanspy.c. (closes issue #16678) Reported by:
+ tim_ringenbach Patches: datastore_destroy_race.diff uploaded by
+ tim ringenbach (license 540) Tested by: dvossel
+
+2010-02-12 18:52 +0000 [r246460] Jason Parker <jparker at digium.com>
+
+ * main/channel.c: Fix some silly formatting, and remove unnecessary
+ option_debug checks
+
+2010-02-10 17:44 +0000 [r246115] David Vossel <dvossel at digium.com>
+
+ * apps/app_queue.c: fixes random deadlock in app_queue with
+ use_weight during reload (closes issue #16677) Reported by:
+ tim_ringenbach Patches: app_queue_use_weight_deadlock.diff
+ uploaded by tim ringenbach (license 540)
+
+2010-02-10 13:37 +0000 [r245944] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/extensions.conf.sample: Include examples of FILTER usage
+ in extension patterns where a "." may be a risk.
+
+2010-02-10 08:24 +0000 [r245909] Olle Johansson <oej at edvina.net>
+
+ * res/res_smdi.c: Make sure that res_smdi loads regardless of
+ configuration, since chan_dahdi depends on res_smdi
+
+2010-02-09 22:55 +0000 [r245792] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Fixes iaxs and iaxsl size off by one issue.
+ 2^15 = 32768 which is the maximum allowed iax2 callnumber.
+ Creating the iaxs and iaxsl array of size 32768 means the maximum
+ callnumber is actually out of bounds. This causes a nasty crash.
+ (closes issue #15997) Reported by: exarv Patches: iax_fix.diff
+ uploaded by dvossel (license 671)
+
+2010-02-08 20:39 +0000 [r245496] Jason Parker <jparker at digium.com>
+
+ * main/ast_expr2.fl, main/ast_expr2f.c: Remove reference of
+ documentation in source directory. People don't always build
+ Asterisk from source (distro packages, anybody?).
+
+2010-02-08 11:57 +0000 [r245422] Olle Johansson <oej at edvina.net>
+
+ * res/res_features.c: Res_features depends on res_adsi in 1.4
+
+2010-02-05 18:32 +0000 [r245044] Kevin P. Fleming <kpfleming at digium.com>
+
+ * contrib/firmware (removed), LICENSE: Remove contrib/firmware
+ directory as it is empty Remove explicit license for IAXy
+ firmware as it is no longer included in the tree
+
+2010-02-05 17:03 +0000 [r244926] Sean Bright <sean at malleable.com>
+
+ * main/asterisk.c: Update main copyright date.
+
+2010-02-04 23:20 +0000 [r244785] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_local.c: Change channel state on local channels for
+ busy,answer,ring. Previously local channels channel state never
+ changed. This became problematic when the state of the other side
+ of the local channel was lost, for example during a masquerade.
+ Changing the state of the local channel allows for the scenario
+ to be detected when the channel state is set to ringing, but the
+ peer isn't ringing. The specific problem scenario is described in
+ 164201. Although this was noted on one of the issues, here is the
+ tested dialplan verified to work: exten =>
+ 9700,1,Dial(Local/*9700 at default&Local/#9700 at default) exten =>
+ *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
+ exten => *9700,n,wait(3) ;3 works, 1 did not exten =>
+ *9700,n,Dial(SIP/5001) exten => #9700,1,Wait(1) ;1 works, 3 did
+ not exten =>
+ #9700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes
+ issue #14992) Reported by: davidw
+
+2010-02-01 23:13 +0000 [r244070-244242] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: Backup and restore original textfile, for
+ prosthesis (gerund of prepend). Also, fix menuselect such that
+ changing voicemail build options correctly causes rebuild.
+ (closes issue #16415) Reported by: tomo1657 Patches:
+ prepention.patch uploaded by tomo1657 (license 484) (with
+ modifications by me to backport to 1.4)
+
+ * res/res_features.c: When a transferer hangs up during an attended
+ transfer BEFORE the transfer is answered, don't stop playing MOH.
+ (closes issue #16513) Reported by: litnimax Patches:
+ atxfer_moh_16513.patch uploaded by gknispel proformatique
+ (license 261) Tested by: litnimax
+
+ * main/channel.c, channels/chan_local.c: Revert previous chan_local
+ fix (r236981) and fix instead by destroying expired frames in the
+ queue. (closes issue #16525) Reported by: kobaz Patches:
+ 20100126__issue16525.diff.txt uploaded by tilghman (license 14)
+ 20100129__issue16525__1.6.0.diff.txt uploaded by tilghman
+ (license 14) Tested by: kobaz, atis (closes issue #16581)
+ Reported by: ZX81 (closes issue #16681) Reported by: alexr1
+
+2010-01-28 18:48 +0000 [r243862-243863] Leif Madsen <lmadsen at digium.com>
+
+ * BUGS: Oops, correct wrong link (https vs. http) in previous
+ commit.
+
+ * BUGS: Update location of bug tracker in documentation.
+
+2010-01-28 15:03 +0000 [r243779] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Fix a bogus third argument to
+ ast_copy_string().
+
+2010-01-27 20:35 +0000 [r243570-243691] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_queue.c: Revert 243570, I should have looked at this
+ closer. Will reopen the issue, but am leaving the review closed
+ as the change was pointless. (issue #16488)
+
+ * apps/app_queue.c: Extend announcement URL used with Queue from 80
+ chars to PATH_MAX. (closes issue #16488) Reported by: syspert
+ Patches: soundfilelen.pacth-2 uploaded by syspert (license 938)
+ Review: https://reviewboard.asterisk.org/r/475/
+
+2010-01-27 18:06 +0000 [r243486] Mark Michelson <mmichelson at digium.com>
+
+ * main/pbx.c: Use a safe list traversal while checking for
+ duplicate vars in pbx_builtin_setvar_helper.
+
+2010-01-26 23:55 +0000 [r243390] David Vossel <dvossel at digium.com>
+
+ * res/res_features.c: fixes bug with channel receiving wrong
+ privileges after call parking (closes issue #16429) Reported by:
+ Yasuhiro Konishi Patches: features.c.diff uploaded by Yasuhiro
+ Konishi (license 947) Tested by: dvossel
+
+2010-01-26 18:19 +0000 [r243258] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c: Remove unnecessary code in ast_read as issue
+ 16058 has been fully solved now.
+
+2010-01-25 21:50 +0000 [r242852-242969] Tilghman Lesher <tlesher at digium.com>
+
+ * main/Makefile, pbx/Makefile: Err, and use the new menuselect
+ define, too.
+
+ * build_tools/cflags.xml, build_tools/menuselect-deps.in,
+ configure, configure.ac: Only rebuild parsers by an option in
+ menuselect
+
+ * configure, main/Makefile, configure.ac, pbx/Makefile: Restore
+ FreeBSD to able-to-compile-ish-mode
+
+2010-01-25 20:08 +0000 [r242850-242851] Olle Johansson <oej at edvina.net>
+
+ * main/manager.c: Remove debugging that indeed should have been
+ gone before commit. Sorry.
+
+ * main/manager.c: Report error when writing to functions returns
+ error in AMI setvar action
+
+2010-01-25 05:42 +0000 [r242520-242728] Tilghman Lesher <tlesher at digium.com>
+
+ * main/Makefile, pbx/Makefile: Buildbot pointed out an error
+ (thanks, buildbot!)
+
+ * main/Makefile, pbx/Makefile: Oops, should have used CMD_PREFIX,
+ not ECHO_PREFIX, for the commands.
+
+ * main/Makefile: Make the build of the Asterisk expression parser
+ match that of the AEL parser.
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ pbx/ael/ael_lex.c, pbx/Makefile, makeopts.in: Only rebuild bison
+ and flex source files on demand, if bison and flex are detected
+ by the configure script. Changed after discussion on the -dev
+ list about possible unnecessary build failures, due to
+ checkouts/untars causing these special source files to possibly
+ be newer than their resulting C files. This should additionally
+ ensure that nobody need learn about extra Makefile arguments to
+ ensure the proper files get rebuilt when changes are made to
+ these special source files.
+
+2010-01-22 21:44 +0000 [r242423] Tilghman Lesher <tlesher at digium.com>
+
[... 27399 lines stripped ...]
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