[asterisk-commits] lmadsen: tag 1.6.0.6-rc1 r175595 - /tags/1.6.0.6-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Feb 13 14:08:06 CST 2009
Author: lmadsen
Date: Fri Feb 13 14:08:06 2009
New Revision: 175595
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=175595
Log:
Importing files for 1.6.0.6-rc1 release
Added:
tags/1.6.0.6-rc1/.lastclean (with props)
tags/1.6.0.6-rc1/.version (with props)
tags/1.6.0.6-rc1/ChangeLog (with props)
Added: tags/1.6.0.6-rc1/.lastclean
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--- tags/1.6.0.6-rc1/ChangeLog (added)
+++ tags/1.6.0.6-rc1/ChangeLog Fri Feb 13 14:08:06 2009
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+2009-02-13 Leif Madsen <lmadsen at digium.com>
+
+ * Released 1.6.0.6-rc1
+
+2009-02-13 16:43 +0000 [r175550] Joshua Colp <jcolp at digium.com>
+
+ * /, apps/app_record.c: Merged revisions 175549 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 |
+ file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add
+ an option to keep the recorded file upon hangup. (closes issue
+ #14341) Reported by: fnordian ........
+
+2009-02-12 21:41 +0000 [r175369] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 175368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 |
+ russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines
+ Remove useless string copy, and make sscanf safe again ........
+
+2009-02-12 21:27 +0000 [r175347] Tilghman Lesher <tlesher at digium.com>
+
+ * main/udptl.c, /: Merged revisions 175334 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009)
+ | 16 lines Merged revisions 175311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009)
+ | 9 lines Fix crashes when receiving certain T.38 packets. Also,
+ increase the maximum size of T.38 packets and warn users when
+ they try to set the limits above those maximums. (closes issue
+ #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: schern ........
+ ................
+
+2009-02-12 20:59 +0000 [r175299-175301] Jeff Peeler <jpeeler at digium.com>
+
+ * main/features.c: Fix mistake in merging conflict from 175299.
+
+ * /, main/features.c: Merged revisions 175298 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009)
+ | 15 lines Merged revisions 175294 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009)
+ | 9 lines Fix ParkedCall event information for From field in the
+ case of a blind transfer If the parker information can not be
+ obtained from the peer, try and see if the BLINDTRANSFER channel
+ variable has been set. Previously, a blind transfer to the
+ ParkAndAnnounce app would return nothing for the From. Closes
+ AST-189 ........ ................
+
+2009-02-12 20:46 +0000 [r175256-175296] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 175295 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 |
+ russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines
+ Avoid using ast_strdupa() in a loop. ........
+
+ * build_tools/cflags.xml, /: Merged revisions 175255 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009)
+ | 4 lines Don't enable something by default that has a dependency
+ on something _not_ enabled by default. menuselect was not happy
+ with this. ........
+
+2009-02-12 18:00 +0000 [r175189] Jeff Peeler <jpeeler at digium.com>
+
+ * /, main/features.c: Merged revisions 175188 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009)
+ | 12 lines Merged revisions 175187 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009)
+ | 6 lines Fix crash in event of failed attempt to transfer to
+ parking The peer may not necessarily exist, such as in the case
+ of a transfer to ParkAndAnnounce. In this case don't try to play
+ a sound to it. ........ ................
+
+2009-02-12 17:03 +0000 [r175126] Russell Bryant <russell at digium.com>
+
+ * main/rtp.c, /: Merged revisions 175125 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009)
+ | 35 lines Merged revisions 175124 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009)
+ | 27 lines Don't send DTMF for infinite time if we do not receive
+ an END event. I thought that this was going to end up being a
+ pretty gnarly fix, but it turns out that there was actually
+ already a configuration option in rtp.conf, dtmftimeout, that was
+ intended to handle this situation. However, in between Asterisk
+ 1.2 and Asterisk 1.4, the code that processed the option got
+ lost. So, this commit brings it back to life. The default timeout
+ is 3 seconds. However, it is worth noting that having this be
+ configurable at all is not really the recommended behavior in RFC
+ 2833. From Section 3.5 of RFC 2833: Limiting the time period of
+ extending the tone is necessary to avoid that a tone "gets
+ stuck". Regardless of the algorithm used, the tone SHOULD NOT be
+ extended by more than three packet interarrival times. A slight
+ extension of tone durations and shortening of pauses is generally
+ harmless. Three seconds will pretty much _always_ be far more
+ than three packet interarrival times. However, that behavior is
+ not required, so I'm going to leave it with our legacy behavior
+ for now. Code from svn/asterisk/team/russell/issue_14460 (closes
+ issue #14460) Reported by: moliveras ........ ................
+
+2009-02-12 16:33 +0000 [r175122] Mark Michelson <mmichelson at digium.com>
+
+ * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
+ 175121 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 |
+ mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11
+ lines Make lock information for ao2_trylock be more useful and
+ gnarly Core show locks information involving an ao2_trylock did
+ not show the function that called ao2_trylock, but would instead
+ show ao2_trylock as the source of the lock. This is not useful
+ when trying to debug locking issues. One bizarre note is that
+ this logic is already in 1.4 but somehow did not get merged to
+ trunk or the 1.6.X branches. ........
+
+2009-02-12 14:27 +0000 [r175059-175090] Philippe Sultan <philippe.sultan at gmail.com>
+
+ * /, channels/chan_gtalk.c: Merged revisions 175089 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r175089 | phsultan | 2009-02-12 15:25:03 +0100 (Thu, 12 Feb 2009)
+ | 6 lines Issue a warning message if our candidate's IP is the
+ loopback address. (closes issue #13985) Reported by: jcovert
+ Tested by: phsultan ........
+
+ * /, channels/chan_gtalk.c: Merged revisions 175058 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r175058 | phsultan | 2009-02-12 11:31:36 +0100
+ (Thu, 12 Feb 2009) | 20 lines Merged revisions 175029 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009)
+ | 12 lines Set the initiator attribute to lowercase in our
+ replies when receiving calls. This attribute contains a JID that
+ identifies the initiator of the GoogleTalk voice session. The
+ GoogleTalk client discards Asterisk's replies if the initiator
+ attribute contains uppercase characters. (closes issue #13984)
+ Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by
+ jcovert (license 551) Tested by: jcovert ........
+ ................
+
+2009-02-11 23:04 +0000 [r174765-174949] Mark Michelson <mmichelson at digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 174948 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174948 | mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb
+ 2009) | 35 lines Fix odd "thank you" sound playing behavior in
+ app_queue.c If someone has configured the queue to play an
+ position or holdtime announcement, then it is odd and potentially
+ unexpected to hear a "Thank you for your patience" sound when no
+ position or holdtime was actually announced. This fixes the
+ announcement so that the "thanks" sound is only played in the
+ case that a position or holdtime was actually announced. There is
+ a way that the "thank you" sound can be played without a position
+ or holdtime, and that is to set announce-frequency to a value but
+ keep announce-position and announce-holdtime both turned off.
+ (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch
+ uploaded by putnopvut (license 60) Tested by: caspy
+ ................
+
+ * apps/app_dial.c, main/channel.c, main/pbx.c, /,
+ apps/app_dictate.c, apps/app_waitforsilence.c,
+ include/asterisk/channel.h: Merged revisions 174945 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb
+ 2009) | 29 lines Fix 'd' option for app_dial and add new option
+ to Answer application The 'd' option would not work for channel
+ types which use RTP to transport DTMF digits. The only way to
+ allow for this to work was to answer the channel if we saw that
+ this option was enabled. I realized that this may cause issues
+ with CDRs, specifically with giving false dispositions and answer
+ times. I therefore modified ast_answer to take another parameter
+ which would tell if the CDR should be marked answered. I also
+ extended this to the Answer application so that the channel may
+ be answered but not CDRified if desired. I also modified
+ app_dictate and app_waitforsilence to only answer the channel if
+ it is not already up, to help not allow for faulty CDR answer
+ times. All of these changes are going into Asterisk trunk. For
+ 1.6.0 and 1.6.1, however, all the changes except for the change
+ to the Answer application will go in since we do not introduce
+ new features into stable branches (closes issue #14164) Reported
+ by: DennisD Patches: 14164.patch uploaded by putnopvut (license
+ 60) Tested by: putnopvut Review:
+ http://reviewboard.digium.com/r/145 ........
+
+ * apps/app_chanspy.c, /: Merged revisions 174805 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174805 |
+ mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11
+ lines Fix potential for stack overflows in app_chanspy.c When
+ using the 'g' or 'e' options, the stack allocations that were
+ used could cause a stack overflow if a spyer stayed on the line
+ long enough without actually successfully spying on anyone. The
+ problem has been corrected by using static buffers and copying
+ the contents of the appropriate strings into them instead of
+ using functions like alloca or ast_strdupa ........
+
+ * main/manager.c, /: Merged revisions 174764 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174764 |
+ mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21
+ lines Fix an fd leak that would occur in HTTP AMI sessions The
+ explanation behind this fix is a bit complicated, and I've
+ already typed it up in the code as a huge comment inside of
+ manager.c, so I'll give the abridged version here. We needed a
+ way to separate action-specific data from session-specific data.
+ Unfortunately, the only way to maintain API compatibility and to
+ not have to change every single manager action was to rename the
+ current mansession structure and wrap it inside a new mansession
+ structure which actually contains action- specific data. (closes
+ issue #14364) Reported by: awk Patches: 14364_better.patch
+ uploaded by putnopvut (license 60) Tested by: putnopvut Review:
+ http://reviewboard.digium.com/r/148/ ........
+
+2009-02-10 20:16 +0000 [r174711] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174710 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174710 |
+ file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines
+ Only decrease inringing count if above zero. (issue #13238)
+ Reported by: kowalma ........
+
+2009-02-10 18:19 +0000 [r174596] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, main/jitterbuf.c: Merged revisions 174584 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb
+ 2009) | 25 lines Merged revisions 174583 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb
+ 2009) | 18 lines Improve behavior of jitterbuffer when
+ maxjitterbuffer is set. This change improves the way the
+ jitterbuffer handles maxjitterbuffer and dramatically reduces the
+ number of frames dropped when maxjitterbuffer is exceeded. In the
+ previous jitterbuffer, when maxjitterbuffer was exceeded, all new
+ frames were dropped until the jitterbuffer is empty. This change
+ modifies the code to only drop frames until maxjitterbuffer is no
+ longer exceeded. Also, previously when maxjitterbuffer was
+ exceeded, dropped frames were not tracked causing stats for
+ dropped frames to be incorrect, this change also addresses that
+ problem. (closes issue #14044) Patches: bug14044-1.diff uploaded
+ by mnicholson (license 96) Tested by: mnicholson Review:
+ http://reviewboard.digium.com/r/144/ ........ ................
+
+2009-02-10 15:39 +0000 [r174544] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174543 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174543 |
+ file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines
+ Make the logic for inuse and inringing manipluation match that of
+ 1.4. The old broken logic would reset the values back to 0 during
+ certain scenarios causing the wrong state to be reported. (closes
+ issue #14399) Reported by: caspy (issue #13238) Reported by:
+ kowalma ........
+
+2009-02-10 05:06 +0000 [r174439] Steve Murphy <murf at digium.com>
+
+ * apps/app_rpt.c: For some strange reason, I didn't think 1.6.0
+ needed this fix. I was wrong. Here it is.
+
+2009-02-09 17:28 +0000 [r174322-174328] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174327 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r174327 |
+ mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3
+ lines Fix something I messed up in the merge I just did ........
+
+ * /, channels/chan_sip.c: Merged revisions 174301 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb
+ 2009) | 20 lines Merged revisions 174282 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb
+ 2009) | 12 lines Don't do an SRV lookup if a port is specified
+ RFC 3263 says to do A record lookups on a hostname if a port has
+ been specified, so that's what we're going to do. See section
+ 4.2. (closes issue #14419) Reported by: klaus3000 Patches:
+ patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000
+ (license 65) ........ ................
+
+2009-02-09 14:50 +0000 [r174220] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_musiconhold.c: Merged revisions 174219 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r174219 | file | 2009-02-09 10:49:24 -0400 (Mon,
+ 09 Feb 2009) | 11 lines Merged revisions 174218 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4
+ lines Don't overwrite our pointer to the music class when music
+ on hold stops. We will use this if it starts again to see if we
+ can resume the music where it left off. (closes issue #14407)
+ Reported by: mostyn ........ ................
+
+2009-02-07 16:17 +0000 [r174151] Russell Bryant <russell at digium.com>
+
+ * /, res/snmp/agent.c: Merged revisions 174149 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174149 | russell | 2009-02-07 10:16:50 -0600 (Sat, 07 Feb 2009)
+ | 10 lines Merged revisions 174148 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009)
+ | 2 lines Fix a race condition that could cause a crash. ........
+ ................
+
+2009-02-06 23:59 +0000 [r174085] Dwayne M. Hubbard <dhubbard at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 174084 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009)
+ | 13 lines Merged revisions 174082 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009)
+ | 5 lines check ast_strlen_zero() before calling ast_strdupa() in
+ sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter
+ didn't actually upload a properly-formed patch, instead a
+ modified chan_sip.c file was uploaded. I created a patch to
+ determine the changes, then modified the suggested changes to
+ create a proper fix. The summary above is a complete description
+ of the changes. (closes issue #13547) Reported by: tecnoxarxa
+ Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258)
+ Tested by: tecnoxarxa ........ ................
+ ------------------------------------------------------------------------
+
+2009-02-06 19:29 +0000 [r173986-174042] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 174041 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4
+ lines Don't subscribe to a mailbox on pseudo channels. It is
+ futile. This solves an issue where duplicated pseudo channels
+ would cause a crash because the first one would unsubscribe and
+ the next one would also try to unsubscribe the same subscription.
+ (closes issue #14322) Reported by: amessina ........
+
+ * /, channels/chan_sip.c: Merged revisions 173974 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) |
+ 15 lines Merged revisions 173967-173968 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4
+ lines Some clients do not put the call-id for replaces at the
+ beginning, so support it being anywhere in the string. (closes
+ issue #14350) Reported by: fhackenberger ........ r173968 | file
+ | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a
+ debug message I put in by accident. ........ ................
+
+2009-02-06 16:33 +0000 [r173963] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 173952 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb
+ 2009) | 14 lines Merged revisions 173917 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb
+ 2009) | 7 lines Limit the addition of the Contact header in SIP
+ responses according to various SIP RFCs. (closes issue #13602)
+ Reported by: hjourdain Tested by: mnicholson ........
+ ................
+
+2009-02-05 23:51 +0000 [r173774-173777] Mark Michelson <mmichelson at digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 173776 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu,
+ 05 Feb 2009) | 14 lines Update extensions.conf.sample to be
+ correct. In trunk, the only necessary change pointed out was that
+ the call to ChanIsAvail uses an option that has been removed. For
+ the 1.6.1 branch, however, it appears that the sample file is
+ badly in need of updating since there are |'s used all over the
+ place there. My tentative plan is just to copy trunk's sample
+ config file to those branches since the info there is most
+ up-to-date and should be correct for use in 1.6.1 Thanks to macli
+ in #asterisk-dev for bringing this up ........
+
+ * apps/app_voicemail.c, /: Merged revisions 173773 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb
+ 2009) | 7 lines Properly set "seen" and "unseen" flags when
+ moving messages from the new to the old folder when using IMAP
+ for voicemail storage (closes issue #13905) Reported by: jaroth
+ Patches: foldermove_v2.patch uploaded by jaroth (license 50)
+ ........
+
+2009-02-05 21:04 +0000 [r173698] Jeff Peeler <jpeeler at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 173697 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r173697 | jpeeler | 2009-02-05 15:00:26 -0600
+ (Thu, 05 Feb 2009) | 18 lines Merged revisions 173696 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009)
+ | 12 lines Add new configuration option to make shared IMAP
+ mailboxes function as expected. The new option is "imapvmshareid"
+ which is an ID to tag multiple mailboxes using the same IMAP
+ storage location to function as one mailbox. This allows all
+ messages to be retrieved for any user in the group. The patch
+ alters the 'X-Asterisk-VM-Extension' header that is responsible
+ for matching voicemails for a given user. (closes issue #13673)
+ Reported by: howardwilkinson ........ ................
+
+2009-02-05 20:34 +0000 [r173590-173694] Mark Michelson <mmichelson at digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 173693 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb
+ 2009) | 20 lines Merged revisions 173692 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb
+ 2009) | 12 lines Fix situations where queue members could be
+ autopaused unexpectedly Specifically, this patch prevents us from
+ autopausing members when we receive a busy or congestion frame
+ from them. (closes issue #14376) Reported by: fiddur Patches:
+ 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur
+ ........ ................
+
+ * apps/app_mixmonitor.c, /: Merged revisions 173593 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r173593 | mmichelson | 2009-02-05 12:48:55 -0600
+ (Thu, 05 Feb 2009) | 11 lines Merged revisions 173592 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb
+ 2009) | 3 lines Add some missing cleanup to app_mixmonitor
+ ........ ................
+
+ * apps/app_mixmonitor.c, /: Merged revisions 173589 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r173589 | mmichelson | 2009-02-05 12:34:06 -0600
+ (Thu, 05 Feb 2009) | 33 lines Merged revisions 173559 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb
+ 2009) | 25 lines Fix a problem where a channel pointer becomes
+ invalid due to masquerading or hanging up. app_mixmonitor runs
+ its own thread to monitor the channel's activity and write the
+ mixed audio to a file. Since this thread runs independently of
+ the channel, it is possible that the mixmonitor thread's channel
+ pointer will point to freed memory when the channel either is
+ masqueraded or hangs up (technically, both cases are hangups, but
+ we need to handle the cases slightly differently). The solution
+ for this is to employ a datastore, which has the nice benefit of
+ allowing us to hook into channel masquerades and hangups and
+ update our pointer as necessary. If this looks familiar, this
+ same technique is employed in app_chanspy. app_chanspy is a bit
+ more involved since it does a lot more operations on the channel
+ that is being spied upon. app_mixmonitor does have an extra touch
+ that app_chanspy doesn't have, though. Since there is a thread
+ race between the channel's thread and the mixmonitor thread on a
+ hangup, we em- ploy a condition-and-boolean combination to ensure
+ that the channel thread finishes with our structure before the
+ mixmonitor thread attempts to free it. No crashes! (closes issue
+ #14374) Reported by: aragon Patches: 14374.patch uploaded by
+ putnopvut (license 60) Tested by: aragon, putnopvut ........
+ ................
+
+2009-02-05 16:23 +0000 [r173554] Jeff Peeler <jpeeler at digium.com>
+
+ * build_tools/menuselect-deps.in: fix WORKING_FORK detection
+
+2009-02-05 00:11 +0000 [r173548] Tilghman Lesher <tlesher at digium.com>
+
+ * build_tools/menuselect-deps.in: regenerate with bootstrap.sh
+
+2009-02-04 23:44 +0000 [r173546-173547] Jeff Peeler <jpeeler at digium.com>
+
+ * /: I messed up and accidentally reverted the trunk-merged prop
+ before committing 173546. Added it manually.
+
+ * main/features.c: Merged revisions 173500 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009)
+ | 23 lines Merged revisions 173211 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009)
+ | 17 lines Parking attempts made to one end of a bridge no longer
+ will hang up due to a parking failure. Parking attempts made
+ using either one-touch, or doing either a blind or assisted
+ transfer to the parking extension now keep up the bridge instead
+ of hanging up the attempted parked party. Normal causes for the
+ parking attempt to fail includes the specific specified extension
+ (via PARKINGEXTEN) not being available or if all the parking
+ spaces are currently in use. To avoid having to reverse a
+ masquerade park_space_reserve was made to provide foresight if a
+ parking attempt will succeed and if so reserve the parking space.
+ (closes issue #13494) Reported by: mdu113 Reviewed by Russell:
+ http://reviewboard.digium.com/r/133/ ........ ................
+
+2009-02-04 22:23 +0000 [r173534] Mark Michelson <mmichelson at digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 173507 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r173507 |
+ mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7
+ lines Fix some areas where the incorrect interface was passed to
+ ast_device_state I swear it feels like I already did this once...
+ (closes issue #14359) Reported by: francesco_r ........
+
+2009-02-04 18:55 +0000 [r173460] Tilghman Lesher <tlesher at digium.com>
+
+ * main/tcptls.c, /: Merged revisions 173458 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r173458 |
+ tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines
+ When using a socket as a FILE *, the stdio functions will
+ sometimes try to do an fseek() on the stream, which is an invalid
+ operation for a socket. Turning off buffering explicitly lets the
+ stdio functions know they cannot do this, thus avoiding a
+ potential error. (closes issue #14400) Reported by: fnordian
+ Patches: tcptls.patch uploaded by fnordian (license 110) ........
+
+2009-02-04 17:46 +0000 [r173355-173398] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 173397 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb
+ 2009) | 11 lines Merged revisions 173396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb
+ 2009) | 3 lines Revert my previous change because it was stupid
+ ........ ................
+
+ * apps/app_chanspy.c, /: Merged revisions 173393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb
+ 2009) | 11 lines Merged revisions 173392 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb
+ 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever
+ matter, but it's needed. ........ ................
+
+ * /, main/file.c: Merged revisions 173354 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r173354 |
+ mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30
+ lines Fix a problem where file playback would cause fds to remain
+ open forever The problem came from the fact that a frame read
+ from a format interpreter was not freed. Adding a call to
+ ast_frfree fixed this. The explanation for why this caused the
+ problem is a bit complex, but here goes: There was a problem in
+ all versions of Asterisk where the embedded frame of a filestream
+ structure was referenced after the filestream was freed. This was
+ fixed by adding reference counting to the filestream structure.
+ The refcount would increase every time that a filestream's frame
+ pointer was pointing to an actual frame of data. When the frame
+ was freed, the refcount would decrease. Once the refcount reached
+ 0, the filestream was freed, and as part of the operation, the
+ open files were closed as well. Thus it becomes more clear why a
+ missing ast_frfree would cause a reference leak and cause the
+ files to not be closed. You may ask then if there was a frame
+ leak before this patch. The answer to that is actually no! The
+ filestream code was "smart" enough to know that since the frame
+ we received came from a format interpreter, the frame had no
+ malloced data and thus didn't need to be freed. Now, however,
+ there is cleanup that needs to be done when we finish with the
+ frame, so we do need to call ast_frfree on the frame to be sure
+ that the refcount for the filestream is decremented
+ appropriately. (closes issue #14384) Reported by: fiddur Patches:
+ 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur,
+ putnopvut ........
+
+2009-02-04 00:45 +0000 [r173312] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c, /: Merged revisions 173311 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r173311 |
+ tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10
+ lines Ensure that commas placed in the middle of extension
+ character classes do not interfere with correct parsing of the
+ extension. Also, if an unterminated character class DOES make its
+ way into the pbx core (through some other method), ensure that it
+ does not crash Asterisk. (closes issue #14362) Reported by:
+ Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: Corydon76 ........
+
+2009-02-03 23:41 +0000 [r173250] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Fixes issue with IAX2 transfer not handing
+ of calls. Fixes issue with IAX2 transfers not taking place. As it
+ was, a call that was being transfered would never be handed off
+ correctly to the call ends because of how call numbers were
+ stored in a hash table. The hash table, "iax_peercallno_pvt",
+ storing all the current call numbers did not take into account
+ the complications associated with transferring a call, so a
+ separate hash table was required. This second hash table
+ "iax_transfercallno_pvt" handles calls being transfered, once the
+ call transfer is complete the call is removed from the transfer
+ hash table and added to the peer hash table resuming normal
+ operations. Addition functions were created to handle storing,
+ removing, and comparing items in the iax_transfercallno_pvt
+ table. (issue #13468) Review:
+ http://reviewboard.digium.com/r/140/
+
+2009-02-03 00:26 +0000 [r173111] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 173104 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r173104 | tilghman | 2009-02-02 18:24:52 -0600
+ (Mon, 02 Feb 2009) | 12 lines Merged revisions 173070 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009)
+ | 5 lines Add warning to standard config, that globals may be
+ overridden by other dialplan configuration files. (closes issue
+ #14388) Reported by: macli ........ ................
+
+2009-02-02 23:59 +0000 [r173068] Terry Wilson <twilson at digium.com>
+
+ * /, main/features.c: Merged revisions 173067 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009)
+ | 9 lines Merged revisions 173066 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009)
+ | 2 lines Fix a feature inheritance bug I added after code review
+ ........ ................
+
+2009-02-02 18:15 +0000 [r172896] Leif Madsen <lmadsen at digium.com>
+
+ * /, configs/res_ldap.conf.sample: Merged revisions 172894 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02
+ Feb 2009) | 7 lines Update the res_ldap.conf file with a better
+ working example. (closes issue #13861) Reported by: scramatte
+ Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage
+ (license 10) Tested by: jcovert ........
+
+2009-02-01 02:45 +0000 [r172707-172742] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 172741 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009)
+ | 4 lines Blank argument crashes Asterisk (closes issue #14377)
+ Reported by: amorsen ........
+
+ * /, funcs/func_strings.c: Merged revisions 172706 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r172706 | tilghman | 2009-01-31 10:40:59 -0600 (Sat, 31 Jan 2009)
+ | 7 lines Don't increment the loop, now that incrementing is
+ taken care of by the decoder function. (closes issue #14363)
+ Reported by: andrew53 Patches: func_strings_filter.patch uploaded
+ by andrew53 (license 519) ........
+
+2009-01-31 00:06 +0000 [r172635-172637] Terry Wilson <twilson at digium.com>
+
+ * configs/features.conf.sample, /: Merged revisions 172581 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r172581 | twilson | 2009-01-30 15:50:03 -0600 (Fri, 30
+ Jan 2009) | 2 lines Remove incorret line from sample config
+ ........
+
+ * configs/features.conf.sample, apps/app_dial.c,
+ main/global_datastores.c, /, main/features.c,
+ include/asterisk/global_datastores.h, CHANGES: Merged revisions
+ 172580 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009)
+ | 44 lines Merged revisions 172517 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009)
+ | 37 lines Fix feature inheritance with builtin features When
+ using builtin features like parking and transfers, the
+ AST_FEATURE_* flags would not be set correctly for all instances
+ when either performing a builtin attended transfer, or parking a
+ call and getting the timeout callback. Also, there was no way on
+ a per-call basis to specify what features someone should have on
+ picking up a parked call (since that doesn't involve the Dial()
+ command). There was a global option for setting whether or not
+ all users who pickup a parked call should have
+ AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or
+ PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan
+ variable which can be set either in the dialplan or with setvar
+ in channels that support it. This variable can be set to any
+ combination of 't', 'k', 'w', and 'h' (case insensitive matching
+ of the equivalent dial options), to set what features should be
+ activated on this channel. The patch moves the setting of the
+ features datastores into the bridging code instead of app_dial to
[... 48766 lines stripped ...]
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