[asterisk-commits] lmadsen: tag 1.6.0.16-rc1 r219207 - /tags/1.6.0.16-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Sep 17 11:02:43 CDT 2009
Author: lmadsen
Date: Thu Sep 17 11:02:38 2009
New Revision: 219207
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=219207
Log:
Importing files for 1.6.0.16-rc1 release.
Added:
tags/1.6.0.16-rc1/.lastclean (with props)
tags/1.6.0.16-rc1/.version (with props)
tags/1.6.0.16-rc1/ChangeLog (with props)
Added: tags/1.6.0.16-rc1/.lastclean
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.6.0.16-rc1/.lastclean?view=auto&rev=219207
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Added: tags/1.6.0.16-rc1/ChangeLog
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.6.0.16-rc1/ChangeLog?view=auto&rev=219207
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--- tags/1.6.0.16-rc1/ChangeLog (added)
+++ tags/1.6.0.16-rc1/ChangeLog Thu Sep 17 11:02:38 2009
@@ -1,0 +1,53356 @@
+2009-09-17 Leif Madsen <lmadsen at digium.com>
+
+ * Released Asterisk 1.6.0.16-rc1
+
+2009-09-17 15:44 +0000 [r219198] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c, /, include/asterisk/cdr.h,
+ include/asterisk/channel.h: Merged revisions 219139 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500
+ (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep
+ 2009) | 10 lines 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 23:52 +0000 [r219064] Tilghman Lesher <tlesher at digium.com>
+
+ * main/config.c, configs/extensions.conf.sample, /: Merged
+ revisions 219061 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009)
+ | 15 lines Merged revisions 219023 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009)
+ | 8 lines 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 19:26 +0000 [r218935] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 |
+ mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12
+ lines Reverse order of args to fread. This way, we don't always
+ write a null byte into byte 1 of the buffer (closes issue #15905)
+ Reported by: ebroad Patches: freadfix.patch uploaded by ebroad
+ (license 878) Tested by: ebroad ........
+
+2009-09-16 18:44 +0000 [r218931] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 |
+ file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On
+ TCP and TLS connections do not attempt to stop retransmission of
+ the packet internally. This was preventing responses from being
+ properly processed because the packet was not being found causing
+ handle_response to return prematurely. ........
+
+2009-09-16 18:11 +0000 [r218869] David Brooks <dbrooks at digium.com>
+
+ * main/pbx.c, /: Merged revisions 218868 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009)
+ | 20 lines Merged revisions 218867 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009)
+ | 13 lines 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 13:36 +0000 [r218800] Russell Bryant <russell at digium.com>
+
+ * contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged
+ revisions 218799 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009)
+ | 16 lines Merged revisions 218798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009)
+ | 9 lines 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 22:39 +0000 [r218732] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 218731 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500
+ (Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009)
+ | 6 lines 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 19:31 +0000 [r218690] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218687 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 |
+ dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines
+ upward bound checking for port string to int conversion ........
+
+2009-09-15 16:21 +0000 [r218601] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218586 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep
+ 2009) | 15 lines Merged revisions 218578 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep
+ 2009) | 8 lines 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 16:05 +0000 [r218580] Tilghman Lesher <tlesher at digium.com>
+
+ * /, apps/app_followme.c: Merged revisions 218579 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009)
+ | 16 lines Merged revisions 218577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009)
+ | 9 lines 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 15:42 +0000 [r218505-218573] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 218566 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 |
+ mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4
+ lines Use a better method of ensuring null-termination of the
+ buffer while reading the SDP when using TCP. ........
+
+ * /, channels/chan_sip.c: Merged revisions 218499,218504 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue,
+ 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent
+ over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53
+ -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP
+ socket is null-terminated. ........
+
+2009-09-15 15:03 +0000 [r218501] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, sounds/Makefile: Merged revisions 218500 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep
+ 2009) | 9 lines Merged revisions 218497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep
+ 2009) | 1 line Use proper hostname for downloading sound files.
+ ........ ................
+
+2009-09-14 22:49 +0000 [r218431] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_dahdi.c: Merged revisions 218430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218430 | jpeeler | 2009-09-14 17:38:25 -0500 (Mon, 14 Sep 2009)
+ | 18 lines Merged revisions 218401 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009)
+ | 11 lines 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 19:49 +0000 [r218362] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, /, configs/voicemail.conf.sample,
+ sounds/Makefile: Merged revisions 218361 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r218361 | tilghman | 2009-09-14 14:29:48 -0500 (Mon, 14 Sep 2009)
+ | 11 lines Recorded merge of revisions 218331 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009)
+ | 4 lines 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 15:22 +0000 [r218244] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 218224 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500
+ (Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep
+ 2009) | 8 lines 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) ........ ................
+
+2009-09-13 19:10 +0000 [r218216] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * channels/chan_phone.c, /: gcc 4.4: Remove a nop memset size 0
+ that annoys gcc This memset doesn't write beyond the end of the
+ buffer. (tmpbuf has size of 4). Merged revisions 218184 via
+ svnmerge from http://svn.digium.com/svn/asterisk/trunk
+
+2009-09-12 13:10 +0000 [r218108] Michiel van Baak <michiel at vanbaak.info>
+
+ * main/rtp.c: Use the ip for the new 'rtp set debug ip <foo>'.
+ Since 1.6.X still has the deprecated 'rtp debug ip <foo>' this
+ patch is different from the fix that went into trunk (closes
+ issue #15711) Reported by: davidw Patches:
+ 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7)
+ Tested by: davidw
+
+2009-09-11 05:58 +0000 [r217920-218051] Tilghman Lesher <tlesher at digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 217990 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009)
+ | 10 lines Merged revisions 217989 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009)
+ | 3 lines 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: Merged revisions 217916 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 |
+ tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines
+ Make calltoken support work with realtime users and peers.
+ ........
+
+2009-09-10 22:31 +0000 [r217858-217913] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: sip peer matching by address only with
+ TCP/TLS This patch removes the contact header matching logic and
+ adds logic to match all tcp/tls connections by ip only. Thanks to
+ oej for finding the issue and suggesting solutions. Review:
+ https://reviewboard.asterisk.org/r/355/
+
+ * /, channels/chan_iax2.c: Merged revisions 217807 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500
+ (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009)
+ | 22 lines 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. (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 19:53 +0000 [r217736] mnick <mnick at localhost>:
+
+ * /, res/res_musiconhold.c: Merged revisions 217730 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r217730 | mnick | 2009-09-10 14:39:41 -0500 (Thu, 10 Sep 2009) |
+ 17 lines Sets the correct musicclass after an announcement
+ (closes issue #15279) Reported by: mbeckwell Patches: patch.txt
+ uploaded by mnick (license ) Tested by: mnick (closes issue
+ #15832) Reported by: mbeckwell Patches: patch.txt uploaded by
+ mnick (license 874) Tested by: mnick ........
+
+2009-09-10 12:16 +0000 [r217596] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 217593 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 |
+ oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines
+ Include ActionID in all events that are responsed to AMI Action
+ SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy
+ Patches: manager_SIPshowregistry_actionid.patch uploaded by nic
+ bellamy (license 299) ........
+
+2009-09-09 20:15 +0000 [r217484] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * /, res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc
+ 4.4 has more strict rules for aliasing. It doesn't like a struct
+ sockaddr_in pointer pointing to a struct sockaddr. So we make it
+ a union. Merged revisions 217445 via svnmerge from
+ http://svn.digium.com/svn/asterisk/trunk
+
+2009-09-09 11:33 +0000 [r217405] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 217368 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 |
+ oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not
+ having any TLS session to write to is a serious XMIT_ERROR.
+ ........
+
+2009-09-08 21:45 +0000 [r217281] Kevin P. Fleming <kpfleming at digium.com>
+
+ * configure: Commit regenerated configure script that I missed
+ earlier.
+
+2009-09-08 20:31 +0000 [r217209] Tilghman Lesher <tlesher at digium.com>
+
+ * /, apps/app_meetme.c: Merged revisions 217199 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009)
+ | 14 lines Merged revisions 217156 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009)
+ | 7 lines 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 ........ ................
+
+2009-09-08 16:38 +0000 [r217075] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, include/asterisk/autoconfig.h.in, configure.ac: Merged
+ revisions 217074 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 |
+ kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9
+ lines Ensure that the default autoconf CFLAGS are not used. A
+ recent change to the configure script that allows the user to
+ specify CFLAGS and/or LDFLAGS to the script had the unfortunate
+ side effect of letting autoconf's default CFLAGS (-g -O2) feed in
+ to the rest of the build system, thereby overriding the
+ DONT_OPTIMIZE setting in menuselect. That problem is now
+ corrected. ........
+
+2009-09-08 15:35 +0000 [r217034] Tilghman Lesher <tlesher at digium.com>
+
+ * /, res/res_limit.c: Merged revisions 217033 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r217033 |
+ tilghman | 2009-09-08 10:30:18 -0500 (Tue, 08 Sep 2009) | 4 lines
+ Remove what appears to be an unnecessary define. (closes issue
+ #15851) Reported by: tzafrir ........
+
+2009-09-08 14:28 +0000 [r216996] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 216993 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 |
+ dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines
+ caller id number empty parse_uri was not being given the correct
+ scheme's, as a result, uri parsing did not parse the username
+ correctly. One of the side effects of this is an empty caller id.
+ (closes issue #15839) Reported by: ebroad Patches:
+ blank_cidv2.patch uploaded by ebroad (license 878)
+ parse_uri_fix.diff uploaded by dvossel (license 671) Tested by:
+ ebroad, dvossel ........
+
+2009-09-07 16:38 +0000 [r216645-216843] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 216842 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 |
+ oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines
+ Make sure we reset global_exclude_static at channel reload
+ ........
+
+ * /, channels/chan_sip.c: Merged revisions 216695 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 |
+ oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If
+ there is no session timer in the INVITE, set it to default value
+ (not unset minimum = -1) Patch by oej closes issue #15621
+ Reported by: fnordian Tested by: atis ........
+
+ * configs/sip.conf.sample: fix documentation so it agrees with code
+
+ * channels/chan_sip.c, CHANGES: Add doc and turn off premature
+ media filter by default
+
+ * apps/app_playback.c, main/pbx.c, /, channels/chan_sip.c,
+ apps/app_disa.c, configs/sip.conf.sample: Merged revisions 216438
+ via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre,
+ 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27
+ lines 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 19:32 +0000 [r216595] Sean Bright <sean at malleable.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 216593 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk ........
+ r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep
+ 2009) | 1 line Use ast_free() instead of free(). ........
+
+2009-09-04 17:32 +0000 [r216548] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c, /, UPGRADE-1.6.txt: Merged revisions 216547 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04
+ Sep 2009) | 3 lines Enable turning off the application delimiter
+ warning with the 'dontwarn' option. Suggested on the -dev list,
+ and implemented in an alternate way by me. ........
+
+2009-09-04 15:07 +0000 [r216507] Michiel van Baak <michiel at vanbaak.info>
+
+ * /, main/utils.c: Merged revisions 216506 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009)
+ | 9 lines Merged revisions 216435 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009)
+ | 2 lines make asterisk compile under devmode with DEBUG_THREADS
+ enabled on OpenBSD ........ ................
+
+2009-09-04 10:49 +0000 [r216265] Russell Bryant <russell at digium.com>
+
+ * doc/IAX2-security.txt (added), /: Merged revisions 216264 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216264 | russell | 2009-09-04 05:48:44 -0500
+ (Fri, 04 Sep 2009) | 16 lines Merged revisions 216263 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216263 | russell | 2009-09-04 05:48:00 -0500
+ (Fri, 04 Sep 2009) | 9 lines 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. ........ ................ ................
+
+2009-09-04 06:11 +0000 [r216223] Michiel van Baak <michiel at vanbaak.info>
+
+ * main/astobj2.c, /: Merged revisions 216222 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r216222 |
+ mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines
+ make sure 'start' is always initialized. Makes asterisk compile
+ with --enable-dev-mode ........
+
+2009-09-03 19:42 +0000 [r216011-216097] Russell Bryant <russell at digium.com>
+
+ * UPGRADE.txt: tweak
+
+ * /, UPGRADE.txt: Merged revisions 216092 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009)
+ | 16 lines Merged revisions 216085 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216085 | russell | 2009-09-03 14:36:46 -0500
+ (Thu, 03 Sep 2009) | 9 lines 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 216009 via
+ svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r216009 | russell | 2009-09-03 13:45:54 -0500
+ (Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r216008 | russell | 2009-09-03 13:44:58 -0500
+ (Thu, 03 Sep 2009) | 9 lines 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 18:40 +0000 [r216003] 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: Merged revisions 215955 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 |
+ dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines
+ Merge code associated with AST-2009-006 (closes issue #12912)
+ Reported by: rathaus Tested by: tilghman, russell, dvossel,
+ dbrooks ........
+
+2009-09-03 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.0.15 released
+
+ * AST-2009-006
+
+2009-08-28 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.6.0.14 released
+
+2009-08-11 Tilghman Lesher <tlesher at digium.com>
+
+ * Released 1.6.0.14-rc1
+
+2009-08-10 19:51 +0000 [r211551-211587] Tilghman Lesher <tlesher at digium.com>
+
+ * doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500
+ (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10
+ Aug 2009) | 1 line Conversion specifiers, not format specifiers
+ ........ ................
+
+ * res/res_config_curl.c, apps/app_waitforring.c,
+ channels/chan_misdn.c, funcs/func_channel.c, apps/app_macro.c,
+ pbx/pbx_config.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
+ res/res_odbc.c, main/asterisk.c, apps/app_voicemail.c,
+ doc/CODING-GUIDELINES, utils/muted.c, apps/app_meetme.c,
+ main/utils.c, cdr/cdr_pgsql.c, res/res_musiconhold.c,
+ apps/app_followme.c, channels/misdn_config.c, utils/frame.c,
+ main/channel.c, main/cdr.c, res/ael/pval.c, funcs/func_enum.c,
+ channels/chan_phone.c, apps/app_osplookup.c,
+ apps/app_setcallerid.c, main/manager.c, funcs/func_odbc.c,
+ apps/app_minivm.c, res/res_agi.c, res/res_config_ldap.c,
+ apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
+ funcs/func_dialplan.c, main/dnsmgr.c, channels/chan_sip.c,
+ res/res_limit.c, apps/app_waitforsilence.c, agi/eagi-test.c,
+ apps/app_waituntil.c, main/acl.c, apps/app_queue.c,
+ channels/chan_oss.c, agi/eagi-sphinx-test.c,
+ channels/chan_usbradio.c, res/snmp/agent.c, pbx/pbx_dundi.c,
+ apps/app_sms.c, utils/extconf.c, apps/app_verbose.c,
+ apps/app_stack.c, apps/app_dahdibarge.c, funcs/func_rand.c,
+ apps/app_readfile.c, main/frame.c, /, apps/app_record.c,
+ funcs/func_strings.c, cdr/cdr_adaptive_odbc.c,
+ apps/app_alarmreceiver.c, channels/chan_iax2.c,
+ main/indications.c, main/config.c, main/cli.c,
+ pbx/pbx_loopback.c, channels/chan_dahdi.c, pbx/pbx_spool.c,
+ res/res_smdi.c, channels/chan_skinny.c, main/features.c,
+ main/http.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, funcs/func_cut.c,
+ channels/iax2-provision.c, pbx/dundi-parser.c,
+ apps/app_talkdetect.c: AST-2009-005
+
+2009-08-10 14:10 +0000 [r211348] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 |
+ file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix
+ retrieval of the port used for the video stream when adding SDP
+ to a SIP message. (closes issue #15121) Reported by: jsmith
+ ........
+
+2009-08-09 15:43 +0000 [r211233-211276] Tilghman Lesher <tlesher at digium.com>
+
+ * /, main/astfd.c: Merged revisions 211275 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009)
+ | 9 lines Merged revisions 211274 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009)
+ | 2 lines Small oops. Clear the flags which have been checked.
+ ........ ................
+
+ * apps/app_stack.c, /: Merged revisions 211232 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 |
+ tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines
+ Check for NULL frame, before dereferencing pointer. (closes issue
+ #15617) Reported by: rain ........
+
+2009-08-07 20:14 +0000 [r211114] Russell Bryant <russell at digium.com>
+
+ * apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009)
+ | 11 lines Recorded merge of revisions 211112 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009)
+ | 4 lines Resolve a deadlock involving app_chanspy and
+ masquerades. (ABE-1936) ........ ................
+
+2009-08-07 18:18 +0000 [r211044] Tilghman Lesher <tlesher at digium.com>
+
+ * /, apps/app_queue.c: Merged revisions 211040 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009)
+ | 21 lines Merged revisions 211038 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009)
+ | 14 lines 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) ........ ................
+
+2009-08-07 13:08 +0000 [r210993] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/udptl.c, /: Merged revisions 210992 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 |
+ kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13
+ lines Workaround broken T.38 endpoints that offer tiny
+ MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as
+ the maximum IFP size that should be sent to them, rather than the
+ maximum packet payload size. If such an endpoint also requests
+ UDPRedundancy as the error correction mode, we'll end up
+ calculating a tiny maximum IFP size, so small as to be unusable.
+ This patch sets a lower bound on what we'll consider the remote's
+ maximum IFP size to be, assuming that endpoints that do this
+ really can accept larger packets than they've offered to accept.
+ (closes issue #15649) Reported by: dazza76 ........
+
+2009-08-06 21:46 +0000 [r210909-210915] Tilghman Lesher <tlesher at digium.com>
+
+ * main/channel.c, /: Merged revisions 210914 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ................
+ r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009)
+ | 14 lines Merged revisions 210913 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009)
+ | 7 lines 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 ........ ................
+
+ * apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged
+ revisions 210908 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 |
+ tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines
+ Allow Gosub to recognize quote delimiters without consuming them.
+ (closes issue #15557) Reported by: rain Patches:
+ 20090723__issue15557.diff.txt uploaded by tilghman (license 14)
+ Tested by: rain Review: https://reviewboard.asterisk.org/r/316/
+ ........
+
+2009-08-06 17:47 +0000 [r210818] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 |
+ file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines
+ Accept additional T.38 reinvites after an initial one has been
+ handled. Discussion of this subject has yielded that it is not
+ actually acceptable to change T.38 parameters after the initial
+ reinvite but declining is harsh and can cause the fax to fail
+ when it may be possible to allow it to continue. This patch
+ changes things so that additional T.38 reinvites are accepted but
+ parameter changes ignored. This gives the fax a fighting chance.
+ (closes issue #15610) Reported by: huangtx2009 ........
+
+2009-08-05 20:07 +0000 [r210647] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/trunk
+ ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500
+ (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009)
+ | 14 lines 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 18:57 +0000 [r210568] Leif Madsen <lmadsen at digium.com>
+
+ * doc/tex/imapstorage.tex, /: Merged revisions 210564 via svnmerge
[... 52659 lines stripped ...]
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