[asterisk-commits] lmadsen: tag 1.8.0-rc4 r292088 - /tags/1.8.0-rc4/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Oct 18 11:06:27 CDT 2010
Author: lmadsen
Date: Mon Oct 18 11:06:24 2010
New Revision: 292088
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=292088
Log:
Importing files for 1.8.0-rc4 release.
Added:
tags/1.8.0-rc4/.lastclean (with props)
tags/1.8.0-rc4/.version (with props)
tags/1.8.0-rc4/ChangeLog (with props)
Added: tags/1.8.0-rc4/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.0-rc4/.lastclean?view=auto&rev=292088
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--- tags/1.8.0-rc4/ChangeLog (added)
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@@ -1,0 +1,25434 @@
+2010-10-18 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.0-rc4 Released
+
+2010-10-18 16:02 +0000 [r292085] David Vossel <dvossel at digium.com>
+
+ * main/netsock2.c: Fixes qos settings for sockets bound to any IPv6
+ or IPv4 address. (closes issue #18099) Reported by: jamesnet
+ Patches: issues_18099_v3.diff uploaded by dvossel (license 671
+
+2010-10-18 15:32 +0000 [r292083] Jeff Peeler <jpeeler at digium.com>
+
+ * pbx/pbx_spool.c: Disable use of inotify for call file handling as
+ it is not working properly. (related to #18089)
+
+2010-10-16 10:47 +0000 [r292050] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * res/res_musiconhold.c, /, configs/musiconhold.conf.sample: Merged
+ revisions 292049 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 ×וק 2010) |
+ 15 lines Base directory for MOH should be ASTDATADIR If the
+ directive 'directory' is relative, make it relative to the
+ datadir, rather than to the varlibdir. In the sample
+ configuration it is relative ('moh'). This has no effect unless
+ you have actively set the datadir explicitly (at build time or at
+ run time). (closes issue #16906) Patches: moh_datadir uploaded by
+ tzafrir (license 46) Review:
+ https://reviewboard.asterisk.org/r/974/ ........
+
+2010-10-15 21:40 +0000 [r292016] Terry Wilson <twilson at digium.com>
+
+ * res/res_srtp.c: Ref/unref res_srtp when we create/destroy a
+ session This avoids unhappy crashing when we try to 'core stop
+ gracefully' and res_srtp tries to unload before chan_sip does.
+ Thanks, Russell! (closes issue #18085) Reported by: st
+
+2010-10-15 20:12 +0000 [r291942] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Fixes peer's host port information being
+ lost on sip reload. (closes issue #18135) Reported by: lmadsen
+ Patches: crazy_ports_v2.diff uploaded by dvossel (license 671)
+ Tested by: lmadsen
+
+2010-10-15 19:50 +0000 [r291940] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * configs/gtalk.conf.sample, /: Merged revisions 291939 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291939 | pabelanger | 2010-10-15 15:35:20 -0400
+ (Fri, 15 Oct 2010) | 9 lines Merged revisions 291938 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri,
+ 15 Oct 2010) | 2 lines Clean up formatting. ........
+ ................
+
+2010-10-15 16:39 +0000 [r291905] Terry Wilson <twilson at digium.com>
+
+ * res/res_jabber.c, /: Merged revisions 291904 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010)
+ | 7 lines Don't crash or deadlock on module unload We can't hold
+ the lock while pthread_join is called since aji_log_hook will
+ attempt to lock from the other therad. We reorder the
+ pthread_join and ast_aji_disconnect so that we don't do an
+ SSL_read() while SSL_shutdown is running, causing a crash.
+ ........
+
+2010-10-14 22:09 +0000 [r291827-291829] David Vossel <dvossel at digium.com>
+
+ * main/netsock2.c: Set TCLASS field of IPv6 header when sip qos
+ options are set. (closes issue #18099) Reported by: jamesnet
+ Patches: issues_18099_v2.diff uploaded by dvossel (license 671)
+ Tested by: dvossel, jamesnet
+
+ * channels/chan_gtalk.c: Safer xml parsing, treat all clients the
+ same, and better local candidate selection. The gtalk channel
+ driver was doing several unsafe operations in regards to how it
+ parsed incoming XML messages. I have cleaned that code up so it
+ should be much safer now. We now treat all clients types the
+ same. We have no reason to distinguish between GMAIL and GOOGLE
+ VOICE clients anymore because they all work the same way. I also
+ modified how the local ip is found. If no bindaddress is provided
+ in the config file, we attempt to determine the local ip we would
+ use to connect to google.com. If that fails, then we fall back to
+ the ast_find_ourip() function as a last resort. Using the new
+ method makes it much less likely that we would ever advertise a
+ local RTP candidate as a loopback address.
+
+2010-10-14 18:45 +0000 [r291791] Jeff Peeler <jpeeler at digium.com>
+
+ * main/stdtime/localtime.c: Add missing ifdefs for test framework
+ and new locale code. (closes issue #18137) Reported by: ovi
+ Patches: 18137_test_framework_ifdef.patch uploaded by wdoekes
+ (license 717) 18137_localelist_warning.patch uploaded by wdoekes
+ (license 717) Tested by: ovi
+
+2010-10-14 15:15 +0000 [r291758] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * channels/chan_gtalk.c, channels/chan_jingle.c,
+ include/asterisk/acl.h, channels/chan_sip.c,
+ channels/chan_h323.c, main/acl.c: Add the ability for
+ ast_find_ourip to return IPv4, IPv6 or both. While testing
+ chan_gtalk I noticed jabber was using my IPv6 address and not
+ IPv4. When using bindaddr=0.0.0.0 it is possible for
+ ast_find_ourip() to return both IPv6 and IPv4 results. Adding a
+ family parameter gives you the ablility to choose. Since
+ jabber/gtalk/h323 do not support IPv6, we should only return IPv4
+ results. Review: https://reviewboard.asterisk.org/r/973/
+
+2010-10-14 12:08 +0000 [r291725] Russell Bryant <russell at digium.com>
+
+ * doc/tex/secure-calls.tex: Fix a typo - s/seucre/secure/
+
+2010-10-13 23:45 +0000 [r291656] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_analog.c, /,
+ channels/sig_analog.h: Merged revisions 291655 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291655 | rmudgett | 2010-10-13 18:36:50 -0500
+ (Wed, 13 Oct 2010) | 27 lines Merged revisions 291643 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010)
+ | 20 lines Deadlock between dahdi_exception() and
+ dahdi_indicate(). There is a deadlock between dahdi_exception()
+ and dahdi_indicate() for analog ports. The call-waiting and
+ three-way-calling feature can experience deadlock if these
+ features are trying to do something and an event from the bridged
+ channel happens at the same time. Deadlock avoidance code added
+ to obtain necessary channel locks before attemting an operation
+ with call-waiting and three-way-calling. (closes issue #16847)
+ Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch
+ uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch
+ uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett
+ Review: https://reviewboard.asterisk.org/r/971/ ........
+ ................
+
+2010-10-13 23:01 +0000 [r291581] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c, /: Merged revisions 291580 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291580 | twilson | 2010-10-13 15:58:43 -0700
+ (Wed, 13 Oct 2010) | 28 lines Merged revisions 291577 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010)
+ | 21 lines Don't ignore frames that have been queued when
+ softhangup'd When an outgoing call is answered and hung up by the
+ far end *very* quickly, we may not read any frames and therefor
+ end up with a call that displays the wrong
+ disposition/DIALSTATUS. The reason is because ast_queue_hangup()
+ immediately sets the _softhangup flag on the channel and then
+ queues the HANGUP control frame, but __ast_read refuses to read
+ any frames if ast_check_hangup() indicates that a hangup request
+ has been made (which it will if _softhangup is set). So, we end
+ up losing control frames. This change makes __ast_read continue
+ to read frames even if a soft hangup has been requested. It
+ queues a hangup frame to make sure that __ast_read() will still
+ eventually return NULL. Much thanks to David Vossel for all of
+ the reviews, discussion, and help! (closes issue #16946) Reported
+ by: davidw Review: https://reviewboard.asterisk.org/r/740/
+ ........ ................
+
+2010-10-13 22:46 +0000 [r291578] David Vossel <dvossel at digium.com>
+
+ * channels/chan_gtalk.c: More fixup for chan_gtalk. This patch
+ makes the xml parsing safer.
+
+2010-10-13 22:24 +0000 [r291575] Terry Wilson <twilson at digium.com>
+
+ * Makefile, static-http/mantest.html (added): Add a simple AMI
+ client web page This patch uses the XML docs to parse all of the
+ available AMI commands and allows you to enter the command name
+ and be presented with a form with the available fields. You can
+ then rapidly tab through the fields and submit the command and
+ view the response. It is much faster/easier than having to use
+ telnet for testing purposes.
+
+2010-10-13 20:21 +0000 [r291469-291541] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: The chan_dahdi faxdetect option only works
+ for the first FAX call. The chan_dahdi faxdetect option only
+ works for the first call. After that the option no longer works.
+ The struct dahdi_pvt.callprogress member is the encoded user
+ config setting for the callprogress and faxdetect config options.
+ Changing this value alters the configuration for all following
+ calls until the chan_dahdi.conf file is reloaded. * Fixed the
+ chan_dahdi ast_channel_setoption callback to not change the users
+ faxdetect config setting except for the current call. * Fixed the
+ chan_dahdi ast_channel_queryoption callback to read the active
+ DSP setting of the faxdetect option. * Made actually disable the
+ active faxdetect DSP setting for the current call on the analog
+ port. my_handle_dtmfup() is used for normal analog ports.
+ dahdi_handle_dtmfup() is the legacy code and is no longer used
+ unless in a radio mode. (closes issue #18116) Reported by:
+ seandarcy Patches: issue18116_v1.8.patch uploaded by rmudgett
+ (license 664) Review: https://reviewboard.asterisk.org/r/972/
+
+ * channels/chan_misdn.c: Merged revision 291504 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed,
+ 13 Oct 2010) | 11 lines Hold off ast_hangup() from destroying the
+ ast_channel. Must get the ast_channel lock before proceeding with
+ release_chan() and release_chan_early() to hold off ast_hangup()
+ from destroying the ast_channel. Missed this change for -r291468.
+ JIRA ABE-2598 JIRA SWP-2317 ..........
+
+ * channels/chan_misdn.c: Merge revision 291468 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed,
+ 13 Oct 2010) | 16 lines Memory overwrites when releasing mISDN
+ call. Phone <--> Asterisk <-- ALERTING --> DISCONNECT <-- RELEASE
+ --> RELEASE_COMPLETE * Add lock protection around channel list
+ for find/add/delete operations. * Protect misdn_hangup() from
+ release_chan() and vise versa using the release_lock. JIRA
+ ABE-2598 JIRA SWP-2317 ..........
+
+2010-10-13 15:46 +0000 [r291394] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 291393 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291393 | russell | 2010-10-13 10:29:21 -0500
+ (Wed, 13 Oct 2010) | 13 lines Merged revisions 291392 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010)
+ | 6 lines Lock pvt so pvt->owner can't disappear when queueing up
+ a frame. This fixes a crash due to a hangup race condition.
+ ABE-2601 ........ ................
+
+2010-10-12 17:20 +0000 [r291284] Leif Madsen <lmadsen at digium.com>
+
+ * configs/phoneprov.conf.sample, /: Merged revisions 291280 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010)
+ | 7 lines Add undocumented variables to phoneprov.conf.sample
+ (closes issue #18107) Reported by: lathama Patches:
+ phoneprov.conf.sample.diff uploaded by lathama (license 1028)
+ ........
+
+2010-10-12 17:06 +0000 [r291265] Tilghman Lesher <tlesher at digium.com>
+
+ * /, main/acl.c: Merged revisions 291264 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291264 | tilghman | 2010-10-12 12:05:31 -0500
+ (Tue, 12 Oct 2010) | 9 lines Merged revisions 291263 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12
+ Oct 2010) | 2 lines Oops, incorrect range (although unallocated
+ at ARIN) ........ ................
+
+2010-10-12 16:08 +0000 [r291230] Leif Madsen <lmadsen at digium.com>
+
+ * configs/manager.conf.sample, /: Merged revisions 291229 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291229 | lmadsen | 2010-10-12 11:07:28 -0500 (Tue, 12 Oct 2010)
+ | 2 lines Add documention that mentions options are defined but
+ not used. (Issue #18101) ........
+
+2010-10-12 15:58 +0000 [r291192-291227] David Vossel <dvossel at digium.com>
+
+ * main/manager.c: Fixes manager.c crash. This issue was caused by
+ improper use of the mansession lock and manession_session lock.
+ These two structures are confusing to begin with so I'm not
+ surprised this occurred. I fixed this by consistently making sure
+ we use each of these locks only to protect the data in the
+ corresponding structure. We had mismatched usage of these locks
+ which resulted in no mutual exclusivity occurring at all. (closes
+ issue #17994) Reported by: vrban Patches:
+ mansession_locking_fix.diff uploaded by dvossel (license 671)
+ Tested by: vrban
+
+ * CHANGES: Update CHANGES to reflect new gtalk.conf options.
+
+ * channels/chan_gtalk.c, include/asterisk/stun.h,
+ configs/gtalk.conf.sample, res/res_stun_monitor.c: Gtalk
+ enhancements and general code cleanup. This patch includes
+ several chan_gtalk enhancements. Two new gtalk.conf options have
+ been added, externip and stunadd. Setting externip allows us to
+ manually specify what the external IP address is outside of a NAT
+ environment. Setting the stunaddr option to a valid stun server
+ allows for that external ip to be retrieved via a STUN server
+ automatically. This external IP is then advertised during call
+ setup as a possible candidate. I have also attempted to clean up
+ chan_gtalk's code so it meets our coding guidelines. During this
+ cleanup I noticed several things that need to be done in the code
+ and made a TODO section at the top of the file.
+
+2010-10-11 18:51 +0000 [r291075-291113] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: Move declaration closer to where now used.
+
+ * /, channels/chan_sip.c: Merged revisions 291110-291111 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500
+ (Mon, 11 Oct 2010) | 9 lines Merged revisions 291109 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11
+ Oct 2010) | 1 line Add missing unlock to an exception condition
+ in reload_config(). ........ ................ r291111 | rmudgett
+ | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit
+ from handle_request_do() consistent. ................
+
+ * main/cli.c, /: Merged revisions 291073 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010)
+ | 15 lines Fixed infinite loop in verbose/debug message output.
+ Setting the module/filename specific message level and then
+ changing it resulted in the linked list being looped on itself.
+ Traversing this linked list is an infinite loop if what you are
+ looking for is not in the list. Also plugged some CLI parsing
+ holes in the associated CLI command: * Removing a nonexistent
+ module from the list actually added it with a level of zero. *
+ Setting the non-module specific level to zero is now equivalent
+ to setting it to "off" as documented. ........
+
+2010-10-09 23:25 +0000 [r291038] Tilghman Lesher <tlesher at digium.com>
+
+ * cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Add missing
+ option to set calls to be logged in GMT/UTC.
+
+2010-10-09 15:00 +0000 [r291005-291037] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.c: small correction for verbose
+ print h.323 packets
+
+ * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c,
+ addons/ooh323c/src/ooh245.c: Added fast start and h.245 tunneling
+ options per user and peer. Added options for faststart/h.245
+ tunneling per user/peer, properly handle these and global
+ options, correction of handling fs/tunneling fields in signalling
+ responses (issue #17972) Reported by: salecha Patches:
+ fs-tunnel-per-point-3.patch uploaded by may213 (license 454)
+ Tested by: may213, salecha
+
+2010-10-08 20:44 +0000 [r290973] David Vossel <dvossel at digium.com>
+
+ * channels/chan_gtalk.c: Make outbound Google Voice calls. This
+ patch allows for outbound Google Voice calls to be dialed from
+ Asterisk using chan_gtalk. Below is an example dialstring. exten
+ -> blah,1,Dial(Gtalk/asterisk/+15552225555 at voice.google.com,,) In
+ this example, 'asterisk' is the jabber.conf profile configured to
+ connect to your gmail account. In order to receive Google Voice
+ calls make sure to enable 'allowguest=yes' in gtalk.conf.
+
+2010-10-08 15:49 +0000 [r290937-290938] Erin Spiceland <erin at thespicelands.com>
+
+ * addons/res_config_mysql.c: Parentheses around assignment used as
+ truth value, introduced in r290937.
+
+ * addons/res_config_mysql.c, addons/app_mysql.c,
+ configs/res_config_mysql.conf.sample: Add option to
+ res_config_mysql and app_mysql to specify a character set that
+ MySQL should use. (closes issue 17948) Reported by qmax.
+
+2010-10-08 02:56 +0000 [r290864] Jeff Peeler <jpeeler at digium.com>
+
+ * main/asterisk.c, /: Merged revisions 290863 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290863 | jpeeler | 2010-10-07 21:45:44 -0500
+ (Thu, 07 Oct 2010) | 16 lines Merged revisions 290862 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010)
+ | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed
+ at control console. A recent change was made to avoid a race
+ condition on shutdown which only called the end functions from
+ the console thread. However, when pressing Ctrl-C the quit
+ handler is called from the signal handler thread. (closes issue
+ #17698) Reported by: jmls ........ ................
+
+2010-10-07 22:38 +0000 [r290828-290829] David Vossel <dvossel at digium.com>
+
+ * channels/chan_gtalk.c: Add Philippe Sultan to chan_gtalk author
+ list. Philippe has made some notable contributions to the gtalk
+ channel driver. His name deserves to be listed amoung the authors
+ of that file. Thanks Philippe!
+
+ * channels/chan_gtalk.c: Outbound gtalk calls now work correctly.
+ There was a problem with how the candidates were being built on
+ an outbound call. This patch fixes that.
+
+2010-10-07 20:58 +0000 [r290752] Jason Parker <jparker at digium.com>
+
+ * autoconf/ast_ext_lib.m4, /, configure,
+ include/asterisk/autoconfig.h.in: Merged revisions 290751 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290751 | qwell | 2010-10-07 15:57:14 -0500
+ (Thu, 07 Oct 2010) | 16 lines Merged revisions 290750 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) |
+ 9 lines Allow PRI to build properly when using --with-pri. Use
+ the directories found for the parent when using lib dependencies.
+ (closes issue #17314) Reported by: tzafrir Patches:
+ 17314-withdeps.diff uploaded by qwell (license 4) ........
+ ................
+
+2010-10-07 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.0-rc3 Released.
+
+2010-10-07 11:00 +0000 [r290713] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c, /: Merged revisions 290712 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290712 | russell | 2010-10-07 12:53:56 +0200 (Thu, 07 Oct 2010)
+ | 4 lines Don't crash when Set() is called without a value.
+ Review: https://reviewboard.asterisk.org/r/949/ ........
+
+2010-10-06 21:22 +0000 [r290648-290674] David Vossel <dvossel at digium.com>
+
+ * channels/chan_gtalk.c: Fixes commented out code to use #if 0
+ instead. Thanks to rmudgett for catching this!
+
+ * channels/chan_gtalk.c: Fixes gtalk outbound DTMF to work
+ properly. Outbound DTMF with gtalk needs to be done within the
+ RTP stream. I discovered this after investigating a packet
+ capture from the gmail client. Instead of performing jingle
+ signaling DTMF, the gtalk servers expect all DTMF to arrive on
+ the RTP stream using RFC2833 way of doing things. Chan_gtalk also
+ had an issue with negotiating RTP payload type 106 for the
+ telephony-event and then sending DTMF as payload 101. This has
+ been resolved by always negotiating 101 as the payload type like
+ we do everywhere else. With this patch, incoming google voice
+ calls forwarded to Asterisk via gtalk work.
+
+2010-10-06 18:50 +0000 [r290614] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_dial.c: Merged revision 290613 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed,
+ 06 Oct 2010) | 5 lines Eliminate a redundant test for
+ AST_CONTROL_REDIRECTING. Eliminate redundant test for
+ AST_CONTROL_REDIRECTING that prevents running the redirecting
+ interception macro if it is defined. ..........
+
+2010-10-06 13:49 +0000 [r290576] Tilghman Lesher <tlesher at digium.com>
+
+ * /, main/file.c: Merged revisions 290575 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010)
+ | 8 lines Allow streaming audio from a pipe. (closes issue
+ #18001) Reported by: jamicque Patches:
+ 20100926__issue18001.diff.txt uploaded by tilghman (license 14)
+ Tested by: jamicque ........
+
+2010-10-06 04:35 +0000 [r290542] Terry Wilson <twilson at digium.com>
+
+ * res/res_rtp_asterisk.c: Don't try to send RTP when remote_address
+ is null It is possible for ast_rtp_stop() to be called which will
+ clear the remote address and cause the sendto to fail and spam
+ warnings. Don't send in this case.
+
+2010-10-05 22:23 +0000 [r290479-290506] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Fixes uninitialized memory problem in 'iax2
+ set debug peer' option.
+
+ * include/asterisk/jingle.h, channels/chan_gtalk.c,
+ res/res_jabber.c, include/asterisk/jabber.h: Fixes chan_gtalk to
+ work with gmail client This patch was written by Philippe Sultan
+ (phsultan). Thanks for keeping this up to date!
+
+2010-10-05 20:23 +0000 [r290408] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_jabber.c, /: Merged revisions 290396 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290396 | tilghman | 2010-10-05 15:21:02 -0500
+ (Tue, 05 Oct 2010) | 15 lines Merged revisions 290392 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010)
+ | 8 lines Fix a crash by ensuring that we don't alter memory
+ after it's freed. (closes issue #17387) Reported by: jmls
+ Patches: 20100726__issue17387.diff.txt uploaded by tilghman
+ (license 14) Tested by: jmls ........ ................
+
+2010-10-05 20:09 +0000 [r290376-290378] David Vossel <dvossel at digium.com>
+
+ * channels/chan_iax2.c: Resolves dnsmgr memory corruption in
+ chan_iax2. (closes issue #17902) Reported by: afried Patches:
+ issue_17902.rev1.txt uploaded by russell (license 2) Tested by:
+ afried, russell, dvossel Review:
+ https://reviewboard.asterisk.org/r/965/
+
+ * /, apps/app_directed_pickup.c: Merged revisions 290375 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010)
+ | 10 lines Fixes PickupChan() not working with full channel name.
+ (closes issue #18011) Reported by: schern Patches:
+ app_directed_pickup.c.2.patch uploaded by schern (license 995)
+ app_directed_pickup.c.trunk.patch uploaded by schern (license
+ 995) Tested by: schern, dvossel ........
+
+2010-10-05 14:15 +0000 [r290066-290289] Tilghman Lesher <tlesher at digium.com>
+
+ * configure, configure.ac: Restore run directory for OS X, as well
+ as standardizing some other paths to Mac OS X.
+
+ * pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5,
+ pbx/ael/ael-test/ref.ael-test19,
+ pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, main/pbx.c,
+ pbx/ael/ael-test/ref.ael-vtest17, /,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+ pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3:
+ Merged revisions 290254 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010)
+ | 11 lines Change new pattern matcher to regard dashes the same
+ as the old pattern matcher -- as visual candy to be ignored. Also
+ change the AEL parser to not generate dashes within extensions,
+ as those dashes would be ignored. Update the AEL tests to match
+ this behavior. (closes issue #17366) Reported by: murf Patches:
+ 20100727__issue17366.diff.txt uploaded by tilghman (license 14)
+ Tested by: tilghman ........
+
+ * /, configure, configure.ac: Merged revisions 290201 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290201 | tilghman | 2010-10-04 15:22:03 -0500
+ (Mon, 04 Oct 2010) | 9 lines Merged revisions 290177 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04
+ Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........
+ ................
+
+ * /, configure, configure.ac: Merged revisions 290101 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r290101 | tilghman | 2010-10-03 16:06:58 -0500
+ (Sun, 03 Oct 2010) | 9 lines Merged revisions 290100 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03
+ Oct 2010) | 2 lines Automatically re-run configure test for
+ menuselect, when the relevant makeopts settings change. ........
+ ................
+
+ * pbx/pbx_spool.c: Get notification only when file is closed, not
+ when created. (closes issue #17924) Reported by: mkeuter Patches:
+ asterisk-1.8-bugid17924.patch uploaded by abelbeck (license 946)
+ Tested by: abelbeck
+
+2010-10-02 17:57 +0000 [r290026] Kevin P. Fleming <kpfleming at digium.com>
+
+ * contrib/scripts/get_mp3_source.sh: Allow users to pass additional
+ arguments to the Subversion command that obtains the MP-3 source
+ code. (reported on IRC by jmls)
+
+2010-10-02 08:56 +0000 [r289951] Olle Johansson <oej at edvina.net>
+
+ * main/manager.c, /: Merged revisions 289950 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289950 | oej | 2010-10-02 10:52:03 +0200 (Lör,
+ 02 Okt 2010) | 9 lines Merged revisions 289949 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2
+ lines Add documentation for undocumented option to AMI action
+ originate ........ ................
+
+2010-10-02 04:46 +0000 [r289875] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, /: Merged revisions 289874 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289874 | tilghman | 2010-10-01 23:45:49 -0500
+ (Fri, 01 Oct 2010) | 15 lines Merged revisions 289873 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010)
+ | 8 lines When forwarding a message, a prepend means that the
+ filesystem will always have a better copy. (closes issue #17803)
+ Reported by: dpetersen Patches: 20100923__issue17803.diff.txt
+ uploaded by tilghman (license 14) Tested by: dpetersen ........
+ ................
+
+2010-10-02 02:43 +0000 [r289840] Jeff Peeler <jpeeler at digium.com>
+
+ * include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c,
+ main/rtp_engine.c, /, channels/chan_sip.c: Merged revisions
+ 289798 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500
+ (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010)
+ | 15 lines Change RFC2833 DTMF event duration on end to report
+ actual elapsed time. The scenario here is with a non P2P early
+ media session. The reported time length of DTMF presses are
+ coming up short when sending to the remote side. Currently the
+ event duration is a running total that is incremented when
+ sending continuation packets. These continuation packets are only
+ triggered upon incoming media from the remote side, which means
+ that the running total probably is not going to end up matching
+ the actual length of time Asterisk received DTMF. This patch
+ changes the end event duration to be lengthened if it is detected
+ that the end event is going to come up short. Review:
+ https://reviewboard.asterisk.org/r/957/ ABE-2476 ........
+ ................
+
+2010-10-01 17:19 +0000 [r289718] Paul Belanger <paul.belanger at polybeacon.com>
+
+ * res/res_jabber.c, /, configs/jabber.conf.sample: Merged revisions
+ 289704 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289704 | pabelanger | 2010-10-01 13:09:03 -0400
+ (Fri, 01 Oct 2010) | 13 lines Merged revisions 289703 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct
+ 2010) | 6 lines Disable debugging by default and reformat .config
+ file. Review: https://reviewboard.asterisk.org/r/929/ ........
+ ................
+
+2010-10-01 16:22 +0000 [r289701] Jeff Peeler <jpeeler at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 289700 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500
+ (Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010)
+ | 14 lines Ensure user portion of SIP URI matches dialplan when
+ using encoded characters. This commit takes a simliar approach to
+ 288112 and checks the dialplan to determine the proper action for
+ an incoming contact header as to whether or not it should be
+ decoded or not. sip_new was blindly always decoding the
+ extension, which also caused the outgoing contact header to be
+ incorrect as well as failing to match the encoded extension in
+ the dialplan. (closes issue #17892) Reported by: wdoekes Patches:
+ bug17892-1.patch uploaded by jpeeler (license 325) Tested by:
+ wdoekes ........ ................
+
+2010-10-01 09:42 +0000 [r289622] Stefan Schmidt <sst at sil.at>
+
+ * channels/chan_sip.c: don't iterate through all dialogs to find
+ and delete old subscribes On every incoming subscribe there is a
+ iteration through all dialogs to find old subscribes and delete
+ them. This is slow and not RFC conform. This was only needed in
+ 1.2 cause a subscribe was not deleted when a dialog was
+ destroyed, after 1.4 a subscribe get removed when its dialog is
+ destroyed. (closes issue #17950) Reported by: schmidts Tested by:
+ schmidts Review: https://reviewboard.asterisk.org/r/901/
+
+2010-09-30 20:23 +0000 [r289581] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_env.c: Solaris fixes.
+
+2010-09-30 19:53 +0000 [r289554] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 289553 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep
+ 2010) | 4 lines Properly handle channel allocation failures duing
+ invites with replaces. ABE-2588 ........
+
+2010-09-30 19:28 +0000 [r289549] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c: Merged revision 289547 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ .......... r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu,
+ 30 Sep 2010) | 10 lines In chan_misdn, the
+ DivertingLegInformation2 DivertingNr is garbage when the number
+ is restricted. The same thing happens with
+ DivertingLegInformation1 DivertedTo number. The
+ misdn_PresentedNumberUnscreened_extract() extracted the
+ Unscreened PartyNumber field unconditionally. It now checks the
+ presented number unscreened type to see if the PartyNumber was
+ even present. JIRA ABE-2595 ..........
+
+2010-09-30 17:50 +0000 [r289543] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/localtime.h, main/stdtime/localtime.c,
+ tests/test_time.c, tests/test_utils.c, res/res_agi.c: More
+ Solaris compatibility fixes
+
+2010-09-30 15:39 +0000 [r289426] Russell Bryant <russell at digium.com>
+
+ * apps/app_sms.c, /: Merged revisions 289425 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289425 | russell | 2010-09-30 10:37:29 -0500
+ (Thu, 30 Sep 2010) | 15 lines Merged revisions 289424 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010)
+ | 8 lines Fix a crash in app_sms. Since the data being passed to
+ the generator callback is on the stack of the SMS() application,
+ we must ensure that the generator is stopped before the
+ application exits. ABE-2587 ........ ................
+
+2010-09-29 21:12 +0000 [r289340] Jason Parker <jparker at digium.com>
+
+ * main/channel.c, /, main/features.c: Merged revisions 289339 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r289339 | qwell | 2010-09-29 16:03:47 -0500
+ (Wed, 29 Sep 2010) | 15 lines Merged revisions 289338 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) |
+ 8 lines Allow a manager originate to succeed on forwarded
+ devices. The timeout to wait for an answer was being set to 0
+ when a device forwarded to another extension. We don't always
+ need the timeout set like this, so make it an optional parameter,
+ and don't use it in this case. ABE-2544 ........ ................
+
+2010-09-29 20:27 +0000 [r289336] Leif Madsen <lmadsen at digium.com>
+
+ * configs/res_ldap.conf.sample, /: Merged revisions 289334 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r289334 | lmadsen | 2010-09-29 15:24:47 -0500 (Wed, 29 Sep 2010)
+ | 1 line Update sample documentation to note md5secret
+ requirements. ........
+
+2010-09-29 20:20 +0000 [r289333] Russell Bryant <russell at digium.com>
+
+ * res/res_config_ldap.c, /: Merged revisions 289332 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r289332 | russell | 2010-09-29 15:15:57 -0500 (Wed, 29
+ Sep 2010) | 4 lines Don't completely ignore md5secret from LDAP
+ if the value does not begin with {md5}. This fixes a problem that
+ lmadsen ran in to where md5secret was not working for him.
+ ........
+
+2010-09-29 17:53 +0000 [r289268-289300] Matthew Nicholson <mnicholson at digium.com>
+
+ * configs/res_fax.conf.sample: Add 'ecm' to the sample fax config
+ file
+
+ * main/channel.c: Update the CDR record when
+ ast_channel_set_caller_event() is called (related to issue
+ #17569) Reported by: tbelder
+
+2010-09-29 16:16 +0000 [r289253] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c: Make development error message indicate which
+ channel.
+
+2010-09-29 15:04 +0000 [r289179] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c, /: Merged revisions 289178 via svnmerge from
[... 24699 lines stripped ...]
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