2010-10-18 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-rc5 Released.
2010-10-18 22:02 +0000 [r292230] Leif Madsen <lmadsen@digium.com>
* sounds/Makefile, /: Merged revisions 292229 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
r292229 | lmadsen | 2010-10-18 17:01:16 -0500 (Mon, 18 Oct 2010)
| 3 lines Fix typo in the sounds/Makefile. (Issue #17426)
........
2010-10-18 21:55 +0000 [r292227] Jeff Peeler <jpeeler@digium.com>
* apps/app_voicemail.c, /: Merged revisions 292226 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r292226 | jpeeler | 2010-10-18 16:54:38 -0500
(Mon, 18 Oct 2010) | 18 lines Merged revisions 292223 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010)
| 11 lines Fix improper operator key acceptance and clean up temp
recording files. This is a fix for when pressing the operator key
after recording an unavailable, busy, name, or temporary message
in mailbox options. The operator key should not be accepted here,
but should be allowed during the message recording. If the
operator key is pressed during ensure the file is saved or
deleted as apporopriate. Also, ensure removal of temporary
recorded files after an early hang up or when message acceptance
confirmation times out. ABE-2518 ........ ................
2010-10-18 21:51 +0000 [r292225] Leif Madsen <lmadsen@digium.com>
* sounds/sounds.xml, sounds/Makefile, /: Merged revisions 292224
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r292224 | lmadsen | 2010-10-18 16:50:47 -0500
(Mon, 18 Oct 2010) | 17 lines Merged revisions 292222 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010)
| 9 lines Add support for the new English (Australian Accent)
sound files. (closes issue #17426) Reported by: camsown Patches:
core-sounds-en_AU.txt uploaded by camsown (license 1050)
add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested
by: camsown, lmadsen, jtodd, qwell ........ ................
2010-10-18 19:50 +0000 [r292188] Russell Bryant <russell@digium.com>
* main/netsock2.c: Resolve some compiler errors in
ast_sockaddr_is_any(). These errors came up once this function
was used from within netsock2.c. The errors were like the
following: netsock2.c:393: error: dereferencing pointer
‘({anonymous})’ does break strict-aliasing rules The usage of a
union here avoids this problem.
2010-10-18 19:16 +0000 [r292155] David Vossel <dvossel@digium.com>
* main/netsock2.c: Fixes build error for systems not supporting
IPV6_TCLASS.
2010-10-18 17:15 +0000 [r292122] Matthew Nicholson <mnicholson@digium.com>
* addons/chan_mobile.c: Fix the cmgr parser. (closes issue 0018152)
Reported by: menschentier
2010-10-18 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-rc4 Released
2010-10-18 16:02 +0000 [r292085] David Vossel <dvossel@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@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@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 אוק 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@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@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@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@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@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@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@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@digium.com>
* doc/tex/secure-calls.tex: Fix a typo - s/seucre/secure/
2010-10-13 23:45 +0000 [r291656] Richard Mudgett <rmudgett@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@digium.com>
* Asterisk 1.8.0-rc3 Released.
2010-10-07 11:00 +0000 [r290713] Russell Bryant <russell@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@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@digium.com>
* funcs/func_env.c: Solaris fixes.
2010-09-30 19:53 +0000 [r289554] Matthew Nicholson <mnicholson@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@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@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@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@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@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@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@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@digium.com>
* main/channel.c: Make development error message indicate which
channel.
2010-09-29 15:04 +0000 [r289179] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c, /: Merged revisions 289178 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r289178 | mnicholson | 2010-09-29 10:04:11 -0500
(Wed, 29 Sep 2010) | 15 lines Merged revisions 289177 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep
2010) | 8 lines Set the caller id on CDRs when it is set on the
parent channel. (closes issue #17569) Reported by: tbelder
Patches: 17569.diff uploaded by tbelder (license 618) Tested by:
tbelder ........ ................
2010-09-28 18:18 +0000 [r289104] Tilghman Lesher <tlesher@digium.com>
* makeopts.in, apps/app_voicemail.c, Makefile, tests/test_time.c,
configure, include/asterisk/autoconfig.h.in,
include/asterisk/compat.h, main/strcompat.c, tests/test_utils.c,
configure.ac: Solaris compatibility fixes Review:
https://reviewboard.asterisk.org/r/942/
2010-09-28 18:18 +0000 [r289099] Brett Bryant <bbryant@digium.com>
* main/channel.c, /: Merged revisions 289095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r289095 | bbryant | 2010-09-28 14:14:19 -0400
(Tue, 28 Sep 2010) | 21 lines Merged revisions 289094 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010)
| 14 lines Fixes an issue with the Newchannel AMI event during
the Masquerading process. Fixes an issue with the Newchannel AMI
event during the Masquerading process, where no Newchannel AMI
event was generated for the psuedo channel used during the
masquerading process. (closes issue #17987) Reported by:
RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish
(license 1122) Tested by: RadicAlish Review:
https://reviewboard.asterisk.org/r/937/ ........ ................
2010-09-28 01:04 +0000 [r289054-289057] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.c: Avoid deadlock processing incoming AOC-E
messages. Deadlock avoidance for the owner channel was not done
when processing incoming AOC-E messages.
* channels/sig_pri.c: Revert stuff not ready for commit in
-r289054.
* channels/sig_pri.c, channels/chan_sip.c: Break up long
ast_manager_event_multichan() event lines.
2010-09-27 18:37 +0000 [r288961] Tilghman Lesher <tlesher@digium.com>
* channels/chan_sip.c: Still build SIP, even if res_crypto cannot
be built (use, not depend). (closes issue #18062) Reported by: a
user on the mailing list
2010-09-27 13:03 +0000 [r288925-288927] Russell Bryant <russell@digium.com>
* res/res_agi.c: Fix some documentation typos and spelling errors.
* res/res_agi.c: Fix a documentation spelling error.
2010-09-24 17:58 +0000 [r288821-288852] David Vossel <dvossel@digium.com>
* channels/chan_sip.c: Append Retry-After header on 500 error
response to Re-INVITE according to RFC3261 section 14.2. ABE-2301
* channels/chan_sip.c: Inspect Require header on BYE transaction
according to RFC3261 section 8.2.2.3. ABE-2293
2010-09-24 16:02 +0000 [r288748] Terry Wilson <twilson@digium.com>
* channels/chan_local.c, /: Merged revisions 288747 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288747 | twilson | 2010-09-24 08:37:39 -0700
(Fri, 24 Sep 2010) | 12 lines Merged revisions 288746 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010)
| 5 lines Don't fail a masquerade if it is already being hung up
This avoids noise on some Local channel situations where we don't
use /n. Thanks to Alec Davis for the suggestion. ........
................
2010-09-24 13:54 +0000 [r288606-288713] Tilghman Lesher <tlesher@digium.com>
* /, funcs/func_strings.c: Merged revisions 288712 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r288712 | tilghman | 2010-09-24 08:53:30 -0500 (Fri, 24
Sep 2010) | 5 lines Solaris won't printf a NULL. (closes issue
#18041) Reported by: asgaroth ........
* main/asterisk.exports.in: Export timersub for platforms which do
not have it
* include/asterisk/channel.h, cdr/cdr_pgsql.c, /, configure,
include/asterisk/autoconfig.h.in, include/asterisk/compat.h,
main/strcompat.c, configure.ac: Merged revisions 288637 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288637 | tilghman | 2010-09-23 22:36:01 -0500
(Thu, 23 Sep 2010) | 9 lines Merged revisions 288636 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23
Sep 2010) | 2 lines Solaris compatibility fixes ........
................
* CHANGES: Add note about the checkhangup option of ${CHANNEL()}
2010-09-23 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-rc2 Released.
2010-09-23 18:05 +0000 [r288507-288572] Terry Wilson <twilson@digium.com>
* main/manager.c: Make AMI honor enabled=no (closes issue #18040)
Reported by: twilson Review:
https://reviewboard.asterisk.org/r/938/
* channels/chan_local.c, /: Merged revisions 288500 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288500 | twilson | 2010-09-22 16:10:09 -0700
(Wed, 22 Sep 2010) | 15 lines Merged revisions 288499 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010)
| 8 lines Don't let a Local channel get bridged to itself If a
local channel gets bridged to itself, it becomes orphaned with no
devices left to actually tell it to hang up. This patch modifies
local_fixup() to detect this case and deny it. Review:
https://reviewboard.asterisk.org/r/934 ........ ................
2010-09-22 Leif Madsen <lmadsen@digium.com>
* Asterisk 1.8.0-rc1 Released.
2010-09-22 17:49 +0000 [r288345-288418] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 288417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288417 | dvossel | 2010-09-22 12:49:05 -0500
(Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010)
| 5 lines RFC3261 section 12.2 explicitly says out of order
requests are responded with a 500 Server Internal Error response.
ABE-2458 ........ ................
* /, channels/chan_sip.c: Merged revisions 288344 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288344 | dvossel | 2010-09-22 11:53:28 -0500
(Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.4
........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22
Sep 2010) | 2 lines During check_pendings, if the dialog is
terminated with a CANCEL, change the invitestate to INV_CANCEL
like in sip_hangup. ........ ................
2010-09-22 16:45 +0000 [r288341] Russell Bryant <russell@digium.com>
* main/asterisk.c, /: Merged revisions 288340 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288340 | russell | 2010-09-22 11:44:13 -0500
(Wed, 22 Sep 2010) | 18 lines Merged revisions 288339 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010)
| 11 lines Fix a 100% CPU consumption problem when setting
console=yes in asterisk.conf. The handling of -c and console=yes
should be the same, but they were not. When you specify -c, it
sets both a flag for console module and for asterisk not to
fork() off into the background. The handling of console=yes only
set console mode, so you would end up with a background process()
trying to run the Asterisk console and freaking out since it
didn't have anything to read input from. Thanks to beagles for
reporting and helping debug the problem! ........
................
2010-09-22 15:14 +0000 [r288268] Tilghman Lesher <tlesher@digium.com>
* UPGRADE.txt, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample, /:
Merged revisions 288267 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288267 | tilghman | 2010-09-22 10:11:09 -0500
(Wed, 22 Sep 2010) | 23 lines Merged revisions 288265-288266 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010)
| 9 lines Allow the encoding to be set, in case local charset
does not agree with database. (closes issue #16940) Reported by:
jamicque Patches: 20100827__issue16940.diff.txt uploaded by
tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt
uploaded by tilghman (license 14) Tested by: jamicque ........
r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010)
| 5 lines Document addition of encoding parameter. (issue #16940)
Reported by: jamicque ........ ................
2010-09-22 00:06 +0000 [r288194] Richard Mudgett <rmudgett@digium.com>
* channels/chan_iax2.c, /: Merged revisions 288193 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288193 | rmudgett | 2010-09-21 19:03:37 -0500
(Tue, 21 Sep 2010) | 33 lines Merged revisions 288192 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010)
| 26 lines In chan_iax2.c:schedule_delivery() calls
ast_bridged_channel() on an unlocked channel. Near the beginning
of schedule_delivery(), ast_bridged_channel() is called on
iaxs[fr->callno]->owner. However, the channel is not locked,
which can result in ast_bridged_channel() crashing should
owner->tech change to a technology that doesn't implement
bridged_channel. I also fixed the other calls to
ast_bridged_channel() in chan_iax2.c since the owner lock was not
held there either. Converted the existing channel deadlock
avoidance to use iax2_lock_owner(). Using the new function
simplified some awkward code. In the process of fixing the
locking on ast_bridged_channel(), I also found a memory leak in
socket_process() for v1.6.2 and v1.8. The local struct variable
ies.vars is not freed on early/abnormal function exits. (closes
issue #17919) Reported by: rain Patches: issue17919_v1.4.patch
uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch
uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch
uploaded by rmudgett (license 664) Review:
https://reviewboard.asterisk.org/r/926/ ........ ................
2010-09-21 22:57 +0000 [r288159] Tilghman Lesher <tlesher@digium.com>
* /, channels/chan_sip.c: Merged revisions 288113 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................ r288113 | tilghman | 2010-09-21 16:59:46 -0500
(Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010)
| 15 lines Try both the encoded and unencoded subscription URI
for a match in hints. When a phone sends an encoded URI for a
subscription, the URI is not matched with the actual hint that is
in decoded format. For example, if we have an extension with a
hint that is named: "#5601" or "*5601", the subscription will
work fine if the phone subscribes with an already decoded URI,
but when it's decoded like "%255601" or "%2A5601", Asterisk is
unable to match it with the correct hint. (closes issue #17785)
Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt
uploaded by tilghman (license 14) Tested by: ramonpeek ........
................
2010-09-21 22:26 +0000 [r288157] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_iax2.c, /: Merged revisions 288147 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........ r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue,
21 Sep 2010) | 9 lines Setup timer before set_config(). (closes
issue #18019) Reported by: Netview Patches: issue_0018019.patch
uploaded by pabelanger (license 224) Tested by: Netview ........
2010-09-21 21:03 +0000 [r288079-288082] Richard Mudgett <rmudgett@digium.com>
* doc/tex/partymanip.tex: Add note in party manipulation chapter on
interception macros.
* apps/app_queue.c, apps/app_dial.c: Simplify locking code for
REDIRECTING interception macro when forwarding a call. Simplified
the locking code by using a local copy of the redirecting party
information in app_dial.c:do_forward() and
app_queue.c:wa