Jason Parker [Fri, 8 Jul 2011 22:25:14 +0000 (22:25 +0000)]
Add .o files to svn:ignore property, since it's only ignored if locally configured to do so.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327246
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Richard Mudgett [Fri, 8 Jul 2011 21:43:49 +0000 (21:43 +0000)]
Merged revisions 327211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) | 9 lines
INVITE 403 Forbidden response always retransmits the maximum times.
Asterisk sends a 403 Forbidden response if authentication fails for an
INVITE as required. However, it ignores the ACK and keeps retransmitting
the response.
* Made not delete the to-tag in the dialog so the expected ACK can be
matched with the dialog and stop the retransmissions.
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David Vossel [Fri, 8 Jul 2011 20:33:49 +0000 (20:33 +0000)]
Adds entry in UPDATES.txt for removal of formats/format_sln16.c. Fixes typo in CHANGES as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327168
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David Vossel [Fri, 8 Jul 2011 20:26:07 +0000 (20:26 +0000)]
Updates CHANGES log to reflect new slinear read/write file interpreters.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327148
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David Vossel [Fri, 8 Jul 2011 20:23:37 +0000 (20:23 +0000)]
Support for writing and reading raw slin files 8khz-192khz.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327137
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David Vossel [Fri, 8 Jul 2011 20:18:39 +0000 (20:18 +0000)]
Moves celt and silk format attribute files into res folder.
It was inconsistent to have the silk and celt format attribute
modules in the format file interpreter folder.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327116
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Matthew Nicholson [Fri, 8 Jul 2011 19:54:23 +0000 (19:54 +0000)]
Merged revisions 327106 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327106 | mnicholson | 2011-07-08 14:52:51 -0500 (Fri, 08 Jul 2011) | 11 lines
Reset our ast_str before passing it on to dialplan function backends.
It is possible for a dialplan backend to not modify the given buffer or ast_str
and still return success. This causes any previous value stored in the buffer
to be used as if the new function call provided it. Some functions also append
to the given buffer assuming it is empty.
The test_substitution unit test has also been modified to detect this problem.
(closes issue ASTERISK-17878)
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Russell Bryant [Fri, 8 Jul 2011 16:00:46 +0000 (16:00 +0000)]
Merged revisions 327046 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327046 | russell | 2011-07-08 11:00:05 -0500 (Fri, 08 Jul 2011) | 2 lines
Fix an error and add more log message info to help see why this fails on FreeBSD.
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Russell Bryant [Fri, 8 Jul 2011 15:39:42 +0000 (15:39 +0000)]
Merged revisions 327044 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327044 | russell | 2011-07-08 10:28:44 -0500 (Fri, 08 Jul 2011) | 2 lines
Resolve some set-but-unused-variable warnings.
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Richard Mudgett [Fri, 8 Jul 2011 01:26:01 +0000 (01:26 +0000)]
Merged revisions 326985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r326985 | rmudgett | 2011-07-07 20:08:05 -0500 (Thu, 07 Jul 2011) | 12 lines
Some code cleanup in pbx.c
* Mostly comment and format changes.
* ast_context_remove_extension_callerid() and ast_add_extension_nolock()
will write lock the found specific context.
* ast_context_find() will now tolerate a NULL name.
* Eliminated some inlined versions of find_context() and
find_context_locked().
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Jason Parker [Thu, 7 Jul 2011 22:39:54 +0000 (22:39 +0000)]
I think reviewboard broke this. The whole file was doubled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326943
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David Vossel [Thu, 7 Jul 2011 22:17:47 +0000 (22:17 +0000)]
Adds the format_attr_celt file which was also missing from the CELT pass through patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326904
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David Vossel [Thu, 7 Jul 2011 22:16:10 +0000 (22:16 +0000)]
Adds missing celt.h file from celt pass-through support patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326900
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David Vossel [Thu, 7 Jul 2011 19:57:06 +0000 (19:57 +0000)]
Fixes spelling errors in CHANGES as well as adding a few entries for CELT and confbridge.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326856
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David Vossel [Thu, 7 Jul 2011 19:39:17 +0000 (19:39 +0000)]
Adds pass-through support for codec CELT.
This patch adds pass-through support for CELT. CELT
formats are defined in codecs.conf and can be configured
to any sample rate a CELT endpoint supports. This patch also
addresses a crash in channel.c resulting from a frame list being
freed incorrectly. This crash was discovered while testing a CELT
translator which had to split encoded audio into multiple frames.
The codec translator is not a part of this patch, but may be
contributed in the future.
Review: https://reviewboard.asterisk.org/r/1294/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326855
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Tilghman Lesher [Thu, 7 Jul 2011 19:20:38 +0000 (19:20 +0000)]
Merged revisions 326830 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r326830 | tilghman | 2011-07-07 14:17:19 -0500 (Thu, 07 Jul 2011) | 1 line
libgen.h is also needed on Darwin for basename(3)
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David Vossel [Thu, 7 Jul 2011 17:24:57 +0000 (17:24 +0000)]
Updates confbridge.conf video documentation and adds dtmf action for releasing video src.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326782
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Terry Wilson [Thu, 7 Jul 2011 16:50:54 +0000 (16:50 +0000)]
Use older functions out of deference to older distros
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326750
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Jonathan Rose [Thu, 7 Jul 2011 16:18:18 +0000 (16:18 +0000)]
Merged revisions 326689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r326689 | jrose | 2011-07-07 11:04:51 -0500 (Thu, 07 Jul 2011) | 10 lines
res_odbc patch by tilghman to fix integers with null values
Addresses some improper sql statements in res_odbc that would cause an update to fail on
realtime peers due to trying to set as "(NULL)" rather than an actual NULL.
(closes issue #1922STERISK-17791)
Reported by: marcelloceschia
Patches:
20110505__issue19223.diff.txt uploaded by tilghman (license 14)
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Matthew Nicholson [Thu, 7 Jul 2011 15:28:47 +0000 (15:28 +0000)]
Merged revisions 326683 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r326683 | mnicholson | 2011-07-07 10:28:25 -0500 (Thu, 07 Jul 2011) | 3 lines
use sips: or sip: depending on the transport in use when building reply digest
URIs
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Matthew Nicholson [Thu, 7 Jul 2011 15:26:42 +0000 (15:26 +0000)]
Merged revisions 326681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r326681 | mnicholson | 2011-07-07 10:25:49 -0500 (Thu, 07 Jul 2011) | 3 lines
make the uri parameter used in reply digests more standards compliant in
certain cases by prepending "sip:" or "sips:" to it
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Tzafrir Cohen [Thu, 7 Jul 2011 09:49:54 +0000 (09:49 +0000)]
live_ast: valgrind: run asterisk under valgrind
Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under
valgrind. The extra command-line parameters are passed to Asterisk as
usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS
in live.conf .
Review: https://reviewboard.asterisk.org/r/1109/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326636
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Terry Wilson [Wed, 6 Jul 2011 20:58:12 +0000 (20:58 +0000)]
Replace Berkeley DB with SQLite 3
There were some bugs in the very ancient version of Berkeley DB that Asterisk
used. Instead of spending the time tracking down the bugs in the Berkeley code
we move to the much better documented SQLite 3.
Conversion of the old astdb happens at runtime by running the included
astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave
identically to the old Berkeley backend, but in the future we could offer a
much more robust interface.
We do not include the SQLite 3 library in the source tree, but instead rely
upon the distribution-provided libraries. SQLite is so ubiquitous that this
should not place undue burden on administrators.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326589
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David Vossel [Wed, 6 Jul 2011 17:39:36 +0000 (17:39 +0000)]
Fixes newlines from being stripped from out of dialog sip MESSAGES.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326544
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David Vossel [Wed, 6 Jul 2011 15:30:28 +0000 (15:30 +0000)]
Merged revisions 326484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r326484 | dvossel | 2011-07-06 10:26:49 -0500 (Wed, 06 Jul 2011) | 10 lines
Reverts fix for timerfd locking issue.
jrose discovered a performance issue with this
fix that prevents his analog phones from working
when using timerfd as a timing source. Until
it is understood what is causing this performance
problem, this patch is being reverted.
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Tilghman Lesher [Tue, 5 Jul 2011 22:11:40 +0000 (22:11 +0000)]
Merged revisions 326411 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines
Add the attribute "type" to each "<use>" for menuselect.
This matters only when autoconf fails to detect that weak linking is supported.
External optional dependencies will become optional in both cases, as they are
removed at compile time when not detected. However, runtime-optional modules
are made mandatory when weak linking is not found. This change affects only
the external optional dependencies; previously, they were incorrectly required
when weak linking support was not detected.
Patches:
20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
Tested by: iasgoscouk
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412
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Kinsey Moore [Tue, 5 Jul 2011 20:25:23 +0000 (20:25 +0000)]
Prompt conversion script
Several variables in the script control which files are converted and the
source and destination formats.
Patch-by: Trey Blancher <support@digium.com>
(closes AST-560)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326368
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Richard Mudgett [Tue, 5 Jul 2011 17:35:54 +0000 (17:35 +0000)]
Merged revisions 326291 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r326291 | rmudgett | 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) | 23 lines
Used auth= parameter freed during "sip reload" causes crash.
If you use the auth= parameter and do a "sip reload" while there is an
ongoing call. The peer->auth data points to free'd memory.
The patch does several things:
1) Puts the authentication list into an ao2 object for reference counting
to fix the reported crash during a SIP reload.
2) Converts the authentication list from open coding to AST list macros.
3) Adds display of the global authentication list in "sip show settings".
(closes issue ASTERISK-17939)
Reported by: wdoekes
Patches:
jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett
Review: https://reviewboard.asterisk.org/r/1303/
JIRA SWP-3526
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Mark Murawki [Tue, 5 Jul 2011 16:46:17 +0000 (16:46 +0000)]
New feature: AMI Action FilterAdd
This adds a new action, FilterAdd to the manager interface that allows control over event filters for the current session
(closes issue ASTERISK-16795)
Reported by: kobaz
Tested by: kobaz,loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326267
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Matthew Jordan [Tue, 5 Jul 2011 13:38:37 +0000 (13:38 +0000)]
Merged revisions 326209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r326209 | mjordan | 2011-07-05 08:23:57 -0500 (Tue, 05 Jul 2011) | 7 lines
Updated filestream destructor to block until move is complete when cache is used
When a cache directory is used, the process is forked and a mv command is executed to move the temporary file to the permanent location. This caused issues with voicemail, where a race condition occurred when the parent expected the file to be in the permanent location prior to the mv command completing. The parent process is now blocked until the mv command completes.
(closes issue ASTERISK-17724)
Reported by: Adiren P.
Tested by: mjordan
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Richard Mudgett [Fri, 1 Jul 2011 21:11:34 +0000 (21:11 +0000)]
Merged revisions 326144 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r326144 | rmudgett | 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) | 16 lines
Better way to get chan and pvt lock for issue ASTERISK-17431.
Redoes -r308945 for issue ASTERISK-17431 deadlock fix for
sip_set_udptl_peer() and sip_set_rtp_peer().
* Lock the channels in the defined order and avoid the need for a deadlock
avoidance loop.
* Lock the channel before getting the pointer to the private structure to
be sure that the pointer will not change due to a masquerade or channel
hangup.
* To preserve sanity, check that chan and p->owner are the same. (Pointer
rearangements should not happen without the protection of locks because
bad things tend to happen otherwise.)
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Gregory Nietsky [Fri, 1 Jul 2011 16:36:29 +0000 (16:36 +0000)]
Change CHANGES move the commits to the right place
r296249 r318141 Application changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326101
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Gregory Nietsky [Fri, 1 Jul 2011 16:16:07 +0000 (16:16 +0000)]
Change CHANGES move the commits to the right place in the file missed in review
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326056
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Matthew Nicholson [Fri, 1 Jul 2011 12:45:09 +0000 (12:45 +0000)]
updated irroots info for the authors section
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326006
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David Vossel [Thu, 30 Jun 2011 21:05:54 +0000 (21:05 +0000)]
Fixes warning message caused by confbridge playback chan not being answered.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325937
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Richard Mudgett [Thu, 30 Jun 2011 20:47:44 +0000 (20:47 +0000)]
Merged revisions 325935 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines
Misc minor changes in chan_sip.
* Add load failure exit if primary SIP container(s) could not get created
in chan_sip.c:load_module().
* Removed a redundant static prototype.
* Some typos.
* Some whitespace.
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David Vossel [Thu, 30 Jun 2011 20:33:15 +0000 (20:33 +0000)]
Video support for ConfBridge.
Review: https://reviewboard.asterisk.org/r/1288/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325931
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Matthew Jordan [Thu, 30 Jun 2011 20:24:00 +0000 (20:24 +0000)]
Merged revisions 325877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r325877 | mjordan | 2011-06-30 15:09:48 -0500 (Thu, 30 Jun 2011) | 9 lines
Patched voicemail user option for emailbody / emailsubject
Incorporated changes per ASTERISK-16795; updated unit tests to check for vmu->emailbody / vmu->emailsubject
(closes issue ASTERISK-16795)
Reported by: mdeneen
Tested by: mjordan
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Jonathan Rose [Thu, 30 Jun 2011 19:31:51 +0000 (19:31 +0000)]
Merged revisions 325821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r325821 | jrose | 2011-06-30 14:17:32 -0500 (Thu, 30 Jun 2011) | 10 lines
Fixes an issue with Music on Hold classes losing files in playlist when realtime is used.
The bug occurs rather intermittently and I relied on the reporters to test the patch.
After a sanity check and some testing, I'm giving it an OK.
(closes issue ASTERISK-17875)
Reported by: David Cunningham
Patches:
res_musiconhold.c.mohrt17875_v1 uploaded by Igor Goncharovsky (license #5009)
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Matthew Nicholson [Thu, 30 Jun 2011 18:22:28 +0000 (18:22 +0000)]
Fax gateway functionality (i.e. translating between a T.30 terminal and a T.38
terminal). Can be enabled on a channel by setting FAXOPT(gateway)=yes in the
dialplan.
Big thanks to irroot for porting this code to use the framehooks api.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325816
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Matthew Nicholson [Thu, 30 Jun 2011 18:19:31 +0000 (18:19 +0000)]
copy all flags on asterisk frames instead of just the timing flag
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325815
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Kinsey Moore [Wed, 29 Jun 2011 21:50:32 +0000 (21:50 +0000)]
Merged revisions 325740 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r325740 | kmoore | 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) | 7 lines
chan_sip: cleanup from the introduction of ast_str
Remove the length field from sip_req and sip_pkt in chan_sip since they are
redundant (ast_str holds its own length) and refactor the necessary functions.
Review: https://reviewboard.asterisk.org/r/1281/
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David Vossel [Wed, 29 Jun 2011 19:02:19 +0000 (19:02 +0000)]
Merged revisions 325673 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r325673 | dvossel | 2011-06-29 13:59:33 -0500 (Wed, 29 Jun 2011) | 6 lines
Fixes timerfd locking issue.
(closes ASTERISK-17867, ASTERISK-17415)
Patches:
fix uploaded by kobaz
Review: https://reviewboard.asterisk.org/r/1255/
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Richard Mudgett [Wed, 29 Jun 2011 18:18:00 +0000 (18:18 +0000)]
Merged revisions 325614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r325614 | rmudgett | 2011-06-29 13:16:45 -0500 (Wed, 29 Jun 2011) | 5 lines
Fixed some error exit cleanup in app_queue.c.
* Fixed error exit cleanup in app_queue.c copy_rules() and
reload_queue_rules().
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Richard Mudgett [Wed, 29 Jun 2011 18:07:26 +0000 (18:07 +0000)]
Merged revisions 325610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r325610 | rmudgett | 2011-06-29 13:05:15 -0500 (Wed, 29 Jun 2011) | 18 lines
Response to QueueRule manager command does not contain ActionID if it was specified.
* Add ActionID support as documented for the QueueRule AMI action.
* Remove documentation for ActionID with the Queues AMI action. The
output does not follow normal AMI response output and there is no place to
put an ActionID header.
(closes issue AST-602)
Reported by: Vlad Povorozniuc
Patches:
jira_ast_602_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Vlad Povorozniuc, rmudgett
Review: https://reviewboard.asterisk.org/r/1295/
JIRA SWP-3575
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Matthew Nicholson [Wed, 29 Jun 2011 16:19:01 +0000 (16:19 +0000)]
Merged revisions 325545 via svnmerge from
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r325545 | mnicholson | 2011-06-29 11:18:39 -0500 (Wed, 29 Jun 2011) | 2 lines
make framehooks prevent native bridging (for real this time)
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Matthew Nicholson [Wed, 29 Jun 2011 15:36:20 +0000 (15:36 +0000)]
Merged revisions 325537 via svnmerge from
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r325537 | mnicholson | 2011-06-29 10:34:47 -0500 (Wed, 29 Jun 2011) | 2 lines
don't do native/remote bridging if a framehook is active on the channel
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Gregory Nietsky [Wed, 29 Jun 2011 06:39:26 +0000 (06:39 +0000)]
Commit "distrotech" app_queue changes to Trunk
* Added general option negative_penalty_invalid default off. when set
members are seen as invalid/logged out when there penalty is negative.
for realtime members when set remove from queue will set penalty to -1.
* Added queue option autopausedelay when autopause is enabled it will be
delayed for this number of seconds since last successful call if there
was no prior call the agent will be autopaused immediately.
* Added member option ignorebusy this when set and ringinuse is not
will allow per member control of multiple calls as ringinuse does for
the Queue.
- Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
- QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
(closes issue ASTERISK-17421)
(closes issue ASTERISK-17391)
Reported by: irroot
Tested by: irroot, jrose
Review: https://reviewboard.asterisk.org/r/1119/
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Kevin P. Fleming [Tue, 28 Jun 2011 21:51:19 +0000 (21:51 +0000)]
Merged revisions 325416 via svnmerge from
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r325416 | kpfleming | 2011-06-28 16:50:43 -0500 (Tue, 28 Jun 2011) | 3 lines
Fix random misspelling noticed on asterisk-users.
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David Vossel [Tue, 28 Jun 2011 20:32:22 +0000 (20:32 +0000)]
Merged revisions 325339 via svnmerge from
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r325339 | dvossel | 2011-06-28 15:31:00 -0500 (Tue, 28 Jun 2011) | 4 lines
Fixes locking inversion caused by holding sip pvt lock during async_goto.
(closes ASTERISK-17352)
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Richard Mudgett [Tue, 28 Jun 2011 17:38:28 +0000 (17:38 +0000)]
Merged revisions 325212 via svnmerge from
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r325212 | rmudgett | 2011-06-28 12:30:16 -0500 (Tue, 28 Jun 2011) | 7 lines
Use the device name and not the channel name to initialize the device state.
Correct ASTERISK-11323 implementation as I don't see how it ever worked as
claimed when it used the channel name and not the device name.
(issue ASTERISK-11323)
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Jonathan Rose [Tue, 28 Jun 2011 16:04:18 +0000 (16:04 +0000)]
Merged revisions 325152 via svnmerge from
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r325152 | jrose | 2011-06-28 10:46:29 -0500 (Tue, 28 Jun 2011) | 5 lines
Fixes moh reload breaking custom mode moh classes when the config file is untouched
(closes issue ASTERISK-17730)
Reported by: sdolloff
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David Vossel [Tue, 28 Jun 2011 15:34:59 +0000 (15:34 +0000)]
Fixes issue with video and text not being reinvited correctly with directmedia
If a SDP does not modify the session, we ignore it. However, we were defaulting
no text and video support to true before checking to see if the sdp modified
anything or not. This would result in process_sdp ignoring an sdp but removing
video and text from the call during direct media reinvites.
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Leif Madsen [Tue, 28 Jun 2011 15:12:34 +0000 (15:12 +0000)]
Merged revisions 325091 via svnmerge from
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r325091 | lmadsen | 2011-06-28 10:12:00 -0500 (Tue, 28 Jun 2011) | 1 line
Remove line from prep_tarball that kills mkrelease.
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Terry Wilson [Tue, 28 Jun 2011 00:07:47 +0000 (00:07 +0000)]
Don't forget to build the Via when sending MESSAGE
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Tilghman Lesher [Mon, 27 Jun 2011 16:32:19 +0000 (16:32 +0000)]
Merged revisions 324955 via svnmerge from
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r324955 | tilghman | 2011-06-27 11:30:50 -0500 (Mon, 27 Jun 2011) | 5 lines
Save and restore errno from within signal handlers.
This is recommended by the POSIX standard, as well as by the sigaction(2) manpage
for various platforms that we support (e.g. Mac OS X).
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Richard Mudgett [Mon, 27 Jun 2011 15:38:44 +0000 (15:38 +0000)]
Merged revisions 324914 via svnmerge from
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r324914 | rmudgett | 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011) | 21 lines
When subscribing MWI to an unsolicited mailbox the first notification is incorrect.
A remote peer subscribed to MWI with the unsolicited option and a local
phone subscribed to the remote mailbox. The notify message-summary events
are sent correctly except for the first one when subscribing, which will
always be 0. This means the phone MWI indicator will be wrong until the
mailbox read/unread count changes and the event is fired.
Looks like this is a regression from ASTERISK-16149.
* Fix the logic to check the cache and if allowed then fallback to
manually counting mailbox messages.
(closes issue ASTERISK-17997)
Reported by: rsw686
Patches:
jira_asterisk_17997_v1.8.patch (license #5621) uploaded by rmudgett
Tested by: rsw686
JIRA SWP-3551
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Richard Mudgett [Fri, 24 Jun 2011 20:50:52 +0000 (20:50 +0000)]
Merged revisions 324849 via svnmerge from
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r324849 | rmudgett | 2011-06-24 15:46:01 -0500 (Fri, 24 Jun 2011) | 15 lines
Syntax errors in dialplan do not display the file name.
When issuing the CLI command "dialplan reload" syntax errors and warnings
are displayed on the console. The offending line number is displayed on
the console, but the file name is not displayed. Errors caught in
main/config.c do display the file name.
(closes issue ASTERISK-17985)
Reported by: ulogic
Patches:
pbx_config.patch uploaded by ulogic (License #5685) modified format
Tested by: rmudgett
JIRA SWP-3554
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Jonathan Rose [Fri, 24 Jun 2011 16:50:49 +0000 (16:50 +0000)]
Merged revisions 324768 via svnmerge from
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r324768 | jrose | 2011-06-24 11:48:06 -0500 (Fri, 24 Jun 2011) | 11 lines
DTMF wasn't being logged on connected consoles when enabled in logger.conf
Previously in order for DTMF to be logged in a connected console session, the user would
have to do logger set channel DTMF on. This corrects that so that it is on by default.
This issue was caused by an off by one error incurred by a logger level count of 6 in
logger.h where it should have been 7.
(closes issue: ASTERISK-17974)
Reported by: Luke H
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Kinsey Moore [Thu, 23 Jun 2011 18:56:05 +0000 (18:56 +0000)]
ConfBridge: redundant code cleanup
There is no reason to clean up features twice.
Review: https://reviewboard.asterisk.org/r/1279/
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Kinsey Moore [Thu, 23 Jun 2011 18:52:59 +0000 (18:52 +0000)]
Merged revisions 324678 via svnmerge from
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r324678 | kmoore | 2011-06-23 13:29:17 -0500 (Thu, 23 Jun 2011) | 11 lines
Merged revisions 324643 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) | 4 lines
Addresses AST-2011-008, memory corruption and remote crash in SIP driver.
AST-2011-008
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David Vossel [Thu, 23 Jun 2011 18:31:42 +0000 (18:31 +0000)]
Merged revisions 324685 via svnmerge from
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r324685 | dvossel | 2011-06-23 13:31:00 -0500 (Thu, 23 Jun 2011) | 8 lines
Fixes sip crash when calling remove_uri_parameters with NULL
AST-2011-009
(closes issue ASTERISK-18017)
Reported by: jaredmauch
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David Vossel [Thu, 23 Jun 2011 18:26:09 +0000 (18:26 +0000)]
Merged revisions 324652 via svnmerge from
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r324652 | dvossel | 2011-06-23 13:23:21 -0500 (Thu, 23 Jun 2011) | 20 lines
Merged revisions 324634 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r324634 | dvossel | 2011-06-23 13:18:46 -0500 (Thu, 23 Jun 2011) | 13 lines
Merged revisions 324627 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) | 7 lines
Addresses AST-2011-010, remote crash in IAX2 driver
Thanks to twilson for identifying the issue and providing the patches.
AST-2011-010
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Terry Wilson [Thu, 23 Jun 2011 03:16:44 +0000 (03:16 +0000)]
Merged revisions 324557 via svnmerge from
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r324557 | twilson | 2011-06-22 22:10:38 -0500 (Wed, 22 Jun 2011) | 5 lines
Remove tests for parsing address with invalid port
getaddrinfo on OS X returns with EAI_NONAME error when passed a port
greater than 65535. Linux throws no error, so remove the tests for now.
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Richard Mudgett [Wed, 22 Jun 2011 19:17:56 +0000 (19:17 +0000)]
Merged revisions 324491 via svnmerge from
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r324491 | rmudgett | 2011-06-22 14:16:29 -0500 (Wed, 22 Jun 2011) | 1 line
Use correct variable for text SRTP media.
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Terry Wilson [Wed, 22 Jun 2011 19:12:24 +0000 (19:12 +0000)]
Merged revisions 324484 via svnmerge from
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r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines
Stop sending IPv6 link-local scope-ids in SIP messages
The idea behind the patch listed below was used, but in a more targeted manner.
There are now address stringification functions for addresses that are meant to
be sent to a remote party. Link-local scope-ids only make sense on the machine
from which they originate and so are stripped in the new functions.
There is also a host sanitization function added to chan_sip which is used
for when peer and dialog tohost fields or sip_registry hostnames are used to
craft a SIP message.
Also added are some basic unit tests for netsock2 address parsing.
(closes issue ASTERISK-17711)
Reported by: ch_djalel
Patches:
asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251)
Review: https://reviewboard.asterisk.org/r/1278/
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Richard Mudgett [Wed, 22 Jun 2011 18:45:24 +0000 (18:45 +0000)]
Merged revisions 324481 via svnmerge from
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Also fixed a reference leak in an error path in sip_msg_send().
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r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) | 19 lines
Timout or error on INFO or MESSAGE transaction causes call to be lost.
When exchanging INFO messages within a call, 4xx error causes the call to
be disconnected although RFC 2976 explicitly states that such transactions
do not modify the state of the dialog.
When exchanging MESSAGE messages within a call, 4xx error causes the call
to be disconnected. To provide least surprise, we should not disconnect
the call since a MESSAGE is like INFO in this case. (Implied by RFC 3428
Section 2)
(closes issue ASTERISK-17901)
Reported by: neutrino88
Review: https://reviewboard.asterisk.org/r/1257/
Review: https://reviewboard.asterisk.org/r/1258/
JIRA SWP-3486
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Richard Mudgett [Wed, 22 Jun 2011 18:27:43 +0000 (18:27 +0000)]
Merged revisions 324479 via svnmerge from
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r324479 | rmudgett | 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011) | 1 line
Comments and whitespace in chan_sip.c
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David Vossel [Tue, 21 Jun 2011 21:55:30 +0000 (21:55 +0000)]
Fixes issue with channel write format being incorrectly restored when MOH is used in confbridge.
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David Vossel [Tue, 21 Jun 2011 20:15:41 +0000 (20:15 +0000)]
Merged revisions 324364 via svnmerge from
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r324364 | dvossel | 2011-06-21 15:11:52 -0500 (Tue, 21 Jun 2011) | 10 lines
Fixes locking inversion issue in ast_async_goto()
During this function we can not hold the "chan" lock while
doing the masquerade, the explicit goto on the tmp chan, or
the channel alloc. Instead we need to get the channel lock,
store off information about the channel that we need, and
then let the channel lock go for the remainder of the function.
Review: https://reviewboard.asterisk.org/r/1275/
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Kinsey Moore [Tue, 21 Jun 2011 16:06:46 +0000 (16:06 +0000)]
ConfBridge does not handle hangup properly
When playing back a prompt to a channel, confbridge neglects to check for
hangup events causing lockup condititions for hangups that occur before
actually joining the conference. This change ensures that the user is removed
from the conference in the event of a premature hangup.
Review: https://reviewboard.asterisk.org/r/1277/
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David Vossel [Tue, 21 Jun 2011 15:49:23 +0000 (15:49 +0000)]
Fixes issue with finding correct extension when message context is used.
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Leif Madsen [Mon, 20 Jun 2011 18:13:02 +0000 (18:13 +0000)]
Merged revisions 324241 via svnmerge from
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r324241 | lmadsen | 2011-06-20 13:12:32 -0500 (Mon, 20 Jun 2011) | 2 lines
Remove extra 'the'.
Reported by Vlad Povorozniuc
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Terry Wilson [Mon, 20 Jun 2011 17:34:45 +0000 (17:34 +0000)]
Merged revisions 324237 via svnmerge from
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r324237 | twilson | 2011-06-20 12:33:07 -0500 (Mon, 20 Jun 2011) | 12 lines
Ignore media offers with a port of 0
Section 5.1 of RFC3264 states:
A port number of zero in the offer indicates that the stream is offered
but MUST NOT be used.
(closes issue ASTERISK-17845)
Reported by: jacco
Patches:
issue19281_2.patch uploaded by jacco (license 1277)
Tested by: jacco, twilson
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Leif Madsen [Fri, 17 Jun 2011 18:52:33 +0000 (18:52 +0000)]
Merged revisions 324178 via svnmerge from
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r324178 | lmadsen | 2011-06-17 14:51:16 -0400 (Fri, 17 Jun 2011) | 2 lines
Add Username and Secret fields to manager Login action.
Pointed out by Vlad Povorozniuc
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Leif Madsen [Fri, 17 Jun 2011 18:39:26 +0000 (18:39 +0000)]
Merged revisions 324176 via svnmerge from
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r324176 | lmadsen | 2011-06-17 14:38:40 -0400 (Fri, 17 Jun 2011) | 2 lines
Fix typo in documentation.
Pointed out by Vlad Povorozniuc
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Richard Mudgett [Fri, 17 Jun 2011 18:23:54 +0000 (18:23 +0000)]
Merged revisions 324174 via svnmerge from
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r324174 | rmudgett | 2011-06-17 13:23:19 -0500 (Fri, 17 Jun 2011) | 5 lines
Add header string to libpri debug output.
Add header string to libpri debug output so the libpri output can be
found/extracted easier from huge debug trace files.
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Leif Madsen [Fri, 17 Jun 2011 15:32:08 +0000 (15:32 +0000)]
Merged revisions 324115 via svnmerge from
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r324115 | lmadsen | 2011-06-17 11:14:54 -0400 (Fri, 17 Jun 2011) | 3 lines
Fix grammar in documentation for Goto() and GotoIf()
(closes issue ASTERISK-18023)
Reported by: Tim Osman
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Terry Wilson [Thu, 16 Jun 2011 22:49:49 +0000 (22:49 +0000)]
Merged revisions 324048 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) | 8 lines
Lock the channel before calling the setoption callback
The channel needs to be locked before calling these callback functions. Also,
sip_setoption needs to lock the pvt and a check p->rtp is non-null before using
it.
Review: https://reviewboard.asterisk.org/r/1220/
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Richard Mudgett [Thu, 16 Jun 2011 18:13:01 +0000 (18:13 +0000)]
Merged revisions 323990 via svnmerge from
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r323990 | rmudgett | 2011-06-16 13:12:32 -0500 (Thu, 16 Jun 2011) | 5 lines
The test_event unit test is occasionally failing.
Wait for the special posted event to process before adding a new
subscription.
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Terry Wilson [Thu, 16 Jun 2011 15:59:17 +0000 (15:59 +0000)]
Merged revisions 323932 via svnmerge from
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r323932 | twilson | 2011-06-16 10:58:22 -0500 (Thu, 16 Jun 2011) | 4 lines
Don't assume ASTDBDIR exists
It most likely doesn't on FreeBSD
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Terry Wilson [Wed, 15 Jun 2011 20:04:55 +0000 (20:04 +0000)]
Merged revisions 323866 via svnmerge from
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r323866 | twilson | 2011-06-15 15:03:58 -0500 (Wed, 15 Jun 2011) | 2 lines
Remove now-useless cast of ARRAY_LEN
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Terry Wilson [Wed, 15 Jun 2011 20:02:30 +0000 (20:02 +0000)]
Merged revisions 323863 via svnmerge from
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r323863 | twilson | 2011-06-15 14:58:18 -0500 (Wed, 15 Jun 2011) | 2 lines
Make ARRAY_LEN() return the same type on x86 and x86_64 systems
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Terry Wilson [Wed, 15 Jun 2011 19:46:46 +0000 (19:46 +0000)]
Merged revisions 323859 via svnmerge from
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r323859 | twilson | 2011-06-15 14:45:20 -0500 (Wed, 15 Jun 2011) | 2 lines
Fix more ARRAY_LEN format string issues
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Terry Wilson [Wed, 15 Jun 2011 18:23:20 +0000 (18:23 +0000)]
Merged revisions 323754 via svnmerge from
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r323754 | twilson | 2011-06-15 13:21:52 -0500 (Wed, 15 Jun 2011) | 23 lines
Merged revisions 323733 via svnmerge from
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r323733 | twilson | 2011-06-15 13:13:00 -0500 (Wed, 15 Jun 2011) | 16 lines
Merged revisions 323732 via svnmerge from
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r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011) | 9 lines
Fix DYNAMIC_FEATURES
DYNAMIC_FEATURES were broken by a recent DTMF change. This patch makes
sure that dynamic features are also checked when deciding whether or not
to pass DTMF through or store it for interpreting.
(closes issue ASTERISK-17914)
Reported by: vrban
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Jonathan Rose [Wed, 15 Jun 2011 17:44:16 +0000 (17:44 +0000)]
Blocked revisions 323730 via svnmerge
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r323730 | jrose | 2011-06-15 12:42:42 -0500 (Wed, 15 Jun 2011) | 11 lines
Adds locking to find_table in res_configure_pgsql to prevent a crash.
Bryonclark described the problem as occuring during this function because of multiple
simultaneous database operations causing corruption against a pgsqlConn object.
(closes issue ASTERISK-17811)
Reported by: byronclark
Patches:
pgsql_find_table_locking.patch uploaded by byronclark (license 1200)
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Terry Wilson [Wed, 15 Jun 2011 17:12:29 +0000 (17:12 +0000)]
Merged revisions 323672 via svnmerge from
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r323672 | twilson | 2011-06-15 10:09:51 -0700 (Wed, 15 Jun 2011) | 5 lines
Cast ARRAY_LEN to size_t for ast_logging
32-bit and 64-bit machines return different types for ARRAY_LEN(), so cast
it before using in a format string.
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Richard Mudgett [Wed, 15 Jun 2011 16:49:34 +0000 (16:49 +0000)]
Merged revisions 323669-323670 via svnmerge from
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r323669 | rmudgett | 2011-06-15 11:43:18 -0500 (Wed, 15 Jun 2011) | 21 lines
[regression] Voicemail MWI is no longer sent.
When leaving a voicemail, the MWI message is never sent. The same thing
happens when checking a voicemail and marking it as read.
If you restart Asterisk, everything comes up at that state correctly, but
changes to the messages in voicemail causes the light to not be set
appropriately. Very easy to reproduce.
* Made ast_event_check_subscriber() return TRUE if there are ANY
subscribers to an event type when there are no restricting ie values
passed. This allows an event being queued to be queued.
(closes issue ASTERISK-18002)
Reported by: lmadsen
Tested by: lmadsen, irroot
Patches:
jira_asterisk_18002_v1.8.patch uploaded by rmudgett (License #5621)
(closes issue ASTERISK-18019)
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r323670 | rmudgett | 2011-06-15 11:43:31 -0500 (Wed, 15 Jun 2011) | 7 lines
Add a test to the event unit tests to catch ASTERISK-18002.
The new tests check to see if there are ANY subscribers to the event type
when ast_event_check_subscriber() is not passed any specific ie values.
(issue ASTERISK-18002)
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Jonathan Rose [Wed, 15 Jun 2011 16:19:38 +0000 (16:19 +0000)]
Merged revisions 323610 via svnmerge from
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r323610 | jrose | 2011-06-15 11:09:24 -0500 (Wed, 15 Jun 2011) | 7 lines
Adds PQclear calls on result to various parts of res_conf_pgsql
(closes issue ASTERISK-17812)
Reported by: byronclark
Patches:
pgsql_pqclear.patch uploaded by byronclark (license 1200)
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Sean Bright [Wed, 15 Jun 2011 15:33:57 +0000 (15:33 +0000)]
Merged revisions 323608 via svnmerge from
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r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines
Merged revisions 323579 via svnmerge from
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r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines
Merged revisions 323559 via svnmerge from
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r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines
Resolve a segfault/bus error when we try to map memory that falls on a page
boundary.
The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the
mmap'd region. The problem with this is that reading/writing to that extra byte
outside of the bounds of the underlying fd causes a bus error.
The real issue is that we are working with both a FILE * and the raw fd
underneath it and not synchronizing between them. The code that was removed in
ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping
the fd.
Looking at the manager code in 1.4 reveals that the FILE * in 'struct
mansession' is never used except to create a temporary file that we immediately
fdopen. This means we just need to write a 0 byte to the fd and everything will
just work. The other branches require a call to fflush() which, while not a
guaranteed fix, should reduce the likelihood of a crash.
This all makes sense in my head.
(closes issue ASTERISK-16460)
Reported by: Ravelomanantsoa Hoby (hoby)
Patches:
issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060)
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Kinsey Moore [Wed, 15 Jun 2011 13:45:41 +0000 (13:45 +0000)]
CONFBRIDGE_INFO function to get conference data
Added the CONFBRIDGE_INFO dialplan function to get information about a
conference bridge including locked status and number of parties, admins, and
marked users.
Review: https://reviewboard.asterisk.org/r/1271/
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Richard Mudgett [Wed, 15 Jun 2011 00:51:01 +0000 (00:51 +0000)]
Merged revisions 323456 via svnmerge from
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r323456 | rmudgett | 2011-06-14 19:50:20 -0500 (Tue, 14 Jun 2011) | 1 line
Add missing break in ast_event_get_cached().
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Richard Mudgett [Tue, 14 Jun 2011 17:22:26 +0000 (17:22 +0000)]
Merged revisions 323392,323394 via svnmerge from
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r323392 | rmudgett | 2011-06-14 12:21:24 -0500 (Tue, 14 Jun 2011) | 6 lines
Add more strict hostname checking to ast_dnsmgr_lookup().
Change suggested in review.
Review: https://reviewboard.asterisk.org/r/1240/
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r323394 | rmudgett | 2011-06-14 12:21:39 -0500 (Tue, 14 Jun 2011) | 2 lines
Made ast_sockaddr_split_hostport() port warning msgs more meaningful.
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Terry Wilson [Tue, 14 Jun 2011 17:03:37 +0000 (17:03 +0000)]
Merged revisions 323370 via svnmerge from
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r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) | 10 lines
Add rtpkeepalives back to 1.8
The RTP-engine conversion left out support for handling rtpkeepalives.
This patch adds them back.
(closes issue ASTERISK-17304)
Reported by: lmadsen
Review: https://reviewboard.asterisk.org/r/1226/
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Jonathan Rose [Tue, 14 Jun 2011 16:47:18 +0000 (16:47 +0000)]
Merged revisions 323371 via svnmerge from
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r323371 | jrose | 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) | 12 lines
Changes contact use in build_peer to use the FORCE_RPORT flag instead of RPORT_PRESENT
It turned out that this was causing NAT=Yes to always use rport when present which was
against 1.6.2 behavior and the check itself was redundant since the only way this
segment of code could be reached was if RPORT_PRESENT was already evaluated as true
earlier.
(closes issue ASTERISK-17789)
Reported by: byronclark
Patches:
use_sip_nat_force_rport.patch uploaded by byronclark (license 1200)
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David Vossel [Tue, 14 Jun 2011 14:37:41 +0000 (14:37 +0000)]
Store sip peer name as var data on a outofcall msg.
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Kinsey Moore [Mon, 13 Jun 2011 20:44:59 +0000 (20:44 +0000)]
Config inheritance doesn't work with ConfBridge() menu definitions
Current behavior in ConfBridge menu definitions is that first definition takes
precedence, even in templated situations. This change allows inheritance and
overriding to work as expected so that the last definition takes precedence.
(closes ASTERISK-17986)
Review: https://reviewboard.asterisk.org/r/1267/
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Leif Madsen [Mon, 13 Jun 2011 19:54:27 +0000 (19:54 +0000)]
Merged revisions 323213 via svnmerge from
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r323213 | lmadsen | 2011-06-13 15:51:52 -0400 (Mon, 13 Jun 2011) | 6 lines
Avoid dividing by zero with L() option to Dial()
Reported by: nicolasom
Patches:
issue-17995.patch - nicolasom (License #5994)
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David Vossel [Mon, 13 Jun 2011 19:43:57 +0000 (19:43 +0000)]
Addition of "outofcall_message_context" sip.conf option.
Review: https://reviewboard.asterisk.org/r/1265/
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Leif Madsen [Mon, 13 Jun 2011 19:03:46 +0000 (19:03 +0000)]
Merged revisions 323154 via svnmerge from
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r323154 | lmadsen | 2011-06-13 15:00:41 -0400 (Mon, 13 Jun 2011) | 6 lines
Tweak documentation for AGI Hangup command.
(closes issue ASTERISK-17999)
Reported by: Ben Klang
Patches:
hangup-doc.diff - uploaded by Ben Klang (License #5876)
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