Russell Bryant [Thu, 13 Dec 2012 16:18:52 +0000 (16:18 +0000)]
Remove compile time check HAVE_DEV_URANDOM.
The code was doing a runtime check, anyway. The compile time check isn't
always valid (cross-compiling, packages).
Review: https://reviewboard.asterisk.org/r/2245/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377977
65c4cc65-6c06-0410-ace0-
fbb531ad65f3
Mark Michelson [Thu, 13 Dec 2012 15:40:03 +0000 (15:40 +0000)]
Re-add taskprocessor cleanup code that was removed by the UUID merge.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377975
65c4cc65-6c06-0410-ace0-
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Sean Bright [Thu, 13 Dec 2012 15:37:55 +0000 (15:37 +0000)]
Use the UUID API to generate and validate UUIDs for res_calendar_exchange.
Currently the res_calendar_exchange module uses its own method of generating
UUIDs using ast_random(). Now that we have a UUID API we should use that
instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377974
65c4cc65-6c06-0410-ace0-
fbb531ad65f3
Mark Michelson [Thu, 13 Dec 2012 15:37:45 +0000 (15:37 +0000)]
The UUID commit removed changes made in res_clialiases.c
This puts back in the changes that are designed to work
around a memory leak fix in the CLI code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377973
65c4cc65-6c06-0410-ace0-
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David M. Lee [Thu, 13 Dec 2012 15:24:22 +0000 (15:24 +0000)]
Fixed configure.ac to look for proper uuid.h file
Introduced in r377846, the configure script was looking for uuid.h instead
of uuid/uuid.h.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377972
65c4cc65-6c06-0410-ace0-
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Brent Eagles [Thu, 13 Dec 2012 15:22:27 +0000 (15:22 +0000)]
This change adds a SIP peer configuration feature to allow the peer's
configured codecs to take precedence on an outgoing call.
This change introduces a new peer configuration property named
'ignore_requested_pref' that causes the requested codec to be ignored when
determining the preferred codec for an outgoing call leg. The consequence is
that Asterisk's usual efforts to prefer avoiding transcoding can be overridden
on a peer-by-peer basis where appropriate.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377971
65c4cc65-6c06-0410-ace0-
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Kinsey Moore [Thu, 13 Dec 2012 14:28:57 +0000 (14:28 +0000)]
Ensure Min-SE is included in outbound INVITEs
Asterisk now includes Min-SE in outbound INVITEs when the value is not
90 (the default) and session timers are not disabled. This has the
effect of Asterisk following RFC4028 more closely with regard to 422
responses and preventing situations in which Asterisk would be forced
to temporarily accept a call to tear it down based on a Session-Expires
below the locally configured Min-SE.
(issue SWP-5051)
Review: https://reviewboard.asterisk.org/r/2222/
Reported-by: Kinsey Moore
Patch-by: Kinsey Moore
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Rusty Newton [Wed, 12 Dec 2012 22:43:40 +0000 (22:43 +0000)]
Incremented EXTRA_SOUNDS_VERSION in sounds/Makefile to 1.4.12 for new Extra Sounds releases
See CHANGES-* files in English extra 1.4.12 tarballs for new sound prompts added.
(closes ASTERISK-20328)
Reported by: Matt Jordan
(closes AST-755)
Reported by: John Bigelow
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Michael L. Young [Wed, 12 Dec 2012 04:43:18 +0000 (04:43 +0000)]
Convert Dynamic Features Buffer To Use ast_str
Currently, the buffer for the dynamic features list is set to a fixed size of
128. If the list is bigger than that, it results in the dynamic feature(s) not
being recognized.
This patch changes the buffer from a fixed size to a dynamic one.
(closes issue ASTERISK-20680)
Reported by: Clod Patry
Tested by: Michael L. Young
Patches:
asterisk-20680-dynamic-features-v2.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2221/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377915
65c4cc65-6c06-0410-ace0-
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Mark Michelson [Wed, 12 Dec 2012 00:02:31 +0000 (00:02 +0000)]
Fix a potential deadlock in chan_sip during transfers.
The issue comes from the fact that transfers may perform
a redirecting update on a channel. The issue is that lock
inversion between the channel and its tech_pvt occurs since
the channel lock is released during the transfer process.
The fix is to move when the redirecting update occurs to a
place where neither the tech_pvt or the channel is locked so
that the two can be locked in the proper order.
(closes issue ASTERISK-20708)
reported by Mark Michelson
patches:
ASTERISK-20708-3.patch uploaded by Mark Michelson (License #5049)
Tested by:
Tim Ringenbach at Asteria Solutions Group
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Mark Michelson [Tue, 11 Dec 2012 22:42:11 +0000 (22:42 +0000)]
Add test events necessary for bridging tests to be able to properly run.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377906
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Richard Mudgett [Tue, 11 Dec 2012 22:03:23 +0000 (22:03 +0000)]
Cleanup CLI commands on exit for several files.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
unregister-cli-multiple-all.patch (license #5909) patch uploaded by Corey Farrell
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Mark Michelson [Tue, 11 Dec 2012 21:53:06 +0000 (21:53 +0000)]
And remove svnmerge-integrated property.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377880
65c4cc65-6c06-0410-ace0-
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Mark Michelson [Tue, 11 Dec 2012 21:47:53 +0000 (21:47 +0000)]
Remove automerge properties.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377878
65c4cc65-6c06-0410-ace0-
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Richard Mudgett [Tue, 11 Dec 2012 21:22:45 +0000 (21:22 +0000)]
Cleanup udptl on exit.
* Cleanup CLI commands on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
udptl-shutdown-1_8-10.patch (license #5909) patch uploaded by Corey Farrell
udptl-shutdown-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
Modified
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Mark Michelson [Tue, 11 Dec 2012 21:04:45 +0000 (21:04 +0000)]
Add UUID support to Asterisk.
This provides a common API for dealing with unique identifiers.
The API provides methods to create, parse, copy, and stringify UUIDs.
An accompanying unit test is provided that tests all operations.
(closes issue ASTERISK-20726)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2217
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377846
65c4cc65-6c06-0410-ace0-
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Mark Michelson [Tue, 11 Dec 2012 20:53:34 +0000 (20:53 +0000)]
Fix crash that can occur if CLI registration fails for an aliased command.
A recent memory leak fix in main/cli.c causes an ast_cli_entry's command
field to be freed and NULLed if ast_cli_register() fails. res_clialiases
was ignoring the return value of ast_cli_register() and was then passing
the NULL command off to a a hash function. This resulted in a crash.
The fix is not to ignore the erroneous return value. If ast_cli_register()
fails, then we do not continue trying to process the current alias.
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Richard Mudgett [Tue, 11 Dec 2012 20:46:23 +0000 (20:46 +0000)]
Cleanup taskprocessor on exit.
* Cleanup CLI commands on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
taskprocessor-cleanup-1_8-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
taskprocessor-cleanup-10-only.patch (license #5909) patch uploaded by Corey Farrell
Modified
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Richard Mudgett [Tue, 11 Dec 2012 20:05:32 +0000 (20:05 +0000)]
Cleanup pbx on exit.
* Cleanup CLI commands on exit.
* Unreference hints and statecbs containers on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
pbx-cleanup-1_8.patch (license #5909) patch uploaded by Corey Farrell
pbx-cleanup-10.patch (license #5909) patch uploaded by Corey Farrell
pbx-cleanup-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
Modified
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Richard Mudgett [Tue, 11 Dec 2012 02:44:53 +0000 (02:44 +0000)]
Cleanup logger on exit.
* Cleanup CLI commands, destroy verbosers and logchannels lists on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
logger-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
Modified
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Richard Mudgett [Tue, 11 Dec 2012 02:13:37 +0000 (02:13 +0000)]
Cleanup indications on exit.
* Made ast_unregister_indication_country() unlink the found tone zone
before selecting a new default_tone_zone to make it impossible to select
the tone zone being unregistered again.
* Ringcadence is no longer parsed twice in store_config_tone_zone().
* Cleanup CLI commands and destroy default_tone_zone on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
indications-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
Modified
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Richard Mudgett [Tue, 11 Dec 2012 01:04:50 +0000 (01:04 +0000)]
Cleanup event on exit.
* Cleanup CLI commands on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
event_shutdown-10-only.patch (license #5909) patch uploaded by Corey Farrell
event_shutdown-1_8-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
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Richard Mudgett [Tue, 11 Dec 2012 00:37:01 +0000 (00:37 +0000)]
Cleanup dnsmgr on exit.
* Cleanup dnsmgr thread and CLI commands on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
dnsmgr-cleanup-1_8.patch (license #5909) patch uploaded by Corey Farrell
dnsmgr-cleanup-10-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
Modified
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Kinsey Moore [Mon, 10 Dec 2012 16:56:37 +0000 (16:56 +0000)]
Ensure ReceiveFax provides a CED tone via T.38
When using res_fax_digium, the T.38 CED tone was not being provided
properly which would cause some incoming faxes to fail. This was not an
issue with res_fax_spandsp since it does not strictly honor the
send_ced flag and sends the CED tone whenever receiving a T.38 fax.
(closes issue FAX-343)
Reported-by: Benjamin Tietz
Patch-by: Kinsey Moore
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Kinsey Moore [Mon, 10 Dec 2012 14:45:52 +0000 (14:45 +0000)]
Handle Session-Expires less than local Min-SE in 200 OK
Ensure that a call is immediately torn down if a Session-Expires value
received in a 200 OK is less than the local Min-SE. This also prevents
Asterisk from allowing calls with Session-Expires below the
RFC4028-mandated minimum (90s).
(closes issue ASTERISK-20653)
Review: https://reviewboard.asterisk.org/r/2237/
Patch-by: Kinsey Moore
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Igor Goncharovskiy [Mon, 10 Dec 2012 07:03:48 +0000 (07:03 +0000)]
Add firmware information to CLI devices listing
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377595
65c4cc65-6c06-0410-ace0-
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Igor Goncharovskiy [Mon, 10 Dec 2012 06:56:04 +0000 (06:56 +0000)]
Fix codec mismatch
Fix code to send in both rx and tx open stream messages correct codecs. Found that on phase 0/1 phones wrong codecs cause to no audio in some situations.
(issue ASTERISK-20183)
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Igor Goncharovskiy [Mon, 10 Dec 2012 05:29:04 +0000 (05:29 +0000)]
Remove trailing whitespaces in number from incoming redial list.
Reported by: Igor Olhovskiy
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Tilghman Lesher [Mon, 10 Dec 2012 01:41:50 +0000 (01:41 +0000)]
Improve documentation by making all of the colors used readable,
no matter what the background color is.
Dark blue on a black background is unreadable, as is yellow on a
light background. This patch turns on the bright attribute for
colors when on a dark background and turns *off* the bright
attribute when the -W command line option is used (indicating a
_light_ background). This ensures that text is readable in both
cases.
Patch by: tilghman
Review: https://reviewboard.asterisk.org/r/2224
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Tilghman Lesher [Mon, 10 Dec 2012 01:29:32 +0000 (01:29 +0000)]
Remove some dead code and additionally handle a case that wasn't handled.
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Joshua Colp [Sun, 9 Dec 2012 01:23:44 +0000 (01:23 +0000)]
Add missing support for "who hung up" to chan_motif.
(closes issue ASTERISK-20671)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2208/
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Richard Mudgett [Sat, 8 Dec 2012 00:30:40 +0000 (00:30 +0000)]
Fix order of SIP allow/disallow in MySQL contrib script.
Using the contrib sippeers.sql script to create the sippeers MySQL table
would result in being unable to place calls if you set the disallow value
to all.
(closes issue ASTERISK-20756)
Reported by: Andre Luis
Patches:
sippeers.patch patch uploaded by Andre Luis
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Richard Mudgett [Fri, 7 Dec 2012 23:45:15 +0000 (23:45 +0000)]
MALLOC_DEBUG: Only wait if we want atexit allocation dumps.
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Kinsey Moore [Fri, 7 Dec 2012 22:08:48 +0000 (22:08 +0000)]
codec_dahdi: Fix output of "transcoder show" CLI command.
In r306010 "Asterisk media architecture conversion - no more format
bitfields", the logic for incrementing encoders and decoders when
opening transcoder channels was changed without making the corresponding
change when decrementing encoder / decoder channels. The result being
that when a channel was destroyed, codec_dahdi couldn't properly tell if
it was an encoder or decoder, and the default case is to assume it was a
decoder.
This could result in negative numbers for decoders in use like in:
VOIP6*CLI> transcoder show
2/-2 encoders/decoders of 92 channels are in use.
(closes issue ASTERISK-19921)
Patch-by: Shaun Ruffell
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Richard Mudgett [Fri, 7 Dec 2012 00:00:39 +0000 (00:00 +0000)]
confbridge: Fix some resource leaks on conference teardown.
* Made destroy_conference_bridge() destroy a missed ast_mutex_t and ast_cond_t.
* Made join_conference_bridge() init the ast_mutex_t's and ast_cond_t so
destroy_conference_bridge() can destroy them unconditionally.
* Made join_conference_bridge() abort if the new conference could not be
added to the conferences container.
* Made leave_conference() discard any post-join actions if
join_conference_bridge() had to abort early.
* Made the join_conference_bridge() diagnostic messages better describe
what happened.
* Renamed leave_conference_bridge() to leave_conference() and made it only
take a conference user pointer. The conference pointer was redundant.
* Made conf_bridge_profile_copy() use struct copy instead of memcpy().
* No need to lock the conference in start_conf_record_thread() since all
of the callers already have it locked.
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Russell Bryant [Thu, 6 Dec 2012 17:29:13 +0000 (17:29 +0000)]
Recorded merge of revisions 377340 from svn.asterisk.org/svn/asterisk/branches/11
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Add CLI tab completion to 'acl show'.
The 'acl show' CLI command allows you to show the details about a specific
named ACL in acl.conf. This patch adds tab completion to the command.
Review: https://reviewboard.asterisk.org/r/2230/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377341
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Russell Bryant [Thu, 6 Dec 2012 15:13:37 +0000 (15:13 +0000)]
Minor code cleanup in named_acl.c.
This patch makes a few little cleanups to named_acl.c. A couple non-public
functions were made static and an opening brace for a function was moved to
its own line, per the coding guidelines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377330
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Russell Bryant [Thu, 6 Dec 2012 15:06:47 +0000 (15:06 +0000)]
Add CLI tab completion to 'acl show'.
The 'acl show' CLI command allows you to show the details about a specific
named ACL in acl.conf. This patch adds tab completion to the command.
Review: https://reviewboard.asterisk.org/r/2230/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377329
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Matthew Jordan [Thu, 6 Dec 2012 14:26:13 +0000 (14:26 +0000)]
Fix memory leak in 'manager show event' when command entered incorrectly
When the CLI command 'manager show event' was run incorrectly and its usage
instructions returned, a reference to the event container was leaked. This
would prevent the container from being reclaimed when Asterisk exits. We now
properly decrement the count on the ao2 object using the nifty RAII_VAR macro.
Thanks to Russell for helping me stumble on this, and Terry for writing that
ridiculously helpful macro.
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Jonathan Rose [Wed, 5 Dec 2012 17:17:06 +0000 (17:17 +0000)]
res_srtp: Fix a crash caused by srtp_dealloc on an already dealloced session
When srtp_create fails, the session may be dealloced or just not alloced. At
the same time though, the session pointer might not be set to NULL in this
process and attempting to srtp_dealloc it again will cause a segfault. This
patch checks for failure of srtp_create and sets the session pointer to NULL
if it fails.
(closes issue ASTERISK-20499)
Reported by: tootai
Review: https://reviewboard.asterisk.org/r/2228/
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Joshua Colp [Wed, 5 Dec 2012 16:51:58 +0000 (16:51 +0000)]
Fix a SIP request memory leak with TLS connections.
During the TLS re-work in chan_sip some TLS specific code was moved
into a separate function. This function operates on a copy of the
incoming SIP request. This copy was never deinitialized causing a
memory leak for each request processed.
This function is now given a SIP request structure which it can use
to copy the incoming request into. This reduces the amount of memory
allocations done since the internal allocated components are reused
between packets and also ensures the SIP request structure is
deinitialized when the TLS connection is torn down.
(closes issue ASTERISK-20763)
Reported by: deti
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Richard Mudgett [Wed, 5 Dec 2012 02:23:10 +0000 (02:23 +0000)]
Remove init_framer(). It no longer does anything.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377246
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Richard Mudgett [Wed, 5 Dec 2012 02:20:57 +0000 (02:20 +0000)]
Fix registering core show codecs/codec CLI commands twice.
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Richard Mudgett [Wed, 5 Dec 2012 01:11:26 +0000 (01:11 +0000)]
confbridge: Fix several small issues.
* Made func_confbridge_helper() allow an empty value when setting options.
You previously could not Set(CONFBRIDGE(user,pin)=) and clear the
configured pin from the dialplan.
* Made func_confbridge_helper() handle its datastore better if multiple
threads attempt to set the first CONFBRIDGE option value on the channel.
* Made the func_confbridge_helper() only output one diagnostic message
concerning the option.
* Made the bridge video_mode able to repeatedly change in the config file
and CONFBRIDGE dialplan function. The video_mode option values are an
enum and not independent of each other.
* Made handle_cli_confbridge_show_bridge_profile() better handle the
video_mode option.
* Simplified datastore handling code in conf_find_user_profile() and
conf_find_bridge_profile().
(closes issue ASTERISK-20655)
Reported by: Birger "WIMPy" Harzenetter
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Richard Mudgett [Tue, 4 Dec 2012 22:34:05 +0000 (22:34 +0000)]
confbridge: Update online XML documentation.
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Russell Bryant [Tue, 4 Dec 2012 13:01:23 +0000 (13:01 +0000)]
Add libuuid to install_prereq for Fedora.
I ran this script and my build failed. pjproject requires this.
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Richard Mudgett [Mon, 3 Dec 2012 23:00:08 +0000 (23:00 +0000)]
Cleanup ast_run_atexits() atexits list.
* Convert atexits list to a mutex instead of a rd/wr lock. The lock is
only write locked.
* Move CLI verbose Asterisk ending message to where AMI message is output
in really_quit() to avoid further surprises about using stuff already
shutdown.
(issue ASTERISK-20649)
Reported by: Corey Farrell
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Richard Mudgett [Mon, 3 Dec 2012 20:46:11 +0000 (20:46 +0000)]
Cleanup core main on exit.
* Cleanup time zones on exit.
* Make exit clean/unclean report consistent for AMI and CLI in
really_quit().
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
core-cleanup-1_8-10.patch (license #5909) patch uploaded by Corey Farrell
core-cleanup-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
Modified
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Richard Mudgett [Mon, 3 Dec 2012 19:33:06 +0000 (19:33 +0000)]
Cleanup config cache on exit.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
config-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
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Richard Mudgett [Mon, 3 Dec 2012 19:17:24 +0000 (19:17 +0000)]
Cleanup CLI resources on exit and CLI command registration errors.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
cli-leaks-1_8-10.patch (license #5909) patch uploaded by Corey Farrell
cli-leaks-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
Modified
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Richard Mudgett [Mon, 3 Dec 2012 18:45:18 +0000 (18:45 +0000)]
Cleanup CDR resources on exit.
* Simplify do_reload() return handling since it never returned anything
other than 0.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
cdr-cleanup-1_8.patch (license #5909) patch uploaded by Corey Farrell
cdr-cleanup-10-11-trunk.patch (license #5909) patch uploaded by Corey Farrell
Modified
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Richard Mudgett [Mon, 3 Dec 2012 17:10:40 +0000 (17:10 +0000)]
Fix CCSS CLI commands and logger level not unregistered.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
ccss-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
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Olle Johansson [Mon, 3 Dec 2012 16:45:49 +0000 (16:45 +0000)]
Formatting fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377035
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Joshua Colp [Mon, 3 Dec 2012 14:56:36 +0000 (14:56 +0000)]
Fix an RTP instance reference count leak in chan_motif.
When setting up an RTP instance the RTCP portion of the instance
keeps a reference to the instance itself. In order to release this
reference and stop RTCP the stop API call must be called before
destroying the instance.
(closes issue ASTERISK-20751)
Reported by: joshoa
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Olle Johansson [Mon, 3 Dec 2012 14:46:02 +0000 (14:46 +0000)]
Move functions to AFTER the block of forward declarations of functions.
It was a mess. The first part of chan_sip.c is constants, declarations, structures and stuff,
then forward declarations and then actual code. It's still a mess, but a bit less messy ;-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377018
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Olle Johansson [Mon, 3 Dec 2012 09:35:55 +0000 (09:35 +0000)]
Formatting changes
Found a large amount of missing {} in the code before patching in another branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376998
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Joshua Colp [Sat, 1 Dec 2012 00:47:42 +0000 (00:47 +0000)]
Tweak extension used for incoming calls received on Motif.
Based on feedback from numerous individuals this patch tweaks incoming calls
to first look for an extension with the name of the endpoint. If no such extension
exists the call will silently fall back to the "s" extension as it previously
did.
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Richard Mudgett [Fri, 30 Nov 2012 21:38:01 +0000 (21:38 +0000)]
chan_misdn: Fix sending RELEASE_COMPLETE in response to SETUP.
Fix sending a RELEASE_COMPLETE in response to a SETUP if chan_misdn does
not have a B channel available to assign to the call.
(closes issue ABE-2869)
Reported by: Guenther Kelleter
Patches:
setup-reject_2.diff (license #6372) patch uploaded by Guenther Kelleter
Modified
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Sean Bright [Fri, 30 Nov 2012 17:08:41 +0000 (17:08 +0000)]
Minor spelling fix to the VOLUME documentation.
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Mark Michelson [Fri, 30 Nov 2012 16:56:53 +0000 (16:56 +0000)]
Fix potential crashes during SIP attended transfers.
The principal behind this patch is simple. During a transfer,
we manipulate channels that are owned by a separate thread than
the one we currently are running in, so it makes sense that we
need to grab a reference to the channels so that they cannot
disappear out from under us.
In the wild, crashes were sometimes seen when the transferring
party would hang up the call before the transfer target answered
the call. The most common place to see the crash occur was when
attempting to send a connected line update to the transferer
channel.
(closes issue ASTERISK-20226)
Reported by Jared Smith
Patches:
ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
Tested by: Jared Smith
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Richard Mudgett [Thu, 29 Nov 2012 23:01:16 +0000 (23:01 +0000)]
chan_local: Fix local_pvt ref leak in local_devicestate().
Regression introduced by ASTERISK-20390 fix.
(closes issue ASTERISK-20769)
Reported by: rmudgett
Tested by: rmudgett
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Richard Mudgett [Thu, 29 Nov 2012 22:34:24 +0000 (22:34 +0000)]
Fix compile error.
(issue ASTERISK-20724)
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Michael L. Young [Thu, 29 Nov 2012 21:58:41 +0000 (21:58 +0000)]
Improve Code Readability And Fix Setting natdetected Flag
For 1.8, 10, 11 and trunk we are are improving the code readability.
For 11 and trunk, auto nat detection was added. The natdetected flag was being
set to 1 when the host address in the VIA header did not specifiy a port. This
patch fixes this by setting the port on the temporary sock address used to
SIP_STANDARD_PORT in order for the sock address comparison to work properly.
(closes issue ASTERISK-20724)
Reported by: Michael L. Young
Patches:
asterisk-20724-set-port-v2.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2206/
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David M. Lee [Thu, 29 Nov 2012 17:16:50 +0000 (17:16 +0000)]
Fixed ast_random's comment about locking.
The original comment was separated from the code at some point, and didn't
reflect the use of libc's other than glibc for Linux.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376821
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Pedro Kiefer [Thu, 29 Nov 2012 16:44:42 +0000 (16:44 +0000)]
Fix chan_sip websocket payload handling
Websocket by default doesn't return an ast_str for the payload received. When
converting it to an ast_str on chan_sip the last character was being omitted,
because ast_str functions expects that the given length includes the trailing
0x00. payload_len only has the actual string length without counting the
trailing zero.
For most cases this passed unnoticed as most of SIP messages ends with \r\n.
(closes issue ASTERISK-20745)
Reported by: Iñaki Baz Castillo
Review: https://reviewboard.asterisk.org/r/2219/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376820
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Richard Mudgett [Thu, 29 Nov 2012 00:48:12 +0000 (00:48 +0000)]
Add MALLOC_DEBUG atexit unreleased malloc memory summary.
* Adds the following CLI commands to control MALLOC_DEBUG reporting of
unreleased malloc memory when Asterisk is shut down.
memory atexit list on
memory atexit list off
memory atexit summary byline
memory atexit summary byfunc
memory atexit summary byfile
memory atexit summary off
* Made check all remaining allocated region blocks atexit for fence
violations.
* Increased the allocated region hash table size by about three times. It
still isn't large enough considering the number of malloced blocks
Asterisk uses.
* Made CLI "memory show allocations anomalies" use
regions_check_all_fences().
Review: https://reviewboard.asterisk.org/r/2196/
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Richard Mudgett [Thu, 29 Nov 2012 00:07:55 +0000 (00:07 +0000)]
Enhance MALLOC_DEBUG CLI commands.
* Fixed CLI "memory show allocations" misspelling of anomalies option.
The command will still accept the original misspelling.
* Miscellaneous tweaks to CLI "memory show allocations" command output
format.
* Made CLI "memory show summary" summarize by line number instead of by
function if a filename is given.
* Made CLI "memory show summary" sort its output by filename or
function-name/line-number depending upon request.
* Miscellaneous tweaks to CLI "memory show summary" command output format.
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Jonathan Rose [Wed, 28 Nov 2012 16:47:44 +0000 (16:47 +0000)]
manager: Make challenge work with allowmultiplelogin=no
Prior to this patch, challenge would yield a multiple logins error if used
without providing the username (which isn't really supposed to be an argument
to challenge) if allowmultiplelogin was set to no because allowmultiplelogin
finds a user with a zero length login name. This check is simply disabled for
the challenge action when the username is empty by this patch.
(closes issue ASTERISK-20677)
Reported by: Vladimir
Patches:
challenge_action_nomultiplelogin.diff uploaded by Jonathan Rose (license 6182)
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Richard Mudgett [Wed, 28 Nov 2012 00:13:10 +0000 (00:13 +0000)]
Fix extension matching with the '-' char.
The '-' char is supposed to be ignored by the dialplan extension matching.
Unfortunately, it's treatment is not handled consistently throughout the
extension matching code.
* Made the old exten matching code consistently ignore '-' chars.
* Made the old exten matching code consistently handle case in the
matching.
* Made ignore empty character sets.
* Fixed ast_extension_cmp() to return -1, 0, or 1 as documented. The only
user of it in pbx_lua.c was testing for -1. It was originally returning
the strcmp() value for less than which is not usually going to be -1.
* Fix character set sorting if the sets have the same number of characters
and start with the same character. Character set [0-9] now sorts before
[02-9a] as originally intended.
* Updated some extension label and priority already in use warnings to
also indicate if the extension is aliased.
(closes issue ASTERISK-19205)
Reported by: Philippe Lindheimer, Birger "WIMPy" Harzenetter
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2201/
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Richard Mudgett [Tue, 27 Nov 2012 20:39:51 +0000 (20:39 +0000)]
Remove unnecessary channel module references.
* Removed call to ast_module_user_hangup_all() in res_config_mysql.c since
it is effectively a noop. No channels can attach a reference to that
module.
* Removed call to ast_module_user_hangup_all() in app_celgenuserevent.c.
The caller of unload_module() has already called it.
* Removed redundant channel module references in pbx_dundi.c. The
registered dialplan function callback dispatchers for the read/read2/write
callbacks already reference the module before calling.
* pbx_dundi: Moved unregistering CLI commands, DUNDi switch, and dialplan
functions to the first thing the unload_module() does. This will reduce
the chance of new channels using DUNDi services while the module is being
torn down.
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Richard Mudgett [Tue, 27 Nov 2012 17:54:25 +0000 (17:54 +0000)]
Made AST_LIST_REMOVE() simpler and use better names.
* Update doxygen of AST_LIST_REMOVE().
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Matthew Jordan [Fri, 23 Nov 2012 00:02:23 +0000 (00:02 +0000)]
Re-initialize logmsgs mutex upon logger initialization to prevent lock errors
Similar to the patch that moved the fork earlier in the startup sequence to
prevent mutex errors in the recursive mutex surrounding the read/write thread
registration lock, this patch re-initializes the logmsgs mutex. Part of the
start up sequence before forking the process into the background includes
reading asterisk.conf; this has to occur prior to the call to daemon in order
to read startup parameters. When reading in a conf file, log statements can
be generated. Since this can't be avoided, the mutex instead is
re-initialized to ensure a reset of any thread tracking information.
This patch also includes some additional debugging to catch errors when
locking or unlocking the recursive mutex that surrounds locks when the
DEBUG_THREADS build option is enabled. DO_CRASH or THREAD_CRASH will
cause an abort() if a mutex error is detected.
(issue ASTERISK-19463)
Reported by: mjordan
Tesetd by: mjordan
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Richard Mudgett [Wed, 21 Nov 2012 18:33:16 +0000 (18:33 +0000)]
Add red-black tree container type to astobj2.
* Add red-black tree container type.
* Add CLI command "astobj2 container dump <name>"
* Added ao2_container_dump() so the container could be dumped by other
modules for debugging purposes.
* Changed ao2_container_stats() so it can be used by other modules like
ao2_container_check() for debugging purposes.
* Updated the unit tests to check red-black tree containers.
(closes issue ASTERISK-19970)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2110/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376575
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David M. Lee [Tue, 20 Nov 2012 22:06:05 +0000 (22:06 +0000)]
Added missing newlines to websocket ast_logs.
Without these newlines, log messages just continue tacking onto the same
line, and do not flush immediately.
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Mark Michelson [Tue, 20 Nov 2012 19:09:37 +0000 (19:09 +0000)]
Add "Require: timer" to 200 OK responses when appropriate.
The method by which the Require header is added to 200 responses is
inspired by the method that Olle Johansson uses in his darjeeling-prack
branch.
(closes issue ASTERISK-20570)
Reported by Matt Jordan, at the behest of Olle Johansson
Review: https://reviewboard.asterisk.org/r/2172
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Alec L Davis [Tue, 20 Nov 2012 17:39:11 +0000 (17:39 +0000)]
Reduce CLI spam of "Extension Changed" device state messages.
Asterisk 11 follows RFC3265 that states that after every subscribe or resubscribe a notify should be sent.
Thus the console if filled continuously with the following after every subscribe;
== Extension Changed 8512[phones] new state IDLE for Notify User cisco1
In Asterisk 1.8 only changes would be sent. Thus only when a device state changed was anything emitted to the console.
fix:
Only print to console when device state isn't forced.
(closes issue ASTERISK-20706)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
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Walter Doekes [Mon, 19 Nov 2012 20:03:56 +0000 (20:03 +0000)]
Fix most leftover non-opaque ast_str uses.
Instead of calling str->str, one should use ast_str_buffer(str). Same
goes for str->used as ast_str_strlen(str) and str->len as
ast_str_size(str).
Review: https://reviewboard.asterisk.org/r/2198
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Matthew Jordan [Mon, 19 Nov 2012 02:14:54 +0000 (02:14 +0000)]
Fix uninitialized in this function error
With some versions of gcc, n_buckets will be flagged as being uninitialized
before use. While its technically impossible (since the switch statement,
even without a default, accounts for all possibilities), we'll initialize the
variable to 0 anyway.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376457
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Matthew Jordan [Sun, 18 Nov 2012 20:27:45 +0000 (20:27 +0000)]
Reorder startup sequence to prevent lockups when process is sent to background
Although it is very rare and timing dependent, the potential exists for the
call to 'daemon' to cause what appears to be a deadlock in Asterisk during
startup. This can occur when a recursive mutex is obtained prior to the
daemon call executing. Since daemon uses fork to send the process into the
background, any threading primitives are unsafe to re-use after the call.
Implementations of pthread recursive mutexes are highly likely to store the
thread identifier of the thread that previously obtained the mutex. If
the mutex was locked prior to the fork, a subsequent unlock operation will
potentially fail as the thread identifier is no longer valid. Since the
mutex is still locked, all subsequent attempts to grab the mutex by other
threads will block.
This behavior exhibited itself most often when DEBUG_THREADS was enabled, as
this compile time option surrounds the mutexes in Asterisk with another
recursive mutex that protects the storage of thread related information. This
made it much more likely that a recursive mutex would be obtained prior to
daemon and unlocked after the call.
This patch does the following:
a) It backports a patch from Asterisk 11 that prevents the spawning of the
localtime monitoring thread. This thread is now spawned after Asterisk has
fully booted.
b) It re-orders the startup sequence to call daemon earlier during Asterisk
startup. This limits the potential of threading primitives being accessed
by initialization calls before daemon is called.
c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called.
Developers should send error messages directly to stderr prior to daemon,
as calls to ast_log may access recursive mutexes that store thread related
information.
d) It reorganizes when thread local storage is created for storing lock
information during the creation of threads. Prior to this patch, the
read/write lock protecting the list of threads in ast_register_thread would
utilize the lock in the thread local storage prior to it being initialized;
this patch prevents that.
On a very related note, this patch will *greatly* improve the stability of the
Asterisk Test Suite.
Review: https://reviewboard.asterisk.org/r/2197
(closes issue ASTERISK-19463)
Reported by: mjordan
Tested by: mjordan
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Matthew Jordan [Sun, 18 Nov 2012 14:31:32 +0000 (14:31 +0000)]
Add a test event that reports changes in ConfBridge state
This patch adds a test event to ConfBridge that reports transitions between
states in ConfBridge. This is used by tests in the Asterisk Test Suite
that verify state changes based on the entering/leaving of conference
participants.
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David M. Lee [Fri, 16 Nov 2012 00:15:30 +0000 (00:15 +0000)]
Fixed extconf.c breakage introduced in r376306.
To quote wdoekes:
> Note that I'm not confirming legitimacy of having that file in tree at
> all. Is anyone using aelparse/conf2ael?
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David M. Lee [Fri, 16 Nov 2012 00:14:00 +0000 (00:14 +0000)]
Somehow I put in svn-1.6 merge information. Oops.
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David M. Lee [Fri, 16 Nov 2012 00:08:00 +0000 (00:08 +0000)]
Migrate hashtest/hashtest2 to be unit tests.
Both hashtest and hashtest2 are manual testing apps that thrash hash
tables (hashtab and ao2 containers, respectively), by spinning up
several threads that randomly insert, delete, lookup and iterate over
the hash table. If the app doesn't crash, the hash table probably passes
the test. Those utils are not a part of the typical Asterisk build, so
they do not usually get compiled. This all makes them less that useful.
This patch removes those manual test programs and replaces them with
Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It also
attempts to make the tests more deterministic.
* Rather than spinning up some number of threads that operate on the
hash table randomly, spin up four threads that concurrenly add,
remove, lookup and iterate over the hash table.
* Each thread checks the state of the hash table both during and after
execution, and indicates a test failure if things are not as expected.
* Each thread times out after 60 seconds to prevent deadlocking the unit
test run.
(closes issue ASTERISK-20505)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2189/
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Jonathan Rose [Thu, 15 Nov 2012 23:10:13 +0000 (23:10 +0000)]
app_meetme: Fix channels lingering when hung up under certain conditions
Channels would get stuck and MeetMe would repeatedly display an Unable
to write frame to channel error in the conf_run function if hung up
during certain sound prompts such as during user count announcements.
This patch fixes that by reintroducing a hangup check in the meetme's
main loop (also in conf_run).
(closes issue ASTERISK-20486)
Reported by: Michael Cargile
Review: https://reviewboard.asterisk.org/r/2187/
Patches:
meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan Rose (license 6182)
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Brent Eagles [Thu, 15 Nov 2012 14:35:01 +0000 (14:35 +0000)]
Patch to prevent stopping the active generator when it is not the silence
generator.
This patch introduces an internal helper function to safely check whether the
current generator is the one that is expected before deactivating it. The
current externally accessible ast_channel_stop_generator() function has been
modified to be implemented in terms of the new function.
(closes issue ASTERISK-19918)
Reported by: Eduardo Abad
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Rusty Newton [Thu, 15 Nov 2012 02:29:40 +0000 (02:29 +0000)]
Patch to play correct sound file when a voicemail's urgent status is removed
We were attempting to play "vm-urgent-removed", which didn't exist. Now we play "vm-marked-nonurgent" which exists
and is the correct sound file. Previous behavior was silence and a warning on the CLI.
(issue ASTERISK-20280)
(closes issue ASTERISK-20280)
Reported by: Tomo Takebe
Tested by: Rusty Newton
Patches:
asterisk20280.patch uploaded by Rusty Newton (license 5829)
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Richard Mudgett [Wed, 14 Nov 2012 19:55:39 +0000 (19:55 +0000)]
Fix call files when astspooldir is relative.
Future dated call files are ignored when astspooldir is relative to the
current directory. The queue_file() assumed that the qdir needed to be
prepended if the given filename did not start with a '/'. If astspooldir
is relative it is not going to start from the root directory obviously so
it will not start with a '/'. The filename used in queue_file()
ultimately results in qdir prepended multiple times.
* Made queue_file() not prepend qdir if the filename contains a '/'.
(closes issue ASTERISK-20593)
Reported by: James Le Cuirot
Patches:
0004-Fix-future-call-files-from-relative-directories.patch (license #6439) patch uploaded by James Le Cuirot
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Jonathan Rose [Tue, 13 Nov 2012 19:42:13 +0000 (19:42 +0000)]
chan_sip: Add SubscribeContext field to SIPshowpeer AMI response
The new field is will show up within the response if the requested peer has a
subscribe context set.
(closes issue ASTERISK-20626)
Reported by: Jaco Kroon
Patches:
asterisk-sip-ami-SubscrContext.patch uploaded by jkroon (license 5671)
-with modifications by jrose to conform to style guidelines
Review: https://reviewboard.asterisk.org/r/2195/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376219
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Joshua Colp [Mon, 12 Nov 2012 20:46:51 +0000 (20:46 +0000)]
Properly check if the "Context" and "Extension" headers are empty in a ShowDialPlan action.
The code which handles the ShowDialPlan action wrongly assumed that a non-NULL return value
from the function which retrieves headers from an action indicates that the header has a
value. This is incorrect and the contents must be checked to see if they are blank.
(closes issue ASTERISK-20628)
Reported by: jkroon
Patches:
asterisk-showdialplan-incorrect-error.patch uploaded by jkroon
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Michael L. Young [Mon, 12 Nov 2012 20:18:47 +0000 (20:18 +0000)]
Fix Dynamic Hints Variable Substition - Underscore Problem
When adding a dynamic hint, if an extension contains an underscore no variable
subsitution is being performed.
This patch changes from checking if the extension contains an underscore to
checking if the extension begins with an underscore.
(closes issue ASTERISK-20639)
Reported by: Steven T. Wheeler
Tested by: Steven T. Wheeler, Michael L. Young
Patches:
asterisk-20639-dynamic-hint-underscore.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2188/
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Joshua Colp [Sun, 11 Nov 2012 17:15:47 +0000 (17:15 +0000)]
Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default.
With ICE support enabled in chan_sip and a large number of interfaces on the system it was
possible for the produced SDP to be truncated due to some fixed size buffers. These buffers
have now been changed so they will dynamically grow as needed.
ICE support is now also enabled by default in res_rtp_asterisk to provide a smoother experience
for chan_motif users where it is required. To maintain the previous behavior in chan_sip it is
no longer enabled by default there.
(closes issue ASTERISK-20643)
Reported by: coopvr
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Mark Michelson [Thu, 8 Nov 2012 22:10:29 +0000 (22:10 +0000)]
Fix a "set but not used" warning on newer gccs.
Turns out the "helpful" setting of ms and res in this
macro is completely useless after the timeout antipattern
fix.
If you're a new guy looking to write code, don't write
a macro like this one.
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Richard Mudgett [Thu, 8 Nov 2012 21:12:35 +0000 (21:12 +0000)]
chan_dahdi/SS7: Made reject incoming call for an in-alarm or blocked channel.
If a SS7 call comes in requesting a CIC that is in-alarm, the call is
accepted and connects if the extension exists in the dialplan. The call
does not have any audio.
* Made release the call immediately with circuit congestion cause.
(closes issue ASTERISK-20204)
Reported by: Tuan Le
Patches:
jira_asterisk_20204_v1.8.patch (license #5621) patch uploaded by rmudgett
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Richard Mudgett [Thu, 8 Nov 2012 17:38:31 +0000 (17:38 +0000)]
Add MALLOC_DEBUG enhancements.
* Makes malloc() behave like calloc(). It will return a memory block
filled with 0x55. A nonzero value.
* Makes free() fill the released memory block and boundary fence's with
0xdeaddead. Any pointer use after free is going to have a pointer
pointing to 0xdeaddead. The 0xdeaddead pointer is usually an invalid
memory address so a crash is expected.
* Puts the freed memory block into a circular array so it is not reused
immediately.
* When the circular array rotates out a memory block to the heap it checks
that the memory has not been altered from 0xdeaddead.
* Made the astmm_log message wording better.
* Made crash if the DO_CRASH menuselect option is enabled and something is
found.
* Fixed a potential alignment issue on 64 bit systems.
struct ast_region.data[] should now be aligned correctly for all
platforms.
* Extracted region_check_fences() from __ast_free_region() and
handle_memory_show().
* Updated handle_memory_show() CLI usage help.
Review: https://reviewboard.asterisk.org/r/2182/
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Mark Michelson [Wed, 7 Nov 2012 19:15:26 +0000 (19:15 +0000)]
Multiple revisions 375993-375994
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r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines
Fix misuses of timeouts throughout the code.
Prior to this change, a common method for determining if a timeout
was reached was to call a function such as ast_waitfor_n() and inspect
the out parameter that told how many milliseconds were left, then use
that as the input to ast_waitfor_n() on the next go-around.
The problem with this is that in some cases, submillisecond timeouts
can occur, resulting in the out parameter not decreasing any. When this
happens thousands of times, the result is that the timeout takes much
longer than intended to be reached. As an example, I had a situation where
a 3 second timeout took multiple days to finally end since most wakeups
from ast_waitfor_n() were under a millisecond.
This patch seeks to fix this pattern throughout the code. Now we log the
time when an operation began and find the difference in wall clock time
between now and when the event started. This means that sub-millisecond timeouts
now cannot play havoc when trying to determine if something has timed out.
Part of this fix also includes changing the function ast_waitfor() so that it
is possible for it to return less than zero when a negative timeout is given
to it. This makes it actually possible to detect errors in ast_waitfor() when
there is no timeout.
(closes issue ASTERISK-20414)
reported by David M. Lee
Review: https://reviewboard.asterisk.org/r/2135/
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r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines
Remove some debugging that accidentally made it in the last commit.
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Richard Mudgett [Tue, 6 Nov 2012 19:05:11 +0000 (19:05 +0000)]
Fix stuck DTMF when bridge is broken.
When a bridge is broken by an AMI Redirect action or the ChannelRedirect
application, an in progress DTMF digit could be stuck sending forever.
* Made simulate a DTMF end event when a bridge is broken and a DTMF digit
was in progress.
(closes issue ASTERISK-20492)
Reported by: Jeremiah Gowdy
Patches:
bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by Jeremiah Gowdy
Modified to jira_asterisk_20492_v1.8.patch
jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2169/
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Joshua Colp [Tue, 6 Nov 2012 12:15:31 +0000 (12:15 +0000)]
Fix a bug where our Motif ICE candidates were not quite proper, and make us more forgiving.
An issue was reported on the mailing list where calling would result in an "Incomplete
ICE-UDP candidate received on session" error message. This is the result of the ICE-UDP
candidate code not placing a "network" attribute within the candidates. This is now done.
To increase compatibility though I have removed the requirement for the "network" attribute
to exist within ICE-UDP candidates that are received since we don't actually require the
value.
Reported on the mailing list by Jean-Denis Girard.
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Matthew Jordan [Mon, 5 Nov 2012 23:10:14 +0000 (23:10 +0000)]
Refactor ast_timer_ack to return an error and handle the error in timer users
Currently, if an acknowledgement of a timer fails Asterisk will not realize
that a serious error occurred and will continue attempting to use the timer's
file descriptor. This can lead to situations where errors stream to the
CLI/log file. This consumes significant resources, masks the actual problem
that occurred (whatever caused the timer to fail in the first place), and
can leave channels in odd states.
This patch propagates the errors in the timing resource modules up through
the timer core, and makes users of these timers handle acknowledgement
failures. It also adds some defensive coding around the use of timers
to prevent using bad file descriptors in off nominal code paths.
Note that the patch created by the issue reporter was modified slightly for
this commit and backported to 1.8, as it was originally written for
Asterisk 10.
Review: https://reviewboard.asterisk.org/r/2178/
(issue ASTERISK-20032)
Reported by: Jeremiah Gowdy
patches:
jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358)
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Richard Mudgett [Mon, 5 Nov 2012 21:42:49 +0000 (21:42 +0000)]
Add safety NULL pointer check in module user references.
Made __ast_module_user_remove() check for NULL pointers.
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Jonathan Rose [Mon, 5 Nov 2012 18:00:39 +0000 (18:00 +0000)]
chan_sip: Document a change to user-field encoding introduced with r303509
The change in question was added to improve compliance with RFC3261, but at
the time of commit, it wasn't adequately documented in the UPGRADE notes.
(closes issue ASTERISK-20561)
Reported by: Deniz
Review: https://reviewboard.asterisk.org/r/2177/
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