Age | Commit message (Collapse) | Author |
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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-fbb531ad65f3
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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Reported by: Igor Olhovskiy
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue ASTERISK-20671)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2208/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377463 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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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* 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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http://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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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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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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* 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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I ran this script and my build failed. pjproject requires this.
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* 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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* 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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(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
config-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
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(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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* 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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(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
ccss-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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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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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Regression introduced by ASTERISK-20390 fix.
(closes issue ASTERISK-20769)
Reported by: rmudgett
Tested by: rmudgett
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(issue ASTERISK-20724)
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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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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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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/
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